“… the power output is phenomenal. Moreover, the detail is incredible and the tonal balance is spot on. The vocals in Antony & The Johnson’ Twilight are intense, the throb of Jeff Beck’s guitar in So Real resonates sublimely, whilst classical works yield profound levels of clarity. The mi-range is highly detailed, the treble is smooth and accurate, and the bass is rich and velvety. You can push the volume without risk: even at high levels the speakers have plenty in reserve.”
I realised this week that I’ve been throwing around words like ” voice coil”, “suspension”, “surround” and “spider” when I talk about loudspeakers, but many people don’t know what these things are – or how a loudspeaker driver works in general… So, this week, I thought I might back up a couple of steps and talk about some basics. There’s nothing here about Bang & Olufsen loudspeakers specifically – it’s just an introduction to how loudspeaker drivers (like woofers, for example) work.
Back in 1820, a Danish guy named Hans Christian Ørsted was in the middle of giving a lecture when he noticed that, when he switched a circuit on and off, a compass sitting nearby on the desk moved a little. Since he had been poking around with experiments in electricity and magnetism for years, it didn’t take him long to put two and two together and come up with the idea that, when you run electrical current through a wire, you get a magnetic field around it. (Interestingly, not only did Ørsted figure this out (unless you believe that Romagnosi did it), but he also wrote papers on aesthetics – and he was the first person to isolate the element aluminium – and he founded Den Polytekniske Læreanstalt which, today we call the Danish Technical University. So, he had a big influence on B&O in many respects.)
Nowadays, we know that, by putting current through a wire, you produce a magnetic field that has magnetic lines of force that encircle the wire. The direction of the lines of force are dependent on the direction of current, and the extent by which the magnetic lines of force extend away from the wire is dependent on the amount of current.
Depending on whether or not you believe Benjamin Franklin you can use your right or left hand to determine the direction of the lines of force. Figure 1, below, shows a right hand (which indicates that we believe Benjamin Franklin and we say that current runs from the positive terminal of a battery to the negative terminal, which is, in fact, incorrect.). The thumb points in the direction of the current and your other fingers wrap around the wire in the same direction as the magnetic lines of force (which go from North to South on a magnet).
Fig 1. The right hand rule shows the direction of the magnetic lines of force around a wire as a result of putting current through it.
If you take that same wire, keep running current through it, and coil it up like a spring, you can make a slightly more useful magnet that actually has a North pole and a South pole. Again, you can use your right hand to figure out which end is which – you wrap your fingers around the spring in the same direction of the current going through the wire, and your thumb will be pointing towards the North pole of the magnet, as is shown in Figure 2.
Fig 2. If you have a coil of wire and you put current in it, you get a magnetic field. Note that, if you run the current the other way (by reversing the battery), the magnetic poles will reverse, so North will be on the left of this diagram.
Now, if you take two magnets and you put them end-to-end with the North of one facing the North of the other (or South to South), they’ll push each other apart. If you put them North-to-South, they’ll pull each other together.
So, let’s do something weird. We’ll make a coil of wire, and we’ll put it in a strange-looking permanent magnet that is a bit like a horseshoe magnet that has been wrapped around itself to make a circular plug in the middle which is one pole (say, North) and a ring around it which is the other pole (say, South) as is shown in Figure 3.
Fig 3. A coil of wire about to be put in the gap of a strange-looking permanent magnet.
Now, if I put current through the wire, I’ll make a magnetic field around it that will either push against or pull towards the magnetic field of the permanent magnet around it. In other words, if it’s free to move, it will.
Now, since a loudspeaker, generally, is a thing that is used to convert electrical energy into acoustical energy; and, since in order to create acoustical energy (i.e. make noises) we need to move air molecules; we can use this strange device we’ve made in Figure 3 to our advantage. However, let’s be a little more methodical about this…
So, let’s build a dynamic moving coil loudspeaker driver, bit by bit. We’ll start by talking about its name. The “dynamic” part means that the basic principle that does the work is electromagnetism (as opposed to electrostatics or some esoteric methods like using plasma, tesla coils, or cats). The “moving coil” part is because, uh, the part of the device that moves in the magnetic field of the permanent magnet is a coil of wire.
What we want to do is to put the wires of the coil inside a magnetic field that is as strong as we can make it (within reason, of course). The easiest way to do this is to make the “gap” the coil sits in as small as possible (and, of course, to use as strong a magnet as we can fit or lift or afford to buy). So, let’s make a small gap for the coil to sit in.
We start by making a “bottom plate” and connect a “pole piece” – this results in a shape that looks like a disc with a cylinder. It’s made out of soft iron because soft iron is a really good magnetic conductor. (In other words, if you stick a magnet on a piece of soft iron, the soft iron basically becomes an extension of the magnet without losing very much magnetic force.) That bottom plate and pole piece assembly is shown in Figure 4, below. I’ve made it red just to keep things clear later. It’s usually not red in real life.
Fig 4. The bottom plate and the pole piece, both typically made of soft iron.
As you can see already, the “plug” in the middle of the magnet in Figure 3 is already visible as part of the pole piece in Figure 4. However, in order to make the strength of the magnetic field greater (in other words, in order to concentrate the magnetic lines of force) we want to make the gap (where the coil is going to sit) narrower. This can be done by making the cylinder on the pole piece a little bigger in diameter – but only where the coil of wire will be. That’s done by putting a ring around it, as is shown by the blue part in Figure 5, below.
Fig 5. A ring has been added around the pole piece to reduce the gap width. (Note that the gap doesn’t exist yet – we’ll need to put in a couple of more pieces first.)
Now we add the magnet as you can see in Figure 6.. This looks like a ring that sits on the disc part of the pole piece. The top of the ring is one pole (say, South) and the bottom is the other pole (say, North) of the magnet. However, this means that the North pole of the magnet is extended to the top of the cylinder on the pole piece because (as I said earlier) the soft iron is a good magnetic conductor.
Fig 6. The blue ring is the permanent magnet, typically made of ferrite or neodymium.
You can see in Figure 6 that the gap between the top of the pole piece and the magnet is pretty big, so let’s make it smaller by putting a “top plate” on the top of the magnet. This is another disc of soft iron, where the hole is just a wee bit bigger than the diameter of the ring around the top of the pole piece as shown in Figure 7. This means that the South pole of the magnet is now the inside edge of the hole in that disc, so we’ve made a circular gap (between the top plate and the ring on the pole piece) that is very small, and therefore has a very concentrated magnetic field.
Fig 7. The top plate, also made of soft iron.
Unfortunately, we can’t just make a coil of wire and stick it in the gap and hope that it’s going to behave. Instead, we take a roll of cardboard (or something else) – a bit like the cardboard tube in the middle of a roll of toilet tissue – and wrap the coil of wire on that. That cardboard roll that supports the coil is called the “former” – it’s shown in Figure 8.
Fig 8. The light blue tube is the former, around which the voice coil is wound. You can’t see the voice coil because it’s hidden by the top plate. (Actually, you can’t see it because I didn’t draw it – but if I had, you’d just see some wires sticking out from the gap – depending on the type of coil we had.)
One little extra piece of information here. Since the voice coil, sitting in a magnetic field is the system that essentially converts electrical energy into movement, we call it the loudspeaker driver’s “motor”. Of course, it isn’t a motor that causes something to spin – but it does cause something to move.
