B&O Tech: But what if my room is Scandinavian?

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

 

The following question recently arrived in my inbox via our customer service department.

“I am an admirer of B&O Hifi products as of over 20 years, but a great mystery for me is how you achieve great sound reproduction in the typically minimalist Scandinavian interior design environmment with polished floors, bare walls and bare glass windows. Effectively such environments are acoustical disasters !?!”

I thought that this was a great question – so it’s the topic of this week’s article.
Of course, you are correct. A room comprised of large flat reflective surfaces with little acoustical absorption has a very different acoustical behaviour from a recording or mastering studio where the final decisions about various aspects of a recording are made. And, consequently, this must have an effect on a listener’s perception of a recording played through a pair (assuming stereo reproduction) of loudspeakers in that room. The initial question to be asked is “what, exactly, are the expected effects of the room’s acoustical behaviour in such a case?” The second is “if the room has too much of an effect, how can I improve the situation (i.e. by adding absorption or changing the physical configuration of the system in the room)?” The third, and possibly final question is “how can we, as a loudspeaker manufacturer compensate (or at least account) for these effects?”
The effect a room’s acoustical behaviour has on a loudspeaker’s sound can, at a simple level, be considered under three general headings:
  1. early reflections
  2. room modes
  3. reverberation

Early Reflections

Early reflections, from sidewalls and the floor and ceiling, have an influence on both the timbre (tone colour) and the spatial characteristics of a stereo reproduction system. Let’s only think about the timbral effects for this article.
Fig 1. The sound arriving at a listener from a loudspeaker in a room with only one wall. Note that the sound arrives from two directions - the first is directly from the loudspeaker. The second is a "first reflection" off the wall.
Fig 1. The sound arriving at a listener from a loudspeaker in a room with only one wall. Note that the sound arrives from two directions – the first is directly from the loudspeaker. The second is a “first reflection” off the wall.
Let’s start by assuming that you have a loudspeaker that has a magnitude response that is perfectly flat – at least from 20 Hz to 20 kHz. We will also assume that it has that response regardless of which direction you measure it in – in other words, it’s a perfectly omnidirectional loudspeaker. The question is, “what effect does the wall reflection have on the measured response of the loudspeaker?”
Very generally speaking, the answer is that you will get a higher level at some frequencies (because the direct sound and the reflection add constructively and reinforce each other) and you will get a lower level at other frequencies (because the direct sound and the reflection work against each other and “cancel each other out”). What is potentially interesting is that the frequencies that add and the frequencies that cancel alternate as you go up the frequency range. So the total result looks like a comb (as in a comb that you use to comb your hair, if, unlike me, you have hair to comb). For example, take a look at Figure 2.
Fig 2. Distance to loudspeaker = 2 m. Distance to wall = 1 m. Wall is perfectly reflective and the loudspeaker is perfectly omnidirectional.
Fig 2. Distance to loudspeaker = 2 m. Distance to wall = 1 m. Wall is perfectly reflective and the loudspeaker is perfectly omnidirectional.
So, you can see in Figure 2 that, at the very low end, the reflection boosts the level of the loudspeaker by a little less than 6 dB (that’s two times the level!) at the listening position. However, as you go up in frequency, the total level drops to about 15 dB less before it starts rising again. As you go up in frequency, the level goes up and down. This alternation actually happens at a regular frequency spacing (i.e. a notch at every 200 Hz) but it doesn’t look regular because the X-axis of my plot is logarithmic (which better represents how we hear differences in frequency).
