Category: Bang & Olufsen

B&O Tech: Beam Width Control – A Primer

#38 in a series of articles about the technology behind Bang & Olufsen loudspeakers

In the last posting, I talked about a little experiment we did that made us realise that control of a loudspeaker’s directivity (or more simply “beam width”) would be a useful parameter in the hands of a listener. And, in a previous article, I talked a little about why that is probably true. This week, let’s get our hands dirty and talk about how Beam Width Control can be accomplished.

Part 1: What is sound?

Okay – we’re really getting back to basics, but it won’t take long. I promise. In order to get to where we’re headed, we have to have a fairly intuitive grasp of what sound is. At a basic level, sound is a change in air pressure over time. Today’s atmospheric (or barometric) pressure is about 100 kiloPascals (give or take). That pressure can be thought of as a measure of the average distances between the air molecules in the room you’re sitting in. It is also important to note that (unless you’re very unlucky and very uncomfortable) the pressure inside your head is roughly the same as the pressure outside your head (otherwise, you’d probably be talking about going to visit a doctor…). The important thing about this is that this means that the pressure that is exerted on the outside of your eardrum (by the air molecules sitting next to it in your ear canal) is almost identical to the pressure that it exerted on the inside of your eardrum (the part that’s inside your head). Since the two sides are being pressed with the same force, then the eardrum stays where it should.

However, if we momentarily change the air pressure outside your head, then you eardrum will be pushed inwards (if the pressure is higher on the outside of your head) or pulled outwards (if the pressure is lower). This small, momentary change in pressure can be caused by many things, but one of those things is, for example, a woofer getting pushed out of its cabinet (thus making a “high pressure” where the air molecules are squeezed closer together than normal) or pulled into its cabinet (thus making a “low pressure” and pulling the air molecules apart).

So, the woofer pushes out of the cabinet, squeezes the air molecules together which push your eardrum into your head which does a bunch of things that result in you hearing a sound like a kick drum.

Part 2: Frequency and wavelength

Take a loudspeaker into your back yard and play a tone with the same pitch as a “Concert A” (the note you use to tune a violin – or, at least, the note violinists use to start tuning a violin…). This note has a frequency of 440 Hz, which means that your loudspeaker will be moving from its resting position, outwards, back, inwards, and back to the resting position (a full cycle of the tone) 440 times every second. This also means that your eardrum is getting pushed inwards and and pulled outwards 440 times every second.

Now, turn off the loudspeaker and walk 344 m away. When you get there, turn on the tone again. You’ll notice that you don’t hear it right away because sound travels pretty slowly in air. In fact, it will take almost exactly one second from the time you turn on the loudspeaker to the time the sound reaches you, since the speed of sound is 344 m/s (you’re 344 m away, so it takes one second for the sound to get to you).

If, at the instant you start to hear the tone, you were able to freeze time – or at least freeze the air molecules between you and the loudspeaker, you’d see something interesting (at least, I think it’s interesting). The instant you hear the start of the tone (the first cycle), the 441st cycle is starting to be generated by the loudspeaker (because it’s producing 440 cycles per second, and it’s been one second since it started playing the tone). This means that, in your frozen-time-world, there are 440 high and low pressure zones “floating” in the 344 m of air between you and the loudspeaker. Given this information, we can calculate how long the length of one wave is (better known as the “wavelength” of the tone). It’s 344 m divided by 440 cycles per second = 78.2 cm.

So, this means that there are 78.2 cm between each high pressure “bump” in the air. This also means that there is half that distance (39.1 cm) between adjacent high and low pressure zones in the air.

Figure 1: High pressure (red) and low pressure (blue) zones radiating outwards from a sound source (the black dot)
Figure 1: High pressure (red) and low pressure (blue) zones radiating outwards from a sound source (the black dot). Note that the scale goes from -1 to 1.

 

Part 3: Interference

Now let’s take a case where we have two woofers. We’ll put them side by side and we’ll send the same 440 Hz tone to them. So, when one woofer goes out, the other one does as well. This means that each one creates a high pressure in front of it. If you’re exactly the same distance from the two woofers, then those two high pressures add together and you get an even higher pressure at the listening position (assuming that there are no reflecting surfaces nearby). This is called constructive interference. This happens when you have two loudspeaker drivers that are either the same distance from the listening position as in Figure 2 or in exactly the same location (which is not possible), as in Figure 3.

 

 

Figure 2: Two sound sources playing the same signal located at the same distance from the listening position.
Figure 2: Two sound sources playing the same signal located at the same distance from the listening position.

 

Figure 2: Two sound sources located in the same position (this is not possible). Note that the pattern of radiation is the same as in Figure 1, it's just twice as much pressure.
Figure 3: Two sound sources located in the same position (this is not possible). Note that the pattern of radiation is the same as in Figure 1, it’s just twice as much pressure – the scale now goes from -2 to 2 because the pressures from the two sources add perfectly in all locations.

 

If we change the wiring to the woofers so that they are in opposite polarity, then something different happens. Now, when one woofer goes outwards, the other goes inwards by the same amount. This means that one creates a high pressure in front of it while the other creates a low pressure of the same magnitude. If you’re exactly the same distance from the two woofers, then those two pressures (one high and one low) arrive at the same time and add together, so you get a perfect cancellation at the listening position (assuming that there are no reflecting surfaces nearby). This is called destructive interference. This is shown in Figure 4 and 5.

 

igure 2: Two sound sources playing the same signal located at the same distance from the listening position.
Figure 4: Two sound sources playing the same signal in opposite polarity located at the same distance from the listening position.

 

 

Figure 4. Two sound sources playing the same signal in opposite polarity. Notice that they cancel each other "on axis" to the pair of sources (to the right of the black dots in the figure).
Figure 4. Two sound sources playing the same signal in opposite polarity. Notice that they cancel each other “on axis” to the pair of sources (to the right of the black dots in the figure). The cancellation is seen as a greenish-colour which corresponds to a maximum pressure of 0, or no sound…

 

However, an important thing about what I just said was that you are the same distance from the two woofers. What if you’re not? Let’s say, for example, that you’ve changed your angle to the “front” of the loudspeaker (which has two loudspeaker drivers). This means that you’re closer to one driver than the other. The result is that, although the two drivers are playing the same signal at the same time, your difference in distance causes their individual signals to have different phases at the listening position – possibly even a completely opposite polarity – as is shown in Figure 5.

 

igure 2: Two sound sources playing the same signal located at the same distance from the listening position.
Figure 5: Two sound sources playing the same signal located at different distances from the listening position. Notice that, although the signals leave the drivers in the same polarity, they are opposite in polarity at the listening position because the difference in distance results in a difference in time of arrival at the listening position. The more distant driver is farther away by one half of a wavelength in this case.

 

If you consider Figure 2 and Figure 5 together (they’re really two different views of the same situation, since both loudspeakers are playing the same signal simultaneously) and you include all other listening positions, then you get Figure 6. As you can see there, there is no sound (again, indicated by the greenish colour which means no change in pressure) at the angle shown in Figure 5.

Figure 3: Two sound sources separated by a distance of one wavelength of the tone they are producing. Note that the two sources add perfectly at all distances directly "in front" (to the right, in the figure) because any position on that line is the same distance from each source.
Figure 6: Two sound sources separated by a distance of one wavelength of the tone they are producing. Note that the two sources add perfectly at all distances directly “in front” (to the right, in the figure) because any position on that line is the same distance from each source.

 

So, Figure 6 shows that, for a position directly in front of the sound sources (to the right of the black dots, in the figure), the result is identical to that in Figure 3 – the two sources add together perfectly if you’re the same distance from them. However, if you start to move away from that line, so that you’re closer to one sound source than the other, then, at some angle, you will get a high pressure from the upper sound source at the same time as you get the previous low pressure from the lower source (because it’s farther away from you, and therefore a bit late in time…) This high+low = nothing, and you hear no sound.

 

Another, slightly more nuanced way to do this is to not just change the polarity of one of the sound sources, but to alter its phase instead. This is a little like applying a delay to the signal sent to the driver, but it’s a delay that is specific to the frequency that we’re considering. An example of this is shown in Figure 7.

Figure 5. Two sound sources playing the same signal with slightly different phases (delay). Notice that they cancel each other in a different direction than the previous cases.
Figure 7. Two sound sources playing the same signal with slightly different phases (another word for “delay”, sort of…). Notice that they cancel each other in a different direction than the previous cases.