Great. Now we have the coil of wire (the “voice coil”) wrapped around the former, sitting in the magnetic field. So far so good. Now we can put current through the wire and it will want to move in or out of the magnetic (depending on which direction we’re sending the current in). Now, our first problem is that, even if the voice coil and former moved out and in, there is nothing there to push and pull the air molecules in front of it – so it won’t make a lot of noise. So, let’s start putting up a surface that can move some air. We’ll start by putting on a “dust cap” which seals off the end of the former. This is the bump that you see on the front of a woofer in the middle of the cone – so we’re starting to get out to the visible “pretty face” of the loudspeaker driver. The dust cap is shown in Figure 9. Note that the dust cap is not always the same diameter as the former. Note as well that it is usually, but not always convex. Note as well that some drivers don’t have a discrete dust cap (like the BeoLab 3 woofer, for example).
Fig 9. The dust cap has been added to the front of the former.
Now we have a problem. We can put current into the coil and get it to move, but there is nothing there to stabilise it. What we need is something to make sure that it doesn’t fall down when you put the loudspeaker on its edge (as most are…). So, we’ll put in a stabiliser. It has to keep the former centred in the magnetic gap, but it also has to be flexible to allow the former to move in and out of the magnet. This part of the loudspeaker is called the “spider” – it looks like a disc that has wiggles in it that can stretch as the former moves in and out. This spider is shown attached to the former in Figure 10. Note that its outside will attached to something else, later.
Fig 10. The spider has been added. It is glued to the former, but is not attached to the coil or the top plate.
Welcome to later. Now we need a frame to attach the outside edge of the spider and some other parts of the loudspeaker to – as well as to allow us to attach the whole loudspeaker to a cabinet. This part is called the “basket” – it doesn’t do much other than act as a structural support for everything – a bit like the steel beams in a building. The basket is shown in Figure 11. It may be interesting to note that the basket for an automotive loudspeaker driver is a little different from one for a home loudspeaker because it has to be able to deal with the possibility of a nasty accident. For example, a friend who knows such things once told me that it’s a bad idea to put a woofer intended for a home loudspeaker in a car door because if you’re ever in a side impact collision, it’s not inconceivable that the magnet will rip away from the basket, shoot across the car and cut your legs off. So now I’ve warned you…
Fig 11. The basket is glued and/or riveted to the top plate. In addition, the outside edge of the spider is glued to the basket.
Now we can put the rest of the loudspeaker parts on. We attach a “diaphragm” or “cone” which makes the moving surface bigger. That’s the medium-dark green part in Figure 12. If we left it at that, when we moved the loudspeaker in and out of the magnet, it would sag, because the spider isn’t strong enough to keep the whole thing vertical. So, we add a “surround” which is usually made of foam or rubber (or fabric, in the old days). The surround is a flexible ring that is glued to the basket and the edge of the diaphragm. It’s the lightest green thing in Figure 12.
Fig 12. An entire moving coil loudspeaker. The light green ring is the surround and the darker green ring inside it is the diaphragm or speaker cone, glued to the top of the former and the dust cap.
So, now when you put current through the voice coil, it pushes out of (or pulls into) the magnet and moves the former, dust cap and diaphragm with it. This causes the spider and the surround (usually grouped into what we call the “suspension”) to stretch.
Fig 13. A cross section of a (not very) simplified model of a moving coil dynamic loudspeaker driver.
If we take the device in Figure 12 and cut it in half, we get a cross section like the one shown in Figure 13. And, just to prove that I’m not lying, I cut apart a real woofer – it’s shown in Figure 14. And then, not satisfied that I had done enough damage, I did it again to a BeoLab 3 woofer – those photos are in Figures 15 to 19. Another good example is this picture.
Fig 14. An actual moving coil dynamic loudspeaker, after I was very mean to it.
Fig 15. A BeoLab 3 woofer – after I was finished with it… You can see here that this particular loudspeaker driver does not have a separate dust cap and diaphragm. Also, you’ll notice that there is a flared cone that is used to connect the former to the outside edge of the diaphragm.
Fig 16. A BeoLab 3 woofer, showing some of the components.
Fig 17. A BeoLab 3 woofer, showing some more of the components. The magnet assembly is hidden inside the silver can at the bottom of the photo.
Fig 18. A BeoLab 3 woofer, showing some more of the components.
Fig 19. A BeoLab 3 woofer, showing some one more component.
That’s about it for this week. If you want to do a little more digging for yourself, you can look into things like the difference between overhung and underhung voice coils, neodymium vs ferrite, or just watch some relaxing, cool, tangentially-related videos like this one or this one or this one or this one. Or maybe just this.
Addendum
For the purposes of this explanation, I said that the top of the pole piece is the North pole of the permanent magnet, and the top plate’s inner edge is the South pole. However, there is no fixed convention for this. Manufacturers will almost always ensure that, when you put a positive voltage on the positive terminal of the loudspeaker, the diaphragm will move outwards. However, the north/south-ness of the magnet and the direction the voice coil is wound, and which end of the wire goes to which terminal vary not only from manufacturer to manufacturer, but model to model within one manufacturer’s portfolio.
One question that I am occasionally asked is why the BeoVision 11 is not able to decode audio signals encoded in Dolby TrueHD and DTS HD-Master Audio. I am not able to answer this question. However I can discuss what the implications are with respect to audio and, more specifically, audio quality in a playback system.
Before we start this discussion, let’s get a couple of terms defined:
LPCM: Linear Pulse Code Modulation. This is what most people call “uncompressed digital audio”. It’s the way digital music is stored on a CD, for example. It’s also the way digital audio is encoded to be able to send it around a computer or digital signal processor and do things like filter it, mix it, or just change the volume. So, at some point, in any system that has any digital audio signals anywhere in it (even if it’s just to change the volume, for example), it will be sending the signals around as LPCM-encoded audio.
CODEC: COmpression-DECompression. This is a way of encoding the LPCM audio signal so that it takes up less space (or less bandwidth). This is the audio equivalent of converting something to a “ZIP” file – sort of. Some CODEC’s are “lossless” – meaning that, if you take a signal, compress it and decompress it, you get back everything you put in – a bit-for-bit match of the original. (If you didn’t, it wouldn’t be “lossless”). Other CODEC’s are “lossy” – meaning that, when you compress the signal, some stuff is thrown away (this is why audio professionals call lossy CODEC’s like MP3 “data reduction” instead of “audio compression”). Hopefully, the stuff that’s thrown away is stuff you can’t hear – but that debate is best left out of this discussion.
Bitstream: Some Blu-ray players allow you to choose whether you send the audio data that’s on the disc directly out its digital output, unchanged OR to decode the audio data to LPCM before sending it out. Typically, this shows up in the menus on the player as a choice between sending “bitstream” or “LPCM”. So, for the purposes of this article, I’ll use these two terms as meaning those two things. However, if we’re going to be accurate, this is an incorrect definition, since an LPCM signal is a stream of bits, and therefore is also a bitstream. But now I’m being purposely punctilious…
Let’s start this discussion by building a fairly standard home theatre system. We have:
A Blu-ray player connected to…
A BeoVision or BeoPlay television or a BeoSystem X OR some other brand’s AVR (Audio Video Receiver) or Surround Processor via an HDMI cable
which is connected to…
A surround sound setup with 5 or 7 main loudspeakers (some of those might be inside the television) and maybe a subwoofer, all connected to the television using power amp, or line level (or PowerLink, for B&O customers) connections
Let’s also assume the following:
The Blu-ray player is connected to the television’s input with an HDMI cable.
