Category: Bang & Olufsen

B&O Tech: Headphone signal flows

#54 in a series of articles about the technology behind Bang & Olufsen products

Someone recently asked a question on this posting regarding headphone loudness. Specifically, the question was:

“There is still a big volume difference between H8 on Bluetooth and cable. Why is that?”

I thought that this would make a good topic for a whole posting, rather than just a quick answer to a comment – so here goes…

Introduction – the building blocks

To begin, let’s take a quick look at all the blocks that we’re going to assemble in a chain later. It’s relatively important to understand one or two small details about each block.

Two start:

  • I’ve used red lines for digital signals and blue lines for analogue signals. I’ve assumed that the digital signal contains 2 audio channels, and that the analogue connections are one channel each.
  • My signal flow goes from left to right
  • I use the word “telephone” not because I’m old-fashioned (although I am that…) but because if I say “phone”, I could be mistaken for someone talking about headphones. However, the source does not have to be a telephone, it could be anything that fits the descriptions below.
  • The blocks in my signal flows should be taken as basic examples. I have not reverse-engineered a particular telephone or computer or pair of headphones. I’m just describing basic concepts here…
Fig 1: A Digital to Analogue converter.
Fig 1: A Digital to Analogue converter.

Figure 1, above, shows a 2-channel audio DAC – a Digital to Analogue Converter. This is a device (these days, it’s usually just a chip) that receives a 2-channel digital audio signal as a stream of bits at its input and outputs an analogue signal that is essentially a voltage that varies appropriately over time.

One important thing to remember here is that different DAC’s have different output levels. So, if you send a Full Scale sine wave (say, a 997 Hz, 0 dB FS) into the input of one DAC, you might get 1 V RMS out. If you sent exactly the same input into another DAC (meaning another brand or model) you might get 2 V RMS out.

You’ll find a DAC, for example, inside your telephone, since the data inside it (your MP3 and .wav files) have to be converted to an analogue signal at some point in the chain in order to move the drivers in a pair of headphones connected to the minijack output.

Fig 2: An Analogue to Digital converter.
Fig 2: An Analogue to Digital converter.

Figure 2, above, shows a 2-channel ADC – and Analogue to Digital Converter. This does the opposite of a DAC – it receives two analogue audio channels, each one a voltage that varies in time, and converts that to a 2-channel digital representation at its output.

One important thing to remember here is the sensitivity of the input of the ADC. When you make (or use) an ADC, one way to help maximise your signal-to-noise ratio (how much louder the music is than the background noise of the device itself) is to make the highest analogue signal level produce a full-scale representation at the digital output. However, different ADC’s have different sensitivities. One ADC might be designed so that 2.0 V RMS signal at its input results in a 0 dB FS (full scale) output. Another ADC might be designed so that a 0.5 V RMS signal at its input results in a 0 dB FS output. If you send 0.5 V RMS to the first ADC (expecting a max of 2 V RMS) then you’ll get an output of approximately -12 dB FS. If you send 2 V RMS to the second ADC (which expects a maximum of only 0.5 V RMS) then you’ll clip the signal.

Fig 3: A Digital Signal Processor.
Fig 3: A Digital Signal Processor.

Figure 3, above, shows a Digital Signal Processor or DSP. This is just the component that does the calculations on the audio signals. The word “calculations” here can mean a lot of different things: it might be a simple volume control, it could be the filtering for a bass or treble control, or, in an extreme case, it might be doing fancy things like compression, upmixing, bass management, processing of headphone signals to make things sound like they’re outside your head, dynamic control of signals to make sure you don’t melt your woofers – anything…

Fig 4: A two-channel analogue amplifier block.
Fig 4: A two-channel analogue amplifier block.

Figure 4, above, shows a two-channel analogue amplifier block. This is typically somewhere in the audio chain because the output of the DAC that is used to drive the headphones either can’t provide a high-enough voltage or current (or both) to drive the headphones. So, the amplifier is there to make the voltage higher, or to be able to provide enough current to the headphones to make them loud enough so that the kids don’t complain.

Fig 5: The headphone drivers - the "business end" of the headphones.
Fig 5: The headphone drivers – the “business end” of the headphones.

The final building block in the chain is the headphone driver itself. In most pairs of headphones, this is comprised of a circular-shaped magnet with a coil of wire inside it. The coil is glued to a diaphragm that can move like the skin of a drum. Sending electrical current back and forth through the coil causes it to move back and forth which pushes and pulls the diaphragm. That, in turn, pushes and pulls the air molecules next to it, generating high and low pressure waves that move outwards from the front of the diaphragm and towards your eardrum. If you’d like to know more about this basic concept – this posting will help.

One important thing to note about a headphone driver is its sensitivity. This is a measure of how loud the output sound is for a given input voltage. The persons who designed the headphone driver’s components determine this sensitivity by changing things like the strength of the magnet, the length of the coil of wire, the weight of the moving parts, resonant chambers around it, and other things. However, the basic point here is that different drivers will have different loudnesses at different frequencies for the same input voltage.

Now that we have all of those building blocks, let’s see how they’re put together so that you can listen to Foo Fighters on your phone.

Version 1: The good-old days

In the olden days, you had a pair of headphones with a wire hanging out of one or both sides and you plugged that wire into the headphone jack of a telephone or computer or something else. We’ll stick with the example of a telephone to keep things consistent.

Figure 6, below, shows an example of the path the audio signal takes from being a MP3 or .wav (or something else) file on your phone to the sound getting into your ears.

The file is read and then decoded into something called a “PCM” signal (Pulse Code Modulation – it doesn’t matter what this is for the purposes of this posting). So, we get to point “A” in the chain and we have audio. In some cases, the decoder doesn’t have to do anything (for example, if you use uncompressed PCM audio like a .wav file) – in other cases (like MP3) the decoder has to convert a stream of data into something that can be understood as an audio signal by the DSP. In essence, the decoder is just a kind of universal translator, because the DSP only speaks one language.

The signal then goes through the DSP, which, in a very simple case is just the volume control. For example, if you want the signal to have half the level, then the DSP just multiplies the incoming numbers (the audio signal) by 0.5 and spits them out again. (No, I’m not going to talk about dither today.) So, that gets us to point “B” in the chain. Note that, if your volume is set to maximum and you aren’t doing anything like changing the bass or treble or anything else – it could be that the DSP just spits out what it’s fed (by multiplying all incoming values by 1.0).

Now, the signal has to be converted to analogue using the DAC. Remember (from above) that the actual voltage at its output (at point “C”) is dependent on the brand and model of DAC we’re talking about. However, that will probably change anyway, since the signal is fed through the amplifiers which output to the minijack connector at point “D”.

Assuming that they’ve set the DSP so that output=input for now, then the voltage level at the output (at “D”) is determined by the telephone’s manufacturer by looking at the DAC’s output voltage and setting the gain of the amplifiers to produce a desired output.

Fig 6: An example of a basic signal flow that occurs when you plus a pair of passive headphones into your phone to listen to music.
Fig 6: An example of a basic signal flow that occurs when you plug a pair of passive headphones into your telephone’s headphone output to listen to music.

