Active vs. passive [loud]speakers

[Biblob]

I just thought I'd use some sims to explain a bit better what I mean here.

All the following is based on a 5.5" SB Acoustics SBNRXC30-4 woofer in a 6-litre ported box with a tuning frequency of 55Hz. This is an affordable, well-performing small midwoofer designed to be used in compact standmount speakers.

Shown is the frequency response of the woofer (black) and port (grey):

View attachment 24543

And now the summed response (black) and port (grey):

View attachment 24544

As you can see, the summed response actually rolls off earlier than the port does. This is because, below the port tuning frequency, the output of the port is out of phase with the output from the woofer. There is absolutely no escaping this effect in a ported speaker.

You can also see that the -3dB point is at about 58Hz, which is just a shade above the port tuning frequency.

Next, here is the diaphragm displacement vs frequency at this SPL (96dB). This woofer has an Xmax of 5mm, so at 96dB, we are just a shade below Xmax at all frequencies from the port tuning frequency up.

View attachment 24545

However, although the woofer does not displace at all at the port tuning frequency, is displaces hugely below it. This is inevitably the case with any passive ported speaker, and is the main argument against them IMHO. Note, however, that displacement at the port tuning frequency and above is actually significantly less than it would be if the box were sealed - but that's another story.

Now this huge displacement below port tuning is definitely a problem. Fortunately, we can tame it quite easily with DSP, firstly by implementing a high pass filter just below the port tuning frequency. In red we see an example effect on our amplitude response of a 4th order high pass filter at 37Hz. Our -3dB has changed very little, but there is now far less stress on the woofer (an excellent thing of course).

View attachment 24547

And here is this filter's effect on diaphragm displacement (as you can see, the woofer will now not extend past Xmax at any frequency at our orginal SPL of 96dB).

View attachment 24548

So we've now used a high-pass filter to hugely reduce diaphragm displacement and resultant distortion, with almost no cost. That's the first and greatest advantage of using DSP to manage the woofer in a ported speaker.

But what if we want to actually lower the -3dB below what it was when the speaker was passive?

We could move our DSP high pass filter lower in frequency and then use some EQ to boost the frequencies between the high pass filter and the port tuning frequency to improve the bass extension. Perhaps we want to aim for a -3dB that is 18Hz lower in frequency, at about 40Hz (this is less of an extension than Devialet claims on the similarly-sized LS50, FWIW):

Here's our resultant SPL:

View attachment 24550

That's great at first glance. But now look at the diaphragm displacement (which FWIW is basically a direct indicator of nonlinear distortion):

View attachment 24551

That type of filter/EQ - even after high-pass filtering - results in a diaphragm displacement of 25mm in the region between the high pass filter and the port tuning frequency (i.e. where we've had to apply the EQ boost). 25mm is 5x this woofer's Xmax. Obviously this is an impossible amount of stress on the woofer.

Our only option if we want to keep the -3dB at 40Hz and not massively increase distortion and almost certainly permanently destroy the woofer, then, is to reduce SPL until the driver stays within Xmax.

Reducing the voltage to the driver by a factor of about 5 or 6 now gives us this diaphragm displacement:

View attachment 24552

That translates into an SPL around 15dB lower than we had with a -3dB point of 58Hz:

View attachment 24553

In other words, we've gained 18Hz of bass extension at the expense of 15dB of SPL. Or, alternatively, our speaker will now produce the same amount of bass distortion at 81dB as it did previously at 96dB.

We're not finished though. Sophisticated DSP can raise and lower the bass extension (i.e. change the corner frequency of the high pass filter) depending on the voltage of the input signal. This is (I suspect) exactly what the Devialet does. But it can't do much, if anything, to reduce the massively increased distortion at any given SPL that results from pushing the -3dB point below the port tuning frequency.

To illustrate, here is the driver displacement at 81dB before we lowered the -3dB point with a lower filter and bass boost:

View attachment 24560

That's 5 or 6 times less displacement, which will translate into 5 or 6 times lower distortion. Such is the cost of trying to make a ported speaker produce sound below the port tuning frequency.

In other words, if this is going to be attempted, it will have to be done extremely conservatively and carefully to avoid increasing the distortion the speaker produces by 5-fold or more.

So, when I see Devialet claim that their SAM system can push the bass extension of an LS50 (which is also a small ported speaker with IIRC a 5.5" woofer) down from 44Hz to 28Hz, I do believe them, but at the same time, I can be certain that the price paid for this is a massive increase in distortion at any given SPL, plus either (a) a huge reduction in max SPL or (b) a -3dB point that rises rapidly as SPLs reach even moderate levels.

If we choose (b), at low levels the speaker will play lower (but with higher distortion), while at high levels the speaker's -3dB will return to what it would have been with our original high-pass filter just below the port tuning frequency.

Personally, if Devialet's statements about e.g. the LS50 are true, I think they have either gone too far for the sake of specsmanship, or (more likely I hope) they are only true at very low SPLs.