This means that, by changing the phase instead of the polarity of the drivers, I can “steer” the direction of the beam and the “null” (the area where the sound is cancelled out)

It should be noted that, at a different separation of the drivers (more accurately stated: at a different relationship between the wavelength of the signal and the distance between the drivers) , the behaviour would be different. For example, look at Figure 8 which shows the same situation as Figure 6 – the only difference is that the two sound sources are half the distance apart (but the wavelength of the signal has remained the same).

Figure 6: Two sound sources separated by a distance of one wavelength of the tone they are producing. Note that the two sources add perfectly at all distances directly "in front" (to the right, in the figure) because any position on that line is the same distance from each source.
Figure 8: Two sound sources separated by a distance of one half wavelength of the tone they are producing. Compare this to Figure 6.

Part 4: Putting it together

So, at a very basic level (so far) we can determine which direction sound is radiated in based on a relationship between

  1. the wavelength of the signal it is producing (which is a function of its frequency)
  2. the distance between the sound sources (the loudspeaker drivers), and
  3. the polarity and/or the phase of the signals sent to the drivers.

All of the examples shown above assume that the two sound sources (the loudspeaker drivers) are playing at the same level, which is not necessarily the case – we can choose to play a given frequency at whatever level we want using a rather simple filter. So, by reducing the level of the “interfering” driver, we can choose how much the directivity of the radiated sound is affected.

In addition to this, all of the examples shown above assume that the sound sources are omnidirectional – that they radiate sound equally at all frequencies – which is not true in real life. In real life, a loudspeaker driver (like a tweeter or a woofer) has some natural directivity pattern. This pattern is different at different frequencies and is influenced by the physical shapes of the driver and the enclosure it’s mounted on.

So, let’s start putting this together.

If I take a loudspeaker driver – say, a midrange unit – and I mount it on the front of a loudspeaker enclosure and measure its directivity pattern  (the geeky term for the width of its sound beam) across its frequency range and beyond I’ll see something like this:

Figure 9: The horizontal directivity (or "beam width") of a midrange driver mounted on the front of a loudspeaker enclosure.
Figure 9: The horizontal directivity (or “beam width”) of a midrange driver mounted on the front of a loudspeaker enclosure. We don’t care about its behaviour in the very low end very high end (outside the red lines), since those frequency bands are taken care of by woofers and tweeters.

As you can see in that plot, in its low frequency range (the red line on the left), the midrange driver is radiating sound with a wide beam – not completely omnidirectional, but very wide. In its high frequency region (around the right-hand red line) (but still not high enough to be re-routed to the tweeter) the midrange is “beaming” – meaning that the beam is getting too narrow. Our goal is to somehow make the beam widths at these frequency bands (and the ones in between) more alike. So, we want to reduce the width of the beam in the low frequencies and increase the width of the beam in the high frequencies. How can we do this? We’ll use more midrange drivers!

Let’s take two more midrange drivers (one on either side of the “front” or “main” one) and use those to control the beam width. In the lower frequencies, we’ll send a signal to the two side drivers that cancel the energy radiated to the sides of the loudspeaker – this reduces the width of the beam compared to using the front midrange by itself. At higher frequencies, we’ll send a signal to the two side drivers that add to the signal from the front driver to make the width of the beam wider. At a frequency in the middle, we don’t have to do anything, since the width of the beam from the front driver is okay by itself, so at that frequency, we don’t send anything to the adjacent drivers.

“Okay”, I hear you cry, “that’s fine for the beam width looking from the point of view of the front driver, but what happens as I come around towards the rear of the loudspeaker?” Excellent question! As you come around the rear of the loudspeaker, you won’t get much contribution from the front midrange, so the closest “side” midrange driver is the “main” driver in that direction. And, as we saw in the previous paragraph, the signal coming out of that driver is pretty strange (because we intentionally made it strange using the filters – it’s in opposite polarity in its low end, it doesn’t produce anything in its midrange, and it’s quiet, but has a “correct” polarity in its high end). So, we’ll need to put in more midrange drivers to compensate again and clean up the signal heading towards the rear of the loudspeaker. (What we’re doing here is basically the same as we did before, we’re using the “rear” drivers to ensure that all frequencies heading in one rearward direction are equally loud – so we make the “rear” drivers do whatever they have to do at whatever frequency bands they have to do it in to make this true.)

“Okay”, I hear you cry, “that’s fine for the beam width looking from the point of view of the rear of the loudspeaker, but what happens as we go back towards the front of the loudspeaker? Won’t the signals from the rear-facing drivers screw up the signal going forwards?” Excellent question! The answer is “yes”. So, this means that after the signal from the rear-facing midrange drivers is applied (which compensate for the side-facing midrange drivers which compensate for the front facing midrange driver) then we have to go back to the front and find out the effects of the compensation of the compensation on the “original” signal and compensate for that – so we start in a loop where we are compensating for the compensation for the for the compensation for the compensation for the… Well, you get the idea…

The total result is a careful balancing act where some things are known or given:

  • the frequency range of the music being played through a given loudspeaker driver (this is limited by the crossover)
  • the natural directivity patterns of the loudspeaker drivers on the loudspeaker enclosure within that frequency range
  • the locations of the loudspeaker drivers in three-dimensional space
  • the orientations of the loudspeaker drivers in three-dimensional space (in other words, which way they’re pointing)

Note that these last two have been calculated and optimised based on a combination of the natural directivity patterns of the loudspeaker drivers and the desired beam widths we want to make available.

As a result each individual loudspeaker driver gets its own customised filter that controls

  • the level of the signal at any given frequency
  • the phase (which includes delay and polarity – sort of…) of the signal at any given frequency

 

Figure X: A conceptual illustration showing the different relationships between the various loudspeaker drivers for two different frequencies.
Figure 10: A conceptual illustration showing the different relationships between the various (in this illustration, seven) loudspeaker drivers for two different frequencies. The sum of all loudspeaker drivers results in the same beam width and the same output level on-axis at both frequencies.

By controlling the individual output levels and phases of each loudspeaker driver at each frequency it produces, we can change the overall level of the combined signals from all the loudspeaker drivers in a given direction. If we want to be loud at 0º (on-axis) and 20 dB quieter at 90º (to the side), we just apply the filters to all of the drivers to make that happen. If we want the loudspeaker to be only 10 dB down at 90º, then this just means changing to a different set of filters. This can be done independently at different frequencies – with the end goal to make all frequencies behave more similarly, as I talked about in this posting and this posting.

Also, since the filters are merely settings of the DSP (the digital signal processing “brain” of the loudspeaker), we can change the beam width of the loudspeaker merely by loading different filters – one for each loudspeaker driver in the system.

The end result is a loudspeaker that can play many different roles, as was shown in the different plots in this posting. In one mode, the beam width can be set to “narrow” for situations where you have one chair, and no friends and you want the ultimate “sweet spot” experience for an hour or two. In another mode, the beam width can be set to “wide”, resulting in a directivity that is very much like an improved version of the wide dispersion of the BeoLab 5. In yet another mode, the beam width can be set to “omni”, sending sound in all directions, making a kind of “party mode”.

 

Part 5: The “side” effect: Beam Direction Control

In order to be able to have a Beam Width Control, a loudspeaker must have a number of identical loudspeaker drivers. A collection of tweeters, midranges and woofers, some on the front, some on the sides and some to the rear of the loudspeaker. In addition, each of these drivers must have its own amplifier and DSP channel so that it is independently controllable by the system.

This means that, instead of using the drivers to merely control the width of the beam in the front of the loudspeaker, they can also be used somewhat differently. We could, instead, decide to “rotate” the sound, so that the “main” tweeter and midrange are the ones on the side (or even the rear) of the loudspeaker instead of the ones on the front. This means that, if you’re sitting in a location that is not directly in front of the loudspeakers (say, to the sides, or even behind), it could be possible to improve the sound quality by “rotating” the “on-axis” direction of the loudspeaker towards your listening position. This by-product of Beam Width Control is called Beam Direction Control.

P.S. Apologies for the pun…

 

 

For more information on Beam Width Control:

What is “Beam Width Control”?