What happens if we move the wall further away? Well, two things will happen. The first is that the reflection will be quieter, so the peaks and notches won’t be as pronounced. The second is that the spacing of the peaks and notches in frequency will get closer together. For example, take a look at Figure 3.
Fig 3. Distance to loudspeaker = 2 m. Distance to wall = 3 m. Wall is perfectly reflective and the loudspeaker is perfectly omnidirectional.
Fig 3. Distance to loudspeaker = 2 m. Distance to wall = 3 m. Wall is perfectly reflective and the loudspeaker is perfectly omnidirectional.
Similarly, if we move the wall closer, we do the opposite, as in Figure 4.
Fig 2. Distance to loudspeaker = 2 m. Distance to wall = 0.25 m. Wall is perfectly reflective and the loudspeaker is perfectly omnidirectional.
Fig 4. Distance to loudspeaker = 2 m. Distance to wall = 0.25 m. Wall is perfectly reflective and the loudspeaker is perfectly omnidirectional.
So, if you have a room with only one wall which is perfectly reflective, and you have a perfectly omnidirectional loudspeaker, then you can see that your best option is to either put the loudspeaker (and yourself) very far or very close to the wall. That way the artefacts caused by the reflection are either too quiet to do any damage, or have an effect that starts at too high a frequency for you to care. Then again, most room have more than one wall, the walls are not perfectly reflective, and the loudspeaker is not perfectly omnidirectional.
So, what happens in the case where the loudspeaker is more directional or you have some fuzzy stuff on your walls? Well, either of these cases will have basically the same effect in most cases since loudspeakers are typically more directional at high frequencies – so you get less high end directed towards the wall. Alternatively, fuzzy stuff tends to soak up high frequencies. So, in either of these two cases, you’ll get less high end in the reflection. Let’s simulate this by putting a low pass filter on the reflection, as shown in Figure 5, 6 and 7 which have identical distances as the simulations in Figures 2, 3, and 4 – for comparison.
Fig 5. Distance to loudspeaker = 2 m. Distance to wall = 1 m. Wall is absorptive at high frequencies and/or the loudspeaker is directional.
Fig 5. Distance to loudspeaker = 2 m. Distance to wall = 1 m. Wall is absorptive and/or the loudspeaker is directional at high frequencies .
Fig 2. Distance to loudspeaker = 2 m. Distance to wall = 3 m. Wall is absorptive at high frequencies and/or the loudspeaker is directional.
Fig 6. Distance to loudspeaker = 2 m. Distance to wall = 3 m. Wall is absorptive and/or the loudspeakers is directional at high frequencies.
Fig 2. Distance to loudspeaker = 2 m. Distance to wall = 0.25 m. Wall is absorptive at high frequencies and/or the loudspeaker is directional.
Fig 7. Distance to loudspeaker = 2 m. Distance to wall = 0.25 m. Wall is absorptive and/or the loudspeaker is directional at high frequencies.
What you can see in all three of the previous plots is that, as the high frequency content of the reflection disappears, there is less and less effect on the total. The bottom plot is basically a proof of the age-old rule of thumb that says that, if you put a loudspeaker next to a wall, you’ll get more bass than if it’s farther from the wall. Since there is not much high frequency energy radiated from the rear of most loudspeakers, Figure 7 is a pretty good general representation of what happens when a loudspeaker is placed close to a wall. Of course, the exact behaviour of the directivity of the loudspeaker will be different – but the general shape of the total curve will be pretty similar to what you see there.
So, the end conclusion of all of this is that, in order to reduce undesirable artefacts caused by a wall reflection, you can do any combination of the following:
  • move the loudspeaker very close to the wall
  • move the loudspeaker farther front the wall
  • sit very close to the wall
  • sit farther away from the wall
  • put absorption on the wall