Then, you plug a pair of headphones into the minijack. The headphone drivers have a sensitivity (a measure of the amount of sound output for a given voltage/current input) that will have an influence on the output level at your eardrum. The more sensitive the drivers to the electrical input, the louder the output. However, since, in this case, we’re talking about an electromechanical system, it will not change its behaviour (much) for different sources. So, if you plug a pair of headphones into a minijack that is supplying 2.0 V RMS, you’ll get 4 times as much sound output as when you plug them into a minijack that is supplying 0.5 V RMS.

This is important, since different devices have VERY different output levels – and therefore the headphones will behave accordingly. I regularly measure the maximum output level of phones, computers, CD players, preamps and so on – just to get an idea of what’s on the market. I’ve seen maximum output levels on a headphone jack as low as 0.28 V RMS (on an Apple iPod Nano Gen4) and as high as 8.11 V RMS (on a Behringer Powerplay Pro-8 headphone distribution amp). This is a very big difference (29 dB, which also happens to be 29  times…).

Version 2: The more-recent past

So, you’ve recently gone out and bought yourself a newfangled pair of noise-cancelling headphones, but you’re a fan of wires, so you keep them plugged into the minijack output of your telephone. Ignoring the noise-cancelling portion, the signal flow that the audio follows, going from a file in the memory to some sound in your ears is probably something like that shown in Figure 7.

Fig 7: An example of a basic signal flow that occurs when you plug a pair of active headphones into your phone's headphone output to listen to music.
Fig 7: An example of a basic signal flow that occurs when you plug a pair of active headphones into your telephone’s headphone output to listen to music.

As you can see by comparing Figures 7 and 6, the two systems are probably identical until you hit the input of the headphones. So, everything that I said in the previous section up to the output of the telephone’s amplifiers is the same. However, things change when we hit the input of the headphones.

The input of the headphones is an analogue to digital converter. As we saw above, the designer of the ADC (and its analogue input stages) had to make a decision about its sensitivity – the relationship between the voltage of the analogue signal at its input and the level of the  digital signal at its output. In this case, the designer of the headphones had to make an assumption/decision about the maximum voltage output of the source device.

Now we’re at point “E” in the signal chain. Let’s say that there is no DSP in the headphones – no tuning, no volume – nothing. So, the signal that comes out of the ADC is sent, bit for bit, to its DAC. Just like the DAC in the source, the headphone’s DAC has some analogue output level for its digital input level. Note that there is no reason for the analogue signal level of the headphones’ input to be identical to the analogue output level of the DAC or the analogue output level of the amplifiers. The only reason a manufacturer might want to try to match the level between the analogue input and the amplifier output is if the headphones work when they’re turned off – thus connecting the source’s amplifier directly to the headphone drivers (just like in Figure 6). This was one of the goals with the BeoPlay H8 – to ensure that if your batteries die, the overall level of the headphones didn’t change considerably.

However, some headphones don’t bother with this alignment because when the batteries die, or you turn them off, they don’t work – there’s no bypass…

Version 3: Look ma! No wires!

These days, many people use Bluetooth to connect wirelessly from the source to the headphones. This means that some components in the chain are omitted (like the DAC’s in the source and the ADC in the headphones) and others are inserted (in Figures 8 and 9, the Bluetooth Transmitter and Receiver).

Note that, to keep things simple, I have not included the encoder and the decoder for the Bluetooth transmission in the chain. Depending mainly on your source’s capabilities, the audio signal will probably be encoded into one of the varieties of an SBC, an AAC, or an aptX codec before transmitting. It’s then decoded back to PCM after receiving. In theory, the output of the decoder has the same level as the input of the encoder, so I’ve left it out of this discussion. I won’t discuss either CODEC’s implications on audio quality in this posting.

Taking a look at Figure 8 or 9 and you’ll see that, in theory, the level of the digital audio signal inside the source is identical to that inside the headphones – or, at least, it can be.

Fig 6: An example of a basic signal flow that occurs when you plus a pair of passive headphones into your phone to listen to music.
Fig 8: One example of a basic signal flow that occurs when you connect a pair of active headphones to your telephone using Bluetooth to listen to music. Note that the volume control, in this example, is shown in the telephone.
Fig 9: One example of a basic signal flow that occurs when you connect a pair of active headphones to your telephone using Bluetooth to listen to music. Note that the volume control, in this example, is shown in the headphones.
Fig 9: One example of a basic signal flow that occurs when you connect a pair of active headphones to your telephone using Bluetooth to listen to music. Note that the volume control, in this example, is shown in the headphones.

This means that the potentially incorrect assumptions made by the headphone manufacturer about the analogue output levels of the source can be avoided. However, it also means that, if you have a pair of headphones like the BeoPlay H7 or H8 that can be used either via an analogue or a Bluetooth connection then there will, in many cases, be a difference in level when switching between the two signal paths.

For example…

Let’s take a simple case. We’ll build a pair of headphones that can be used in two ways. The first is using an analogue input that is processed through the headphone’s internal DSP (just as is shown in Figure 6). We’ll build the headphones so that they can be used with a 2.0 V RMS output – therefore we’ll set the input sensitivity so that a 2.0 V RMS signal will result in a 0 dB FS signal internally.

We then connect the headphones to an Apple MacBook Pro’s headphone output, we play a signal with a level of 0 dB FS, and we turn up the volume to maximum. This will result in an analogue violate level of 2.085 V RMS coming from the computer’s headphone output.

Now we’ll use the same headphones and connect them to an Apple iPhone 4s which has a maximum analogue output level of 0.92 V RMS. This is less than half the level of the MacBook Pro’s output. So, if we set the volume to maximum on the iPhone and play exactly the same file as on the MacBook Pro, the headphones will have half the output level.

A second way to connect the headphones is via Bluetooth using the signal flow shown in Figure 8. Now, if we use Bluetooth to connect the headphones to the MacBook Pro with its volume set to maximum, a 0 dB FS signal inside the computer results in a 0 dB FS signal inside the headphones.

If we connect the headphones to the iPhone 4s via Bluetooth and play the same file at maximum volume, we’ll get the same output as we did with the MacBook Pro. This is because the 0 dB FS signal inside the phone is also producing a 0 dB FS signal in the headphones.

So, if you’re on the computer, switching from a Bluetooth connection to an analogue wired connection using the same volume settings will result in the same output level from the headphones (because the headphones are designed for a max 2 V RMS analogue signal). However, if you’re using the telephone, switching from a Bluetooth connection to an analogue wired connection will results in a drop in the output level by more than 6 dB (because the telephone’s maximum output level is less than 1 V RMS).

Wrapping up

So, the answer to the initial question is that there’s a difference between the output of the H8 headphones when switching between Bluetooth and the cable because the output level of the source that you’re using is different from what was anticipated by the engineers who designed the input stage of the headphones. This is likely because the input stage of the headphones was designed to be compatible with a device with a higher maximum output level than the one you’re using.

