“So how did the BeoLab 90 make us feel? When we closed our eyes in Bang & Olufsen’s special listening room, the pair of master reference speakers (#2 and #3 ever made)—along with the room—seemed to vanish the instant a song played. We weren’t listening to sound emanating from two specific points; instead, the Weeknd was singing his heart out right in front of us. Benny Goodman’s band performed an intimate set, and you could picture where each musician was sitting. The BeoLab 90’s ability to create such a lifelike three-dimensional sound stage is unparalleled when you’re sitting in the sweet spot. It certainly brings up the question of whether a speaker can be “too” good for the music—some now-classic albums weren’t necessarily well-recorded and mastered (think of when the Rolling Stones turned the basement of a rented French mansion into a makeshift studio slash drug den). But when all the variables align perfectly, the music engulfs listeners entirely and hits the guts. The result of such incredible technology and engineering happens to be a very visceral human experience.”
“Die Abbildung war phänomenal, jedes Instrument der gewählten Musik nahm ganz selbstverständlich den für sich bestimmten Platz im Raum ein, jedes Element war von Anfang bis Ende verfolg- und erlebbar. Aber nicht nur die Ortung verblüffte, auch die Detailgenauigkeit, mit der selbst kleinste Feinheiten bis zum erkälteten Backgroundsänger aufgedeckt wurde, sucht ihresgleichen.”
– modernhifi.de
“Vi kan bevidne, at effekten er besnærende. Højttalerne spiller sammen med lytterummet på en måde, vi ikke har oplevet før. Personligt har jeg aldrig hørt et mere holografisk realistisk lydbillede, hverken i eller uden for sweet spot. BeoLab 90 er også en fuldblods, fullrange-højttaler, der ikke overlader noget til tilfældighederne.”
– lydogbillede.dk
“I found that the size of the soundstage was consistently proportional to the size of the ensemble and the recording. I found that the bass was very well extended, taut, and satisfying. Most of all, I was impressed by the prototypes’ reproduction of detail throughout the audio band, and the uniformity of that quality across the soundstage.”
– Kalman Rubinson, Stereophile magazine (print version, October, 2015)
“Wohl noch nie haben Lautsprecher die musikali schen Akteure so scharf ins Wohnzimmer projiziert. Ganz gleich, ob grosse Orchester oder kleine Jazz-Formationen – jedes ein zelne Instrument hat seinen exakten Platz im virtuellen Raum, der auch seine Tiefen dimension verblüffend genau zu erkennen gibt: Der Hörer kann zum Beispiel fast in Zentimetern abzählen, wie weit das Schlag zeug hinter dem Kontrabass placiert ist. Dass der Beolab 90 auch für schwärzeste BassTiefe, überbordende Dynamik und feinen, luftigen Obertonglanz steht, müssen TestHörer der Vollständigkeit halber natür lich ebenfalls zu Protokoll geben, aber das eigentlich Spektakuläre des Lautsprechers ist tatsächlich seine überragende räumliche Abbildung.”
– NZZ am Sonntag 18. Oktober 2015
This video is from the press event, recorded by the journalist from recordere.dk.
Jakob Dyreby designed/engineered the DSP that controls the beam width (or “directivity”, if you’re a geek) of the BeoLab 90. In this video, he discusses some of the processes involved in arriving at those filters.
One of the questions that has come up with regards to the specifications of the new BeoLab 90 is about the size of the woofers. The specifications state that it has one 13″ front woofer and three 10″ woofers for the sides and rear. However, if you look in the technical specifications in the Technical Sound Guide, you’ll see that the “effective diameter” of the front woofer is 260 mm (about 10″) and the remaining woofers is 212 mm (about 8″). Why is there a discrepancy?
The difference is in how a woofer – or any loudspeaker driver – is measured. When you say 13″ woofer, the measurement is the external diameter of the circular metal frame around the front of the driver. If you look on the first page of the datasheet shows that this diameter is 320 mm for the BeoLab 90’s front woofer – so it’s a 13″ driver. (I’ve copied the technical drawing from the datasheet below – see the two dimensions given on the left side of the drawing.) However, this diameter includes non-moving parts (at least they should not move – they’re screwed to the enclosure).
If you measure the moving parts of the woofer, then we have to decide on where to measure – what is the actual diameter of the diaphragm? Normally, the way to measure this is from the high points on the surround that encircles the diaphragm and connects it to the loudspeaker frame. As you can see in that same drawing, this diameter is 258.8 mm, which, in the “official” datasheet is rounded to 260 mm.
The technical drawing of the Scan-Speak 13″ Revelator front woofer from the official datasheet.
