Limiting causing distortion help

Posted: Mon Mar 31, 2008 6:36 am

Hi there all,

I've come across this problem before but resolved it. I've purchased a new card since and it's returned and i can't remember how i fixed it lol.

Anyways...

I mix my project in cubase sx 3 and have done a mixdown on it to move it for mastering.

When i get down to the final process, *the limiting* i get horrible crackling and distortion.

I know this just sounds like im pushing the threshold too much, but the distortion starts almost immediately after getting some attenuation on the track. Basically as soon as the limiter starts working. I used to be able to get the threshold a lot lower, and even then it wasnt the same kind of audio breakup i was getting.

I've tried several limiters and sequencers (thanks to a friend), all resulting in the same problem.

It's happened on both the sound cards i have owned, different sequencers and different plugins. So im assuming its a hardware problem.

I was just hoping someone might have had a similar problem or might know how i could start going about fixing it.

Thanks in advance for any help,

Sorry for the rambling but its not easy to describe lol

Posted: Mon Mar 31, 2008 6:41 am

First step -

If you're preparing your track for mastering - don't use a limiter. Period.

Let your mastering engineer use his/her limiter.

Your mix should have headroom prior to going to the ME. If it doesn't, there's very little they can do to your mix except fix (to varying degrees) the horrible mess made by the plug-in limiter you're using.

Sorry - this doesn't really address the problem of the clipping limiter. I'm just saying, don't use it.

Cheers-
Jeremy

Posted: Mon Mar 31, 2008 8:24 am

Look for "mystery bass" - something may have some serious low end stuff that isn't showing up on your speakers or in your tracks. If you can't hear it, you may be able to see it with a spectral analysis viewer, etc. It might be a track that shouldn't normally have much bass anyway, so it could be fooling you.

I'm with Jeremy though - don't do any comp/limiting at all if someone else is going to master your tracks.

Posted: Mon Mar 31, 2008 10:00 am

This also could be your computer crapping out on you. Make sure you aren't using up all your CPU power. A limiter on the master output can really take up some resources

Posted: Mon Mar 31, 2008 12:33 pm

I'm pretty confident that your problem is intersample peaks. Short answer: Don't hit -0 dB, go to something like -0.2 dB or possibly a little lower. That should clean it up a lot.

Long answer: When your sound card converts the digital samples that make up your waveform into a smooth, analog wave, the analog wave does not end up looking exactly like it did when it was digital. If there are two samples together that are higher than the ones on either side of them, the analog wave after conversion can (and probably will) actually go higher than those samples did when they were digital at the point in between where those two higher samples were. So if the digital samples were hitting -0 dB, you might actually end up with a positive number, which means clipping! I believe that's what you are hearing.

Getting new converters might (depending on the design) get rid of the problem...on YOUR machine. This can still be a problem on other people's machines, since their converters might do what yours are doing now. That's why my suggestion is just to master it at -0.2 dB or so, since that will provide a bit of extra headroom for intersample peaks to fall into and still not clip.

There is software that will show you if you are likely to have intersample peaks. And I'm sure analog equipment exists as well that could tell for certain. I'm not sure what that software is (sorry, I've totally forgotten), but I'd recommend using it if you can find it. You might also want to try a different limiter, if one exists that is designed to prevent intersample peaks from happening (again, I have no idea if that exists).

Hope this helps.

Posted: Mon Mar 31, 2008 1:56 pm

Not quite.

The sampling rate has nothing to do with amplitude peaks. That is a function of the bit rate. There is no way to exceed 0dBFS.

If I understand what you're saying, it's that the sample points occur on either side of a sinusoidal wave and that the peak itself extends higher than those sample points...

This isn't correct.

Again - amplitude is a function of bit rate, not sampling frequency.

A signal that is represented as:
111111111111111111111111

is clipped.

A signal that is represented as:
111111111111111111111110
is not
(assuming 24 bit processing).

Actually - edit of sorts - the first representation (24 '1s') is not technically "clipped" until there are two samples (or 3 if you listen to Sony's definition of clipping) that are successive which contain all 1's.

This too is an oversimplification since not all of the bits are truly used for the audio portion, but it's a good enough representation to be used as an example.

Posted: Mon Mar 31, 2008 2:42 pm

Hi

Thanks all for your replies and in answer...

Cucco: I've been mastering my own tracks now for years. The limiter is not put on my main mixes. I mix down, then start a brand new project where apply the mastering chain.

Crankitup: Im getting the problem with only a limiter in the chain. My CPU can handle that..it has before. Which is why i dont understand it. It happened on my old card but went away on its own lol. Now i have an emu 1212m it has returned, and yes it has been fine on my emu previously.

danbronson: This is exactly what i thought at first too, but in experimenting ive dropped it as far as -7 and yet it still is yeilding the same horrible as soon as the limiter kicks in.

I really do appreciate all your answers, this is driving me crazy! lol. It's happening right in the middle of recording an acoustic album. Doing my head in!!

If you can help further id be very greatful.

Posted: Mon Mar 31, 2008 3:55 pm

Acoustic album, eh?

Perhaps it's the inherent wider dynamic range of the instruments that's causing the problem?