B&O at Munich High-End

I’m on my way home from the Munich High End audio show where we were running continuous demos of the Beolab 90’s (an 8’45” mix of tracks on repeat from 10:00 a.m. to 6:00 p.m. for 4 days… It will be a while before I get the Chris Jones, Lyle Lovett and the Foo Fighters out of my head…)

We used an intentionally bare room – one Oppo Blu-ray player playing the .wav file off a USB stick, connected via S/P-DIF to the Beolab 90’s. Nothing else was in the room except three chairs, a lamp, and the people. The floor was carpeted, the ceiling was absorptive, and the walls were EXTREMELY reflective. (This helped people to hear the difference between Narrow and Wide mode – an intentional decision on my part… The room looks like a cell for someone in a straight jacket, but the listeners didn’t mind… In fact, many came back for a second listening session after hearing some of the other loudspeakers at the show…)

 

IMG_0387

 

IMG_0391

We spent a while the day before the show doing the Active Room Compensation measurements, and that was it – we were good to go!

The outside of the booth was simple – but the tree of Scan Speak drivers certainly attracted attention – audiophiles like drivers, apparently. :-)

 

IMG_0393

Although it was nice to get out of the listening room in Struer for a while and meet some normal people – after four days on a 8’45” loop, it’s time to head home and listen to nothing for a day or two. :-)

 

 

B&O Tech: The Beogram 4002 Design Story

#53 in a series of articles about the technology behind Bang & Olufsen products

I was cleaning up around my desk over the past week and I came across a booklet called the “Beogram 4002 Design Story” published by Bang & Olufsen in 1975. Briefly reading through it, I thought that there was a lot of general information in there that would be worth sharing for people interested in turntables. So, I got permission to post it as a PDF here. Click here to download the file.

 

 

 

B&O Tech: Location, Location, Location

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

 

I often have to do demos of loudspeakers for people. Also, I frequently have to make recommendations on how to do  (either for Bang & Olufsen dealers, or for things like press events). One of the problems that I face every time I have to do this is how to arrange the chairs so that everyone gets a reasonable impression of how a loudspeaker sounds. The problem is that this is basically impossible, due to the significant influence things like the loudspeakers’ locations, the listener location, and the room, have on the overall sound.

One aspect of how-a-loudspeaker-sounds is its magnitude response (often called a “frequency response”). A (perhaps too-simple) definition of a magnitude response is “a measure of how loud the output signal is at different frequencies if you put in a signal that is the same level at all frequencies”.

If we wanted to make a measurement of a loudspeaker’s magnitude response in a room at a particular position, we just have to put in a signal that contains all frequencies at the same magnitude (or level), capture that output with a  microphone somewhere in the room, and compare how loud the signal is at different frequencies. Of course, in order to do this, we have to take care of some details. We have to make sure that the microphone (and everything else in the measurement part of the signal path) has a flat magnitude response. If it doesn’t, then at least we should know what its response is, so we can subtract it from the measurement to remove its influence on the result.

However, for the purposes of this posting, I’m not really interested in the absolute response of a loudspeaker. I’m more interested in how that response changes as you move in a room. Specifically, I want to show how much the magnitude response can change with very small changes in listening position.

Let’s start by measuring the magnitude response of a loudspeaker in a room at a location. For the purposes of this example, I’ve used a full-range, multi-way loudspeaker without a port. It’s placed roughly 1 m from the side wall and 1 m from the front wall, aimed at a listening position. The listening position is in the centre of the room’s width, and closer to the rear wall than the front wall by about a metre. The details of the location for the microphone (a 1/4″ omnidirectional measurement mic) for this measurement are shown in Figure 1, below.

 

Fig 1. The reference position for the microphone. All other measurements compared to this one. Height of mic diaphragm from floor was 1015 mm.
Fig 1. The reference position for the microphone. All other measurements compared to this one. Height of mic diaphragm from floor was 1015 mm.

 

I did an impulse response measurement (using an MLS signal with 4 averages (to improve SNR) and 4 sequences (to reduce the effect of distortion)) of the loudspeaker, the result of which is shown below in Figure 2. As you can see there, there are many obvious reflections after the initial impulse, and there is some kind of ringing in the room’s response.

The extremely long time before the onset of the impulse is arbitrary. The microphone was not actually 40 m away from the loudspeaker…

 

Fig 2: The impulse response of the loudspeaker at the Reference measurement position. Reflections from the room surfaces (walls, floor, ceiling) can be seen as "spikes" after the initial impulse.
Fig 2: The impulse response of the loudspeaker at the Reference measurement position. Reflections from the room surfaces (walls, floor, ceiling) can be seen as “spikes” after the initial impulse.

 

As I said above, I’m not interested in the resulting magnitude response of this measurement. I can tell you that it’s messy. There are bumps and dips in the low end caused (primarily) by room modes. The top end is messy due to the reflections. The overall curve is not flat due to the loudspeaker’s response, the microphone’s orientation, and various components in the signal path. However, I don’t care, since I’m not here to measure how the speaker behaves at one location in the room. I’m here to find out how its behaviour changes when you change location. So, let’s move the microphone.

As you can see in Figure 3, below, I started by moving the microphone only 100 mm, directly forwards in the room.

 

Fig 2. Microphone Position 2 - forward 100 mm. Height of mic diaphragm from floor was 1015 mm.
Fig 3. Microphone Position 2 – forward 100 mm. Height of mic diaphragm from floor was unchanged at 1015 mm.

 

Again, I measured the impulse response, converted that to a magnitude response (reflections and all!), smoothed it with a running 1/3 octave smoothing and subtracted the magnitude response measured at the Reference position (also smoothed to 1/3 octave). The resulting difference is shown in Figure 4, below.

Fig X: Difference in the magnitude response measurement at Position 2 and the Reference Position.
Fig 4: Difference in the magnitude response measurement at Position 2 and the Reference Position.

 

As you can see in Figure 3, moving the listening position only 100 mm results in a magnitude response deviation of about -2 to +4 dB. This is easily within the threshold of audibility for most people…

Now, let’s move the microphone sideways instead, as shown in Figure 5.

 

Fig 2. Microphone Position 1 - sideways 113 mm. Height of mic diaphragm from floor was 1015 mm.
Fig 5. Microphone Position 3 – sideways 113 mm. Height of mic diaphragm from floor was unchanged at 1015 mm.

Again, a roughly 100 mm movement results in a large change in the magnitude response – although now the most significant changes have happened in the low end, as can be seen in Figure 6.

 

 

Fig X: Difference in the magnitude response measurement at Position 3 and the Reference Position.
Fig 6: Difference in the magnitude response measurement at Position 3 and the Reference Position.

 

If we have more than one listener attending the demo, then I prefer to seat them “bus” style – one directly in front of the other – to ensure that everyone is getting a reasonably good phantom centre image. Sitting off-centre results in the time of arrival of signals from the two loudspeakers being mis-matched which will result in phantom images pulling towards the closer (and therefore earlier) loudspeaker.

Let’s say the we have a person roughly a half-metre behind the “good” chair, as shown in Figure 7. How different is the sound in that location?