(IIRC, Devialet also implements limiting to ensure the driver never exceeds Xmax. This and a voltage-dependant variable high pass filter can basically be used complementarily to achieve the same purpose of protecting the woofer at moderate-high SPLs while letting it play lower at lower SPLs. The downside of limiting is of course that it is a form of compression and its implementation will therefore reduce dynamic range.)

In all cases, of course, we have a speaker that performs better than it would have if it were completely passive. But the ported design places a pretty hard limit on just how much extra performance can be squeezed out of it.

Thank you for this elaborate post and example. I wil use this knowledge for my own design

 

[b1daly]

I would say that this implies that every active speaker can have 'a passive version', which I would dispute. One of the advantages of active speakers is that options are opened up that are simply not possible with passive speakers. For example, in the passive world desperate measures are taken to minimise the number of crossovers, particularly "in the middle of the vocal range", etc. resulting in a plethora of over-stretched two-way speakers which have many of their own problems.

In audiophile applications, straightforward DSP can make crossovers transparent, meaning that not only do they sound better per se, but that more 'ways' can be added without penalty. This is a virtuous circle because the demands on the individual drivers are reduced, constancy of directivity is improved and so on.

Well sure, my observation is just anecdata, I mention it only to counter what I see as a bias towards preferring systems that should be "superior" in theory.

More anecdata:

Amir posted this link from a recent listening test between JBL M2 and Revel Salon2:
https://www.avsforum.com/forum/89-s...iewed-speakers-ever-made-12.html#post54628832

In this shootout, the speaker with the passive crossover won decisively

Obviously, it's silly to reduce any such victory to a single factor like crossover design. But it's food for thought.

I have not heard any active crossover speaker that I like. Not that I've heard that many in the scheme of things. Maybe 20 or 30. In any case, that's crazy.

I use them for work, and they are OK, but I am frustrated by working on speakers that I think sound bad. My tastes seem to be an outlier, and I have only some notions of what might be behind my subjective experience.

My preferred listening speakers are old designs where perhaps speaker design necessarily required more "design by ear." Maybe including this element of subjective listening provides some benefits to the overall sound of a speaker?

These speakers are often far from neutral. I tend to do an in-depth EQ to address some aspects of speakeer sound that bother me, and wind up with a very enjoyable experience. Such speakers would be woefully unequipped as studio monitors. And the aspects of more modern bi-amped designs that I dislike I am not able to address with EQ.

As I've discussed in other comments, my fun listening tends towards rock, especially older rock, which was produced on, and for, speakers that have drivers, in boxes, with passive crossovers. I think it makes sense that such speakers would be good for listening to this type of music.

One function that I think the more "perfect sounding," bi-amped active speakers do not do well is as final "sound integrator." This makes sense in a way, because theoretically we just want a speaker to simply reproduce the signal it is presented with.

Hypothetically, the recording producers would manage to perfectly "integrate" their mix in the mix. But...

Part of this "integration" function is performed by distortions. Passive crossovers add distortion. The different bands of the crossover also interact with each other based on the program frequencies and overall levels.

A way to see the problem is to consider that overdubbed music often combines signals that were originally at very different SPL's and that have different levels of distortion.

A drumset in a room will completely overwhelm an unamplified vocal. The vocal will not energize the room enough to imprint its sonic signature enough to matter. But in an analog device, the vocal can energize the physical attributes of the overall playback system to a similar degree as the much louder drums. I think this helps "complete the illusion" that these sounds were played together, and that the musical expression of each instrument was really related to the other(s).

This issue has gotten worse as most mixing has moved to digital platforms.

You can see users responding to this issue by cranking the shit out playback systems in cars and using little boom box systems that are inherently distorted.

This issue is especially important at lower listening SPLs, where the sound just does not energize the room to a significant degree.

The other day I was listening to random music on our studios Genelec 8030/sub system, which is a rather pristine sounding system. The Rolling Stone's "Can't You Hear Me Knocking" came on and sounded just wrong.

Basically, I could hear each guitar amp being close mic'd, in the left and right speakers. The drums were too quiet and too far back (a lot of room sound). The Stones tended to record with amps set relatively quiet. In reality, the drums were much louder. I've listened to this song hundreds of times, and have a good sense of how it should sound, and it simply did not sound correct on these speakers.

I was listening to my whole Spotify collection on random, so the styles were all over the place. Mostly I felt that the reproduction experience of these speakers was pretty bad. However, on vintage synth pop like Human League, Depeche Mode, Cabaret Voltaire it sounded sublime. I have no idea what to make of that

Whether what I'm discussing could be generalized into speaker design I'm not sure. But I think it points to a significant problem in the whole concept of "hi-fi." The history of the hi-fi sensibility is dominated by the reproduction of acoustic instruments, recorded in a reverberant space. I find this annoying, as this is one of the least popular forms of music.