Shark Fins and the birth of Beam Width Control

B&O Tech: Shark Fins and the birth of Beam Width Control

#37 in a series of articles about the technology behind Bang & Olufsen loudspeakers

In the last posting, I described a little experiment that we did where we could easily be accused of asking a stupid question and getting an obvious answer. This posting is a little different, although I’ll also describe a little experiment where we asked a much-less-stupid question and got an answer we weren’t looking for… However, before I describe the experiment itself, we’ll have to start with a little background information.

Let’s say that we wanted to build a “simple” two-way loudspeaker with a tweeter and a woofer in a box like the one shown in Figure 1.

Figure 1. A simple two-way loudspeaker
Figure 1. A simple two-way loudspeaker.

Since we’re Bang & Olufsen, it is implied that this will be a fully active, DSP-based loudspeaker. Let’s also, for the sake of simplicity, say that we used loudspeaker drivers and other components with absolutely no linear distortion artefacts (I can dream, can’t I?). This means that we can apply any kind of filter we like to each of the two loudspeaker drivers to result in whatever response we want (in a given direction in a “free field” (a space with no reflecting surfaces, extending to infinity in all directions) – for more information on this, please read this posting and/or this posting).

Now, it’s fairly reasonable to assume that one portion of a good starting point for a loudspeaker’s acoustic response is to have a flat magnitude response when measured on-axis in a free field. This means that, if you have the loudspeaker in a space that has no reflections, and you are directly in front of it, no frequency will be louder than any other (within some limits, of course…). In fact, this is the way a lot of studio monitor loudspeakers are tuned (which makes sense, since many studio control rooms don’t have a lot of reflections as was discussed here).

The problem is that, if you do make a loudspeaker that is flat-on-axis, you’ll probably have problems in its power response (for a description of what a loudspeaker’s “power response” is, see here). This is dependent on a lot of things like the crossover frequency, the sizes and shapes of the drivers, the phase responses of the crossover filters, the shape of the cabinet, and other things. However, if we were to simplify, we could say that a two-way loudspeaker (say, with a 4th order Linkwitz-Riley crossover, just to pick one type…) that is flat on-axis, will have a dip in its power response at the crossover frequency. This is because, although the two drivers (the tweeter and the woofer) add together nicely in one location (on-axis, directly in front of the loudspeaker) they do not add together nicely anywhere else (because the distances to the two drivers from the listening position are probably not matched, particularly when you go vertical…).

So, one basic problem with building a “simple” two-way loudspeaker is that you have to choose to either have a flat magnitude response on-axis, or a smooth power response (without a dip) – but you can’t have both. (If you want to really dig into this, I’d recommend starting with J. Vanderkooy & S.P. Lipshitz, “Power Response of Loudspeakers with Noncoincident Drivers — The Influence of Crossover Design,” J. Audio Eng. Soc., Vol. 34, No. 4, pp. 236-244 (Apr. 1986).)

This basic problem raised a question in the head of one of our acoustical engineers, Gert Munch. He started to wonder how we could build a loudspeaker that could have a flat magnitude response on-axis and still have a smooth power response that didn’t suffer from a dip at the crossover. One possible solution is to build a loudspeaker with the desired on-axis response, and then somehow create an additional portion of the loudspeaker that could “fill up” the dip in the power response by sending energy into the room without it affecting the on-axis response.

One possible way to do this is to use an extra loudspeaker with a “dipole” characteristic – a two-sided loudspeaker where the same signal is sent to both sides, but in opposite polarity. This means that, when one side of the loudspeaker produces a high pressure, the other side produces a low. When you sit on one side of the loudspeaker or the other, then you hear a signal. However, if you sit “on edge” to the loudspeaker, you hear nothing, since the two sides of the loudspeaker cancel each other out (in theory).

So, Gert’s idea was to add a two-way dipole loudspeaker on the top of a normal two-way loudspeaker and to just use the dipole (which became known as a “shark fin” – shown in Figure 2)  to add energy around the crossover frequency of the “normal” loudspeaker. Since the edge of the dipole was facing forwards, the theory was that none of its sound would have an effect on the on-axis response of the two-way loudspeaker below it.

Figure 2. A two-way loudspeaker with a side-facing dipole loudspeaker on top.
Figure 2. A two-way loudspeaker with a side-facing dipole loudspeaker on top. Note that the other side of the dipole looks identical to the one you see.

 

So, the question to answer was:

“Which sounds better:

  • a loudspeaker tuned to be flat on-axis, but with no correction for its power response
  • a loudspeaker that was tuned to have a smooth power response (in other words, with a “bump” around the crossover)
  • a loudspeaker that was tuned to be flat on-axis and had a power response correction provided by the dipole

?”

Since each of the 6 loudspeaker drivers in our model loudspeaker was independently controllable, this was an easy comparison to do on-the-fly, so we tried it out.

Also, just to be certain, we tried two different orientations of the dipole, both of which had the “null” – the edge of the loudspeaker – facing the listening position. The first orientation is shown in Figures 2 and 3. The second orientation is shown in Figure 4.

 

Figure 3. The dipoles in the first orientation.
Figure 3. The dipoles in the first orientation.

 

Figure 3. The dipoles in a second orientation.
Figure 4. The dipoles in a second orientation.

 

So, a bunch of us sat in the listening room for a week or so, listening to the various possible tunings of the loudspeakers, in various positions in the room, with various types of music. We each had an opinion about the whether we liked one configuration more or less than another. However, one thing that was noticeable was that there was an effect not only on the timbral impression of the loudspeakers (the tone colour) – but also on the spatial presentation of the pair. As we switched between the different tunings, phantom images changed in width and distance, and envelopment (when it existed in the recording) also changed.

This was particularly noticeable when the loudspeakers were closer to a reflecting surface, as is shown in Figure 5.

Figure 5. A listening configuration where one loudspeaker was intentionally placed close to a reflecting surface.
Figure 5. A listening configuration where one loudspeaker was intentionally placed close to a reflecting surface.

 

The end result was a bunch of different conclusions:

  1. The three different configurations of the loudspeaker (listed above) sounded different. (this was not a big surprise)
  2. A two-way loudspeaker tuned to have a smooth power response and a loudspeaker with a flat on-axis response with a shark fin smoothing the power response sounded different – but mostly spatially.
  3. Switching between those two loudspeaker configurations listed in Point #2 basically meant that we were switching between two different loudspeakers with the same power response, but different directivities (a fancy word meaning “directional behaviours”).
  4. The dipole loudspeaker’s output was audible at the listening position because of the sidewall reflections in the case where the dipole was on top.
  5. Switching the dipole on and off had a noticeable effect on the phantom image distance and width… When the dipole was turned on, the sidewall reflections were obviously louder, and the phantom images moved to a distance from the listening position roughly the same as the distance to the loudspeakers themselves. At the same time, the phantom image width increased and became “cloudy” or “fuzzy” and less precise.

 

All of those little conclusions could be folded into two big ones:

  1. The original idea of using a dipole to fill in the power response of the loudspeaker doesn’t work well enough in all situations because we don’t know where the reflective surfaces around the loudspeaker will be in any given configuration.
  2. Being able to control the directivity of the loudspeaker (specifically, how much energy is coming from the sidewall reflections) is a very interesting control parameter that we should look into…

 

So, this is where the concept of the shark fin died, but where the idea of “Beam Width Control” began.

Well… more accurately: it was the beginning of our thinking that being able to change the beam width (or the “directivity”, if you’re a geek) of the loudspeaker would be a useful “handle” in the hands of our customers. The idea that was born was something along the lines of the following: “If a customer could change the beam width of a loudspeaker, then (s)he could change it from being a narrow-directivity loudspeaker for people with one chair and no friends to a wide-directivity loudspeaker for people with a sofa and some friends to an omni-directivity loudspeaker for people throwing a party – all just by the flick of a switch…”

 

For more information on Beam Width Control:

What is “Beam Width Control”?

Beam Width Control – A Primer

B&O Tech: It’s lonely at the top

#36 in a series of articles about the technology behind Bang & Olufsen loudspeakers

 

“In all affairs it’s a healthy thing now and then to hang a question mark on the things you have long taken for granted.”

-Bertrand Russell

 

If you were to get your hands on Harry Potter’s invisibility cloak and you were to walk into the acoustics department at Bang & Olufsen and eavesdrop on conversations, you’d sometimes be amazed at some of the questions we ask each other. That’s particularly true when we’re working on a new concept, because, if the concept is new, then it’s also new for us. The good thing is that all of my colleagues (and I) are ready and willing at any time to ask what might be considered to be a stupid (or at least a very basic) question.