However, there is one interesting effect that sits on top of all of this – that is the fact that what you’ll see in a measurement with a microphone is not necessarily representative of what you’ll hear. This is because a microphone does not have two ears. Also, the direction the reflection comes from will change how you perceive it. A sidewall reflection sounds different from a floor reflection. This is because you have two ears – one on each side of your head. Your brain uses the sidewall reflections (or, more precisely, how they relate to the direct sound) to determine, in part, how far away a sound source is. Also, since, in the case of sidewall reflections, your two ears get two different delay times on the reflection (usually), you get two different comb-filter patterns, where the peaks in one ear can be used to fill in the notches in the other ear and vice versa. When the reflection comes from the floor or ceiling, your two ears get the same artefacts (since your two ears are the same distance to the floor, probably). Consequently, it’s easily noticeable (and it’s been proven using science!) that a floor or ceiling reflection has a bigger timbral effect on a loudspeaker than a lateral (or sideways) reflection.

Room modes

 Room modes are a completely different beast – although they exist because of reflections. If you pluck a guitar string, you make a deflection in the string that moves outwards until it hits the ends of the string. It then bounces back down the string, bounces again, etc. etc. See the diagrams and animations on this page – they might help. As the wave bounces back and forth, it settles in to a total result where it looks like the string is just bouncing up and down like a skipping rope. The longer the string, the lower the note, because it takes longer for the wave to bounce back and forth on the string. You can also lower the note by lowering the tension of the string, since this will slow down the speed of the wave moving back and forth on it. The last way to lower the note is to make the string heavier (i.e. by making it thicker) – since a heavier string is harder to move, the wave moves slower on it.
The air in a pipe behaves exactly the same way. If you “pluck” the air in the middle of a pipe (say, by clapping our hands, or coughing, or making any noise at all) then the sound wave travels along the pipe until it hits the end. Whether the end of the pipe is capped or not, the wave will bounce back and travel back through the pipe in the opposite direction from whence it came. (Whether the pipe is closed (capped) or open only determines the characteristic of the reflection – there will be a reflection either way.) It might help to look at the animations linked on this page to get an idea of how the air molecules behave in a pipe. As the wave bounces back and forth off he two ends of the pipe, it also settles down (just like the guitar string) into something called a “standing wave”. This is the pipe’s equivalent of the skipping rope behaviour in the guitar string. The result is that the pipe will resonate or ring at a note. The longer the pipe, the lower the note because the speed of the sound wave moving in air in the pipe stays the same, but the longer the pipe, the longer it takes for the wave to bounce back and forth. This is basically how all woodwind instruments work.
What’s interesting is that, when it comes to resonating, a room is basically a pipe. If you “pluck” the air in the room (say, by putting sound out of a loudspeaker) the sound wave will move down the room, bounce off the wall, go back through the room, bounce of the opposite wall, etc. etc. etc. (other things are happening, but we’ll ignore those). This effect is most obvious on a graph by putting some sound in a room and stopping suddenly. Instead of actually stopping, you can see the room “ringing” at a frequency that gradually decays as time goes by. However, it’s important to remember that this ringing is always happening – even while the sound is playing. So, for example, a kick drum “thump” comes out of the speaker which “plucks” the room mode and it rings, while the music continues on. You can see this in Figure 8, below.
Fig 8. The concept of the effect of a room mode. The sound coming out of the loudspeaker is shown on the top plot, in black. The response in the room is shown in blue. You can see there that the room keeps "ringing" at a frequency after the sound from the loudspeaker stops. The red plot on the bottom is the difference between the two plots - in other words, the "sound" of the room mode in isolation.
Fig 8. The concept of the effect of a room mode. See the text below for an explanation.
Figure 8 shows the concept of the effect of a room mode. The sound coming out of the loudspeaker is shown on the top plot, in black. The response in the room is shown in the middle plot in blue. You can see there that the room keeps “ringing” at a frequency after the sound from the loudspeaker stops. The red plot on the bottom is the difference between the two plots – in other words, the “sound” of the room mode in isolation (note that it’s at a different scale than the top two plots to make things easier to see).

There are two audible effects of this. The first is that, if your music contains the frequency that the room wants to resonate at, then that note will sound louder. When you hear people talk of “uneven bass” or a “one-note-bass” effect, one of the first suspects to blame is a room mode.

The second is that, since the mode is ringing along with the music, the overall effect will be muddiness. This is particularly true when one bass note causes the room mode to start ringing, and it keeps ringing when the next bass note is playing.  For example, if your room rings on a C#, and the bass plays a C# followed by a D – then the room will be ringing at C#, conflicting with the D and resulting in mud. This is also true if the kick drum triggers the room mode, so you have a kick drum “plucking” the room ringing on a C# all through the track. If the tune is in the key of F, then this will not be pretty.

 

If you would like to calculate a prediction of where you’ll have a problem with a room mode, you can just do the following math:

metric version: room mode frequency in Hz = 172 / (room length in metres)

imperial version: room mode frequency in Hz = 558 / (room length in feet)

Your worst modes will be the frequencies calculated using either of the equations above, and multiples of them (i.e. 2 times the result, 3 times the result, and so on).

So, for example, if your room is 5 m wide, your worst-case modes will be at 172 / 5 = 34.4 Hz, as well as 68.8 Hz, 103.2 Hz and so on. Remember that these are just predictions – but they’ll come pretty close. You should also remember that this assumes that you have completely immovable walls and no absorption – if this is not true, then the mode might not be a problem at all. (If you would like to do a more thorough modal analysis of your listening room, check out this page as a good start.)

Sadly, there is not much you can do about room modes. There are ways to manage them, including, but not exclusive to the following strategies:

  • make sure that the three dimensions of your listening room are not related to each other with simple ratios
  • put up membrane absorbers or slot absorbers that are tuned to the modal frequencies
  • place your loudspeaker in a node – a location in a room where it does not couple to a problematic mode (however, note that one mode’s node is another mode’s antinode)
  • sit in a node – a location in a room where you do not couple to a problematic mode (see warning above…)
  • use room correction DSP software such as ABC in the BeoLab 5

 

Reverberation

Reverberation is what you hear when you clap your hands in a big cathedral. It’s the  collection of a lot of reflections bouncing from everywhere as you go through time. When you first clap your hands, you get a couple of reflections that come in separated enough in time that they get their own label – “early reflections”. After that, there are so many reflections coming from so many directions, and so densely packed together in time, that we can’t separate them, so we just call them “reverberation” or “reverb” (although you’ll often hear people call it “echo” which is the wrong word to use for this.