In any case, the metaphorical aims of hi-fi reproduction simply do not translate well outside of this narrow style of recording. The bulk of recorded material in the world are studio creations and the idea of reproducing the acoustic event of the original performance is meaningless.

Instead, I think of the overall system of recorded music production and consumption as a cultural activity in which software (music recordings) is produced to be played on music production systems (amps and speakers). The creators of this cultural product target implicitly and explicitly the generalized environment of playback systems that exist at the time of creation.

The musical "signal" is transmitted at a much different level of "resolution" than is expressed in the typical concerns of hi-fi enthusiasts. Issues like "signal to noise" performance of amps is largely irrelevant to the effective communication of the musical signal.

One of the main issues is that because these works of art were created on particular monitor systems, the coloration of that system is imprinted in an inverse form in the signal. This is more than something that can be addressed in mastering, or with DSP, because the feedback loop of the creative process happens over an extended time. It's a complex and subtle component of the final product.

This is a tangent, but to tie it back to the original subject of the post, a large body of recorded material has been produced which was not intended to be listened to on super accurate, controlled, bi-amped active crossover speaker, systems. So I'm doubtful that such speaker designs will be the best approach for listeners whose tastes fall in the category I like. (Rock and pop music from 1950 to 2000 or so).

I think about the playback system as a three-dimensional object that occupies space in the listening environment. As you've discussed elsewhere, the listeners move about the space and can integrate the sound of speaker in the room. When it works well a stereo system can generate a kind of stable "image" that is not so sensitive to the listening position.

I think people also integrate the visual, kinesthetic, and user interface aspects of the system. (They learn where the system is so they don't bump into it!)

A playback system is something we live with. It's a real object, making real sound, as opposed to something that creates a facsimile of another other reality.

 

[Ron Texas]

@b1daly your post touches on a lot of interesting areas.

The M2/Salon 2 comparison might not be the best example because the two speakers are so different in concept. An LS50 vs LS50W might be interesting, with the bass boost off on the LS50W. One may note the LS50W has a setting which is said to make it sound more like the LS50 passive. It probably turns off the delay on the LF driver.

When Zeos reviewed the Genelec 8341 he mentioned hearing background noises in a recording which he had never heard before. Your description of the 8030 seems in line with this. The speaker is so transparent it reveals that which was not supposed to be heard. That might why the LS50W has the sound like an LS50 passive setting. Back when CD's first came on the market there were complaints that the low noise floor made it possible to hear people coughing in the audience of live classical music recordings.

Perhaps we have to be careful what we wish for.

 

[Daverz]

Back when CD's first came on the market there were complaints that the low noise floor made it possible to hear people coughing in the audience of live classical music recordings.

Perhaps we have to be careful what we wish for.

The one I've read about is the rumble of the London Underground becoming audible in recordings made in Kingsway Hall.

 

[DoDe]

Nope, it's Canton GLE 496.2 vs. Canton GLE 496.2 BT. These are identical speakers, just passive vs. active. Feel free to find other measurements if you like, but I know for sure about the different measurements about these speakers since last year when I purchased the passive ones.

Also, read some SVS technical papers too, also some measurements as well. You'll see what a good active amp combined with DSP can do from speaker drivers. You can resolve deeps and peaks related to drivers and case resonant frequencies, but you can also correct phase too.

At least theoretically, well engineered active speakers with DSP built-in could do wonders.

Hi,
Is the DSP processing used if I connect the signal from an external DAC throw analog inputs to GLE 496.2 BT?

 

[trl]

I'm sure it is, because if you look on the measurements published by few reviewers you'll find out that peak and deeps were corrected in the BT version and I'm pretty sure they haven't changed the crossover inside.

 

[Chromatischism]

Back when CD's first came on the market there were complaints that the low noise floor made it possible to hear people coughing in the audience of live classical music recordings.

I hear this all the time and I don't listen to CDs. I think it's just the recording.

 

[DoDe]

I'm sure it is, because if you look on the measurements published by few reviewers you'll find out that peak and deeps were corrected in the BT version and I'm pretty sure they haven't changed the crossover inside.

Well on the measurements website there's no data about the configuration and we only can guess it's the same for every input.

Nevertheless I checked the sound from my Oppo UDP-203 connected throw HDMI and throw AUX input and I can't hear any difference. The interlink cable was the monster MC400 i2.
Does that mean the internal DAC is as good as the DAC inside Oppo (Oppo uses the AK4458VN)? Or does that mean my hearing is poor?

 

[AnalogSteph]

Nevertheless I checked the sound from my Oppo UDP-203 connected throw HDMI and throw AUX input and I can't hear any difference. The interlink cable was the monster MC400 i2.
Does that mean the internal DAC is as good as the DAC inside Oppo (Oppo uses the AK4458VN)? Or does that mean my hearing is poor?

Quite possibly neither. Both DACs may well be good enough by the standards of human hearing in general. It is not at all uncommon. Audible transparency was achievable in A/D-D/A chains 30 years ago.