 

Fig 6. Microphone Position 4 - backwards 444 mm. Height of mic diaphragm from floor was 1015 mm.
Fig 7. Microphone Position 4 – backwards 444 mm. Height of mic diaphragm from floor was 1015 mm.

Now we can see in Figure 8 that, by moving backwards in the room, we get more than ± 10 dB of variation in the magnitude response, with significant deviations happening as high as 1 kHz (depending on how you define “significant”).

 

Fig X: Difference in the magnitude response measurement at Position 4 and the Reference Position.
Fig 8: Difference in the magnitude response measurement at Position 4 and the Reference Position.

 

Similarly, moving forwards by a half metre from the Reference position (shown in Figure 9) results in a similar amount of change in the magnitude response, shown in Figure 10.

 

Fig 2. Microphone Position 5 - forward 509 mm. Height of mic diaphragm from floor was 1015 mm.
Fig 9. Microphone Position 5 – forward 509 mm. Height of mic diaphragm from floor was 1015 mm.

 

Fig X: Difference in the magnitude response measurement at Position 5 and the Reference Position.
Fig 10: Difference in the magnitude response measurement at Position 5 and the Reference Position.

 

Just for comparison, I’ve re-plotted the 4 magnitude response differences shown above in one plot. This is to show that the changes are not necessarily easily predicable with a simple knowledge of room layout. In other words, it would be almost impossible, without some serious simulation software, to predict these changes just by looking at a floorplan of the room and the chairs.

 

Fig X: Difference in the magnitude response measurement at Positions 2 to 5 and the Reference Position.
Fig 11: Difference in the magnitude response measurement at Positions 2 to 5 and the Reference Position.

 

What’s the moral of the story here? There are many – but I’ll just mention three.

The first is the message that, even a very small change in location (like leaning to one side in your chair – or leaning forwards to rest your elbows on your knees and your chin on your hands) can dramatically change the simple magnitude response of a loudspeaker (we won’t get into the effects on the spatial behaviour of the system).

The second is that, when you’re sitting with a friend, auditioning a pair of loudspeakers, switch chairs now and again. It is extremely unlikely that you’re both hearing the same thing at the same time.

Thirdly, the fact that there are significant differences between magnitude responses at different listening positions (even within a half-metre radius) means that, if you’re doing measurements for a room compensation system using a microphone around the listening position, it’s always smarter to make more than one measurement. In fact, there are some people who argue that, in this case, having only one measurement is worse than having no measurements, since you can easily get distracted by something in the magnitude or the time response that is a problem at only that location and nowhere else.

Finally, it’s worth considering that first point a different way. Let’s say you’re the type of person who likes to tweak a stereo system by upgrading components like wires. And, let’s say that you have incredible powers of “sonic memory” – in other words, you can listen to a system, take a break, listen to it again, and you are able to detect extremely small changes in system performance (like the magnitude response). So, you listen to your system – then you get out of your chair, change the component, sit down again, and start listening to the same tune at the same level. Remember that, unless you are in exactly the same location that you were before (not just “in the same chair”…), it could be that there is a larger difference in the magnitude response of the loudspeaker due your change in position than there is due to the component you just changed… So if you are a tweaker – get someone else to do the dirty work for you so you can sit there, in your chair, with your head in a clamp, waiting to evaluate the “upgrade”…

 

B&O Club

 

This is a short Danish documentary about the B&O Club of retired B&O employees that meets weekly to restore older products for the Struer Museum.

B&O Tech: The Cube

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

Figure X: A Beolab 8000 woofer undergoing near field measurements.
Figure 1: A Beolab 8000 woofer undergoing near field measurements in The Cube.

Sometimes, we have journalists visiting Bang & Olufsen in Struer to see our facilities. Of course, any visit to Struer means a visit to The Cube – our room where we do almost all of the measurements of the acoustical behaviour of our loudspeaker prototypes. Different people ask different questions about that room – but there are two that come up again and again:

  • Why is the room so big?
  • Why isn’t it an anechoic chamber?

Of course, the level of detail of the answer is different for different groups of people (technical journalists from audio magazines get a different level of answer than lifestyle journalists from interior design magazines). In this article, I’ll give an even more thorough answer than the one the geeks get. :-)

Introduction #1 – What do we want?

Our goal, when we measure a loudspeaker, is to find out something about its behaviour in the absence of a room. If we measured the loudspeaker in a “real” room, the measurement would be “infected” by the characteristics of the room itself. Since everyone’s room is different, we need to remove that part of the equation and try to measure how the loudspeaker would behave without any walls, ceiling or floor to disturb it.

So, this means (conceptually, at least) that we want to measure the loudspeaker when it’s floating in space.

Introduction #2 – What kind of measurements do we do?

Basically, the measurements that we perform on a loudspeaker can be boiled down into four types:

  1. on-axis frequency response
  2. off-axis frequency responses
  3. directivity
  4. power response

Luckily, if you’re just a wee bit clever (and we think that we are…), all four of these measurements can be done using the same basic underlying technique.

The very basic idea of doing any audio measurement is that you have some thing whose characteristics you’re trying to measure – the problem is that this thing is usually a link in a chain of things – but you’re only really interested in that one link. In our case, the things in the audio chain are electrical (like the DAC, microphone preamplifier, and ADC) and acoustical (like the measurement microphone and the room itself).

Figure 2: A block diagram of the measurement system in The Cube.
Figure 2: A block diagram of the measurement system in The Cube. The acoustical signals in the room are shown in red. Analogue electrical connections around shown in blue. (That thing on the bottom is supposed to look like a computer showing a magnitude response plot…)

The computer sends a special signal (we’ll come back to that…) out of its “sound card” to the input of the loudspeaker. The sound comes out of the loudspeaker and comes into the microphone (however, so do all the reflections from the walls, ceiling and floor of the Cube). The output of the microphone gets amplified by a preamplifier and sent back into the computer. The computer then “looks at” the difference between the signal that it sent out and the signal that came back in. Since we already know the characteristics of the sound card, the microphone and the mic preamp, then the only thing remaining that caused the output and input signals to be different is the loudspeaker.

Introduction #3 – The signal

There are lots of different ways to measure an audio device. One particularly useful way is to analyse how it behaves when you send it a signal called an “impulse” – a click. The nice thing about a theoretically perfect click is that it contains all frequencies at the same amplitude and with a known phase relationship. If you send the impulse through a device that has an “imperfect” frequency response, then the click changes its shape. By doing some analysis using some 200-year old mathematical tricks (called “Fourier analysis“), we can convert the shape of the impulse into a plot of the magnitude and phase responses of the device.

So, we measure the way the device (in our case, a loudspeaker) responds to an impulse – in other words, its “impulse response”.

Figure X: Simplified example of a loudspeaker impulse response from the Cube.
Figure 3: Simplified example of a loudspeaker impulse response from the Cube.

There are three things to initially notice in this figure.

