Too loud? My experiments with Limiters = square waves...

Posted: Sat Jan 07, 2006 6:43 am

Here is one more stupid question, for ya. Very Happy

Is there any difference between clipping inside eg an application like Sound forge/Wavelab at 0dB making them square tops, compared to clipping inside eg a VST plug, that limits it's output? Eg if a VST plug is set to -0.1dB max output, then I guess that Sound forge/Wavelab etc dont consider the peaks to be to be clipped? Confused

But looking at the wave form
(that is some kind of dynamic representation of all the dynamics in all frequencies bands) it shows clippings anyway. It might sound okay or not, listening to it? But the eye is fooled in some way, IMHO. And the audio editor programs stats might lie?

Moderator Joined: Sep 12, 2002 Posts: 1867 Location: NYC

there is a difference in sound between the two. I find that plugs do a much worse job in handling clips. probably because they aren't written to handle them well.
Nova (apogee UV22 option) was basically just taking the overs down a hair so they wouldn't look like overs.

Posted: Sat Jan 07, 2006 10:46 am

Michael Fossenkemper wrote:

there is a difference in sound between the two. I find that plugs do a much worse job in handling clips. probably because they aren't written to handle them well.
Nova (apogee UV22 option) was basically just taking the overs down a hair so they wouldn't look like overs.

Okay, thanx for the answer, I am gonna ask some more. Very Happy

1) The audio editor programs, what do they do? take down the overs or
just cut the output?

I guess the recommendation is probably to use some built in dynamic plug/algoritm/VST, to do the job? To get the best result.

2) Are there any VST plugs/built in algoritms that can reverse the process or is a clipp not reversible, sound wise? I guess it could be possible to get some dynamics back from a very squashed mix? If eg the clipp gets diminished in time, it might not 'rattle speakers' so much, that it will be heard? This is a very hypothetical topic. Wink

3) Also within a software, digital distortion seems like an abstract, IMHO. At least for an offline algoritm, that can analyse the dynamics and handle
potential clipps causing digital distortion in a better way.

Moderator Joined: Sep 12, 2002 Posts: 1867 Location: NYC

I'm not sure why you are asking, but i'll give it a go.

Clipping is not a linear process, whenever there is a non linear process, it produces junk, (harmonics). Now there are units out there that do a better job at controlling them then others. Now these frequencies that are produced are beyond the sampling frequency. When they go beyond this point they are aliased back into the audible frequency. this is dependent on what frequency was clipped of coarse but there is a formula for figuring out where that alias would end up, but I don't know it. If you were to use a unit that did this process in a higher sampling rate, then filtered out these frequencies before they could be aliased back into the audible spectrum, then you would have a better chance at reconstructing the waveform. Otherwise you could kind of reconstruct the waveform but you'd still be left with these aliased frequencies that don't sound very good. Now I would imagine that different audio engines in various programs deal with these generated non linear frequencies differently. Some do it very well and other don't. Plugins are about the worst I've heard.

Moderator Joined: Sep 12, 2002 Posts: 1867 Location: NYC

Actually Nika used to hang around here. he wrote some really informative papers on digital that you might find usefull.

Posted: Sun Jan 08, 2006 4:12 am

Does limiting on wave cycles, or half wave cycles have anything to do with this? For example, would limiting on a bass peak have different distortion harmonics compared to limiting on a treble peak?

Posted: Sun Jan 08, 2006 8:03 am

In short - yes. The harmonics would be significantly different.

Posted: Sun Jan 08, 2006 8:32 am

Michael Fossenkemper wrote:

I'm not sure why you are asking, but i'll give it a go.

Clipping is not a linear process, whenever there is a non linear process, it produces junk, (harmonics). Now there are units out there that do a better job at controlling them then others. Now these frequencies that are produced are beyond the sampling frequency. When they go beyond this point they are aliased back into the audible frequency. this is dependent on what frequency was clipped of coarse but there is a formula for figuring out where that alias would end up, but I don't know it. If you were to use a unit that did this process in a higher sampling rate, then filtered out these frequencies before they could be aliased back into the audible spectrum, then you would have a better chance at reconstructing the waveform. Otherwise you could kind of reconstruct the waveform but you'd still be left with these aliased frequencies that don't sound very good. Now I would imagine that different audio engines in various programs deal with these generated non linear frequencies differently. Some do it very well and other don't. Plugins are about the worst I've heard.

Hi Michael

Thanx, this make very good sense to me. Smile

Now I think I understand why some clipps sounds worse than others. Or at least I have something
to hang up my reasoning on. Very Happy

I will start to try track down some of Nikas writings. Smile

Must try to get hold of a copy of his book. Very Happy

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