One of those questions that we asked each other recently seemed like a basic one – why do we always put the tweeter on the top? It seems like there are very few loudspeakers that don’t do this (of course, there are exceptions – take the BeoLab Penta, for example, which has the tweeter in the middle). However, more often than not, when we (and most other people) make a loudspeaker, we put the loudspeaker drivers in ascending order of frequency – woofers on the bottom, tweeters on the top. Maybe this is because woofers are heavier, and if you stand a BeoLab 5 on its head, it will fall over – but that’s not really the question we were asking…

The REAL question we were asking ourselves at the time was: if we were to build a multi-way loudspeaker – let’s say a 3-way, with a woofer, a midrange and a tweeter, and if the crossovers were such that the bulk of the “interesting” information (say, the vocal range) was coming from the midrange, then why would we not put the midrange at ear height and put the tweeter between it and the woofer? For example, imagine we made BeoLab 20 without an Acoustic Lens on top, would it be better to arrange the drivers like the version on the left of Figure 1 or the version on the right? Which one would sound better?

 

Figure 1: Two versions of a BeoLab 20-like loudspeaker with different driver arrangements.
Figure 1: Two versions of a BeoLab 20-like loudspeaker with different driver arrangements.

 

After answering that question, there’s a second question would follow closely behind: how close together do the drivers with adjacent frequency bands (i.e. the woofer and the midrange or the tweeter and the midrange) have to be in order for them to sound like one thing? Of course, these two questions are inter-related. If your midrange and tweeter are so far apart that they sound like different sound sources, then you would probably be more interested in where the voice was coming from than where the cymbals were coming from…

Of course, step one in answering the second question could be to calculate/simulate the response of the loudspeaker, based on distance between drivers, the crossover frequency (and therefore the wavelengths of the frequency band we’re interested in), the slopes of the crossover filters, and so on. It would also be pretty easy to make a prototype model out of MDF, put the loudspeaker drivers in there, do the vertical directivity measurements of the system in the Cube, and see how well the theory matches reality.

However, the question we were really interested in was “but how would it sound?” – just to get a rough idea before going any further. And I have to stress here that we were really talking about getting a rough idea. What I’m about to describe to you is a little undertaking that we put together in a day – just to find out what would happen. This was not a big, scientifically-valid experiment using a large number of subjects and intensive statistics to evaluate the results. It was a couple of guys having a chat over coffee one morning when one of them said “I wonder what would happen if we put the midrange on top…” and then, off they went to a listening room to find out.

One thing we have learned in doing both “quick-n-dirty” listening comparisons and “real, scientifically valid listening tests” is that the placement of a loudspeaker in a room, has a huge effect on how the loudspeaker sounds. So, when we’re comparing two loudspeakers, we try to put them as close together as possible. So, we tried different versions of this. In the first, we took two pairs of loudspeakers, and put the left loudspeakers in each pair side-by-side, with one pair upside down and the other right-side up, as shown in Figure 2.

 

Figure X: Another arrangement of the loudspeakers used in the first part of the experiment.
Figure 2: One arrangement of the loudspeakers used in the first part of the experiment. “Pair A” is in black and “Pair B” is in red.

 

We then switched between the right-side up pair and the upside down pair, listening for a change in vertical position of the image. Note that we tried two arrangements of this – one where both right-side up loudspeakers were to the left of the upside-down loudspeakers. The other where the “right-side up” loudspeakers were the “outside” pair, as shown in Figure 3.

Figure X: Another arrangement of the loudspeakers used in the first part of the experiment.
Figure 3: Another arrangement of the loudspeakers used in the first part of the experiment.

 

There are advantages and disadvantages of both of these arrangements – but in both cases, there is a lateral shift in the stereo image. When switching between pairs, either you get a left-right shift in image, or a change in width… It turned out that this change was more distracting than the vertical arrangement of the drivers, so we changed configuration to the one shown in Figure 4, below.

Figure X: Another arrangement of the loudspeakers used in the first part of the experiment.
Figure 4: Another arrangement of the loudspeakers used in the first part of the experiment. Note that, in this version, we used the loudspeaker pairs differently. “Pair A” used the red loudspeakers as tweeters and the black loudspeakers as woofers using an external crossover. “Pair B” used the red loudspeakers as woofers and the black loudspeakers as tweeters.

 

Now, instead of switching between loudspeakers, we pretended that one of them was a tweeter and other was a mid-woofer, and switched which was which, on the fly. Our “virtual” crossover was close-ish to the real crossover in the loudspeakers (our crossover was at 3.5 kHz, if you’re curious), so you could say that we were sort-of changing between using the upper tweeter + the lower woofer and using the upper woofer + lower tweeter, the “acoustical centres” of which are roughly at the same height. (remember – this was a quick-n-dirty experiment…)

 

Figure X: A photo of the loudspeakers used in the first part of the experiment.
Figure 5: A photo of one configuration of the  loudspeakers used in the first part of the experiment. Note that the BeoLab 9 and the BeoPlay V1 were not part of the experiment – we just didn’t bother to clean up before getting busy…

 

 

Figure X: A photo of the loudspeakers used in the first part of the experiment.
Figure 6: A photo of one configuration of the loudspeakers used in the first part of the experiment.

 

After having listened to these three configurations of loudspeakers, we decided that the vertical arrangement of the drivers was not important with the vertical separation we were using.

This brought us to the second part of the question… If the tweeter and the midrange were further apart, would we have a preference? So, we kept our virtual crossover, using one loudspeaker as the “tweeter” and the other as the “mid-woofer”, and we moved the loudspeakers further apart, one example of which is shown in Figure 7. (One thing to note here is that when I say “further apart” I’m talking about the separation in the vertical angles of the sources – not necessarily their actual distance from each other. For example, if the loudspeakers were 1 m apart vertically, and you were level with one of the loudspeakers, but the listening position was a kilometre away, then the vertical angular separation (0.057 degrees) would be much smaller than if you were 1 m away (45 degrees)…)

 

Figure X: A photo of the loudspeakers used in the second part of the experiment.
Figure 7: A photo of the loudspeakers used in the second part of the experiment. This was one (extreme) example of a vertical separation.

The answer we arrived at at the end was that, when the vertical separation of the drivers gets extreme (perhaps I should use the word “silly” instead), we preferred the configuration where the “mid-woofer” was at ear-height. However, this was basically a choice of which version we disliked less (“preference” is a loaded word…). When the drivers get far enough apart, of course, they no longer “hang together” as a single device without some extra signal processing tricks.

So, we went to lunch having made the decision that, as long as the tweeters and the midranges are close enough to each other vertically, we really didn’t have a preference as to which was on top, so, if anyone asked, we would let the visual designers decide. (Remember that “close enough” is not only determined by this little experiment – it is also determined by the wavelengths of the crossover frequencies and whether or not there are more drivers to consider. For example, in the example in Figure 1, it might be that we don’t care about the relative placement of the tweeter and midrange in isolation – but perhaps putting the tweeter between the midrange and woofer will make them too far apart to have a nice vertical directivity behaviour across the lower crossover…)

 

Addendum

This isn’t the first time we asked ourselves such a question. Although I was not yet working at B&O at the time, my colleagues tell me that, back in the days when they were developing the BeoLab 3500 and the BeoLab 7-1 (both of which are “stereo” loudspeakers – essentially two multi-way loudspeakers in a single device) , they questioned the driver arrangement as well. Should the tweeters or the midranges / woofers be on the outside? You can see in the photos of the 3500 on this page that they decided to put the lower-frequency bands wider because the overall impression with stereo signals was wider than if the tweeters were on the outside.

 

B&O Tech: Active Room Compensation

#35 in a series of articles about the technology behind Bang & Olufsen loudspeakers

 

Introduction:
Why do I need to compensate for my room?

Take a look at Figure 1. You’ll see a pair of headphones (BeoPlay H6‘s, if you’re curious…) sitting under a lamp that is lighting them directly. (That lamp is the only light source in the room. I can’t prove it, so you’ll have to trust me on this one…) You can see the headphones because the light is shining on them, right? Well… sort of.

Figure X: An object being lit with direct and reflected light.
Figure 1: An object being lit with direct and reflected light.