Both the Blu-ray player and the television (or AVR or Surround Processor) have been certified by Dolby and DTS to decode all the encoding formats in which we’re interested for this discussion
(note that this is not true of all devices in the real world – but we’ll use the assumption for now).
The player does what you tell it to do. In other words, if you set its output to bitstream, it sends the stream of bits that are on the disc it’s playing – and nothing else.
The question is: What format should I use to send the audio signals from the Blu-ray player to the television? Should I set my Blu-ray player to send a Bitstream to the television and decode the audio signals there? Or maybe it’s better to tell the Blu-ray player to decode the signals and send LPCM to the television.
The answer, unfortunately, is potentially complicated… but let’s look at what the implications are for the two options.
Audio Quality
If you do a search on the web, you’ll find all sorts of answers to this question. Some of the answers are correct, some are partly correct, and some are just plain wrong – actually, some aren’t even wrong.
The basic reason for this is that Bang & Olufsen, like every other company that manufacturers audio/video components that support these formats must adhere to a very strict set of regulations that are set by Dolby and DTS in order to receive certification for our products. Also, before we can deliver a new product to our dealers, it must be thoroughly tested by both Dolby and DTS to ensure that we meet their exacting standards for their formats. In other words, a Dolby TrueHD decoder is a Dolby TrueHD decoder – regardless of what the box it’s in looks like.
This means that every device that is certified to decode Dolby TrueHD or DTS-HD Master Audio and convert that format to an uncompressed PCM audio signal (footnote: the “standard” format that is used by the internal Digital Signal Processing (DSP) of the device) does that decoding in the same way. This is true whether the device is a Blu-ray player, a DVD player, an AVR, or surround sound processor. Note that this does NOT, however, mean that a loudspeaker connected to any of these devices will give you the same sound quality – there are many, many other links in the audio chain that have an effect on sound. It merely means that the conversion from a compressed (either lossy or lossless) signal to an uncompressed signal will be identical, regardless of where it’s done.
So, whether:
you decode in the player and send PCM to the television, OR
you send the bitstream to the television and decode the signal there,
there is no difference in audio quality
Now, you might tell me “But I went to a user forum and saw a posting from a person who did a test with his Acme Blu-ray player and his Flybynight AVR and he reports that he can EASILY hear the difference in quality in the audio when he changes from LPCM to Bitstream on his player.” This may, in fact, be true. However, the reason that there is an audible difference is not due to one of the devices having a better decoder or some very minor issue like jitter on the HDMI signal. There are at least two very basic and very simple explanations as to why this might happen.
The first is a simple level difference. If you do an ABX test of a system where A or B is 1 dB louder than the other, but are identical in every other aspect, your listening test subjects will have no difficulty identifying what X is. (An ABX test is one where you can listen to 3 things – an “A”, a “B” and an “X”. “X” is guaranteed to be identical to either “A” or “B” (which are guaranteed to be different). Your task is to identify whether “X” matches “A” or “B”. So, if the Bitstream is just 1 dB (or less!) louder than than the LCPM version, then you will hear a difference – although you might not notice that the difference is a simple loudness. It’s likely that you will perceive the louder one as just being “better”.
The second is that some AVR’s can be set to apply different post processing to LPCM signals than they do to signals decoded internally. So, it’s possible that there could be something as simple as a bass or treble difference between the two signals – but it could be something as complicated as a lot of spatialisation and reverberation with a big EQ curve – or anything in-between. So, in this case, the difference between Bitstream and LPCM coming in from the player is in the post-processing differences in the AVR.
However, as I said, this is not the full story. Let’s look at more pieces of the puzzle to see what the differences are.
Audio during fast-forward / fast-rewind
Different players behave differently when you are fast-forwarding or fast-rewinding through a disc. If you are sending an encoded bitstream from the player, many devices will not deliver any audio when you are moving through your disc at a higher speed. This behaviour is different from brand to brand and model to model. However, in many cases, these same players WILL deliver audio if the output is set to PCM.
Listening Levels
In a theory, there should be no difference in audio levels (i.e. how loud the signals are) caused by moving from a decoder inside your player to one inside your television. However, as I mentioned above, this world isn’t perfect – and one of the results of that imperfection is that there may, indeed, be a difference in level caused by switching from a bitstream to a PCM output from the Blu-ray player. If this does happen, then it’s likely because of some extra (or different) post-processing that is happening to the signal(s).
Mixing of Extra Audio Channels
One of the cool features of DVD and Blu-ray (that I, personally, never use) is that you can watch a movie whilst listening to the director (or someone else) talk about the movie – a feature usually called “Commentary” or something like that. This is fun, because there’s nothing that can make IronMan 3 more interesting than hearing about how someone accidentally spilled a cup of coffee on their computer keyboard while they were rendering a 3D CGI version of an Audi falling into the ocean.
In order for this feature to work, the system has to take the audio signals for the movie and mix in an additional audio channel. However, some players are not able to mix these two sounds together if they’re outputting a bitstream. They can only do it if they’re decoding the movie AND the commentary separately, and then mixing the two LPCM streams. Of course, to then re-encode the result, just to send out a bitstream would be silly.
Another example of these extra sounds that may not make it through a bitstream output are the “clicks” or cute noises that are assigned to menu items.
So, with some players, if you want to hear these extra sounds, you’ll have to output a decoded LPCM stream.
Channel Allocation and Routing
This is where things get a little debatable. Usually, when people say “5.1” they mean the following channel allocations (in no particular order)
Left Front
Centre Front
Right Front
Left Surround
Right Surround
LFE
And, when they say “7.1” they mean
Left Front
Centre Front
Right Front
Left Surround
Right Surround
Left Back
Right Back
LFE
We’ll call these the “standard audio channel allocations” for this article. However, as I talked about it a previous article, “7.1” actually has seven “legal” variants that include non-standard audio channel allocations like front-wide and height channels.
Let’s assume, temporarily, that you have a disc that has some audio channels on it that should be routed to, say, a ceiling loudspeaker. The question is: “IF your surround processor has a “ceiling speaker” output, and IF there is a “ceiling” audio channel on the disc, can the signal get to the correct loudspeaker?” The answer is complicated. . .
Firstly, the question is “how can the disc ‘know’ that the audio channel is a ‘ceiling’ channel?” Well, both Dolby TrueHD and DTS HD-Master Audio (for example) have the option to include “metadata” (this is a fancy word meaning “data about the data” – or, in other words “information about the audio signals”) that can tell the decoder something like “audio channel #6 should be sent to a ceiling loudspeaker”. Other CODEC’s (like Dolby Digital, for example) do not have the possibility to have non-standard audio
Secondly, the question is “if the player decodes the signal to LPCM, and its decoder knows that the channel should be routed to the ceiling, can it ‘tell’ the surround processor via the HDMI metadata where to route the signal?” The answer to this question is dependent on the version numbers of the HDMI transmitter in the player and the HDMI receiver in the surround processor. If you are using HDMI 1.3 or earlier, then you cannot have non-standard channel allocations with an LPCM signal. If you have HDMI 1.4 or higher (for both the transmitter and the receiver) you might be able to get the correct metadata across from device to device. (if you look here, you can see that, if your HDMI transmitter or receiver is version 1.2 or earlier, then you cannot send the Dolby or DTS lossless codec’s – so this will also not work for non-standard channel allocations.)