Most loudspeaker projects at Bang & Olufsen are conceived in the design or the Product Definition department. This means that someone decides something like “we’re going to make a loudspeaker, this size with this design” and then the project arrives at the Acoustics Department to find out whether or not the idea is feasible. If it is, then it continues through the development process until we reach the end where the is a product in a store. A better description of this process is in this posting.
BeoLab 90 was different. Instead of being a single project that began, evolved, and ended, it was more like a number of little streams coming together to form a river. Each stream was an idea that contributed to the final product.
One of the early “streams” was an idea that was hatched in the Acoustics Department itself around 2009. I went to the head of the department at the time, and offered to make a deal. If I were to pay for all the components personally, could I use my work hours and B&O resources (like the Cube) to build a pair of loudspeakers for home. These would be a “one chair – no friends” style of loudspeaker – so it would not really be a good candidate for a B&O loudspeaker (our customers typically have friends…). In return, I would keep the loudspeakers in the listening room at B&O for an extended time so that we could use them to demo what we are capable of creating, without our typical restraints imposed by design, development time, size, “normal” product requirements (like built-in amplifiers and DSP), and cost of components.
By early 2011, these loudspeakers were built (although not finished…) and ready for measurements and tuning. The photo below shows the “raw” loudspeaker on the crane in the Cube going out to be measured.
Figure 1. One of the original “seeds” that began the discussions about whether BeoLab 90 should be considered. The woofers are identical to the BeoLab 9 10″woofer. The mid-bass is a Scan-Speak 5 1/4″ Illuminator. The tweeter is a Fountek Pro5i ribbon.
Those loudspeakers lived in Listening Room 1 for about a year. We’re bring people in for a “special demonstration” of a loudspeaker behind the curtain. The general consensus was that the loudspeakers sounded great – but when the curtain was opened, many people started laughing due to the sheer size (and the ugliness of my design, apparently…) of the loudspeakers.
Figure 2: The finished version.
The second “stream” was an idea that was born from Gert Munch’s goal of building a loudspeaker with a smooth power response as well as a flat on-axis magnitude response. The experiment was based on a “normal” two-way loudspeaker that had an additional side-firing dipole on it. The basic idea was that the two-way loudspeaker could be equalised to deliver a flat on-axis response, and the dipole could be used to correct the power response without affecting the on-axis sound (since the on-axis direction is in the “null” of the dipole). For more details about this project, please read this post.
Figure 3: The “Shark Fin” loudspeaker – an experiment attempting to independently tune the power response without changing the on-axis frequency response. See this posting for more information.
Figure 4: The “shark-fin” loudspeaker – side view.
Figure 5: The “shark-fin” loudspeaker during the listening portion of the experiments in Listening Room 1.
After the “shark fin” experiment, we knew that we wanted to head towards building a loudspeaker with some kind of active directivity control to allow us to determine the amount of energy we sent to the nearby walls. Two members of the Acoustics Department, Gert Munch and Jakob Dyreby, had been collaborating with two graduate students (both of whom started working at B&O after they graduated), Martin Møller and Martin Olsen on exactly this idea. They (with Finn Agerkvist, a professor at DTU) published a scientific paper in 2010 called “Circular Loudspeaker Arrays with Controllable Directivity”. In this paper they showed how a barrel of 24 small loudspeaker drivers (each with its own amplifier and individualised DSP) could be used to steer a beam of sound in any direction in the horizontal plane, with a controlled beam width. (That paper can be purchased from the Audio Engineering Society from here.)
Figure 6: The results of a research project using 24 drivers placed radially symmetrically on a barrel. The top box is the DSP which feeds the bottom two boxes containing the amplifiers, one for each driver. The assembly is placed on the crane in the Cube.
The next step was to start combining these ideas (along with other, more developed technologies such as Thermal Compression Compensation and ABL) into a single loudspeaker. The first version of this was an attempt to reduce the barrel loudspeaker shown in Figure 6 to a reasonable number of loudspeaker drivers. The result is shown below in Figure 7.
Figure 7: Prototype 1. The idea for this version was a loudspeaker that could act a little like a lighthouse. The beam would be rotatable in any direction. There are 6 tweeters and 6 midranges arranged in a hexagon. The 4 woofers are arranged as a square. This version failed due to the lobing of the drivers – essentially, they are too far apart to result in an adequately-controlled beam.
This first prototype had a hexagonal arrangement of tweeters and midranges (6 of each) and a square arrangement of woofers. Each driver had its own DSP and amplification with customised filters to do the “usual” clean-up of magnitude response in addition to the beam steering much like what is described in the AES paper.