What happens if we put something between the lamp and the headphones? Take a look at Figure 2, which was taken with the same camera, the same lens, the same shutter speed, the same F-stop, and the same ISO (in other words, I’m not playing any tricks on you – I promise…).

Figure X: An object being lit with reflected light only.
Figure 2: An object being lit with reflected light only.

Notice that you can still see the headphones, even though there is no direct light shining on them. This probably does not come as a surprise, since there is a mirror next to them – so there is enough light bouncing off the mirror to reflect enough light back to the headphones so that we can still see them. In fact, there’s enough light from the mirror that we can see the shadow caused by the reflected lamp (which is also visible in Figure 1, if you’re paying attention…).

If you don’t believe me, look around the room you’re sitting in right now. You can probably see everything in it – even the things that do not have light shining directly on them (for example, the wall behind an open door, or the floor beneath your feet if you lift them a little…)

Exactly the same is true for sound. Let’s turn the lamp into a loudspeaker and the headphones on the floor into you, in the listening position and send a “click” sound (what we geeks call an “impulse”) out of the loudspeaker. What arrives at the listening position? This is illustrated in Figure 3, which is what we call an “impulse response” – how a room responds to an impulse (a click coming from a loudspeaker).

Figure X: The Impulse Response of a loudspeaker in a room at one location. The top plot shows the impulse (a "click" sound) sent to the loudspeaker. The bottom plot shows the sound received at the listening position.
Figure 3: The Impulse Response of a loudspeaker in a room at one location. The top plot shows the impulse (a “click” sound) sent to the loudspeaker. The bottom plot shows the sound received at the listening position.

 

 

The top plot in Figure 1 shows the signal that is sent to the input of the loudspeaker. The bottom plot is the signal at the input of the microphone placed at the listening room. If we zoom in on the bottom plot, the result is Figure 4. This makes it much easier to see the direct sound and the reflections as separate components.

Figure X: A zoom of the bottom plot in Figure X.
Figure 4: A zoom of the bottom plot in Figure 3.

If we zoom in even further to the beginning of the plot in Figure 4, we can see individual reflections in the room’s response, as is shown in Figure 5.

Figure X: A zoom of the plot in Figure X showing the direct sound and some of the reflections off of surfaces in the room. Note that the first reflection is only about 12 dB quieter than the direct sound.
Figure 5: A zoom of the plot in Figure 4 showing the direct sound and some of the reflections off of surfaces in the room. Note that the first reflection is only about 12 dB quieter than the direct sound.

 

Let’s take the total impulse response and separate the direct sound from the rest. This is shown in Figure 6.

 

Figure X: The impulse response of a loudspeaker in a room, separating the direct sound (in red) from the reflections (in blue).
Figure 6: The impulse response of a loudspeaker in a room, separating the direct sound (in red) from the reflections (in blue).

 

We can then calculate the magnitude responses of the two separate components individually to see what their relative contributions are – shown in Figure 7.

 

Figure X: The magnitude responses of the separate components of the total impulse response in Figure X. Red: Direct sound, Blue: Reflections.
Figure 7: The magnitude responses of the separate components of the total impulse response in Figure 6. Red: Direct sound, Blue: Reflections. 1/3 octave smoothed

 

Now, before you get carried away, I have to say up-front that this plot is a little misleading for many reasons – but I’ll only mention two…

The first is that it shows that the direct sound is quieter than the reflected sound in almost all frequency bands, but as you can see in Figure 6, the reflected energy is never actually louder than the direct sound. However, the reflected energy lasts for much longer than the direct sound, which is why the analysis “sees” it as containing more energy – but you don’t hear the decay in the room’s response at the same time when you play a click out of the loudspeaker. Then again, you usually don’t listen to a click – you listen to music, so you’re listening to the end of the room decay on the music that happened a second ago while you’re listening to the middle of the decay on the music that happened a half-second ago while you’re listening to the direct sound of the music that happened just now… So, at any given time, if you’re playing music (assuming that this music was constantly loud – like Metallica, for example…), you’re hearing a lot of energy from the room smearing music from the recent past, compared to the amount of energy in the direct sound which is the most recent thing to come out of the loudspeaker.

The second is in the apparent magnitude response of the direct sound. It appears from the red curve in Figure 7 that this loudspeaker has a response that lacks low frequency energy. This is not actually true – the loudspeaker that I used for this measurement actually has a flat on-axis magnitude response within about 1 dB from well below 20 Hz to well above 20 kHz. However, in order to see that actual response of the loudspeaker, I would have to use a much longer slice of time than the little red spike shown in Figure 6. In other words, the weirdness in the magnitude response is an artefact of the time-slicing of the impulse response. The details of this are complicated, so I won’t bother explaining it in this article – you’ll just have to trust me when I say that that isn’t really the actual response of the loudspeaker in free space…

The “punch line” for all of this is that the room has a significant influence on the perceived sound of the loudspeaker (something I talked about in more detail in this article). The more reflective the surfaces in the room, the more influence it has on the sound. (Also, the more omnidirectional the loudspeaker, the more energy it sends in more directions in the room, which also will mean that the room has more influence on the total sound at the listening position… but there’s more information about that in the article on Beam Width Control.)

So, if the room has a significant influence on the sound of the loudspeaker at the listening position, then it’s smart to want to do something about it. In a best case (and very generally speaking…), we would want to measure the effects that the room has on the overall sound of the loudspeaker and “undo” them. The problem is that we can’t actually undo them without changing the room itself. However, we can make some compensation for some aspects of the effects of the room. For example, one of the obvious things in the blue curve in Figure 7 is that the listening room I did the measurement in  has a nasty resonance in the low end (specifically, it’s at about 57 Hz which is the second axial mode for the depth of the room which is about 6 m). It would certainly help the overall sound of the loudspeaker to put in a notch filter at that frequency – in a best case, we should measure the phase response of the room’s resonance and insert a filter that has the opposite phase response. But that’s just the beginning with one mode – there are lots more things to fix…

 

A short history

Almost all Bang & Olufsen loudspeakers have a switch that allows you to change its magnitude response to compensate for the position of the loudspeaker in the room. This is typically called a Free/Wall/Corner switch, since it’s designed to offset the changes to the timbre of the loudspeaker caused by the closest boundaries. There’s a whole article about this effect and how we make a filter to compensate for it at this link.

In 2002, Bang & Olufsen took this a step further when it introduced the BeoLab 5 which included ABC – Automatic Bass Calibration. This was a system that uses a microphone to measure the effects of the listening room’s acoustical behaviour on the sound of the loudspeaker, and then creates a filter that compensates for those effects in the low frequency band. As a simple example, if your room tends to increase the apparent bass level, then the BeoLab 5’s reduce their bass level by the same amount. This system works very well, but it has some drawbacks. Specifically, ABC is designed to improve the response of the loudspeaker averaged over all locations in the room. However this follows the philosophy first stated Spock said in Star Trek II: The Wrath of Kahn when he said “the needs of the many outweigh the needs of the few, or the one.” In other words, in order to make the averaged response of the loudspeaker better in all locations in the room, it could be that the response at one location in the room (say, the “sweet spot” for example…) gets worse (whatever that might mean…). This philosophy behind ABC makes sense in BeoLab 5, since it is designed as a loudspeaker that has a wide horizontal directivity – meaning it is designed as a loudspeaker for “social” listening, not as a loudspeaker for someone with one chair and no friends… Therefore an improved average room response would “win” in importance over an improved sweet spot.

 

Active Room Compensation

We are currently working on a taking this concept to a new level with Active Room Compensation. Using an external microphone, we can measure the effects of the room’s acoustical behaviour in different zones in the room and subsequently optimise compensation filters for different situations. For example, in order to duplicate the behaviour of BeoLab 5’s ABC, we just need to use the microphone to measure a number of widely-space locations around the room, thus giving us a total average for the space. However, if we want to create a room compensation filter for a single location – the sweet spot, for example – then we can restrict the locations of the microphone measurements to that area within the room. If we want to have a compensation filter that is pretty good for the whole room, but has emphasis on the sweet spot, we just have to make more measurements in the sweet spot than in the rest of the room. The weighting of importance of different locations in the room can be determined by the number of microphone measurements we do in each location. Of course, this isn’s as simple a procedure as pressing one button, as in ABC on the BeoLab 5, but it has the advantage in the ability to create a compensation filter for a specific location instead of for the whole listening space.