So, the only way to guarantee that your complete system can support non-standard audio channel allocations (assuming that your surround processor has the ability to output them) is to send a bitstream from the player and decode at the end of the chain.
However, the question to ask after you’ve answered all of that is “how many commercially-available recordings include non-standard audio channel allocations?” The answer to this, as far as I’ve been able to figure out is “none”. (If you know of any examples of this that prove me wrong, please leave a note in the “replies” – I’m looking for materials! But read the rest of this paragraph first…) Of course, there are SACD’s where the LFE channel should be directed to a height channel – but SACD’s don’t include metadata to tell the player about the routing. Dolby ProLogic IIz and dts Neo:X have height channel outputs, but that’s a different system than a CODEC with discrete output channels. There are some other formats like Imax, Auro3D, and Dolby Atmos that use height channels, but they’re not available in consumer media.
So, this, at least for now, is an solution without a problem.
Bass Management
Some Blu-ray players (and even some good ol’ DVD players) include a bass management system that will filter the bass out of the main channels and add it to the LFE output.
IF your player can do this, then:
it can only do it to a decoded signal – so the bass management in the player will not work with a bitstream output
you should be sure that it is doing it (if you want it to do so) or that it is not doing it (if you don’t)
Latency
Some forum discussion groups have highlighted an issue with some specific players that exhibit lip-synch problems when they decode the signal to LPCM internally. The few comments about this that I have read in these fora indicate that switching the player’s output to bitstream appears to fix this problem. However, this should be considered a “work-around” for a bug in the player’s software. There should be no difference in synchronisation of sound and picture whether you’re decoding in the player or the surround processor. If there is a lip synch problem, then the people that made the player haven’t done their jobs properly.
Conclusion
So, to wrap up: the big things to remember here are that,
in any audio playback system, audio that is stored (and/or transmitted) in a CODEC has to be converted to LPCM somewhere
there is no difference in quality of decoder – in other words a Dolby (or DTS) decoder in one device won’t be better than a Dolby (or DTS) decoder in another device. If it were, then Dolby (or DTS) would not have approved one of them
there are some issues not related to audio quality that are affected by the location of the decoder in an audio chain, but these are typically very small (or even non-existent) issues for almost all consumers
Finally, I didn’t talk about jitter – which is term that a lot of people throw around as being one of those evils in every audio system where you can place all of your blame for everything that is bad about the system. This puts it in a league with things like “society“, “television“, “Canada“, and “The Boogie“. I’ll talk about that sometime in the future.
One last thing
Everything I’ve said above is only true for an HDMI connection. If you have an S/P-DIF or TOSLINK connection, then the discussion will be quite different, since you cannot have more than 2 channels of LPCM-encoded audio on those systems – so the only way to get multichannel audio through it is to use a lossy CODEC. So, if you have a multichannel source and you decode to LPCM before sending it out on S/P-DIF or TOSLINK, you will wind up with a 2.0 channel downmix.
Makes: One individual with reduced faith in loudspeaker reviews
Total time: approximately 2 hours
Directions
Take a woofer and put it in a cabinet
Connect an amplifier to it
Put it in a sauna
Set the room temperature to 20° C and wait until everything in the room is the same temperature
Measure the woofer’s on-axis response with a microphone
Look at the pretty plot of its magnitude response
Turn up the thermostat to 100° C and wait until the woofer warms up
Measure the response again
Look at the new pretty plot of its magnitude response
Scratch your head while you ask yourself why the two measurements look so different.
The setup
When you read a magazine review of a loudspeaker, it will include a measurement of its “frequency response” (more accurately called its “magnitude response”) which shows (ignoring a bunch of things) how loud different frequencies are when they come out of the loudspeaker assuming that they all came in at the same level.
However, as we saw in a previous article, for a Bang & Olufsen loudspeaker, this magnitude response is dependent not only on the loudspeaker, but how loudly you’re playing the signal.
Unfortunately, it gets much worse than this… For example, if we take a woofer (say, the one from the recipe above, for example) we can explain its electromechanical characteristics by breaking it down into different components (both actual and analogical). For example, the suspension (which is comprised of the surround and the spider) can be thought of as a spring. The electrical analogy for this is a capacitor. If you take all of the moving parts in the loudspeaker driver, they all add up to a mass that has to be moved – the electrical analogy for that mass is an inductor (since an inductor has some electrical “inertia” just like the mass of a bunch of loudspeaker bits). Some of the components are not an electrical analogy – they are real electrical components. For example, the voice coil, since it’s a coil, acts as an inductor. And, since it is a thin bit of wire, it also has some resistance to the flow of electrical current through it, so it’s also a resistor.
Fig 1. A simplified version of the actual electrical and electrical analogies of mechanical components in a loudspeaker driver.
If you look at the diagram above, you’ll see a very simplified “circuit” that shows the components of a moving coil dynamic loudspeaker. If these components don’t look familiar to you, don’t worry, it’s not important. Some components in the circuit are actual electrical things (like the resistance of the voice coil, shown in red) and others are analogies – electrical representations for a mechanical component in the system (such as a capacitor representing the “spring” of the surround and the spider).
If you know how each of these components behaves, and you know the correct values to put in for a given loudspeaker, and you know how to do the right math, then you can come pretty close to getting a decent prediction of the response of the loudspeaker that you’re modelling with the circuit. However, if you just put in one value for each component, then you’re assuming that they never change – in other words that you’re dealing with a “linear” system.
The problem is that this assumption is incorrect. For example, the Voice Coil Resistance – the amount that the wire in the voice coil resists the flow of current through it when the loudspeaker driver is not moving – changes with temperature. The hotter the wire gets, the higher the resistance goes. (This is a normal behaviour for most resistors.) If the voice coil resistance changes, then the whole system behaves differently, since it isn’t the only component in the circuit. So, as we change the temperature of the voice coil, the total response of the loudspeaker changes.
Sadly, the temperature of the voice coil isn’t only dependent on the room temperature as it seemed to be in our recipe for a Befuddled Speaker Enthusiast. As soon as you start playing sound with a loudspeaker, it starts heating up. The louder the signal (either because you turned up the volume or because your Metallica album just came on) the hotter it gets. So as you play music, it heats and cools. The speed with which it heats up and cools down is dependent on its “thermal time constant” – a big woofer with a giant magnet will take longer to heat up and cool down (and therefore have a longer thermal time constant) than a little tiny tweeter.
So, now you should have at least three questions that deserve answers:
How much does the temperature vary when I play music?
How does the response of the loudspeaker change with temperature?
How much does the response of the loudspeaker change with temperature?
What are you going to do about it?
1. Voice coil temperature
As I’ve talked about in a previous article, a loudspeaker driver is, give or take, about 1% efficient. That means that 99% of the power that you push into it (from the amplifier) is not converted into sound. Unfortunately, all of that power is lost as heat – almost all of it at the voice coil of the loudspeaker. So, the louder your music, the hotter your voice coil gets. Of course, if you have a way of cooling it (by using other parts of the loudspeaker as a radiator to your listening room) then it won’t get as hot, and it will cool down faster.
Let’s take a BeoLab 5 as an example (since that’s where we’re headed anyway…). Let’s take some relatively new-ish pop music (which has been mastered to be fairly loud due to a war that has been going on for years) and play it on a B&O player through Power Link (B&O’s version of a line level signal) at maximum volume on a BeoLab 5 whilst monitoring the temperature of the voice coils. What you’ll see if you do this is something like the plot below.