Unfortunately, this version was not a success. The basic problem when trying to do directivity control actively is that you need the loudspeaker drivers to be as close together as possible to have control of the beam width in their high-frequency band – but as far apart as possible to be able to control their low-frequency band. In the case of prototype 1, the drivers were simply too far apart to result in an acceptably constant directivity. (In other words, the beam width was different at different frequencies.) So, we had to try to get the drivers closer together.
For the second prototype (shown below in Figure 8, 9, and 10) we decided to try to forget about a steerable beam – and just focus (forgive the pun) on a narrow beam with constant directivity (the same beam width at all frequencies). In addition to this, we experimented with a prototype 8mm supertweeter that would take care of the band from about 15 kHz and up.
Figure 8: Prototype 2. This version did not have a rotatable beam, but it did have the capability of creating a very narrow beam due to the cluster of 5 midranges and 5 tweeters. Notice as well the cluster of prototype 8 mm super tweeters on the top. Also note that this version used 3 woofers instead of 4.
Figure 9: Prototype 2: side view
Figure 10: Prototype 2: Back view
Although Prototype #2 sounded great in the sweet spot, it lacked the versatility of the first prototype. In other words, it was an amazing loudspeaker for a person with one chair and no friends – but it was not really a good loudspeaker for sharing… So, we started working on a third prototype that merged the two concepts – now called “Beam Width Control” and “Beam Direction Control”. The result in shown below in Figures 11, 12, and 13.
As you can see there, the “cluster” of 3 tweeters and 3 midranges comes from prototype 2 – but we re-gained side-firing drivers and rear-firing drivers to be able to steer the sound beam in either of 4 directions. The Beam Width could only be controlled for the front-firing beam, since it is a product of the cluster. You’ll also notice that the super tweeter was still there in this prototype. However, we also changed to a different tweeter and were starting to question whether the extra 8mm driver (and its amplifier, DAC and DSP path) would be necessary.
Figure 11: Prototype 3. This version had almost as narrow a beam as the previous prototype, with the ability to steer the beam in 4 directions. The wired hanging from the front of the woofer cabinet connect to a thermal sensor on the magnet. At this point, we were still wondering whether to use an 8 mm supertweeter (on the top). It was eventually decided to not use this driver, since the Scan-Speak tweeters measure up to 40 kHz.
Figure 12: Prototype 3: Side view
Figure 13: Prototype 3: back view
One thing that any good acoustical engineer knows is that corners cause diffraction. This is well-known at B&O as you can read here. Looking at the enclosure for the midranges and tweeters in the three previous figures, you can see many flat surfaces and corners – which, we assumed, were bad. So, we set about on an informal experiment to find out what would happen if we smoothed out the corners in an effort to reduce diffraction – or at least to change it. This was initially done by applying putty to the MDF enclosures and measuring the off-axis response of the result. One example of this (in progress of bring coated with putty – this was not the final version – it’s just there to show the process) is shown below in Figure 14.
Figure 14. One of the tests for Prototype 3 was to find out whether there was a problem with diffraction due to the hard corners in the top cluster. Smoothing the shape with putty was done by hand.
Surprisingly, we found out in these tests that the “smoothing” of the structure around the midranges and tweeters either made no difference or made things worse. So, we continued on, knowing that the final result would be “smoother” anyway…
While that work was going on in the Cube, a third “stream” for the project was underway – the development of the Active Room Compensation algorithm. In the early versions, it was called “ASFC” or “Active Sound Field Control” – but as time went on and the algorithm evolved, we changed to a different system and gave it a different name. Photo 15, below, shows the listening room during one of the tests for the original algorithm. I count 9 microphones in there – but there may be more hiding somewhere.
Figure 15: One of the early Active Room Compensation experiments. Notice the many microphones distributed around the room. (I count at least 8 mic’s there…)
All of the prototypes shown above are just loudspeaker drivers in MDF enclosures. All of the DSP and amplification (in a worst-case, 17 channels in total per loudspeaker) were outside the loudspeakers. In addition, the amplifiers needed active cooling (a fancy way to say “fans”) so they had to be in a different room due to noise. The photo below shows the rack of DSP boxes (on top) and amplifiers (4 channels per box) used for the prototyping.
Figure 16. The electronics needed to drive the prototypes. The left stack is for the left loudspeaker, the right stack is identical. The top box is the DSP prototype board with 1 analogue input channel and 20 line outputs. The remaining 5 boxes contain ICEpower amplifiers – 4 channels per enclosure.
While this equipment was being used to evaluate and tune the complete prototypes, a parallel project was underway to find out whether we could customise the amplifiers to optimise their behaviour for the use. For example, if you know that an amplifier will only be used for a midrange driver, then it doesn’t need to behave the same as if it were being used for a full-range loudspeaker. I’ll describe that development procedure in a future blog posting, since it’s interesting enough to deserve its own story.