  • The first is the time before the first impulse comes in. This is the time it takes the sound to travel from the loudspeaker to the microphone. Since we normally make measurements at a distance of 3 m, this is about 8.7 ms.
  • The second is the fact that the impulse doesn’t look perfect. That’s because it isn’t – the loudspeaker has made it different.
  • The third is the presence of the wall reflection. (In real life, we see 6 of these – 4 walls, a ceiling and the floor – but this example is a simulation that I created, just to show the concept.)

In order to get a measurement of the loudspeaker in the absence of a room, we have to get rid of those reflections… In this case, all we have to do is to tell the computer to “stop listening” before that reflection arrives. The result is the impulse response of the loudspeaker in the absence of any reflections – which is exactly what we want.

  • We make this measurement “on-axis” (usually directly in front of the loudspeaker, at some specific height which may or may not be the same height as the tweeter) to get a basic idea to begin with.
  • We can then rotate the crane (the device hanging from the ceiling that the loudspeaker is resting on) with a precision of 1º. This allows us to make an off-axis measurement at any horizontal angle.
  • In order to find the loudspeaker’s directivity (like Figure 5, 6, and 7 in this posting), we start by making a series of measurements (typically, every 5º for all 360º around the loudspeaker). We then tell the computer to compare the difference between each off-axis measurement and the on-axis measurement (because we’re only interested in how the sound changes with rotation – not its actual response at a given off-axis angle – we already got that in the previous measurement.
  • Finally, we have to get the power response. This one is a little more tricky. The power response is the behaviour of the loudspeaker when measured in all directions simultaneously. Think of this as putting the loudspeaker in the centre of a sphere with a diameter of 3 m made of microphones. We send a signal out of the loudspeaker and measure the response at all points on the sphere, add them all together and see what the total is. This is an expensive way to do this measurement, since microphones cost a lot… An easier way is to use one microphone and rotate the loudspeaker (both horizontally and vertically) and do a LOT of off-axis measurements – not just rotating around the loudspeaker,  but also going above and below it. We do each of these measurements individually, and then add the results to get a 3D sum of all responses in all directions. That total is the power response of the loudspeaker – a measurement of the total energy the loudspeaker radiates into the room.
Figure X: A prototype loudspeaker made of MDF on the crane, on its way out to be measured.
Figure 4: A prototype loudspeaker made of MDF on the crane, on its way out to be measured.
Figure 5: A BeoVision 11 during off-axis response measurements.
Figure 5: A BeoVision 11 during off-axis response measurements.

The original questions…

Great. That’s a list of the basic measurements that come out of The Cube. However, I still have’t directly answered the original questions…

Let’s take the second question first: “Why isn’t The Cube an anechoic chamber?”

This raises the question: “What’s an anechoic chamber?” An anechoic chamber is a room whose walls are designed to be absorptive (typically to sound waves, although there are some chambers that are designed to absorb radio waves – these are for testing antennae instead of loudspeakers). If the walls are perfectly absorptive, then there are no reflections, and the loudspeaker behaves as if there are no walls.

So, this question has already been answered – albeit indirectly. We do an impulse response measurement of the loudspeaker, which is converted to magnitude and phase response measurements. As we saw in Figure 5, the reflections off the walls are easily visible in the impulse response. Since, after the impulse response measurement is done, we can “delete” the reflection (using a process called “windowing”) we end up with an impulse response that has no reflections. This is why we typically say that The Cube is “pseudo-anechoic” – the room is not anechoic, but we can modify the measurements after they’re done to be the same as if it were.

Now to the harder question to answer: “Why is the room so big?”

Let’s say that you have a device (for example, a loudspeaker), and it’s your job to measure its magnitude response. One typical way to do this is to measure its impulse response and to do a DFT (or FFT) on that to arrive at a magnitude response.

Let’s also say that you didn’t do your impulse response measurement in a REAL free field (a space where there are no reflections – the wave is free to keep going forever without hitting anything) – but, instead, that you did your measurement in a real space where there are some reflections. “No problem,” you say “I’ll just window out the reflections” (translation: “I’ll just cut the length of the impulse response so that I slice off the end before the first reflection shows up.”)

This is a common method of making a “pseudo-anechoic” measurement of a loudspeaker. You do the measurement in a space, and then slice off the end of the impulse response before you do an FFT to get a magnitude response.

Generally speaking, this procedure works fairly well… One thing that you have to worry about is a well-known relationship between the length of the impulse response (after you’ve sliced it) and the reliability of your measurement. The shorter the impulse response, the less you can trust the low-frequency result from your FFT. One reason for this is that, when you do an FFT, it uses a “slice” of time to convert the signal into a frequency response. In order to be able to measure a given frequency accurately, the FFT math needs at least one full cycle within the slice of time. Take a look at Figure 6, below.

Figure 4: A 100 Hz and a 20 Hz sine tone, plotted for the first 20 ms.
Figure 6: A 100 Hz (blue) and a 20 Hz (red) sine tone, plotted for the first 20 ms.

As you can see in that plot, if the slice of time that we’re looking at is 20 ms long, there is enough time to “see” two complete cycles of a 100 Hz sine tone (in blue). However, 20 ms is not long enough to see even one half of a cycle of a 20 Hz sine tone (in red).

However, there is something else to worry about – a less-well-known relationship between the level and extension of the low-frequency content of the device under test and the impulse response length. (Actually, these two issues are basically the same thing – we’re just playing with how low is “low”…)

Let’s start be inventing a loudspeaker that has a perfectly flat on-axis magnitude response but a low-frequency limitation with a roll-off at 10 Hz. I’ve simulated this very unrealistic loudspeaker by building a signal processing flow as shown in Figure 7.

Figure X:
Figure 7: Signal Processing created to simulate a loudspeaker described above.

If we were to do an impulse response measurement of that system, it would look like the plot in Figure 8, below.

Figure X:
Figure 8: As you can see, this is almost a perfect impulse. It goes to a value of 1 at time 0 (therefore there’s no delay) and then it’s a value of 0 for the rest of all time… Or is it? Look very carefully at the line just after the impulse. Notice how it dips just below the 0 line? Let’s zoom in on that…
Figure X:
Figure 9: Zooming in on Figure 8.

Figure 9, above shows a closeup of what happens just after the impulse. Notice that the signal drops below 0, then swings back up, then negative again. In fact, this keeps happening – the signal goes positive, negative, positive, negative – theoretically for an infinite amount of time – it never stops. (This is why the filters that I used to make this high pass are called “IIR” filters or “Infinite Impulse Response” filters.)

The problem is that this “ringing” in time (to infinity) is very small. However, it’s more easily visible if we plot it on a logarithmic scale, as shown below in Figure 10.

Figure X:
Figure 10: The same plot as shown in Figure 8, plotted on a logarithmic scale by converting to decibels.

As you can see there, after 1 second (1000 ms) the oscillation caused by the filtering has dropped by about 400 dB relative to the initial impulse (that means it has a level of about 0.000 000 000 000 000 000 01 if the initial impulse has a value of 1). This is very small – but it exists. This means that, if we “cut off” the impulse to measure its frequency response, we’ll be cutting off some of the signal (the oscillation) and therefore getting some error in the conversion to frequency. The question then is: how much error is generated when we shorten the impulse length?