Fig 2. The temperatures (in °C) of the voice coils of the four drivers in a BeoLab 5 as a result of playing pop music at full volume on a BeoSound 5. The X-axis is the time in minutes. (green = tweeter, light blue = midrange, dark blue = mid woofer, red = woofer)
As you can see in the screenshot in Figure 2, the lower woofer (a 15″ driver connected to a 1000 W Ice Power amplifier) varied with a maximum of about 205° C. While it was playing this music at this level, it rarely dropped below 120°C.
This means that the difference in temperature of the woofer was 185°C at a maximum (205°C – 20°C) and rarely below 100°C.
In case you are wondering, this temperature cannot be measured directly, since it would destroy the voice coil if we tried to do so. Instead, what we do is to measure the temperature of the loudspeaker driver magnets, and use that real-time data input in addition to the signal that we’re sending to the drivers to calculate the temperatures of the voice coils based on thermal models of each of them. As you can see in Figure 3, below, the magnet temperatures are very different, and react much more slowly. These measurements were taken at exactly the same time as the ones shown in Figure 2. (Note that, although the mid woofer and woofer voice coils are roughly the same temperature, the mid woofer magnet is hotter than the woofer magnet by about 20°C or so. This just proves that their thermal models are different.)
Fig 3. The temperatures (in °C) of the magnets of the four drivers in a BeoLab 5 as a result of playing pop music at full volume on a BeoSound 5. The X-axis is the time in minutes. (green = tweeter, light blue = midrange, dark blue = mid woofer, red = woofer)
2 & 3. Loudspeaker response changes
So, now the question is “what does this change in temperature do to the response of the driver?” We’ll only deal with one driver – the woofer.
As I mentioned above, the thing that changes most in the model shown in Figure 1 is the loudspeaker driver’s voice coil resistance. For those of you with a background in reading electrical circuits, you may notice that the one shown in Figure 1 has some reactive components in it which will result in a resonance at some frequency. For those of you without a background in reading electrical circuits, what this means is that a loudspeaker driver (like a woofer) has some frequency that it “wants” to ring at – if you thump it with your thumb, that’s the frequency that you will hear ringing – a bit like a bell with a low pitch.
As the voice coil resistance goes up, its resistance increases, and we generally lose sensitivity (i.e. level or loudness) from the woofer. In other words, the hotter it gets, the quieter it gets. However, this only happens at the frequencies where the resistor is not “overridden” by another component – say the mechanical resonance of the woofer or the inductance of the voice coil.
The total result is shown for various temperature differences in Figure 4. Notice that these plots show the change in magnitude response of the driver with CHANGES in temperature. So, the blue curve at the top is the change in magnitude response (which is 0 dB at all frequencies – in other words no change) when the loudspeaker is playing at the same temperature it was measured at (let’s say, 20°C or room temperature). As the temperature of the voice coil increases above that temperature, you can see that you lose output in two frequency bands on either side of a “bump” in the response – this is at the resonant frequency of the loudspeaker driver.
So, the louder you play, the more low end you lose, apart from a peak in the response (which also rings in time) at the resonant frequency of the driver.
Fig 4. Sensitivity of an example woofer vs. the change in temperature of its voice coil in degrees Celsius
In case you’re wondering, the plot shown in Figure 4 is pretty close to the actual response of the 15″ woofer in the BeoLab 5 at different temperatures above room temperature.
The solution
Interestingly, all of the stuff I said above is true for every loudspeaker. So, if you’re the kind of person who believes that the only proper loudspeaker is one where you have nothing but a loudspeaker driver (in a cabinet of any kind, or not) and an amplifier – and no weird filtering or mucking-about going on, then you’ll have to live with the kind of unpredictable behaviour that you see above. This happens all the time to every dynamic loudspeaker. Since, in a passive loudspeaker, there’s nothing you can do about this (except for trying to keep the drivers cool somehow) you don’t often hear passive loudspeaker manufacturers talking about this little skeleton in their closet…
However, since a BeoLab 5 “knows” the temperature of the voice coil of the woofer, and since it has been programmed with the curves shown above in Figure 4, we can do something about it.
In essence all we need to do is to take Figure 4, flip it upside down and make a filter that “undoes” the effect of temperature on the loudspeaker’s response. In other words, if (because the woofer gets 160°C above “normal”) it drops 3 dB at 20 Hz, the BeoLab 5 knows this and adds 3 dB at 20 Hz. So, built into the BeoLab 5 is a set of filters that are used, depending on the temperature of the woofer’s voice coil. These filters are shown in Figure 5.
Fig 5. Magnitude responses of the compensating filter for the woofer from the previous plot vs. the temperature of its voice coil in degrees Celsius
It’s important to note three things here.
This can be done because we know the behaviour of the woofer at different temperatures (this was measured as part of the development process)
This can be done because the loudspeaker “brain” (the DSP) knows the temperature of the voice coil in real time as you’re playing music
This filter should only be applied to the woofer. The mid woofer and the other drivers have different behaviours and should not be affected by this correction curve. Therefore, this filtering can only be done because the BeoLab 5 is an active loudspeaker with independent filtering for each loudspeaker driver.
Some extra information
You should be left with at least one question. I said above that, as the music gets loud, the woofer heats up, so you lose output, so we add a filter that compensates by putting more signal into the driver.
“Waitaminute!” I hear you cry… “The problem is caused by the signal being too loud, so you make it louder!?” Well… yes.
However, there is one more trick up our sleeve. In a previous posting, I mentioned in passing that we also have Thermal Protection in almost all of the loudspeakers in the B&O portfolio. This means that the DSP brain knows the temperature of the drivers and, in a worst-case situation, turns the levels down to protect things from burning up. So, if we go back to the example of a BeoLab 5 playing at full volume, let’s see what’s happening to the signal levels.
Fig 6. The gains (in dB) applied to the signals sent to the four drivers in a BeoLab 5 as a result of playing pop music at full volume on a BeoSound 5. The X-axis is the time in minutes. (green = tweeter, light blue = midrange, dark blue = mid woofer, red = woofer)
These curves in Figure 6 show the gains applied to the four loudspeaker drivers in a BeoLab 5 at the same time as the measurements shown in Figures 2 and 3 were being made. In fact, if you look carefully at Figure 2 around the 23 minute mark, you’ll see that the temperature dropped – which is why the gain in Figure 6 increases (because it can!) in response.
Now, don’t panic. The BeoLab 5 isn’t screwing around with the gains of the drivers all the time. Remember that this test was done at FULL VOLUME – which, for a BeoLAb 5 is VERY LOUD. The gains shown in Figure 6 are a last-ditch effort of the loudspeaker to protect itself from a very mean customer (or the very mean children of a customer who is away for the weekend). This is the equivalent of the airbags deploying in your car. You know that if the airbags are outside the steering wheel (or if the gains in the BeoLab 5 dropped by 15 dB or so) something significant occurred…
Thanks to Gert Munch for his help in cleaning up the mistakes I made in the drafts up to and including the penultimate version of this article.
I listen to a lot of recordings – usually to find problems in loudspeakers, so it’s not very often that an album makes the hair on the back of my neck stand up.
This one does.
I haven’t yet been able to find out who the recording engineer(s) was(were) for this album, but to whomever it was – Thanks!