Finally, we were at a point where we built a first prototype of the “real thing”. This was hand-built using 3D-printed parts and a lot of time and effort by a lot of people. The first example of this stage is shown below on the crane in the Cube, sitting next to Prototype #4 for comparison. Notice that, by now, we had decided that the supertweeter was unnecessary, since the Scan-Speak tweeter we’re using was reliable up to at least 40 kHz. The only significant difference between the 4th prototype and the mechanical sample is that the wooden version has only one tweeter and one midrange pointing directly backwards. The “real thing” has two, aimed slightly towards the Left Back and Right Back. (See the Technical Sound Guide for more detailed information about this.)
Figure 17 Prototype 4 (on the floor) was basically the same as Prototype 3, but with a different woofer arrangement. Instead of 4 matched 13″ drivers, this version used three 10″ drivers and one 13″ in the front. On the crane is the first sample with the original mechanical design.
Once the measurement of the first mechanical samples were done and the correct filters programmed into it, it was time to move a pair into the listening room to see (or, more importantly, to hear) if they performed the same as the wooden prototypes. The first setup of device numbers 2 and 3 (the first one stayed with the electronics team for testing) in Listening Room 1 in Struer is shown below in Figure 18. For reference, the room is 6 m deep x 5 m wide – and that’s a 55″ BeoVision 11 on the wall.
Figure 18: Samples #2 and 3 in the listening room (they’re still there today, since this is the “master reference” pair). These have internal electronics and did not require the external DSP or amplification.
When we did the measurements on the samples shown in Figure 18 – both in the Cube and in the listening room, we could see that there was an unusual (and unexpected) dip in the on-axis magnitude response of about 1 dB at around 8o0 Hz. Unfortunately, it did’t seem to be easily correctable using filtering in the DSP, which meant that it was probably the result of a reflection somewhere off the loudspeaker, cancelling the direct sound at the listening position. After a day or two of playing with putty placed in various locations around the loudspeaker, we found that the problem was caused by a reflection off the “shelf” just below the face of the top unit. That can be seen in Figure 19, below.
Figure 19: The “shelf” below the midranges caused a reflection at the listening position from the top midrange. The result was an approximately 1 dB dip at about 800 Hz. This was redesigned, resulting in a new shape, shown below.
The way to correct this problem was to bring the height of the shelf up, which also meant that it was closer to the face of the top cluster. (Note that the front panel is missing in Figures 19 and 20 – the actual face is the pink panel seen in Figure 22.) This fixed the problem, but it meant changing the mould for the aluminium enclosure. In the meantime, while that change was happening, we were able to 3D-print an insert of the same shape that could be used for the listening reference pair of loudspeakers. This meant that we didn’t have to wait for the new aluminium versions to start tuning.
Figure 20: The repaired “shelf”. The pink 3D-printed portion raised the shelf closer to the midranges and is almost flush with the faceplate that is mounted in front of the loudspeaker cluster. The bad paint job on the top loudspeaker enclosure is a sealant. The enclosure is 3D-printed plastic which (we found out) is not airtight, to it had to be sealed to eliminate leaks.
Of course, the electronics team developed their components on a test bench, piece by piece. Eventually, all of those pieces came together into a single unit (minus the loudspeaker drivers and enclosures) which could be used for testing and software development. An example of one of those test boards (actually, the first one of its kind to be made – and one of the few with all of the amplifiers attached…) is shown below in Figure 21.
I’ll probably show some better photos of the DSP board in a later posting.
Figure 21: The entire electronics assembly for the BeoLab 90. The top portion in the brown MDF box is the power supply. Moving down the photo, the 4 PCB’s are the woofer amplifiers. Next are the input and DSP boards. On the bottom are the 14 ICEpower amplifiers. The circle not he right is the light ring comprised of 72 LED’s.
Finally, everything came together into a product that, acoustically and electrically, was identical to the production model. This is the version that we use for sound design. It’s shown (about to go out into the Cube for yet another round of measurements) in Figures 22 and 23, below.
Figure 22: The early sample, on the crane in the cube, undergoing measurements to ensure that it is ready to start the sound design process.
B&O received a question on one of its social media sites this week, and I was asked to write up an answer. The question was:
Hi Bang & Olufsen
I just wanna be sure of a myth that’s been going around my audio community recently. The myth is that condenser microphones are more prone to produce feedback than dynamic microphones as a result of higher sensitivity in (and reproduction of) the treble.
Is this true of false? Thanks.