We won’t do an analysis of how to answer this question – I’ll just give some examples. Let’s take the total impulse response shown in Figure 6 and cut it to different lengths – 10, 15, 20, 25, 30 and 1000 ms. For each of those versions, I’ll take an FFT and look at the resulting magnitude response. These are shown below in Figure 11.

Figure X:

Figure 11: The magnitude responses resulting from taking an FFT of a shortened portion of a single impulse response plotted in Figure 8.

We’ll assume that the light blue curve in Figure 9 is the “reference” since, although it has some error due to the fact that the impulse response is “only” one second long, that error is very small. You can see in the dark blue curve that, by doing an FFT on only the first 10 ms of the total impulse response, we get a strange behaviour in the result. The first is that we’ve lost a lot in the low frequency region (notice that the dark blue curve is below the light blue curve at 10 Hz). We also see a strange bump at about 70 Hz – which is the beginning of a “ripple” in the magnitude response that goes all the way up into the high frequency region.

The amount of error that we get – and the specific details of how wrong it is – are dependent on the length of the portion of the impulse response that we use.

If we plot this as an error – how wrong is each of the curves relative to our reference, the result looks like Figure 12.

Figure X:
Figure 12: The magnitude of the error resulting from using too-short an impulse response to calculate the magnitude response. This is the “reference” curve in Figure 9 subtracted from each curve in the same plot.

So what?

As you can see there, using a shorted impulse response produces an error in our measurement when the signal has a significant low frequency output. However, as we said above, we shorten the impulse response to delete the early reflections from the walls of The Cube in our measurement to make it “pseudo-anechoic”. This means, therefore, that we must have some error in our measurement. In fact, this is true – we do have some error in our measurement – but the error is smaller than it would have been if the room had been smaller. A bigger room means that we can have a longer impulse response which, in turn, means that we have a more accurate magnitude response measurement.

“So why not use an anechoic chamber and not mess around with this ‘pseudo-anechoic’ stuff?” I hear you cry… This is a good idea, in theory – however, in practice, the problem that we see above is caused by the fact that the loudspeaker has a low-frequency output. Making a room anechoic at a very low frequency (say, 10 Hz) would be very expensive AND it would have to be VERY big (because the absorptive wedges on the walls would have to be VERY deep – a good rule of thumb is that the wedges should be 1/4 of the wavelength of the lowest frequency you’re trying to absorb, and a 10 Hz wave has a wavelength of 34.4 m, so you’d need wedges about 8.6 m deep on all surfaces… This would therefore be a very big room…)

Appendix 1 – Tricks

Of course, there are some tricks that can be played to make the room seem bigger than it is. One trick that we use is to do our low-frequency measurements in the “near field” which is much closer than 3 m from the loudspeaker, as is shown in Figure 13 below. The advantage of doing this is that it makes the direct sound MUCH louder than the wall reflections (in addition to making the difference in their time of arrival at the microphone slightly longer) which reduces their impact on the measurement. The problem with doing near-field measurements is that you are very sensitive to distance – and you typically have to assume that the loudspeaker is radiating omnidirectional – but this is a fairly safe assumption in most cases.

Figure X: A closeup of a Beolab 8000 woofer undergoing near field measurements.
Figure 13: A closeup of a Beolab 8000 woofer undergoing near field measurements.

Appendix 2 – Windowing

Those of you with some experience with FFT’s may have heard of something called a windowing function which is just a fact way to slice up the impulse response. Instead of either letting signal through or not, we can choose to “fade out” the impulse response more gradually. This changes the error that we’ll get, but we’ll still get an error, as can be seen below.

Figure X:
Figure 14: Windowing functions showing the gain vs. time for a 1-second impulse response, assuming that the impulse is at time=0.
Figure X:
Figure 15: The magnitude responses for the impulse response in Figure 8, using a Hann window to cut the impulse response to various lengths.
Figure 16:
Figure 16: The magnitude responses for the impulse response in Figure 8, using a Hamming window to cut the impulse response to various lengths.
Figure X:
Figure 17: The magnitude responses for the impulse response in Figure 8, using a Blackman window to cut the impulse response to various lengths.
Figure X:
Figure 18: The magnitude responses for the impulse response in Figure 8, using a Blackman-Harris window to cut the impulse response to various lengths.
Figure X:
Figure 19: The magnitude responses for the impulse response in Figure 8, using a Bartlett window to cut the impulse response to various lengths.

So, as you can see with all of those, the error is different for each windowing function and impulse response length – but there’s no “magic bullet” here that makes the problems go away. If you have a loudspeaker with low-frequency output, then you need a longer impulse response to see what it’s doing, even in the higher frequencies.

B&O Tech: Active Room Compensation – Some More Details, Part 2

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

 

Before we begin, if you’re unfamiliar with the concept of a “room mode”, it might be useful to review three other postings:

But what if my room is Scandinavian?

Active Room Compensation

and

Active Room Compensation – Some Details, Part 1

At the end of that last posting, I said:

“P.S. As I mentioned briefly in this posting, Active Room Compensation has one additional feature – multichannel processing. That will be explained later in Part 2.”

So, that’s the theme for this article – the multichannel aspect of Active Room Compensation.

 

A monophonic view

Let’s start by looking at Figure 1, below.

Figure 1

This is a basic representation of the fundamental room mode in one dimension of a room. If a loudspeaker is placed at the location of the red circle and it produces energy at the same frequency of the room mode, then the room will resonate, just like a closed pipe, as can be seen in the animation.

One important thing to notice in the figure above is that when the pressure on one side of the room goes positive, the opposite side goes negative. In other words, the two sides of the room are in opposite polarity. This information will come in handy later.

 

Now let’s look at the second harmonic – this is a resonance that has a frequency that is two times that of the fundamental mode. Its behaviour in one dimension of the room is shown below. Again, the loudspeaker is in the position of the red circle.

1spkr_mode2

Now, you’ll notice, the two sides of the room have the same polarity – when one goes positive, so does the other. It’s the centre of the room that is in opposite polarity to the sides.

If you measure the natural response of a loudspeaker that is otherwise flat (let’s not split hairs over whether we’re talking about on-axis magnitude response or the power response – for the purposes of this discussion, it’s irrelevant) in a one-dimensional room like the one shown above, you’ll see that there is a natural peak in the response at each frequency where you have a room mode. The result will look something like Figure 3, below

Figure 3
Figure 3: The simplified response of a loudspeaker in a one-dimensional room, showing the effects of room modes, but nothing else… Figures 1 and 2 correspond to the frequencies “f” and “2f” in this plot.

As you can see there, every mode is excited by the loudspeaker (we’re assuming that the loudspeaker is not sitting on a “node” and therefore not coupling to the room mode at all).

So, if you build a room compensation system that only takes one loudspeaker into account, then it will measure a response similar to the one in Figure 3, and it will create a compensation filter that looks something like the one shown in Figure 4.

Figure 4.
Figure 4: A simple compensation filter created to “undo” the effects of the room response displayed in Figure 3. In a simplest implementation, this is just the inverse of the room response.