In a previous post, I talked about why a subwoofer might be a smart addition to a sound system – and why a subwoofer brings something different to a Bang & Olufsen loudspeaker configuration than it does for other companies’ loudspeakers.
Usually, in a loudspeaker system that includes a subwoofer, the signal that is sent to that subwoofer is either
coming in directly from the medium (say, the Blu-ray disc) from the LFE (Low Frequency Effects) channel OR
created by something called a “bass management system” which is basically a mixer (something that adds audio signals) and some frequency division OR
all of the above
Let’s assume, for the purposes of this article, that we’re talking about #3. So, let’s start by talking about how a system like that would work. The simple version is that you take an audio input, send it to two different filters, one called a “high pass filter” which lets the high frequencies pass through it and it makes the lower frequencies quieter. The second filter is called a “low pass filter” – you can figure that one out. The output of the high pass filter is sent to your “main” loudspeaker, and the output of the low pass filter is sent to the subwoofer.
Fig 1. Simple bass management algorithm for one audio channel
That’s what happens if you have a good ol’ fashioned monophonic system with only one audio channel, one main loudspeaker and one subwoofer. Most people nowadays, however, have more than one main loudspeaker and lots of channels coming out of their players. So, in cases like that, we have to take the low end (the bass) out of each of the main channels using low pass filters, add the results all together, add the LFE channel to that, and send the total to the subwoofer. A simple version (with some important details left out, since we’re only talking about basic concepts here…) is shown in the diagram below.
Fig 2. Simple bass management system for a 5.1 system
Basics of Signal Addition
Let’s take an audio signal and add it to another audio signal – and, just to keep things simple, we’ll make them both sine waves. This is done by looking at the amplitude of the two signals at a given moment in time, adding those two values, and you get a result. For example:
Fig 3. The bottom plot is the result of adding the signals shown in the top and middle plots, moment by moment in time. For example, the red stars show the values at one moment in time. One plus one (the top two plots) equals two (the bottom plot).
If you take a look at the example above, there are a couple of things that you can see. The first is that, if you add two sine waves, you get a sine wave. Also, if you add two sine waves of the same frequency, you get a sine wave of the same frequency. Next, if the two sine waves have the same amplitude and are “in phase”- basically meaning that they have the same value at the same time (sort of, but not really like a delay difference of 0) – the result is a sine wave that is in phase with the other two with double the amplitude of the two inputs.
Now, let’s move one of the two signals in time. We’ll make it late by half the length of the sine wave (in time) and see what happens.
Fig 4. The bottom plot is the result of adding the signals shown in the top and middle plots, moment by moment in time. For example, the red stars show the values at one moment in time. One plus one (the top two plots) equals two (the bottom plot).
Now you can see that, since the bottom plot is the negative of the top plot at any given moment in time (because we’ve delayed it by half a “wave”), when you add them together, you get no output.
Let’s look at one last example, where the two input signals have some phase difference that is not quite so simple. This is shown in the figure below.
Fig 5. The bottom plot is the result of adding the signals shown in the top and middle plots, moment by moment in time. For example, the red stars show the values at one moment in time. One plus one (the top two plots) equals two (the bottom plot).
Again, you can see that, when you add two sine waves of the same frequency, you get another sine wave of the same frequency. You can also see that the phase (remember – phase is something like delay) of the result is not the same as the phase of either of the two inputs. Finally, you can see that the amplitude (the maximum value) of the result is neither 0 nor 2 – it’s something in between.
So, the moral of this story is that, if you have two sine waves then the result (what you hear) is (at least partly) dependent on how the two signals add – and that result might mean that you get something twice as loud as either input – or it could mean that you get nothing – or you get something in between.
The real world
The three plots shown above illustrate some simple examples of what happens when you have two sound sources that are added together to produce a result. For the purposes of this article, the two sources are two loudspeakers – the “main” loudspeaker and the subwoofer, and the result is the sum of those two signals at your listening position. Let’s go back to thinking about what frequency ranges are produced by these two loudspeakers. The plot below shows the magnitude responses of the filters in a BeoVision 11’s internal bass management system at its default crossover frequency of 120 Hz. The red curve shows the response of the low pass filter whose output is sent to the subwoofer output. The black curve shows the response of the high pass filter whose output is sent to a main loudspeaker.
Fig 6. The magnitude responses of the filters in a BeoVision 11’s bass management system.
Take a look at the response of the signal that is processed by the low pass filter. You’ll notice that, although the “crossover frequency” is set to 120 Hz, there is still signal coming out of the filter (and therefore out of the subwoofer) above that frequency – it’s just getting quieter as you go further up and away from the crossover.
You might also notice that, at the crossover frequency, the level of the signal coming out of the low pass filter is identical to the level of the signal coming out of the high pass signal. That’s (more or less) what makes it the crossover frequency. As you move down from that frequency, you gradually get more out of the low pass (the sub) than the high pass (the main). As you move up in frequency, the opposite happens. However, there is a region around the crossover where both loudspeakers are contributing roughly equally (within reason) to the signal that you get at the listening position. So, the “truth” is a little more like the plot below:
Fig 7. A conceptual way of thinking about which loudspeaker is playing the audio signal.
I’ve chosen -20 dB as the point where I can start ignoring a signal, but that’s a pretty arbitrary decision on my part. I could have set my threshold higher or lower and still argued that I was right. So, if you disagree with my choice of -20 dB as the threshold of “I don’t care any more” then I agree with you. :-)
By now, you should start to worry a little. You should be asking something like “hmmmm… you’re telling me that there is a big band of frequencies (say, roughly between 1 and 2 octaves) right around where human voice fundamental frequencies sit (well, at least my voice sits there – but I sing bass…) where a bass management system will send the signal out of two loudspeakers!? AND, to add insult to injury, you told me (in the previous section) that if the phases and amplitudes of those signals from those loudspeakers aren’t perfectly aligned, the result at the listening position won’t be the same as the input of the whole system?” If you ARE asking something like that, then you’re in good shape. As has been said by many other people in the past: the first step in fixing a problem is admitting you have one.
So, let’s ask a different question: what parts of the audio signal chain could affect either the amplitude or the phase of the signals coming out of the loudspeakers? Brace yourself… This list includes, but is not exclusive to:
The characteristics of the filters in the bass management system
The characteristics of the filtering in the loudspeakers
The physical principal of the loudspeaker (i.e. sealed cabinets will be different from ported loudspeakers which are different from passive radiators)
Diffraction (although this might be a small issue)
Latencies (total delay) of the loudspeakers (for example, digital loudspeakers have a bigger delay than analogue ones typically)
Distances of the loudspeakers to the listening position
The characteristics of the room itself
And more!
All of these issues (including the ones that fall under the “And more” category) have some effect on the phase (and amplitude) of the signal that you hear at the listening position. And, since some of these (like the distances and the room characteristics) are impossible for us (as a manufacturer) to predict, we have to give you, the end user, some way of adjusting your signals so that you can compensate for misbehaviour in your final system.
Now, although this is a “technical” article, I think that it would be too technical to start looking at the specifics of the phase responses of sealed cabinet vs ported loudspeakers, for example, since the details will be messed up by the listening room anyway. So, instead of getting into too many details, let’s just say that “you can’t expect your system to work perfectly without tweaking it” (see the reasons above) and just talk about some strategies for setting up your system so that it behaves as well as it can (without going out and hiring an acoustical consultant).