The short answer
This is false.
The long answer
Feedback happens when you have a system where the input to a microphone is amplified and sent to the output of a loudspeaker, AND the output of the loudspeaker is received at the microphone at a level loud enough to cause the signal to get louder (instead of quieter) each time it circulates through the system. The result is a “howling” or “squealing” sound from the loudspeaker. This effect will happen first at whatever frequency has the highest gain (amplification) in the system.
That frequency could be due to a peak in the magnitude response of the microphone or the loudspeaker, or some acoustical effect of the room (such as a room mode), or something else. (For example, if you put your hand over the microphone diaphragm, making a resonant cavity, you could result in a peak in the total system’s magnitude response that would not be there if you moved your hand away.)
So, the basic problem is one of signal gain. The higher the gain (or “amplification”) of the signal, the more likely you are to have feedback. The question is: what determines this total loop gain in a typical sound reinforcement system?
the sensitivity of the microphone
This is frequency-dependent, since the magnitude response of the microphone is likely not perfectly flat.
It is also spatially dependent. If you have an omnidirectional microphone and a cardioid microphone that have the same sensitivity on-axis (in front of the microphone), they will be very different behind the microphone. Also note that this directional pattern is also frequency-dependent. A cardioid is not a cardioid at all frequencies…
the gain of the microphone pre-amplifier
the additional gain applied after the microphone preamplifier
this may be frequency-dependent, like an EQ applied to the microphone signal, or an EQ applied to the entire mix sent to the loudspeaker amplifiers
the gain of the loudspeaker amplifier(s)
the sensitivity of the loudspeaker(s)
the distance between the loudspeaker(s) and the microphone
the radiation pattern of the loudspeaker(s)
many loudspeakers are directional, so they’re louder in front than behind
this is also frequency-dependent. Bass is usually omnidirectional, high frequencies are usually directional
the orientation of the microphone relative to the loudspeakers (i.e. is the loudspeaker in front of, or to the rear of the microphone), especially if the microphone is directional (like a cardioid or a hypercardioid pattern)
the coupling to room modes due to
the strength of the modes themselves (a function of the room construction and its materials)
the location of the loudspeaker(s)
the location of the microphone
There may be some other things – but that’s certainly enough to worry about.
IF you have two microphones, one is a dynamic microphone and the other is a condenser microphone, and they both have the same polar patterns, the same magnitude responses, the same sensitivities, they’re both in the same location in the room with the same orientation to the loudspeakers, and all other components in the system are identical, THEN the risk of getting feedback with the two mic’s is identical.
IF you have two microphones with different polar patterns, different magnitude responses, different different sensitivities, etc. etc. THEN the risk of getting feedback with the two mic’s is different. Whether the basic electromechanical construction is based on a condenser or a dynamic design is not the cause of the difference.
That said, it is true that microphones (both condenser and dynamic) are built with particular uses in mind. For example, (dynamic) Shure SM58 is designed to be tolerant of noises caused by it being hand-held (the diaphragm assembly is vibration-isolated from the housing) this is not true of a (condenser) AKG 451 which is designed to be mounted on a stand and not touched while you’re using it. However, this difference is not caused by the fact that one is dynamic and the other is a condenser – it’s a result of the mechanical designs of the microphones housing the “business end” of the devices. (Note, however, that this example has nothing to do with feedback – it’s just an example of microphones being designed for different purposes.)
It is also true that many condenser microphones have a magnitude response that extends to the high frequency bands with less roll-off than many dynamic microphones (there are exceptions to this statement – but I used the word “many” twice…). And, a higher sensitivity in any frequency band will result in a greater risk of feedback. However, this increased risk is a result of the magnitude response of the microphone – not its electromechanical construction. If you have a condenser microphone with a roll-off in the high end (say, an older, large-diaphragm mic, especially off-axis) and a dynamic microphone with an extended high-frequency range (i.e. a ribbon microphone, which typically has a flatter high-frequency response than a moving-coil microphone), then the dynamic will be at higher risk of feedback.
So, like I said at the start – the myth is false. If you get feedback in your system, it’s because
the person running the system was not paying attention to the gain
the person with the microphone moved too close to a loudspeaker while the person running the system was not paying attention to the gain
Either way, it’s the fault of the person controlling the system – not the construction of the microphone. As the old saying goes: “It’s a poor craftsman that blames his tools.” Or, as a friend of mine once told a class he was teaching: “If it’s too quiet, you turn it up. If it’s too loud, you turn it down. That’s the way I remember it.”