Reverb is what you get when you have a lot of reflective surfaces in your room – but since it’s so irregular in time and space, it just makes a wash of sound rather than a weird comb-filter effect like we saw with a single reflection. So, although it makes things “cloudy” – it’s more like having a fog on your glasses instead of a scratch. Think of it like the soft focus effect that was applied to all attractive alien women on the original Star Trek – you lose the details, but it’s not necessarily a bad thing.

 

So what are you gonna do about it?

Fine, this is a short-form version of what a room’s acoustics does to the sound of a loudspeaker, but how do we, as a manufacturer of loudspeakers, ensure that our products can withstand the abuse that your listening room will apply to the sound? Well, there are a number of strategies that we use to do what we can…

1. ABC. The BeoLab 5 has a proprietary system built-in called Adaptive Bass Control or ABC. Pressing a button at the top of the loudspeaker starts a measurement procedure that is performed using a built-in microphone that measures the loudspeaker’s behaviour in two locations. Actually, what it’s doing is looking at the difference in the loudspeaker’s response in those two positions of the microphone to determine the radiation resistance that the loudspeaker “sees” as a result of reflective surfaces in the room. The ABC algorithm then creates a filter that is used to “undo” the effects of some of the low-frequency effects of the room’s acoustics. For example, if the radiation resistance indicates that the loudspeaker is close to a wall (which, as we saw above, will boost the bass) then the filter will reduce the bass symmetrically. That way, the loss in the filter and the gain due to the wall will cancel each other.

2. Position switches. ABC in the BeoLab 5 is a very customised filter that, in part, will adjust the loudspeaker’s response for placement near a wall or in a corner. Almost all of the other BeoLab loudspeakers (and other sound systems such as the BeoPlay A8 and A9, for example), include a manual-adjusted “position switch”. This allows you to use one of three filters that we have customised in the development of the loudspeaker to account for its behaviour according to whether you have placed it away from a reflective surface (“Free”), near one surface (“Wall”) or in a Corner. This is not just a filter that adjusts the bass level. The three filters for “free”, “wall”, and “corner” have been calculated using three dimensional measurements of the acoustical behaviour of the loudspeaker. So, the filters for the BeoLab 3 are completely different from those for the A9, for example, because they have very different directivity characteristics.

3. Sound design in multiple rooms. As I talked about in a previous posting, when we do the sound design of all of our loudspeakers, we tune each of them in at least 4 or 5 rooms with very different acoustical behaviours ranging from a very “dead” living room with lots of absorptive and diffusive surfaces to a larger and very “live” space with a minimalistic decorating, and large flat surfaces (just like the description in the original question). Once we have a single sound design that is based on the common elements those rooms, we test the loudspeakers in more rooms to ensure that they’ll behave well under all conditions.

 

Wrap-up

Of course, I haven’t covered everything there is to know about room acoustics here. And, of course, you can’t expect a loudspeaker to sound exactly the same in every room. If that were true, there would be no such thing as a “good”concert hall. A room’s acoustical behaviour affects the sound of all sound sources in the room. On the other hand, humans also have an amazing ability to adapt – in other words you “get used to” the characteristics of your listening room. Back when I was working as a part-time recording engineer in Montreal, I did a lot of recordings in churches. Typically, we (the producer and I) would set up a control room with loudspeakers in a back room, and the musicians would sit out in the church. When we arrived to set up the gear, the first thing was to set up the monitor loudspeakers and a CD player, and we would play CD’s that we knew well while we set up everything else. That way, we would “learn” the characteristics of the control room (since we already knew what was on the discs and the characteristics of the monitor loudspeakers). So, if all of our CD’s sounded like that had too much bass, then we should do a recording with too much bass – it was the fault of the control room.

However,  there is no debate that, due to lots of issues (the first two that come to mind are frequency range and directivity) two different loudspeakers will behave differently from each other in two different rooms. In other words, if you listen to loudspeaker “A” and loudspeaker “B” in a showroom of a shop, you might prefer loudspeaker “A” – but if you took them home, you might prefer loudspeaker “B”. This would not be surprising, since what you hear is not only the loudspeaker but the loudspeaker “filtered” by the listening room. This is exactly why, when you are buying a loudspeaker, you should audition it in your home in order to ensure that you will be happy with your purchase. And THIS is why you can arrange a home demonstration of Bang & Olufsen loudspeakers through your dealer.