As part of this work, we are developing a new concept in acoustical room compensation: multichannel processing. This means that the loudspeakers not only “see” each other as having an effect on the room – but they help each other to control the room’s acoustical influence. So, if you play music in the left loudspeaker only, then some sound will also come out of the right loudspeaker. This is because both the left and right loudspeakers are working together to control the room (which is being “activated” by sound only from the left loudspeaker.

 

B&O Tech: What is “Beam Width Control”?

#34 in a series of articles about the technology behind Bang & Olufsen loudspeakers

 

A little background:
Distance Perception in “Real Life”

Go to the middle of a snow-covered frozen lake with a loudspeaker, a chair, and a friend. Sit on the chair, close your eyes and get your friend to place the loudspeaker some distance from you. Keep your eyes closed, play some sounds out of the loudspeaker and try to estimate how far away it is. You will be wrong (unless you’re VERY lucky). Why? It’s because, in real life with real sources in real spaces, distance information (in other words, the information that tells you how far away a sound source is) comes mainly from the relationship between the direct sound and the early reflections that come at you horizontally. If you get the direct sound only, then you get no distance information. Add the early reflections and you can very easily tell how far away it is. If you’re interested in digging into this on a more geeky level, this report is a good starting point.

A little more background:
Distance perception in a recording

Recording engineers use this information as a trick to simulate differences in apparent distance to sound sources in a stereo recording by playing with the so-called “dry-wet” ratio – in other words, the relative levels of the direct sound and the reverb. I first learned this in the little booklet that came with my first piece of recording gear – an Alesis Microverb (1st generation… It was a while ago…). To be honest – this is a bit of an over-simplification, but it’s good enough to work (for example, listen to the reverberation on Grover’s voice change as he moves from “near” to “far” in this video). The people at another reverb unit manufacturer know that the truth requires a little more details. For example, their flagship reverb unit uses correctly-positioned and correctly-delayed early reflections to deliver a believable room size and sound source location in that room.

Recording Studios vs. Living Rooms

When a recording engineer makes a recording in a well-designed studio, he or she is sitting not only in a carefully-designed acoustical space, but a very special area within that space. In many recording studios, there is an area behind the mixing console where there are no (or at least almost no) reflections from the sidewalls . This is accomplished either by putting acoustically absorptive materials on the walls to soak up the sound so it cannot reflect (as shown in Figure 1), or to angle the walls so that the reflections are directed away from the listening position (as shown in Figure 2).

Figure 1: A typical floorplan for a recording studio that was built inside an existing room. The large rectangle is the recording console. The blue triangles are acoustically absorptive materials.
Figure 1: A typical floorplan for a recording studio that was built inside an existing room. The large rectangle is the recording console. The blue triangles are acoustically absorptive materials.

 

Figure 1: A typical floorplan for a recording studio that was designed for the purpose. The large rectangle is the recording console.
Figure 2: A typical floorplan for a recording studio that was designed for the purpose. The large rectangle is the recording console. Note that the side walls are angled to reflect energy away from the listening position.

Both of these are significantly different from what happens in a typical domestic listening room (in other words, your living room) where the walls on either side of the listening position are usually  acoustically reflective, as is shown in Figure 3.

Figure 3: A typical floorplan for a living room used as a listening room.
Figure 3: A typical floorplan for a living room used as a listening room.

 

In order to get the same acoustical behaviour at the listening position in your living room that the recording engineer had in the studio, we will have to reduce the amount of energy that is reflected off the side walls. If we do not want to change the room, one way to do this is to change the behaviour of the loudspeaker by focusing the beam of sound so that it stays directed at the listening position, but it sends less sound to the sides, towards the walls, as is shown in Figure 4.

Figure 4: A representation of a system using loudspeakers that send less energy towards the sidewalls.
Figure 4: A representation of a system using loudspeakers that send less energy towards the sidewalls. Note that there are still sidewall reflections – they’re just less noticeable.

So, if you could reduce the width of the beam of sound directed out the front of the loudspeaker to be narrower to reduce the level of sidewall reflections, you would get a more accurate representation of the sound the recording engineer heard when the recording was made. This is because, although you still have sidewalls that are reflective, there is less energy going towards them that will reflect to the listening position.

However, if you’re sharing your music with friends or family, depending on where people are sitting, the beam may be too narrow to ensure that everyone has the same experience. In this case, it may be desirable to make the loudspeaker’s sound beam wider. Of course, this can be extended to its extreme where the loudspeaker’s beam width is extended to radiate sound in all directions equally. This may be a good setting for cases where you have many people moving around the listening space, as may be the case at a party, for example.

For the past 5 or 6 years, we in the acoustics department at Bang & Olufsen have been working on a loudspeaker technology that allows us to change this radiation pattern using a system we call Beam Width Control. Using lots of DSP power, racks of amplifiers, and loudspeaker drivers, we are able to not only create the beam width that we want (or switch on-the-fly between different beam widths), but we can do so over a wide frequency range. This allows us to listen to the results, and design the directivity pattern of a loudspeaker, just as we currently design its timbral characteristics by sculpting its magnitude response. This means that we can not only decide how a loudspeaker “sounds” – but how it represents the spatial properties of the recording.

 

What does Beam Width Control do?

Let’s start by taking a simple recording – Susanne Vega singing “Tom’s Diner”. This is a song that consists only of a fairly dryly-recorded voice without any accompanying instruments. If you play this tune over “normal” multi-way loudspeakers, the distance to the voice can (depending on the specifics of the loudspeakers and the listening room’s reflective surfaces) sound a little odd.  As I discussed in more detail in this article, different beam widths (or, if you’re a little geeky – “differences in directivity”) at different frequency bands can cause artefacts like Vega’s “t’s” and “‘s’s” appearing to be closer to you than her vowel sounds, as I have tried to represent in Figure 5.

Figure X: A spatial map representing the location of the voice in Suzanne Vega's recording of Tom's Diner. Beam Width = off.
Figure 5: A spatial map representing the location of the voice in Suzanne Vega’s recording of Tom’s Diner. Beam Width Control = off. Note that the actual experience is that some frequency bands in her voice appear closer than others. This is due to the fact that the loudspeakers have different directivities at different frequencies.

 

Figure 6: The directivity of the system as a "normal" multi-way loudspeaker.
Figure 6: The directivity of the system as a “normal” multi-way loudspeaker. 3 dB per contour to -12 dB relative to on-axis.

If you then switch to a loudspeaker with a narrow beam width (such as that shown in the directivity plot in Figure 7 – the beam width is the vertical thickness of the shape in the plot – note that it’s wide in the low frequencies and narrowest at 10,000 Hz), you don’t get much energy reflected off the side walls of the listening room. You should also notice that the contour lines are almost parallel, which means that the same beam width doesn’t change as much with frequency.

Figure 7: The directivity of the system in "narrow" beam width.
Figure 7: The directivity of the system in “narrow” beam width. 3 dB per contour to -12 dB relative to on-axis.

Since there is very little reflected energy in the recording itself, the result is that the voice seems to float in space as a pinpoint, roughly half-way between the listening position and the loudspeakers – much as was the case of the sound of your friend on the snow-covered lake. In addition, as you can see in Figure 7, the beam width of the loudspeaker’s radiation is almost the same at all frequencies – which means that, not only does Vega’s voice float in a location between you and the loudspeakers, but all frequency bands of her voice appear to be the same distance from you. This is represented in Figure 8.

Figure X: A spatial map representing the location of the voice in Suzanne Vega's recording of Tom's Diner. Beam Width = narrow.
Figure 8: A spatial map representing the location of the voice in Suzanne Vega’s recording of Tom’s Diner. Beam Width = narrow.

If we then switch to a completely different beam width that sends sound in all directions, making a kind of omnidirectional loudspeaker (with a directivity response as is shown in Figure 9), then there are at least three significant changes in the perceived sound. (If you’re familiar with such plots, you’ll be able to see the “lobing” and diffraction caused by various things, including the hard corners on our MDF loudspeaker enclosures. See this article for more information about this little issue… )

Figure 8: The directivity of the system in "omni" beam width.
Figure 9: The directivity of the system in “omni” beam width. 3 dB per contour to -12 dB relative to on-axis.