Once-upon-a-time, I wrote a posting explaining why a loudspeaker’s biggest enemy is your listening room. The basic problem is as follows:
when you listen to a loudspeaker in a room, roughly 99% or more of what you hear is sound that goes outwards from the loudspeaker, fills up the room, bounces around and then gets to you. Only about 1% or less of what you hear actually comes directly to you from the loudspeaker…
rooms have early reflections that mess up the tone colour and the imaging of the loudspeakers
rooms have room modes that make the bass response extremely uneven, and very different from place to place in the room
rooms have reverberation that, generally, makes some frequency bands sound louder because they last longer in time
So, to borrow a phrase from South Park’s Mr. Mackey, “rooms are bad, m’kay?
For this posting, we’re going to focus only on the third issue there: that of “room modes” and a couple of ways to deal with them. To begin with, let’s talk about what a room mode is.
The short version
The PowerPoint-single-slide-of-bullet-points version of an explanation of room compensation is as follows:
A room mode is a resonant frequency where the room “sings along” with the sound of the loudspeaker, making that pitch (or note) sound louder than others.
This resonance can be measured using a microphone
To reduce the effect of this problem, the loudspeaker can either
reduce the amount of energy it emits at that particular frequency
actively absorb energy at that frequency
This means that one detrimental aspect of the room’s acoustical behaviour on the sound of the loudspeaker will be reduced
That’s it. However, of course, the real story is a little more nuanced (or maybe just more complicated) than that. So, if you’re interested in knowing a little more, read on!
Resonance
If you put a kid on a swing and push him (not too hard… we don’t want anyone to get hurt here…) and stand back, you’ll watch him swing back and forth, decreasing a little in distance from the resting position each time, until eventually, he’ll come to a stop. Now, do it again, and pay attention to his movement. He moves forwards, away from the resting position until he reaches a high point, then stops, comes backwards past the resting position to a high point, stops, and then moves forwards again. We’ll call that total return trip from going forwards at the resting position to being back again, going forwards at the resting position, one cycle. The number of times the kid does this for each period of time (let’s say, in this case, each minute) is called the frequency – how frequently he’s repeating the motion. The maximum distance he moves from the resting position (to the highest point in the cycle) is called the amplitude.
Now, if you pay really close attention to the movement of the kid on the swing, you might notice that, even though the amplitude of the cycle decreases over time, the frequency doesn’t change.
Another example of this kind of motion is shown in the video in this posting. However, that video is not for the faint-of-heart. Watch it at your own risk…
Okay, if you watched that video and you’re still awake, let’s move on.
There are a couple of important concepts to glean from this discussion so far.
If you have a system that can resonate (like a spring and a mass, or a kid on a swing), it will “want to” resonate at a fundamental frequency.
You can trigger that system to resonate at its fundamental frequency by injecting energy into the system (like lifting the mass, or pushing the kid)
Another example of a system that “wants” to resonate is a string that’s fixed on both ends – like a guitar string. It has a fundamental frequency that is determined by the string’s mass and tension. The cool thing about a string is that it also resonates at multiples of that fundamental frequency (better known as harmonics or overtones – which almost mean the same thing, but not quite – but the difference is irrelevant here). So, if you have a guitar string that’s tuned to 100 Hz (an abbreviation for “Hertz” which is the word we use to mean “cycles per second”) then it will also resonate at harmonics with frequencies of 200 Hz, 300 Hz, 400 Hz, and so on. If you inject energy into the string (by plucking it with your finger, for example), then the string’s total vibration will include resonances at all those frequencies, on up to ∞ Hz (in theory… not in practice…). You’ll have a resonance at each individual harmonic, each with its own amplitude (how much the string is vibrating or moving away from its resting position – the higher the amplitude, the louder it is) and the total sum of all of these results in the shape (and sound) of the string.
At this point, you’re probably wondering “what does this have to do with my room?” We’re getting there… I promise.
One last example of a system that resonates is the air in a pipe (like an organ pipe, for example). If you could shrink yourself down to the size of a molecule and get inside an organ pipe, you’d see that you’re looking down a long tube that’s capped at both ends (we won’t talk about the other kind of pipe that’s open on one end… let’s pretend those don’t exist…). If you face one end of the pipe and push the air molecule next to you towards it, it will push the one in front of it, which will push the one in front of it, and so on, until the shoving match ends at the cap at the end of the pipe. (See this page for a good animation of the story so far…) That last molecule can’t push the “wall” so it bounces back, which winds up in a return shoving match (or pressure wave…) that will eventually push against you, and you’ll push the molecule behind you, which keeps repeating until the wavefront gets to the cap at the opposite end of the pipe, which reflects, and sends wavefront back to you again.