Assuming that you’re careful about your measurements, and you consider things like phase response in your creation of the filter, this system will work very well with just one small problem: most people don’t use only one loudspeaker – they use at least two.

So, how does this change things?

 

Now, in stereophonic sound*!

Let’s go back and consider our room modes again, this time with two loudspeakers.

2spkr_mode1

 

Figure 5, above, shows the same room as in Figure 1, but now I’ve plotted the locations of two loudspeakers, one on the left (atypically, in red) and one on the right (in black). Notice that these two points in the room, when the room mode is ringing, are opposite in polarity (or “out of phase” as many people say…). However, consider that, in most recordings, the bass (which is, in most cases, a good estimation of the frequency band of the fundamental room mode) is panned to the centre, and therefore is “in phase” in the two channels.

In other words, in almost all cases, the two loudspeakers are producing the same signal at the same level, in phase (and therefore with the same polarity). However, the room is ringing in opposite polarity at the two loudspeakers.

What’s the result of this conflict? It’s simple – the room mode is naturally cancelled by the signals in the loudspeakers. In other words, there is no need to apply room correction for a recording like this, with the loudspeakers in the locations that they’re shown in, for this room mode.

So, if you measure each loudspeaker individually, you’ll put a dip in their responses that should not be there to compensate for a room mode that is not ringing. You must consider both loudspeakers playing a correlated signal, and how that will interact with the room mode.

Now let’s look at the next harmonic, shown below in Figure 6.

2spkr_mode2

Now you can see that the mode is ringing in the same phase at the two loudspeaker positions. So, if the bass (still a good guess…) is in phase in the two channels (also a good guess…) then this resonance will be twice as bad as it would be with only one loudspeaker. Again, we need to consider the behaviour of the room mode with a correlated signal in the two loudspeakers – but this time things are worse instead of non-existent.

So, this means that we have to re-consider our room compensation filter. Instead of measuring each loudspeaker independently and building a filter for each one and ignoring that people rarely listen in mono, we have to measure the two (or more) loudspeakers and analyse the way that different signals will interact with the room modes.

For example, in the simple case shown above, we might wind up with the two filters shown in Figure 7 and 8.

Figure 7
Figure 7: A compensation filter for a signal that is directed to two loudspeakers with the same polarity in positions shown in Figure 6. Notice that it is only necessary to compensate for the even-order modes.

 

Figure 7
Figure 8: A compensation filter for a signal that is directed to two loudspeakers in opposite polarity in positions shown in Figure 6. Notice that it is only necessary to compensate for the odd-order modes.

 

Now, you might be saying “I understand Figure 7 – lots of signals have the same polarity in the two loudspeakers (like the vocals, the bass, the kick drum – anything panned to the centre). But what signals are out-of-phase if I’ve connected by loudspeakers correctly?”

The answer to this comes mainly from classical recordings where it is normal to use microphones (usually omnidirectional) that are spaced apart. In this case, signals enter the two microphones are different times (depending on the angle to the sound source) – and a time difference results in a phase difference.

 

This processing is done in Beolab 90‘s Active Room Compensation to ensure that the loudspeakers are best optimised, not only to the room they’re in, but their locations within it, and its interaction with the recordings you’re playing. The end result is that each loudspeaker “knows” that the other one is in the room – each not only considers the other’s effect on the room’s response, but they “help each other” to control the room modes.

Of course, I’ve left out a lot of details in this description – for example, the actual responses of the correlated and negatively correlated signals will not really look like the ones I’ve shown here; most rooms contain more than one dimension; and I haven’t talked about boundary effects. In addition, everything I’ve said here is just an example using a very simplified view of the universe. The measurements of the loudspeakers at the microphone positions will result in very different responses than the ones shown here, which will, in turn, result in very different compensation filters.

 

*Most people don’t seem to know it, but “stereophonic sound” (or “stereo” if you’re into the whole brevity thing) means that you have two or more audio channels. Just thought I’d be explicit here – in case anyone was wondering.

 

 

 

BeoLab 90 reviews

Stereophile

“B&O has also succeeded in using all that technology to serve the music: the BeoLab 90’s sound quality equaled the general performance of any other speaker I’ve heard, and exceeded them in the resolution and stability of the sound-stages the pair of them presented.” (Kal Rubinson)

“For now, I exhort every caring music listener to listen to the BeoLab 90 and hear what is now possible. It’s that good. (KR)

“To say how impressed with how the BeoLab 90 measured in its Narrow mode would be an understatement. This loudspeaker demonstrates just how much can be achieved with intelligent use of DSP to optimise its acoustic performance. As Kal Rubinson summed up, it is a tour de force. Wow! (John Atkinson)

January, 2017 edition

Lyd og Bilde Norway

“Lyden av BeoLab 90 er vanskelig å forklare, den må egentlig bare oppleves. Personlig har jeg aldri hørt en mer livaktig musikkgjengivelse, og flere som har vært på besøk reagerte med å klype seg i armen eller felle en tåre når de hørte et opptak de kjente, eller rettere sagt trodde de kjente!”

“The sound of the BeoLab 90 is hard to explain, it must really be experienced. Personally I have never heard a more lifelike music reproduction, and several who have been visiting reacted by pinching their own arm or shedding a tear when they heard a recording they knew, or rather thought they knew!”

link

After Munich High End 2016

From fineartofliving.de: “Fazit: Bang & Olufsen BeoLab 90 – Die Zukunft des Klangs? Die Antwort darauf: Nach unserer Meinung zurzeit eine, wenn nicht sogar die beste Möglichkeit für optimales Klangerlebnis im High End-Bereich. Mit einer Einschränkung: Selbst angesichts des nicht minder beachtlichen Preises kann es aufgrund der hohen Nachfrage zu Lieferengpässen kommen…..

 

From www.hificlube.net:  High End 2016 – Munich – Best Sound: Runner-up – “My colleague Martin Colloms wrote a long feature article about the Beolab 90 for Hificritic. Still I wasn’t ready for what I heard. An immersive holographic sound even when I was sitting no more than 5 feet away from the loudspeakers. Excellent focus, amazing localization of musicians and singers in the huge and deep soundstage. Maybe a hint of upper mid emphasis. To say that I am amazed is an understatement. It gets an A verging on the A+…”

Press reviews from CES 2016

201602_bestofces

soundstagehifi.com: “…this demonstration was, by far, the best I heard – and the BeoLab 90 was my favorite product – at CES 2016.”

 

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stereophile.com: “The sound was incredibly dynamic but relaxed. This was the best bass I heard at the show, and the most startlingly different soundstaging and midrange I experienced during CES 2016.

“The question I’ve been asked at every CES more often than any other is “Which sound system impressed you the most?” It’s a question that’s often difficult to answer. Typically, I’ve been impressed by several systems, often for different reasons, and it’s hard to say which was “the best.” That is, except at CES 2016. This time, I had no trouble answering this question: the BeoLab 90. The sound was utterly natural, effortless, with outstanding soundstage and imaging. I also couldn’t help but be impressed by the technology represented by this speaker system: 18 drivers of the highest quality, 18 amplifiers, full DSP control, cast aluminum enclosure. Given the components and the extensive research that went into the design of the speaker, $75,000 for the pair does not seem excessive.”

