Category: Bang & Olufsen

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

bo_beoplay_2

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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B&O Tech: Reading Spec’s – Part 1

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

 

Introduction

Occasionally, I read other people’s blogs and forum postings to see what’s happening outside my little world. This week, I came across this page in which one of the contributors made some comments about B&O’s loudspeaker specifications – or, more precisely, the lack of them – or the lack of precision in them – particularly with respect to the Frequency Range specifications.

So, this posting will be an attempt to explain how we determine the frequency range of our loudspeakers.

 

Bandwidth (also known as “Frequency Range”)

Ask any first-year electrical engineering student to explain how to find the “bandwidth” of an audio product and they’ll probably all tell you the same thing – which will be something like the following:

  1. Measure the magnitude response (what many people call the “frequency response”) of the product.
  2. Find the peak in the magnitude response
  3. Going upwards in frequency, find the point where the magnitude drops by 3.01 dB relative to the peak.
  4. Going downwards in frequency, find the point where the magnitude drops by 3.01 dB relative to the peak.
  5. Subtract the lower frequency from the upper frequency and you get the bandwidth.

(If you’re curious, the 3.01 dB threshold is chosen because -3.01 dB is equivalent to one-half of the power of the peak. This is why the -3.01 dB points are also known as a the “half-power points”)

Fig 1: An example of how bandwidth is measured on a typical audio device. See text for more information
Fig 1: An example of how bandwidth is measured on a typical audio device. See text above for more information

 

Figure 1 shows a pretty typical looking curve for an audio device (admittedly, not a very good one…). The magnitude response is flat enough, and it extends down to 34 Hz and up to 15.6 kHz.

This same technique can be used to find the bandwidth of an audio processing device, as is shown in Figure 2.

 

Fig 2: An example of how bandwidth is measured on a filter - in this case a peaking filter.
Fig 2: An example of how bandwidth is measured on a filter – in this case a peaking filter. The Bandwidth of this filter ranges from 770 Hz to 1300 Hz – a total bandwidth of 530 Hz.

Loudspeaker Frequency Range

Let’s try applying that same method used for audio “black boxes” on a loudspeaker. We’ll measure the on-axis magnitude response of the loudspeaker in a free field (one without reflections), and find the frequencies where the magnitude drops 3.01 dB below the peak value. An example of this is shown below.

Fig 3: An unsmoothed on-axis magnitude response of a real loudspeaker. If we use the same technique to measure the bandwidth on this loudspeaker as we did in Fig 2, the bandwidth will range from 7.3 kHz to 13.9 kHz.
Fig 3: An unsmoothed on-axis magnitude response of Loudspeaker #1. If we use the same technique to measure the bandwidth on this loudspeaker as we did in Fig 2, the bandwidth will range from 7.3 kHz to 13.9 kHz.

 

Hmmmm… that didn’t turn out as nicely as I had hoped. It seems that (using this definition) the loudspeaker whose response is shown in Figure 3 has a Frequency Range of 7.3 kHz to 13.9 kHz. This is unfortunate, since it is not a tweeter – it’s a rather large, commerically-available, floor-standing loudspeaker with a rather good reputation.

Okay, maybe we’re being too stringent. Let’s say that, instead of defining the Frequency Range as the area between the – 3 / + 0 dB points, we’ll make it ± 3 dB instead. Figure 4 shows the same loudspeaker with that version of Frequency Range.

 

Fig 4: An unsmoothed on-axis magnitude response of a real loudspeaker. We now define the Frequency Range as being within the ± 3 dB points, which results in a range of 2.2 kHz to 15.9 kHz.
Fig 4: An unsmoothed on-axis magnitude response of Loudspeaker #1. We now define the Frequency Range as being within the ± 3 dB points, which results in a range of 2.2 kHz to 15.9 kHz.

 

Great- it got better! Now the Frequency Range of this loudspeaker (under the new ±3 dB definition) is from 2.2 kHz to 15.9 kHz. On paper, that still makes it a tweeter – so we’re still in trouble here. Note that I have scaled the magnitude response here to “help” the loudspeaker as much as I can by putting the peak in the magnitude response on the + 3 dB line. If I had not done this (for example, if I had said “±3 dB relative to the magnitude at 1 kHz” the numbers would certainly not get better…)

Let’s try the same definition on a different loudspeaker – shown in Figure 5.