Now, what happens if, while all of that is happening, you are pushing repeatedly? Every time the wave bounces back at you, you push it again in the same direction that the particles want to move in (this is exactly the same as pushing a kid on a swing at exactly the right time to make him swing higher and higher – you time the push so that he’s moving in the same direction as you’re pushing).
Take a look at the top part of the animation below.
Let’s say that you’re the red molecule and you’re in a pipe. You push and pull the two adjacent particles (one in front, one behind) exactly in sync with the wave that’s reflecting off the two ends of the pipe, and you’re helping to inject energy into the fundamental resonance of the air column inside the pipe. The result is something called a “standing wave”. it’s called this because it looks like the wave is “standing still” and just changing in instantaneous amplitude over time, but the reality is that this is just an illusion. What is actually happening is that the right-going wave is aligned with the left-going wave to perfectly cancel each other out at all times in the middle of the pipe, and to result in big changes in pressure at the ends of the pipe. Check out the third animation on this page for a good explanation of this summing or “superposition” of waves resulting in a standing wave.
Just like a string, a pipe will also resonate at harmonics of the fundamental, in multiples of the frequency of the lowest resonance. So, if the pipe resonates at 150 Hz, then you will also have resonances at 300 Hz, 450 Hz, and so on… I’ll show some animations with examples of these later in the posting.
Room Modes
The nice thing (at least it’s nice in that it helps our understanding – it’s not actually nice in real life…) is that a room behaves just like a very big, very wide, organ pipe. So, for example, when I sit in a listening room that is 5 m wide by 6 m long, some of the resonances that I hear “singing along” with every sound source in the room are exactly the same frequencies as the ones that would come out of organ pipes 5 m long and 6 m long.
Take a look at that animation above once again. You can think of the red dot as the loudspeaker, pushing and pulling the adjacent air particles in time. You sit at the listening position at the black dot – the adjacent air particles push and pull your eardrum in and out of your head. The pressure of the sound wave (the only thing we’re going to worry about in this posting) can be thought of as a measure of how close two adjacent air particles are (shown by the three vertical lines in the top part, and represented by the entire bottom part of the animation).
So, as the loudspeaker woofer (for example) moves in and out of the enclosure on a kick drum hit, it injects energy into the room at many different frequencies. If the frequency of the resonance in the room is one of the frequencies in the woofer’s signal, then that note will sound much louder than the other notes, making for a very uneven bass. This is because the room “wants” to resonate (or “sing along”) at that particular frequency, so a little energy coming into it will give a large result (just like a series of small, but well-timed, pushes of a kid on a swing can build up over time to result in the kid moving back and forth much more than you’re pushing).
This will happen not only at the fundamental frequency of the resonance, but its harmonics. The second harmonic is shown below.
The third harmonic looks like the animation below.
And the fourth harmonic looks like the animation below.
We won’t go any higher than this.
There are some things to notice in these animations.
The air particles at the walls on either end of the room experience the maximum pressure change. This is because one of the particles cannot push the wall, so it receives the full effect of the pressure.
There are places in the room where the pressure does not change over time. For example, in the case of Mode #1 in the first animation, the point at the exact middle of the room doesn’t have a change in pressure. This doesn’t mean that the air particles are not moving – they are. However, the ones in the middle of the room are staying the same distance from each other – so the pressure doesn’t change over time.
If you are sitting at a point in the room where there is no change in pressure caused by the mode (say, for example, you’re the black dot in Mode #2, above), then you will not be able to hear the room mode. It’s still happening around you – but at your exact position, since there is no change in pressure over time, it doesn’t exist. However, this doesn’t mean that you won’t hear anything at that particular frequency – you will – it’s just that the sound you hear (from the loudspeaker) doesn’t have the room mode singing along with it at your location.
Similarly, if the loudspeaker is placed at a location where there is no change in pressure when the room mode is ringing (for example, the location of the red circle in Mode #3), then the loudspeaker will not trigger this mode to ring. Energy will come out of the loudspeaker at that frequency, and you’ll be able to hear it – but the room mode will not “sing along” with the signal, since the loudspeaker can’t inject energy into it. (Note that this is statement is based on at least two assumptions: (1) that the loudspeaker is a pressure source, and (2) that it is infinitely small. At least one of these assumptions is incorrect.)
This means that, looking at the four animations above, and assuming that the red dot is the loudspeaker and the black dot is the listener:
We need to fix Mode #1, since the loudspeaker and the listener are “coupled” to it (meaning that they are not sitting on a point where there is no pressure change in the mode).
We do not need to fix Mode #2 since, although the loudspeaker is triggering the mode to ring, the listener cannot hear it. (However, if the loudspeaker was placed somewhere else, this mode could be a problem.)