The first big change is that the timbre of the voice is considerably different – particularly in the mid-range (although you could easily argue that this particular recording only has mid-range…). This is caused by the “addition” of reflections from the listening room’s walls at the listening position (since we’re now sending more energy towards the room boundaries). The second change is in the apparent distance to the voice. It now appears to be floating at a distance that is the same as the distance to the loudspeakers from the listening position. (In other words, she moved away from you…). The third change is in the apparent width of the phantom image – it becomes much wider and “fuzzier” – like a slightly wide cloud floating between the loudspeakers (instead of a pin-point location). The total result is represented in Figure 10, below.

Figure X: A spatial map representing the location of the voice in Suzanne Vega's recording of Tom's Diner. Beam Width = omni.
Figure 10: A spatial map representing the location of the voice in Suzanne Vega’s recording of Tom’s Diner. Beam Width = omni.

 

All three of these artefacts are the result of the increased energy from the wall reflections.

Of course, we don’t need to go from a very narrow to an omnidirectional beam width. We could find a “middle ground” – similar to the 180º beam width of BeoLab 5 and call that “wide”. The result of this is shown in Figures 11 and 12, with a measurement of the BeoLab 5’s directivity shown for comparison in Figure 13.

Figure 8: The directivity of the system in "wide" beam width.
Figure 11: The directivity of the system in “wide” beam width. 3 dB per contour to -12 dB relative to on-axis.

 

Figure X: A spatial map representing the location of the voice in Suzanne Vega's recording of Tom's Diner. Beam Width = wide.
Figure 12: A spatial map representing the location of the voice in Suzanne Vega’s recording of Tom’s Diner. Beam Width = wide.

 

Figure 8: The directivity of a BeoLab 5.
Figure 13: The directivity of a BeoLab 5.

If we do the same comparison using a more complex mix (say, Jennifer Warnes singing “Bird on a Wire” for example) the difference in the spatial representation is something like that which is shown in Figures 14 and 15. (Compare these to the map shown in this article.) Please note that these are merely an “artist’s rendition” of the effect and should not be taken as precise representations of the perceived spatial representation of the mixes. Actual results will certainly vary from listener to listener, room to room, and with changes in loudspeaker placement relative to room boundaries.

 

Figure X: A spatial map representing the locations of some of the sound sources in Jennifer Warnes's recording of Bird on a Wire. Beam Width = narrow.
Figure 14: A spatial map representing the locations of some of the sound sources in Jennifer Warnes’s recording of Bird on a Wire. Beam Width = narrow.

 

Figure X: A spatial map representing the locations of some of the sound sources in Jennifer Warnes's recording of Bird on a Wire. Beam Width = wide.
Figure 15: A spatial map representing the locations of some of the sound sources in Jennifer Warnes’s recording of Bird on a Wire. Beam Width = wide.

 

Of course, everything I’ve said above assumes that you’re sitting in the “sweet spot” – a location equidistant to the two loudspeakers at which both loudspeakers are aimed. If you’re not, then the perceived differences between the “narrow” and “omni” beam widths will be very different… This is because you’re sitting outside the narrow beam, so, for starters, the direct sound from the loudspeakers in omni mode will be louder than when they’re in narrow mode. In an extreme case, if you’re in “narrow” mode, with the loudspeaker pointing at the wall instead of the listening position, then the reflection will be louder than the direct sound – but now I’m getting pedantic.

 

Wrapping up…

The idea here is that we’re experimenting on building a loudspeaker that can deliver a narrow beam width so that, if you’re like me – the kind of person who has one chair and no friends, and you know what a “stereo sweet spot” is, then you can sit in that chair and hear the same spatial representation that the recording engineer heard in the recording studio (without having to make changes to your living room’s acoustical treatment). However, if you do happen to have some friends visiting, you have the option of switching over to a wider beam width so that everyone shares a more similar experience. It won’t sound as good (whatever that might mean to you…) in the sweet spot, but it might sound better if you’re somewhere else. Similarly, if you take that to an extreme and have a LOT of friends over, you can use the “omni” beam width and get a more even distribution of background music throughout the room.

 

For more information on Beam Width Control

Shark Fins and the birth of Beam Width Control

Beam Width Control – A Primer

 

Post-script

For an outsider’s view, please see the following…

“Ny lydteknikk fra Bang & Olufsen” – Lyd & Bilde (Norway)

Stereophile magazine (October 2015, Print edition) did an article on their experiences hearing the prototypes as well.

“BeoLab 90: B&O laver banebrydende højttaler” – Lyd & Bilde (Norway)

BeoPlay H2 Headphones

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I was part of the development team, and one of the two persons who decided on the final sound design (aka tonal balance) of the B&O H2 headphones. So, I’m happy to share some of the blame for some of the comments (at least on the sound quality) from the reviews.

 

from Techradar India

“Bass accuracy is right on and just as powerful as it needs to be. Mids and highs also shine through in the sound with a subtle warmness that’s hard to find in a set of headphones.”

 

from What Hi-Fi?

“The 40mm driver and bass port in each earcup provide an easily accessible sound, as is appropriate for headphones intended to be worn outdoors.

“It’s warm without being overbearing, and the presentation is even across the frequency range, so no one area stands out as prominent. Treble is clear without being too sharp or bright, and the midrange is a strength, with vocals coming across warm and intimate.

“The bass is a touch tubby, but only compared with our current class favourites, the Award-winning Philips M1 MkIIs (indeed the H2s’ bass is reminiscent of the original M1s’).

“It’s not overblown, though, and that character trait certainly doesn’t hurt in a pair of headphones designed to be worn in the open.”

 

from International Business Times

“Overall, the sound quality is great, as you’d expect, and users will notice a stark difference between these and an entry-level set.”

B&O Tech: Video Engine Customisation Part 1: Bass Management

#33 in a series of articles about the technology behind Bang & Olufsen loudspeakers

 

Earlier today (just after lunch), I had a colleague drop by to ask some technical questions about his new BeoVision 11. He’s done the setup with his external loudspeakers, but wanted to know if (mostly “how”…) he could customise his bass management settings to optimise his system. So, we went into the listening room to walk through the process. Since he was discussing this with some other colleagues around the lunch table, it turned out that we discovered that he wasn’t the only one who wanted to learn this stuff – so we wound up with a  little group of 8 or 10 asking lots of questions that usually started with “but what if I wanted to…” After a half an hour or so, we all realised that this information should be shared with more people – so here I am, sharing.

This article is going to dive into the audio technical capabilities of what we call the “Video Engine” (the processing hardware inside the BeoPlay V1, the BeoVision 11, BeoVision Avant, and BeoSystem 4). For all of these devices, the software and its capabilities with respect to processing of audio signals are identical. Of course, the hardware capabilities are different – for example, the number of output channels are different from product to product – but everything I’m describing below can be done on all of those products.

One other warning: Almost of the menus that I direct you to in the text below are accessed as follows:

“MENU” button on remote control -> Setup -> Sound -> Speaker Group -> NAME -> Advanced Settings -> Bass Management

Fig 1: Video Engine 1 menu map
Fig 1: Video Engine 1 menu map

You can see this in the menu map, below.

 

 

So, let’s start by looking at a signal flow block diagram of the way the audio is processed inside the bass management processing.

 

Fig 1: Block diagram of the bass management signal flow in the Video Engine
Fig 2: Block diagram of the bass management signal flow in the Video Engine

 

So, let’s start at the input to the system:

The “Speaker” signals coming in on the left are coming from the 16 output channels of the True Image upmixer (if you’re using it) or they are the direct 2.0, 5.1, or 7.1 (or other formats, if you’re weird like me) channels from the source. (We’ll ignore the LFE input for now – but we’ll come back to it later…)

The first thing the signals encounter is the switch labelled Enable Filtering (which is also the name of the menu item where you control this parameter). This where you decide whether you want the bass removed from the signal or not. For smaller loudspeakers (assuming that you have a big some in the same system) you will want to Enable the Filtering and remove the bass to re-direct it to the larger loudspeaker. If you have larger loudspeakers in the same system, you may not want to re-direct the signals. So, let’s say that you have BeoLab 20’s as the Left Front and Right Front (Lf / Rf for the rest of this article), BeoLab 17’s as the Left Surround and Right Surround (Ls / Rs), and the internal loudspeakers as the Centre Front (Cf), you will want to Enable Filtering on the Ls, Rs, and Cf signals. This will remove the bass from those channels and re-direct it somewhere else (to the 20’s – but we make that decision downstream…). You will not want to Enable Filtering for the 20’s since you’ll be using them as the “subwoofers”.