 

Fig 4: An unsmoothed on-axis magnitude response of a real loudspeaker. Using a Frequency Range defined by the ± 3 dB points, the range is 5.4 kHz to 18.0 kHz.
Fig 4: An unsmoothed on-axis magnitude response of Loudspeaker #2. Using a Frequency Range defined by the ± 3 dB points, the range is 5.4 kHz to 18.0 kHz.

 

This loudspeaker (also a commerically-available floor-standing model with a good reputation) has a Frequency Range (using the ± 3 dB points) of 5.4 kHz to 18.0 kHz. This ±3 dB definition isn’t working out very well. Let’s try one more loudspeaker to see what happens.

 

Fig 6: An unsmoothed on-axis magnitude response of a real loudspeaker. Using a Frequency Range defined by the ± 3 dB points, the range is 5.4 kHz to 18.0 kHz.
Fig 5: An unsmoothed on-axis magnitude response of Loudspeaker #3. Using a Frequency Range defined by the ± 3 dB points, the range is 83 Hz to 20.9 kHz.

 

Yay! It worked! For loudspeaker #3, shown in Figure 5, the Frequency Range is from 83 Hz to 20.9 kHz. Ummmm… except that, of the 3 loudspeakers I measured for this experiment, this is, by far the smallest. It’s a little 2-way loudspeaker that I can easily lift with one hand whilst sipping a cup of coffee from the other (and I don’t work out – ever!)

So, what we have learned so far is that it’s better to buy a small loudspeaker than a big one, since it has a much wider frequency range. No, wait. that can’t be right…

Hmmmm… what if we were to smooth the magnitude responses? Maybe that would help…

 

Fig 6: The on-axis magnitude responses of the 3 loudspeakers (Black=#1, Blue=#2, Red =#3), 1/3 octave smoothed.
Fig 6: The on-axis magnitude responses of the 3 loudspeakers (Black=#1, Blue=#2, Red =#3), 1/3 octave smoothed.

 

That’s a little better, but Loudspeaker #1 (the black curve), still bottoms out in the midrange, causing its frequency range to resemble that of a tweeter. And Loudspeaker #2 loses out on the high end due to the peaks. (Note that, in this plot, I’ve scaled them all to have the same magnitude at 1 kHz – just trying something out to see if that helps. It didn’t.) How about more smoothing?

 

Fig 7: The on-axis magnitude responses of the 3 loudspeakers (Black=#1, Blue=#2, Red =#3), 1 octave smoothed.
Fig 7: The on-axis magnitude responses of the 3 loudspeakers (Black=#1, Blue=#2, Red =#3), 1 octave smoothed.

 

Got it! Now all of the loudspeakers’ responses have been smeared out enough… uh… adequately smoothed… to make our definition of Frequency Range have a little meaning. The “big” loudspeakers have wider Frequency Ranges than the “little” loudspeaker. So, we’ll just octave-smooth all of our measurements. Well…  at least until we find another loudspeaker that needs even more smoothing…

 

Sidebar: In case you’re wondering: the three loudspeakers I’m talking about above are all commercially available products. One of the three is a Bang & Olufsen loudspeaker. Don’t bother asking which is which (or which B&O loudspeaker it is) – I’m not telling – mostly because it doesn’t matter.

The moral thus far…

Of course, the point that I’m trying to make here is that Frequency Range, like any specification for any audio device, needs to make sense. If we arbitrarily set some test method (i.e. “measure the on-axis magnitude response – and then smooth it”) and apply arbitrary criteria (i.e. ± 3 dB) then we may not get a useful description of the device’s behaviour.

So, at this point, you’re probably asking “Well… how does B&O measure Frequency Range?” Well, I’m glad you asked!

After we’re done with the sound design of the loudspeaker, we have to make sure it has the correct sensitivity. This is a measure of how loud it is for a given voltage at its input.  So, we put the loudspeaker in the Cube and measure its final on-axis magnitude response.