We do not need to fix Mode #3 since the loudspeaker cannot trigger the mode, so no one hears it (although the mode will occur for other sound sources in the room). (Also, if the listening position was placed somewhere else, this mode could be a problem.)
We need to fix Mode #4.
Figure 5: A simulated, over-simplified example of the total effect on the magnitude response of two modes, the fundamental at 100 Hz, and the 4th harmonic at 400 Hz (see the above text to explain why 200 Hz and 300 Hz are missing…)
How can we fix this?
As I said in the PowerPoint version above, we have two basic strategies for dealing with room modes.
The first is a “symptom attenuation” approach: the room mode makes some frequencies louder than others, so we’ll just reduce the level of those frequencies by the same amount. If the room boosts 100 Hz by 6 dB, then we’ll put a dip of -6 dB at 100 Hz in the signal sent to the loudspeaker. This doesn’t actually correct the problem – it just covers it up. It’s a bit like taking a pain killer because you’ve broken your leg. You are no longer in pain, but the problem still exists… The trick here is to measure the response of the loudspeaker at the listening position (this is easy) and find out which frequencies appear to be boosted by the room. We then make an equalisation which is that same response turned “upside down” so a peak in the measured response becomes a dip in the loudspeaker’s processing.
As I said, this covers up the problem but it doesn’t solve it. The thing to remember is that a room mode appears to be louder because the room moves the energy in time. For example, if you pluck a guitar string, you “activate it” for a very short period of time. The resonance of the string takes that energy and extends it in time so you hear a note that goes on for seconds after the pluck. A room mode does the same thing – it “rings” in time, smearing the energy coming out of your loudspeakers (say, a kick drum hit) and extending it like a guitar string ringing. If you send less energy out of the loudspeaker at the same frequency of the room mode, then the total result at the listening position will be quieter, but it’s still ringing in time. (So, instead of plucking the guitar string and then stopping it, we’re just plucking with less force.)
The second method is a little more complicated, but works better. Since we have control of the loudspeaker, what we can do is to send the sound we want out of it – but we’ll add an extra signal that cancels the room mode by working against it. If we go back to the original description where we were a little air particle in a pipe, we send the signal out, and when it comes back, reflected off the cap, we’ll send a signal out of the loudspeaker that “sucks it back in”. This way, the loudspeaker acts as an active absorber. (This is a little like the way a person on a trampoline can stop jumping by using his/her legs to absorb the push-back from the springs. See this video at 0:53, for example.)
The great thing is that, if you’re smart and a little lucky, both of these approaches are the same thing. If we consider the room mode to have the same characteristics as a “minimum phase” peaking filter (which, in isolation, it does), then, if we can measure it correctly, we can implement its exact reciprocal. This means that the magnitude response of the filter we insert into the loudspeaker’s processing will be the exact opposite of the room’s response. In addition to this, the phase response of the filter is, at any frequency, opposite in polarity to the phase response of the room mode (in isolation). This means that both approaches are rolled into one solution.
Figure 6: The same two modes shown in Figure 5, decomposed into the two individual resonances.
Figure 7: The correction filter we would need to cancel the effects of the two modes shown in Figure 6. Note that the phase responses are polarity-inverted copies of the modes’ phase responses. If we take the signals shown in Figure 6 and process them with a filter with the characteristics shown in Figure 7, the result would be a perfectly flat signal, as if nothing had happened to it.
Of course, this is very theoretical. However, in practice the concept holds true enough that it results in a noticeable improvement in the sound of a loudspeaker in the listening position, assuming that you are careful about the measurements you make around that position. If you choose to make a room correction filter that applies to the whole room, and not just one listening position, then you should measure the room in many different and varied locations. This will give you a “general” compensation that is better, on average, for everyone in the room – although it might make one specific location (like, say, the sweet spot) worse. If, instead, you just measure one location by placing the microphone in a tight area around the sweet spot, for example, then the compensation filter will be optimised for that location, but it will probably make the rest of the room worse as a result.
Active Room Compensation gives you this option to customise the room compensation filters as they best suit your needs. If you’re the kind of person with one chair and no friends, then you measure around that chair and you never leave it while you’re listening. If you’re the kind of person who has lots of friends, or who moves around while you listen, then you create a compensation filter that fixes the entire room, on average.
Of course, in a perfect world, you would be able to make both of these measurements in advance and save each one as a preset – then you would be able to select the one that best suits the situation when the time arises… Or, you could combine them and build a filter that corrects the room, with emphasis on the sweet spot…
P.S. As I mentioned briefly in this posting, Active Room Compensation has one additional feature – multichannel processing. That will be explained later in Part 2.
