Dynamic speaker processing

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Primoz Vozelj

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Apr 21, 2014
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I've heard various rumors online about some manufacturers using dynamic speaker processing. I don't mean protective limiters but frequency response that changes with drive level/SPL (dynamic EQ). In theory it makes sense since our ears hearing response also changes with SPL (equal loudness curve), so why not compensate for this with cheaper than ever DSP.
Has anyone actually measured a speaker whose TF changes with level, or is everything just rumors?
 
Primoz Vozelj said:
I've heard various rumors online about some manufacturers using dynamic speaker processing. I don't mean protective limiters but frequency response that changes with drive level/SPL (dynamic EQ). In theory it makes sense since our ears hearing response also changes with SPL (equal loudness curve), so why not compensate for this with cheaper than ever DSP.
Has anyone actually measured a speaker whose TF changes with level, or is everything just rumors?
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I know that Meyer and Renkus Heinz both implemented this in their analog processors back in the day. I haven't seen it implemented in digital before though.
 
Actually measured, no. Are manufacturers doing it? Hell yes they are. Yamaha's DSR series is one, I believe that Turbosound is doing it as well on some models and I am certain others as well. I just read the other day ( but can't remember the name of the vendor ) that one company is doing an active low cut based on SPl to help keep the speaker in line at high levels. It is being done and is done for good reason.
 
David Sturzenbecher said:
I know that Meyer and Renkus Heinz both implemented this in their analog processors back in the day. I haven't seen it implemented in digital before though.
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Urban legend, at least with the Meyer stuff - there was never any "sliding crossover". Could the different bandpasses go into limit independently? Yes. Would doing so make it sound as if the crossover point was moving? Yes, of course - the acoustic crossover would be moving if one passband could get louder but the other stayed static. But electronically the crossover was fixed. And any other speaker/processor that did this, new or old, would have the same effect.
 
Brian Bolly said:
Urban legend, at least with the Meyer stuff - there was never any "sliding crossover". Could the different bandpasses go into limit independently? Yes. Would doing so make it sound as if the crossover point was moving? Yes, of course - the acoustic crossover would be moving if one passband could get louder but the other stayed static. But electronically the crossover was fixed. And any other speaker/processor that did this, new or old, would have the same effect.
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Yep, reflects my thinking too.
I don't think any special processing, in terms of sliding crossover points, or tone control contours vs SPL, or etc,
is being done with any of the common self-powered boxes. Maybe not any boxes period.

Just plain ole limiting is all I've encountered, ...and that shifts acoustic xover as you say.

The only exception I've seen, is the Linea/Danley SC-48 processor, which does have a sliding-freq high-pass for the sub band.
They call it a x-max limiter, and I've watched the transfer function shift the high pass freq up with increasing input levels. It works.
 
Mark Wilkinson said:
The only exception I've seen, is the Linea/Danley SC-48 processor, which does have a sliding-freq high-pass for the sub band.
They call it a x-max limiter, and I've watched the transfer function shift the high pass freq up with increasing input levels. It works.
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That's my understanding of what Meyer did back in the last century for the original UPA. I've been told that was correct, and told it was incorrect. If I ever see Jon Meyer in person, I'll ask him. :eek:
 
It would be interesting to see compensation (not just EQ) re equal loudness contours (ie Fletcher-Munson) implemented in a truly dynamic/real-time DSP (and not being comparable to the home stereo garbage that tries to account for this)

The curves are VERY drastic in terms of human hearing of low and high frequencies across a volume range - it is ANYTHING but flat at low volumes, and gradually flattens as volume increases.

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While the idea seems quite reasonable on the surface, as demonstrated by the "loudness" button on many home and car stereos, even the more recent solutions such as Dolby Volume are still too simple and don't do it justice IMO.

Sure, you can do things like dynamically boost the bass depending on volume level. But at lower levels these big compensating boosts can make the low end sound kind of disconnected and unnatural. Consumer audio makers went all-in on putting such "dynamic" features on by default in their products a few years ago, but this seems to be a declining fad lately.

Doing it "right" might be contentiously subjective, and would present quite a number of challenges to implement. As an example, how to consider for auditory masking thresholds, keeping phase linear through changes, etc.

Some of it would likely be in the realm of implementing dynamically changing FIR filters - is that even technically possible right now?








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