B&O Tech: The Naked Truth II

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

 

As I mentioned in an early posting about two months ago, if I didn’t have time to write enough, I’d cop out and post some photos of development models of some of our loudspeakers. It’s been a busy week. So, here are some photos of the BeoLab 18 and BeoLab 14.

Early BeoLab 18 prototype.
Early BeoLab 18 prototype. This photo was taken on the floor of the Cube.

 

Early BeoLab 18 prototype - acoustic lens closeup
Early BeoLab 18 prototype – acoustic lens closeup. The lens itself is an SLA prototype and is attached to the top of the milled plastic cabinet using modelling putty.

 

Early BeoLab 18 prototype showing a closeup of the port. Putty was added to the bottom of the port opening to smooth the curve as part of an investigation about reduction of turbulence noise.
Early BeoLab 18 prototype showing a closeup of the port. More modelling putty was added to the bottom of the port opening to smooth the curve as part of an investigation about reduction of turbulence noise.

 

The first BeoLab 14 prototype - subwoofer and satellite. Note that the DSP and amplifiers would be external for this prototype.
The first BeoLab 14 prototype – subwoofer and satellite. Note that the DSP and amplifiers would be external for this prototype. The white modelling putty at the top of the pipe used to port the cabinet was put there to reduce turbulence noise. It is not a prototype of the final geometry of the port flare – it was just a quick-and-dirty solution for the initial demos.

 

BeoLab 14 subwoofer innards.
BeoLab 14 subwoofer innards.

 

 

The business end of a BeoLab 14 subwoofer.
The business end of a BeoLab 14 subwoofer.

 

 

Bang & Olufsen: BeoLab 17 reviews

beolab17

 

 

 

homecinemachoice.com‘s review (March 2014 edition)

“… 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.”

 

 

B&O Tech: How to Make a Loudspeaker Driver (A primer)

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

 

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).

 

The right hand rule shows that, when you put current through a wire, you get a magnetic field around it.
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.

 

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, the magnetic poles will reverse, so North will be on the left of this diagram.
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.

 

A coil of wire about to be put in the gap of a strange-looking magnet.
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.

 

The bottom plate and the pole piece, both typically made of soft iron.
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.

A ring has been added around the pole piece to reduce the gap width.
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.

 

The blue ring is the permanent magnet, typically made of ferrite or neodymium.
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.

 

The top plate, also made of soft iron.
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.

 

The light blue tube is the former, around which the voice coil is wound.
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).

 

The dust cap has been added to the front of the former.
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.

 

The spider has been added. It is glued to the former, but is not attached to the coil or the top plate.
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…

 

The basket is glued and/or riveted to the top plate.
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.

 

An entire moving coil loudspeaker. The green ring is the surround and the greyish-purple ring inside it is the diaphragm or speaker cone, glued to the top of the former.
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.

 

A cross section of a (not very) simplified model of a moving coil dynamic loudspeaker driver.
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.

 

An actual moving coil dynamic loudspeaker, after I was very mean to it.
Fig 14. An actual moving coil dynamic loudspeaker, after I was very mean to it.

 

A BeoLab 3 woofer - after I was finished with 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.

 

A BeoLab 3 woofer, showing some of the components.
Fig 16. A BeoLab 3 woofer, showing some of the components.

 

A BeoLab 3 woofer, showing some more 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.

 

A BeoLab 3 woofer, showing some more of the components.
Fig 18. A BeoLab 3 woofer, showing some more of the components.

 

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.

B&O Tech: Where should I decode?

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

 

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:

  1. A Blu-ray player connected to…
  2. 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…
  3. 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:

  1. The Blu-ray player is connected to the television’s input with an HDMI cable.
  2. 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).
  3. 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.

If you go to a reasonably reputable source such as www.hdmi.org, you’ll read that “There is no inherent difference in quality between Dolby TrueHD/DTS-HD being sent over HDMI as decoded PCM vs. encoded bit stream“. This is, in fact, true (although it’s not the full story). It’s true if your audio is encoded in Dolby TrueHD, Dolby Digital Plus, Dolby Digital, DTS-HD Master Audio, DTS-HD High Resolution Audio, or DTS Digital Surround.

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:

  1. you decode in the player and send PCM to the television, OR
  2. 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.

  1. 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”.
  2. 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.