 

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digitaltrends.com: “When we sat down for a short audition, we were gobsmacked by what we were hearing. Though we were in the very back of the room with enough people blocking our sight of the speakers that we could barely see the system, we heard a perfectly balanced presentation with the vocals seemingly coming from directly in front of us, just 2 feet away (someone’s head was in that spot).

 

 

 

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audio-head.com: “From common audiophile test tracks to a sweet spot rotation, whoever was leading the charge at the company certainly seemed vastly familiar with the hobby’s eccentric standards. The sound of the BEOLAB 90 followed in suit, the quick listening session yielded surprisingly tight image and responsive dynamic range. The bass was big, properly proportioned and beautiful in a way that reminded me of other high end rigs that play in this price range. Overall, it felt like a very impressive presentation that didn’t disappoint, checking off boxes left and right for both the audiophile and the luxury lifestyle listener.”

 

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AVSforum.com: “These speakers have real guts, and yet possess more finesse than passive speakers I’ve heard that cost even more. In the rarefied price point that they occupy, Bang and Olufsen’s flagship speakers actually strike me as a bargain, as outrageous as that may sound when discussing $80,000 speakers. But, that’s exactly how good they are. Respect.”

 

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soundstageglobal.com: “what I heard from the pair at CES 2016 absolutely blew me away — the way the pair imaged, coupled with the fullness, richness, and presence exhibited, had me talking about them for the rest of the day. Now I feel a little foolish for at first thinking they were a stupid design. Don’t make the same mistake — hear a pair of 90s if you can.”

 

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bloomberg.com: “We got a brief demo and the sound is incredible.”

 

 

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wired.com: “The soundstage is breathtaking, the low end is positively massive. Possibly the best speakers I’ve ever heard. Possibly the very speakers Crom has in his throne room in Valhalla.”

 

highdefdigest.com: “Having two of these exquisite pieces focused on me made for a smooth, powerful sound that will likely standout in my memory for the rest of my life. For those who can indulge, I strongly advise doing so.”

 

 

 

From the early press events

“So how did the BeoLab 90 make us feel? When we closed our eyes in Bang & Olufsen’s special listening room, the pair of master reference speakers (#2 and #3 ever made)—along with the room—seemed to vanish the instant a song played. We weren’t listening to sound emanating from two specific points; instead, the Weeknd was singing his heart out right in front of us. Benny Goodman’s band performed an intimate set, and you could picture where each musician was sitting. The BeoLab 90’s ability to create such a lifelike three-dimensional sound stage is unparalleled when you’re sitting in the sweet spot. It certainly brings up the question of whether a speaker can be “too” good for the music—some now-classic albums weren’t necessarily well-recorded and mastered (think of when the Rolling Stones turned the basement of a rented French mansion into a makeshift studio slash drug den). But when all the variables align perfectly, the music engulfs listeners entirely and hits the guts. The result of such incredible technology and engineering happens to be a very visceral human experience.”

– Coolhunting.com

 

“Die Abbildung war phänomenal, jedes Instrument der gewählten Musik nahm ganz selbstverständlich den für sich bestimmten Platz im Raum ein, jedes Element war von Anfang bis Ende verfolg- und erlebbar. Aber nicht nur die Ortung verblüffte, auch die Detailgenauigkeit, mit der selbst kleinste Feinheiten bis zum erkälteten Backgroundsänger aufgedeckt wurde, sucht ihresgleichen.”
– modernhifi.de

 

“Vi kan bevidne, at effekten er besnærende. Højttalerne spiller sammen med lytterummet på en måde, vi ikke har oplevet før. Personligt har jeg aldrig hørt et mere holografisk realistisk lydbillede, hverken i eller uden for sweet spot. BeoLab 90 er også en fuldblods, fullrange-højttaler, der ikke overlader noget til tilfældighederne.”
– lydogbillede.dk

 

“I found that the size of the soundstage was consistently proportional to the size of the ensemble and the recording. I found that the bass was very well extended, taut, and satisfying. Most of all, I was impressed by the prototypes’ reproduction of detail throughout the audio band, and the uniformity of that quality across the soundstage.”
– Kalman Rubinson, Stereophile magazine (print version, October, 2015)

 

“Wohl noch nie haben Lautsprecher die musikali­ schen Akteure so scharf ins Wohnzimmer projiziert. Ganz gleich, ob grosse Orchester oder kleine Jazz­-Formationen – jedes ein­ zelne Instrument hat seinen exakten Platz im virtuellen Raum, der auch seine Tiefen­ dimension verblüffend genau zu erkennen gibt: Der Hörer kann zum Beispiel fast in Zentimetern abzählen, wie weit das Schlag­ zeug hinter dem Kontrabass placiert ist. Dass der Beolab 90 auch für schwärzeste Bass­Tiefe, überbordende Dynamik und feinen, luftigen Obertonglanz steht, müssen Test­Hörer der Vollständigkeit halber natür­ lich ebenfalls zu Protokoll geben, aber das eigentlich Spektakuläre des Lautsprechers ist tatsächlich seine überragende räumliche Abbildung.”
– NZZ am Sonntag 18. Oktober 2015

For more comments and reviews:

How Bang & Olufsen’s BeoLab 90 Became a Reality: www.coolhunting.com

Skønheden eller udhyret? Bag om B&O BeoLab 90: www.recordere.dk

BeoLab 90: B&O laver banebrydende højttaler: www.lydogbillede.dk

Beolab 90 is Bang & Olufsen’s striking 90th anniversary speaker: www.whathifi.com

Der Traum vom Raum: Frankfurter Allgemeine

Bang & Olufsen BeoLab 90: Erster Hörtest: www.modernhifi.de

 

 

Outside the mainstream press…

diyaudio.com: “The Beolab 90 is a BEAST; I’m surprised Bang & Olufsen wasn’t kicked out of CES. I never even got close to the SPL limits of my Summas, but if the Beolab 90 had a limit, I couldn’t tell what it was. B&O cranked it up for a couple of tracks, and you could FEEL the bass. … The sound is clean and dynamic. The soundstage is whatever you give it; give it a good recording and the soundstage is pinpoint and huge, give it a crummy recording and it’s crummy. … If you couldn’t tell yet, this is the best speaker I’ve ever heard, bar none. In fact, I feel a little silly lumping it in with other loudspeakers.”

 

beoworld.org: I have heard a lot of powerful speakers, veeeery expensive speakers, crazy speakers and well, very much different setups. But the Beolab90 blew my mind!  … The power in them, the presence of all the tones, the clarity of the sound, the smoothness, the details…. I could go on for ever. It was like I had smooth whipped butter made out of diamonds inside my ears. I will probably never again, or at least for sure not in a long time hear a speaker like this again. I will surely hear the BL90 again, but I doubt something else will beat this. It could very well be the best speaker ever made. Simple as that. Mind you, that they had only the basic software in them, so they are just getting better.”