What limiter(s) have you tried so far that are giving you trouble?

Posted: Tue Apr 01, 2008 2:05 am

Well everything really lol.

I have UAD-1 package with the precise limiter. Also i tried waves L2 and voxengo one from a friend.

I dont think it's the software though, they all seem to give the same results.

I've tried sweeping frequencies in the mastering process before the limiter, also doing a heavy compression before going to the limiter to try and squash any naughty dynamics but doesnt seem to make any difference.

I'd start pointing the finger at my alesis monitors if it wasnt for the fact that i had the problem before and it went away. Speakers don't fix themselves. Shame.

Posted: Tue Apr 01, 2008 6:41 am

Oh yes they do...
Poke the cable about a bit, hear it burst into life. Low end connections, grr.

Maybe your limiter is being set unbelievably harshly or some setting has snuck under your radar and is screwing up the settings. Maybe something like the input gain being up or auto adjustments...

Posted: Tue Apr 01, 2008 7:02 am

Hmmm...all sorts of possibilities.

Perhaps the output of the limiter is set to a value greater than 0.0?

Perhaps the routing of limiter is the problem? Maybe it's before another effect in the chain? Something I found interesting about my UAD Precision Limiter is that for some reason put into the bus prior to the master output. If you put a master buss EQ on, it comes after the limiter. This will cause clipping for sure.

My point is - don't just assume that because it's turned on, it's assigned in the right order. I'm not sure what software you use, but with Sequoia, you simply click the routing button and you can move the limiter to the last piece in the chain and problem solved.

Posted: Tue Apr 01, 2008 9:09 am

Cucco wrote:

Not quite.

The sampling rate has nothing to do with amplitude peaks. That is a function of the bit rate. There is no way to exceed 0dBFS.

If I understand what you're saying, it's that the sample points occur on either side of a sinusoidal wave and that the peak itself extends higher than those sample points...

This isn't correct.

Again - amplitude is a function of bit rate, not sampling frequency.

A signal that is represented as:
111111111111111111111111

is clipped.

A signal that is represented as:
111111111111111111111110
is not
(assuming 24 bit processing).

Actually - edit of sorts - the first representation (24 '1s') is not technically "clipped" until there are two samples (or 3 if you listen to Sony's definition of clipping) that are successive which contain all 1's.

This too is an oversimplification since not all of the bits are truly used for the audio portion, but it's a good enough representation to be used as an example.

Perhaps I didn't explain it well. Here is a link that explains with helpful diagrams what I'm referring to.
http://www.hometracked.com/2007/11/08/prevent-intersample-peaks/

But nick1982, if you brought the output to -7 and the distortion was still there, I'm not sure what the problem is. Make sure the software you're using can handle the sample rate and bit depth of your recording.

Posted: Tue Apr 01, 2008 9:19 am

No - the information on that page is plainly incorrect.

They are assuming that the amplitude and frequency are represented by a single numerator (operating in time as a straight line).

It's not. It is represented by a plot of two factors (frequency on the x domain, and amplitude on the y domain.) Since this is sampled 44,100 times per second (for 44.1kHz sampling), the time line is drawn by the assembly of each of these samples.

Seriously - that page is completely wrong and its author should be beaten with a rubber hose.

What he's referring to is simple clipping - if an analog signal goes beyond the capabilities of the digital representation (greater than all 1's), then the signal is clipped and you have 2 or more concurrent 0dBFS samples. There is nothing that occurs between samples. Period. Yes, in analog, there is. In digital, there is not.

This page propogates bad information that is often misunderstood.

Posted: Tue Apr 01, 2008 9:44 am

Cucco wrote:

No - the information on that page is plainly incorrect.

They are assuming that the amplitude and frequency are represented by a single numerator (operating in time as a straight line).

It's not. It is represented by a plot of two factors (frequency on the x domain, and amplitude on the y domain.) Since this is sampled 44,100 times per second (for 44.1kHz sampling), the time line is drawn by the assembly of each of these samples.

Seriously - that page is completely wrong and its author should be beaten with a rubber hose.

What he's referring to is simple clipping - if an analog signal goes beyond the capabilities of the digital representation (greater than all 1's), then the signal is clipped and you have 2 or more concurrent 0dBFS samples. There is nothing that occurs between samples. Period. Yes, in analog, there is. In digital, there is not.

This page propogates bad information that is often misunderstood.

I don't understand what you mean. How is it wrong? Are you saying intersample peaks don't exist (created in the DAC process)?

Keep in mind, nobody (by that I mean myself or the page I linked to) has claimed that the digital level can go higher than 0 dB. Just that if two samples next to each other hit 0 dB, the converted analog signal at the point in between those samples can exceed 0 dB, depending on the converters used. You're saying that's incorrect?

I'm far from the most educated person out there when it comes to this stuff, so I appreciate any correction to this you can give me!

Posted: Tue Apr 01, 2008 10:03 am

It's just it doesnt explain why it has worked for a year previously. This is what is baffeling me. It's not new hardware or software. It's all worked fine, then it didnt...then it did again for a long while and now it doesnt lol.

That's why im having trouble finding the problem as its worked flawlessy with my current setup for months on end.

Sad

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