Assuming that you’ve enabled the filtering, then the signal takes the lower path and splits to go in two directions. The upper direction is to a high-pass filter. This is a 4th order highpass that is 6.02 dB down at the frequency chosen in the Crossover Frequency menu. We use a 4th order filter because the crossover in our bass management system is a 4th order Linkwitz-Riley design. As you can see in the block diagram, the output of the highs filter is routed directly to the output to the loudspeaker. The low path in the split goes to a block called Panning. This is where you decide, on a channel-by-channel basis, whether the signal should be routed to the left or the right bass channel (or some mix of the two). For example, if you have a Ls loudspeaker that you’re bass managing, you will probably want to direct its bass to the Left bass channel. The Rs loudspeaker’s bass will probably direct to the Right bass channel, and the Cf loudspeaker’s bass will go to both. (Of course, if, at the end of all this, you only have one subwoofer, then it doesn’t matter, since the Left and Right bass channels will be summed anyway.) The outputs of all the panning blocks are added together to form the two bass channels – although, you may notice in the block diagram, they are still full-range signals at this point (internally) in the signal flow, since we haven’t low-pass filtered them yet.

Next, the “outputs” of the two bass channels are low-pass filtered using a 4th order filter on each. Again, this is due to the 4th order Linkwitz-Riley crossover design. The cutoff frequency of these two low pass filters are identical to the highpass filters which are all identical to each other. There is a very good reason for this. Whenever you apply a minimum-phase filter (which ours are, in this case) of any kind to an audio signal, you get two results: one is a change in the magnitude response of the signal, the other is a change in its phase response. One of the “beautiful” aspects of the Linkwitz-Riley crossover design is that the Low Pass and High Pass filters are 360° out-of-phase with each other at all frequencies. This is (sort of…) the same as being in-phase at all frequencies – so the signals add back together nicely. If, however, you use a different cutoff frequency for the low and high pass components, then the phase responses don’t line up nicely – and things don’t add back together equally at all frequencies. If you have audio channels that have the same signals (say, for example, the bass guitar in both the Lf and Rf at the same time – completely correlated) then this also means that you’ll have to use the same filter characteristics on both of those channels. So, the moral of the story here is that, in a bass management system, there can be only one crossover frequency to rule them all.

You may be wondering why we add the signals before we apply the low-pass filter. The only reason for this was an optimisation of the computing power – whether we apply the filter on each input channel (remember, there are up to 16 of those…) or on two summed outputs, the result is the same. So, it’s smarter from a DSP MIPS-load point of view to use two filters instead of 16 if the result is identical (all of our processing is in floating point, so there’s no worry of overloading the system internally).

Now comes the point where we take the low-frequency components of the bass-managed signals and add them to the incoming LFE channel. You may notice a little triangle on that LFE channel before it gets summed. This is not the +10 dB that is normally added to the LFE channel – that has already happened before it arrived at the bass management system. This gain is a reduction, since we’re splitting the signal to two internal bass channels that may get added back together (if you have only one subwoofer, for example). If we didn’t drop the gain here, you’d wind up with too much LFE in the summed output later.

Now we have the combined LFE and bass management low-frequencies on two (left and right) bass channels, ready to go somewhere – but the question is “where?” We have two decisions left to make. The first is the Re-direction Balance. This is basically the same as a good-old-fashioned “balance” control on your parents’ stereo system. Here you can decide (for a given loudspeaker output) whether it gets the Left bass channel, the Right bass channel, or a combination of the two (you only have three options here). If you have a single subwoofer, you’ll probably be smart to take the “combination” option. If you have separate Left and Right subwoofers, then you’ll want to direct the Left bass channel to the left subwoofer and the right to the right.

Finally, you get to the Bass Re-direction Level menu. This is where you decide the gain that should be applied to the bass channel that is sent to the particular loudspeaker. If you have one subwoofer and you want to send it everything, then its Redirection level will by 0 dB and the other loudspeakers will be -100 dB. If you want to send bass everywhere, then set everything to 0 dB (this is not necessarily a good idea – unless you REALLY like bass…).

 

It’s important to note that ANY loudspeaker connected in your system can be treated like a “subwoofer” – which does not necessarily mean that it has a “subwoofer” Speaker Role. For example, in my system, I use my Lf and Rf loudspeakers as the Lf and Rf channels in addition to the subwoofers. This can be seen in the menus as the Lf and Rf Re-direction levels set to 0 dB (and all others are set to -100 dB to keep the bass out of the smaller loudspeakers).

 

Special Treatment for Subwoofers

As you can see in the block diagram, there are two “subwoofer” outputs which, as far as the bass management is concerned, are identical to other loudspeakers. However, the Subwoofer outputs have two additional controls downstream for customising the alignment with the rest of the system. The first is a Time Alignment adjustment which can be set from -30 ms to +30 ms. If this value is positive, then the subwoofer output is delayed relative to the rest of the system. If it’s negative, then the rest of the system is delayed relative to the subwoofer. There are lots of reasons why you might want to do either of these on top of your Speaker Distance adjustment – but I’m not going to get into that here.

The second control is a first-order Allpass filter. This will be 90° out of phase at the frequency specified on the screen – going to 180° out at a maximum in the high frequencies. The reason to use this would be to align for phase response differences between your subwoofer’s high end and your main “main loudspeakers'” low-end. Say, for example, you have a closed-box subwoofer, but ported (or slave driver-based) main loudspeakers. You may need some phase correction in a case like this to clean up the addition of the signals across the crossover region. Of course, if you have different main loudspeakers, then one allpass filter on your subwoofer can’t correct for all of the different responses in one shot. Of course, if you don’t want to have an allpass filter in your subwoofer signal path, you can bypass it.

 

For more information about the stuff I talked about here (including cool things like phase response plots of the allpass filter), check out the Technical Sound Guide for the Video Engine-based products. This is downloadable from this page for example.

B&O Tech: The Naked Truth V

#32 in a series of articles about the technology behind Bang & Olufsen loudspeakers

This posting: something new, something old…

First, the insides of the BeoLab 14 subwoofer. The obvious part is the port curling around to get the right length in a somewhat shorter package. This concept has been around for a while as you can see when you look at a trumpet or a tuba…

The silver-coloured disc right below the bottom of the port is the pole piece of the woofer. The black ring around this is the ferrite magnet. In the background you can see the circuit boards containing the power supply, DSP and amplifiers for the sub and the satellites. For a better view of this, check out this page.

The reasons the end of the port is flared like a trumpet bell is to reduce the velocity of the air at the end of the pipe. This reduces turbulence which, in turn, means that there is less noise or “port chuffing” at the resonant frequency of the port. Of course, the other end of the port at the top of the subwoofer is also flared for the same reason.

As I mentioned in a previous posting, the DSP is constantly calculating the air velocity inside the port and doesn’t allow it to exceed a value that we determined in the tuning. This doesn’t mean that it’s impossible to hear the turbulence – if you test the system with a sine tone, you’ll hear it – but that was a tuning decision we made. This is because we pushed the output to a point that is almost always inaudible with music – but can be heard with sine tones. If we hadn’t done that, the cost would have been a subwoofer with less bass output.

 

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Now for something a little older… This is a BeoLab 3500 (we’re not looking at the BeoLab 7-4 on the shelf below)

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Below is a close-up of the tweeter and woofer. You may notice that you can see light through the edge of the surround of the woofer. This is because we cut it with a knife for a different demonstration – it’s not normal… You can also see the fins which help to keep the electronics cool.

 

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As you can see in the photos below, all the electronics are inside the woofer enclosures. The tweeter has its own built-in chamber, so it’s sealed from the woofer enclosure.

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B&O Tech: Naked Truth IV

#29 in a series of articles about the technology behind Bang & Olufsen loudspeakers

 

Sorry – I’ve been busy lately, so I haven’t been too active on the blog.

 

Here are some internal shots of the BeoLab 17 and BeoLab 20 loudspeakers. As you can see in the shot of the back of the BeoLab 17, the entire case is the enclosure is for the woofer. The tweeter has its own enclosure which seals it from the woofer cabinet.

 

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What’s not obvious in the photos of the BeoLab 20 is that the midrange and woofer cabinets are separate sealed boxes. There is a bulkhead that separates the two enclosures cutting across the loudspeaker just below the midrange driver.

 

 

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