Category: loudspeakers

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)

Speakers and Sneakers

I recently received an email from someone asking the following question:

“I’ve been reading your blog for a while and a question popped up. What do you think of the practice of ”breaking in” speakers? Is there any truth to it or is it simply just another one of the million myths believed by audiophiles?”

To answer this question, I’ll tell a story.

At work, we have a small collection of loudspeakers – not only current B&O models, but older ones as well. In addition, of course, we have a number of loudspeakers made by our competitors. Many loudspeakers in this collection don’t get used very often, but occasionally, we’ll bring out a pair to have a listen as a refresher or reminder. Usually, the way this works is that one of us from the acoustics department will sneak into the listening room with a pair of loudspeakers, and set them up behind an acoustically transparent, but visually opaque curtain. The rest of us then get together and listen to the same collection of recordings at the same listening level, each of sitting in the same chair. We talk about how things sound, and then we open the curtain to see what we’ve been complaining about.

One day, about three years ago, it was my turn to bring in the loudspeakers, so I set up a pair of passive loudspeakers (not B&O) that have a reasonably good reputation. We had a listen and everyone agreed that the sound was less than optimal (to be polite…). No bass, harsh upper midrange, everything sounded like it was weirdly compressed. Not many of us had anything nice to say. I opened the curtain, and everyone in the room was surprised when they saw what we had listened to – since we would have all expected things to sound much better.

Later that day, I spoke with one of our colleagues who was not in the room, and I told him the story – no one liked the sound, but those speakers should sound better. His advice was to wait until next week, and play the same loudspeakers again – but the next time, play pink noise through them at a reasonably high level for a couple of hours before we listened. So, the next week, the day before we were scheduled to have our listening session, I set up the same speakers in the same locations in the room, and played pink noise at about 70 dB SPL through them overnight. The next morning, we had our blind listening session, and everyone in the room agreed that the sound was quite good – much better than what we heard last week. I opened the curtains and everyone was surprised again to see that nothing had changed. Or had it? I was as surprised as anyone, since my religious belief precludes this story from being true. But I was there… it actually happened.

So, what’s the explanation? Simple! Go to the store and buy two identical pairs of sneakers (or “running shoes” or “trainers”, depending on where you’re from). When you get home, take one pair out of the box, and wear them daily. After three or four months, take the pair that you left in the box and try them on. They will NOT feel the same as the pair you’ve been wearing. This is not a surprise – the leather and plastic and rubber in the sneakers you’ve been wearing has been stretched and flexed and now fits your foot better than the ones you have not been wearing. In addition, you’ll probably notice that the “old” ones are more flexible in the places where your foot bends, because you’ve been bending them.

It turns out (according to the colleague who suggested the pink noise trick who also used to design and make loudspeaker drivers for a living) that the suspension (the surround and spider) of a loudspeaker driver becomes more flexible by repeated flexing – just like your sneakers. If you take a pair of loudspeakers out of the box, plug them in, and start listening, they’ll be stiff. You need to work them a little to “loosen them up”.

This is not only true of new sneakers (and speakers) but also of sneakers (and speakers). For example, I keep my old sneakers around to use when I’m mowing the lawn. When I stick my foot into the sneakers that I haven’t worn all winter, they feel stiff, like a new pair, because they have not been flexed for a while. This is what happened to those speakers that I brought upstairs after sitting in the basement storage for years. The suspensions became stiff and needed to be moved a little before using them for listening to music.

 

A small problem that compounds the complexity of evaluating this issue is that we also “get used to” how things sound. So, as you’re “breaking in” a loudspeaker by listening to it, you are also learning and accommodating yourself to how it sounds, so you’re both changing simultaneously. Unless you have the option of playing a trick on people like I did with my colleagues, it’s difficult to make a reliable judgement of how big a difference this makes.

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.

 

_GGM1646

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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)

_GGM1657

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.

 

_GGM1660

 

_GGM1656

 

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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_GGM1652

 

 

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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