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Limiter DSP for LFE channel

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Matt:
Are you sure it makes sense to split the signal based on frequency, limit part of it, and add it back together?  Is this common practice?

I'm trying to understand if this is better than just turning all frequencies down equally.  This approach has the advantage that it limits what actually goes to the speaker (voltage) where the other method still adds multiple signals and coherence / incoherence will cause you trouble.

Matt:
It's worth mentioning that an adaptive limiter will limit low frequencies more than high frequencies in practice.  This is because low frequencies are the only frequencies that use a voltage / speaker excursion large enough to hit the limiter.

There is an issue of what happens if there's a 20 Hz hit and 50 Hz it at the same time.  Each approach would sound different, but I'm not sure which approach would sound better.

mojave:

--- Quote from: Matt on September 11, 2012, 02:08:39 pm ---Are you sure it makes sense to split the signal based on frequency, limit part of it, and add it back together?  Is this common practice?
--- End quote ---


--- Quote ---It's worth mentioning that an adaptive limiter will limit low frequencies more than high frequencies in practice.  This is because low frequencies are the only frequencies that use a voltage / speaker excursion large enough to hit the limiter.
--- End quote ---

There is a recent thread at AVS that gets into a discussion about limiting and might be worth reading. Mark Seaton says the following in that thread: "While I know the upper octave of the subwoofer range are fairly easy for most woofers to produce, a 60Hz high level signal requires just as much Voltage as a 16Hz signal, and if you produce equal signals at the same time, that requires 2x the peak Voltage. This reality with program material is not well demonstrated with sweeps which only sweep a single frequency, but can be hinted at in the maximum capabilities, and is also where the compromise of using EQ to achieve a response vs. picking a driver with deeper natural response plays out, particularly if real use will tickle the limits of the subwoofer system."

Mark Seaton uses DSP amplifiers from Speaker Power. Danley Sound Labs also use these DSP amplifiers. I think Mark mentioned once that he splits the bass signal so he can specify where the limiting occurs. I also think I've read of others doing the same. If you look at the testing of subs at data-bass.com you can see that some of the consumer subwoofers with DSP amps have limiting that occurs below a certain frequency when a high level is reached.

I have tried to read the manuals for the Peavey IPR DSP amp and the Behringer iNuke DSP amp, but they don't say much about how their limiter works. I don't think it is common practice, but I think it is done.

I experimented with a clip from a movie (I think Iron Man's Jericho scene) and unless I split the frequencies, I still got limiting at the higher frequencies even though I didn't need it. I was going to split the frequencies and combine them again without and limiting and see if they were identical, but I never got around to it. In other words I wanted to see if splitting the signal was a detriment in any way.

Matt:

--- Quote from: mojave on September 11, 2012, 03:18:55 pm ---a 60Hz high level signal requires just as much Voltage as a 16Hz signal
--- End quote ---

That's true, unless you're adding a Linkwitz.  Then you're boosting the 16Hz signal a lot more than the 60Hz signal.



--- Quote ---I experimented with a clip from a movie (I think Iron Man's Jericho scene) and unless I split the frequencies, I still got limiting at the higher frequencies even though I didn't need it.
--- End quote ---

I'm intrigued by the idea of frequency based limiting.  But I'm not sure how to handle the math.  Imagine:

Split [Input] to [Low] and [High].  Imagine the goal is to limit [Low] + [High] to -1 to 1.

Do you limit both [Low] and [High] to -0.5 to 0.5?  This would mean [Low] couldn't use the value 1, even if [High] was at zero.

Or do you wait for [Low] + [High] to cross the limit, and then turn down _only_ [Low] until you're under the limit?  

The second approach seems reasonable, but what if [High] alone puts you over the limit?

And do you want to take [Low] right down to 0 to preserve [High]?

So maybe you should have a limiting ratio where you limit 80% [Low] and 20% [High]?

The slope of filter used to make [Low] / [High] would naturally allow some [Low] through in [High] and vice-versa, so that also has to be considered.

Matt:
The general idea is that limiting is applied as a smooth frequency curve that limits lower frequencies more strongly.

One way to do this might be to base the limiter around a high-pass filter (probably at 24 dB/octave, but we would have to test).

The limiting would be accomplished by varying the frequency of the filter.

If no limiting was happening, the frequency would be at 0 or 1 Hz.

If lots of limiting was happening, the frequency might be like 100 Hz.  This would make 50 Hz -24 dB and 25 Hz -48 dB.

The only question I have is if continuously adjusting the frequency of the high-pass would lead to phasing artifacts, or have a delay that causes problems (so a peak would slip through before it could adapt).

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