BeoLab 19 Controls
The BeoLab 19 has a number of controls that have not been available on previous B&O subwoofers. As a result they may cause a little confusion and playing with them without knowing what to expect or listen for could result in your system not performing as well as it could. On the other hand, it could be that these controls could help you improve your system if you’re finding that it’s not really behaving. Let’s take each control, one by one, and explain what it does, and talk about strategy afterwards.
Fig 8. BeoLab 19’s control panel
Gain
The gain knob (on the left in the diagram above) is basically just a volume knob that controls how loud the subwoofer is overall. Let’s say, for example, that you have a perfectly configured system, then the outputs of the subwoofer and the main loudspeaker mate perfectly and result in a perfectly flat response below, through and above the crossover region. (this never happens in real life – but we can pretend). The diagram below shows an example of this, where the top plot shows the outputs of the subwoofer and main loudspeaker and the bottom plot shows the total result at the listening position.
Fig 9. The output of a subwoofer and a main loudspeaker, with a “correct” crossover, at the same distance, with the same gain, with no room acoustics to bother anyone…
If you do nothing but change the gain of the subwoofer, (using the Gain knob on the BeoLab 19, for example) then the result would be something like the plot below.
Fig 10. The output of a subwoofer and a main loudspeaker. The subwoofer’s gain has been increased by 6 dB. The distance to both loudspeakers is the same.
You can see in the plot above that all you do is to boost a region of low frequencies without doing anything strange through the crossover region. So, if you like bass, this might be a nice tweak for you. However, in theory, your goal is to get a response like the one in the first plot, where the outputs of the subwoofer and main loudspeakers have the same level (at the listening position).
LP Filter
One possible configuration of the BeoLab 19 is to connect it in parallel with your main loudspeakers and to not use and external bass management system.
Fig 11. A block diagram of the parallel method of connecting a subwoofer to a 2-channel stereo system.
If you do this, then you are relying on the fact that the main loudspeakers have a high pass filter built-in, and you will align the low pass filter inside the BeoLab 19 to have approximately the same frequency so that the total result is a smooth-ish crossover region. In order for the low pass filter to work, you will have to turn it ON using the switch. (Note that, if you’re using an external bass management system as in the BeoVision 11, for example, then you should turn the low pass filter OFF, thus removing it from the signal path of the subwoofer.)
In theory, the goal here is to match the cutoff frequencies so the two loudspeakers behave nicely together across the crossover region. For example, if the natural low frequency cutoff is about 50 Hz, and you set the LPF in the subwoofer to 50 Hz, then you get the result below
Fig. 12. The theoretical responses of a subwoofer with a low pass of 50 Hz (black curve) a main loudspeaker with a high pass of 50 Hz (blue) and the total sum at the listening position (red) in an imaginary world.
What would happen if you set the LPF incorrectly – let’s say that you make it higher than the correct value, since you would think that, by overlapping the sub with the main speaker, you’ll get more output and impress the neighbours. Well, the result would be the plot below.
Fig 13. The theoretical responses of a subwoofer with a low pass of 120 Hz (black curve) a main loudspeaker with a high pass of 50 Hz (blue) and the total sum at the listening position (red) in an imaginary world.
As you can see, although the sub is now delivering more signal (because it’s going all the way up to 120 Hz instead of 50 Hz in the previous plot), you actually get a reduction in the total output of the system. This may be initially counterintuitive, but it’s true in our example, since (as you may remember from something I said earlier in this article) the phase of the subwoofer is, in part, determined by the characteristics of the filtering in the loudspeaker. By changing the low pass filter frequency, we change the phase of the subwoofer in the crossover region and result in a cancellation with the main loudspeaker instead of a summing. In essence, both the sub and the main loudspeaker are now working very hard to cancel each other (especially around 80 Hz or so) and you hear very little at the listening position.
On the other hand, I have assumed here that the main loudspeaker’s high pass filter is a very specific type. A different main loudspeaker with a low frequency cutoff of 50 Hz would have had a completely different behaviour as you can see below.
Fig 14. The theoretical responses of a subwoofer with a low pass of 50 Hz (black curve) a different main loudspeaker with a high pass of 50 Hz (blue) and the total sum at the listening position (red) in an imaginary world. (Note that the slope of the high pass filter in the blue curve is different from Figures 12 and 13.)
So, the moral of the story here is that setting the low pass filter frequency will have some effect on your total response. However, you should not jump to the conclusion that you can predict what the frequency should be – you will have to fiddle with the knob whilst listening to or measuring the total output of the system. You should also not jump to the conclusion that increasing the frequency range that is covered by the subwoofer in a parallel configuration will result in more output from your system. Overlap is not necessarily a good thing – sometimes, more is less…
Phase
Go back up and take a look at the two sine wave in the plots in Figure 4. One way to describe these two waves is to say that the middle one is half a wave later than the upper one – in other words, they are 180º out of phase. Another way to describe them is to say that the middle one is the inverse of the upper one – they have the same instantaneous value at any time, except that they are the negative of each other (in other words, signal 2 = signal 1 * -1).
So, intuitively, you can see that shifting the phase of a signal by 180º is the same as flipping it upside down. This could mean that, for example, all other things being ignored, that when a kick drum tells your subwoofer to push outwards, shifting the phase by 180º will result in your subwoofer sucking inwards instead. However, this is only true if all other things are being ignored. As soon as your subwoofer has a high pass filter (i.e. a low frequency limit) and a low pass filter (a high frequency limit) and it’s a loudspeaker driver in a cabinet in a room, all bets are off. All of those aspects (and more!) will have some effect on the phase of the system, so you can’t predict whether the kick drum will cause the woofer to put out or suck inwards.
So, instead of worrying about the “absolute phase” of the subwoofer, it’s more interesting to worry, once again, how it matches up with the main loudspeaker. Let’s take exactly the same responses from the plots shown in Figure 14 above (which didn’t add together so well for some reason) and shift the phase of the sub by 180º using the Phase switch. The result is shown below in Figure 15.
Fig 15. The theoretical responses of a subwoofer with a low pass of 50 Hz (black curve) a different main loudspeaker with a high pass of 50 Hz (blue) and the total sum at the listening position (red) in an imaginary world. These are the same as the loudspeakers shown in Figure 14, however, in this case, the subwoofer’s polarity has been inverted by changing the “phase” switch to 180.
As you can see, the big dip in the total response of the system (seen in Figure 14) has been corrected, and we now have more output (actually, a little too much) below that. So, the result is that the total system response is much better than it was without flipping the phase switch.
Of course, if we flipped the phase switch in a system that was behaving nicely, then bad things might happen. Let’s flip the phase on the system shown in Figure 9, for example. That total result would look like the one in Figure 16, below.
Fig 16. The theoretical responses of a subwoofer with a low pass of 120 Hz (black curve) a different main loudspeaker with a high pass of 120 Hz (blue) and the total sum at the listening position (red) in an imaginary world. In this case, the subwoofer’s polarity has been inverted by changing the “phase” switch to 180.
As you can see, you get the same amount of low bass in Figures 9 and 16. However, there is a nasty dip at the crossover frequency of 120 Hz when the two loudspeakers are cancelling each other.