I’ll illustrate this using a magnitude response that I invented, shown in Fig 8. Note that this response is not a real loudspeaker – it’s one that I invented just for the purposes of this discussion.

 

Step 1: We measure the on-axis magnitude response of the loudspeaker at 1 m after the sound design is finished. We then look at the average level of the response between 200 Hz and 2 kHz.
Fig 8: Step 1: We measure the on-axis magnitude response of the loudspeaker at 1 m after the sound design is finished. We then look at the average level of the response between 200 Hz and 2 kHz. (Note that the response shown here is NOT a measurement of a real loudspeaker.)

 

 

We then look at the average level of the magnitude response between 200 Hz and 2 kHz and adjust the gain in the signal processing of the loudspeaker to make the sensitivity what we want it to be. For almost all B&O loudspeakers, that sensitivity corresponds to an output level of 88 dB SPL for an input with a level of 125 mV RMS. (The only exceptions are BeoLab 1, BeoLab 5, and BeoLab 9 which produce 91 dB SPL for a 125 mV RMS input.)

So, after the gain has been adjusted, the magnitude response looks like Figure 9, below.

 

Step 2: The gain of the loudspeaker has been adjusted so that the average output between 200 Hz and 2 kHz is 88 dB SPL for a 125 mV RMS input.
Fig 9: Step 2: The gain of the loudspeaker has been adjusted so that the average output between 200 Hz and 2 kHz is 88 dB SPL for a 125 mV RMS input.

 

We then look for the frequencies that have a magnitude that are 10 dB lower than the average level between 200 Hz and 2 kHz. This is illustrated in Figure 10.

 

Step 3: The frequencies with magnitudes that are 10 dB lower than the average between 200 Hz and 2 kHz are found. These are the values used in the Frequency Range specification.
Step 3: The frequencies with magnitudes that are 10 dB lower than the average between 200 Hz and 2 kHz are found. These are the values used in the Frequency Range specification.

 

The values that correspond to the -10 dB points (relative to the average level between 200 Hz and 2 kHz) are the frequencies stated in the Frequency Range specification.

This is how B&O specifies Frequency Range for all of its loudspeakers. That way, you (and we) can directly compare their specifications to each other. Of course, some other manufacturer may (or probably will) use a different method – so you cannot use B&O’s Frequency Range specifications to compare to another company’s products. We don’t use a ±3 dB threshold, not only because this would require arbitrary smoothing in order to prevent weird things from happening (as I showed above) but also because the on-axis magnitude response of B&O loudspeakers is a result of the loudspeakers’ sound design (which includes a consideration of its power response) which means that, if you just look at the on-axis response, it might not be as flat as a magazine would lead you to believe it should be.

 

 

The Fine Print

1. The method I described above is a slightly simplified explanation of what we actually do – but the difference between what I said and the truth is irrelevant. The details are in the method we use to do the averaging – so it’s not really a big deal unless you’re actually writing the software that has to do the work.

2. We do this measurement of the Frequency Range using a signal with a level of 125 mV at the input of the loudspeaker. So, if your music is playing with a similar level (or lower) then you will have a loudspeaker that is performing as specified. However, if you play the music louder, the frequency range will change. In most cases, the low frequency limit will increase due to the ABL and thermal protection algorithms. The details of (1) how much it will increase, (2) what level of music will cause it to increase, and (3) what frequency content in the music will cause it to increase, are different from loudspeaker model to loudspeaker model.  This was the root of some confusion for some people when they compare the frequency range of the BeoLab 12-3 to the BeoLab 12-2. These two loudspeakers have almost identical low frequency cutoffs, despite the fact that one of them has 2 woofers and the other has only 1. At “normal” listening levels, they have both been tuned to have similar magnitude responses – however, as you turn up the volume, the BeoLab 12-2 will lose bass earlier than the BeoLab 12-3.

3. Subwoofers are different – since it doesn’t make sense to try and find the average magnitude response of a subwoofer between 200 Hz and 2 kHz.