So, the moral of the story here is that, if you have a problem in the crossover region between the main loudspeaker and the subwoofer, flipping the phase of one of the two might help the situation – although it might make things worse…
Pos (aka Position)
Almost every loudspeaker in the Bang & Olufsen portfolio has a switch that lets you change the characteristics of the loudspeaker to compensate for the differences in its response as a result of its placement in a room. Generally speaking, the closer you put a loudspeaker to a wall, the more bass you’ll get out of it. If you put it closer to two walls (i.e. in a corner) you’ll get even more bass. However, that is a very general characterisation – the reality is that you’ll get a little more at some frequencies and a little less in other frequencies – and that behaviour is dependent on the diameters of the loudspeaker drivers, the crossover frequencies, and the physical shape of the loudspeaker.
So, without getting into the details of exactly what is being changed in a BeoLab 19 (or BeoLab 2 or BeoLab 11 – or any other loudspeaker for that matter), let’s say that you should put the position switch in whatever setting best corresponds to the location of the loudspeaker in your room. However, if you want, you could cheat a little and fiddle with the switch to see if you like another setting more.
For example, if your loudspeaker is in the corner, and you put it in “free” mode, you’ll get LOTS of bass – too much bass. But if you like bass, this is one way to get it. Of course, there are other implications to this decision, but if you like bass enough, that might be reason enough to change to the “incorrect” the switch setting.
Wired / Wireless
The BeoLab 19 has the ability to receive its input via the analogue or digital input OR via the wireless receiver module that is built into it. This switch merely tells the loudspeaker whether it should “listen” to the wired input (either analogue or digital) or the wireless one. (Note that the BeoLab 2 and the BeoLab 11 do not have wireless receivers.)
L / R/ L+R
Most subwoofers (including the BeoLab 2 and the BeoLab 11) are built with the assumption that you will use them either:
as a stand-alone subwoofer in a multichannel (i.e. 5.1 or 7.1) system where it gets the “.1” output from the source (that may, or may not have bass management) and so you just send one audio channel into it OR
in a 2.1 setup where you want the left and right channels coming into the subwoofer where they are added together produce a mono bass signal internally
Consequently, most subwoofers either have 1 input (assuming that they are to be connected to the “subwoofer out” on something like an AVR) or a 2-channel stereo input (assuming that they should “see” left and right) that is summed to mono. BeoLab 2 and 11 are built based on the second assumption.
BeoLab 19 allows you to use the subwoofer in either of these configurations. So, in either “L” or “R” mode, it is only “listening to” the Left or Right audio channel on the Power Link input. In “L+R” mode, the input of the subwoofer is taking both audio input channels and summing them to make a mono input to the subwoofer. Note that, if you send exactly the same signal on the Left and Right audio channels on the Power Link cable, and then you switch the BeoLab 19 from either L or R to L+R, you’ll find that you get a doubling in the output level. This is because a signal plus itself is twice as loud. Since this is what BeoLab 2 and 11 do all the time, if you simply replace a BeoLab 2 or 11 with a BeoLab 19, you should put the 19 in “L+R” mode – otherwise you’ll lose some bass in your system.
However, if you want to use a single Power Link cable to run to the Subwoofer and to another loudspeaker (say, a centre channel, for example), then you should put the BeoLab 19 in either L or R mode (and the other loudspeaker in the opposite mode) so that you can access both loudspeakers independently. This is also the case if you want to run two BeoLab 19’s on the same Power Link cable and use the 2-channel LFE output option in a BeoVision 11. In this case, you se one BeoLab 19 to “L”, the other to “R” and set the Speaker Roles in the BeoVision 11 to “Sub Left” and “Sub Right” (or “Sub Front” and “Sub Back”) appropriately.
Note that, if you are in Wireless mode, the “L/R/L+R” switch does nothing.
How to do it (Finally!)
Method for an Externally Bass Managed Configuration
If your main loudspeakers and your subwoofer are connected to a system that has a bass management system, then you should use it. There are a number of reasons for this:
the main loudspeakers may behave better (for example, with respect to distortion or port noise) if they are not being pushed by a lot of bass
a bass management system will work for a multichannel loudspeaker system
a bass management system (for example, in a BeoVision 11) will be capable of making some “intelligent” decisions with respect to your entire system
a good bass management system (for example, in a BeoVision 11) will allow you to make fine adjustments to accommodate your configuration and room
So, your procedure here (assuming that you have a BeoVision 11 and a BeoLab 19 and some main loudspeakers) is as follows:
Turn off the LP Filter on the BeoLab 19, set the Phase to 0, set the Gain to 0, and set the other switches to whatever is best for your particular configuration.
Put the correct loudspeaker models into the Speaker Connections menu on the BV11.
This will compensate for differences in the latencies and sensitivities of the loudspeakers, in addition to making some intelligent decisions about where to route the bass.
Set your Speaker Distances correctly
This will ensure that you do not have phase differences in the loudspeakers at the listening position as a result of problems caused by the speed of sound and mis-matched distances.
Set your Speaker Levels correctly
On a BeoVision 11, this is done by making sure that, at the same volume level, all loudspeakers produce the same level in “dB SPL, C-weighted, Slow” on an SPL meter like this one, or this one, or this one, for example.
Turn on a piece of music that has a constant bass level
The opening of Freddy Mercury’s “Living on my Own” or Santanta’s “You Are My Kind” are a possible tunes. Claire Martin singing “Black Coffee” is also a good candidate. If you want to look like a professional, then I suppose that you could use pink noise or this track instead of music.
Sit in the listening position and listen to the total behaviour of the system. Pay particular attention to “unevenness in the bass”. In other words, listen to the bass and pay attention to whether some notes are quieter or louder than others.
In theory, if you performed Step 4 correctly, then you shouldn’t have to play with the Speaker Level in the TV or the Gain on the subwoofer.
If there is a general area somewhere in the middle of the bass where lots of notes are too quiet, try flipping the phase on the subwoofer.
If some individual frequencies (or notes) are quiet then playing with the allpass filter on the TV might help.
If some individual frequencies (or notes) are louder than others, this is probably caused by the room, and you might be able to deal with it by moving the subwoofer. If, when you put the sub in the corner, you make this problem worse, it is almost certainly the room acoustics that you’re dealing with, so moving the sub is your best bet.
Method for a Parallel Connection Configuration
Since, in a configuration where the sub and the loudspeaker are connected in parallel, the behaviour of the transition between the BL19 and the main loudspeakers (let’s say that there are only two of them for this example) in the system is not only dependent on the loudspeaker models themselves, but also the distances to the 3 loudspeakers and the behaviour of the room, the best thing to do is to do a bunch of acoustical measurements, interpret the results and then make adjustments to the system, evaluating the measurements repeatedly as you go along. If you can’t do this, then you can tune it by ear. Unfortunately, this will take more time, and might require an extra person to help, but it will probably result in better results than doing nothing. Here is how I would do it:
Turn the LPF frequency as low as you can go
Turn on a piece of music that has a constant bass level
The opening of Freddy Mercury’s “Living on my Own” or Santanta’s “You Are My Kind” are a possible tunes. Claire Martin singing “Black Coffee” is also a good candidate. If you want to look like a professional, then I suppose that you could use pink noise or this track instead of music.
Sit in the listening position and ask someone to turn the LPF as low as it will go.
You should notice that there is a “hole” in the level of the bass between the subwoofer and the main loudspeaker. Turn up the LPF frequency and pay attention whether the “hole” fills up or gets worse. If it gets worse, flip the phase switch and start Step 4 again.
If the hole did not get worse, then keep turning up the LPF frequency until it sounds like there the hole is filled up.
One you’re done playing with the LPF frequency, try moving the Gain to adjust the bass to the level that you like.
