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NeverClip... in simple pictures


soundslikejustin

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This is something that I have made to simply show how and what NeverClip does. I appreciate that Glenn has tried to do this as well in his presentation, and this is not to diminish what information he has put out, only to support it, and simplify it. There are a number of people on this board that are confused by or mis-understand what NeverClip actually does (there's a billion pages of the 'NeverClip Explained' thread and there's still confusion)...

 

Note: This diagram does NOT take into account the variation on NeverClip that is now being put into transmitters - it works differently - but it is most like using an output compressor in the diagram provided.

 

NeverClip in Nomad works by producing a signal via it's dual 24-bit A-D that would be distorted if fed into a single 24-bit A-D.

 

NeverClip input does not suffer from input limiter distortion, because there isn't one.

 

NeverClip can create this signal because the internal mixer of the Nomad is 32 bit - it has room for this 'extra' digital signal.

 

If nothing is done to this extra signal and it is output or recorded through a 24-bit D-A output or card track, it will be clipped. A single 24-bit D-A or 24-bit file cannot handle this 'extra' signal.

 

If you compress this extra signal on it's way to the outputs or card tracks, you won't clip, though you may get some compressor/limiter distortion - BUT - the Nomad's output/card track compressor is digital, has look ahead and is very smooth.

 

Using ISO attenuation, the entire signal is gained down by a set amount - this includes the still-clean 'extra' digital information, and in doing so, you fit the high-dynamic range audio into a 24-bit file by replacing the Least Significant Bits (LSB) - which are traditionally filled with noise - with audio. This ONLY works for ISO tracks - it will not work for inputs routed directly to outputs.

 

I hope this helps.

 

post-2617-0-88707000-1367722047_thumb.jp

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so neverclip is already active but to fully utilize it and the headroom, you have to use the ISO attenuation? ive been trying to wrap my mind around this.  

NeverClip is always active, but you need to somehow deal with the 'extra' audio. Your options are:

1. using the faders (this will save your outputs but not your ISO's)

2. using output/track limiting - with possible (but unlikely) distortion from heavy limiting.

3. using ISO attenuation - which is only going to save your ISO tracks.

 

You can (and should) use combinations of the above options to the best effect.

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Great graph, but your track one with ISO attenuation is missing. As it is now one gets the impression that its level will be the same as in the other modes. But it will be attenuated too, so it will have to be adjusted in post.

Sure, but this graph was mainly to do with what happens to the signal that would normally clip. I will add this in, though. Thanks.

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Thanks for your picture and explanation.

 

Question 1 : your schema is done at a precise moment. If at a moment the level is normal, maybe a little low, and then, at another moment, the level is too strong, is there a transition between those two moments, any cross fade ? Or does NeverClip lowered the level for the whole track on all the take ?

 

Question 2 : When you write"with ISO attenuation", do you mean "with ISO attenuation by Neverclip" ?

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Thanks for your picture and explanation.

Question 1 : your schema is done at a precise moment. If at a moment the level is normal, maybe a little low, and then, at another moment, the level is too strong, is there a transition between those two moments, any cross fade ? Or does NeverClip lowered the level for the whole track on all the take ?

Question 2 : When you write"with ISO attenuation", do you mean "with ISO attenuation by Neverclip" ?

1. There is no transition between loud and soft sounds coming through the input - just think of it as a preamp with a high dynamic range - higher than what the outputs can be.

2. ISO attenuation is an option that the user can enable to fit this high dynamic range audio within the confines of a 24bit audio file. It gains the entire signal down (by an amount set by the user) to bring the extended dynamic range down. It's not compression, though.

If you have any recent DAW, you can try it for yourself.

Set up your session for mixing in 32 bit.

Send 6 tracks of tone or noise that peak at 0 out the same output - I suggest turning your speakers down first - it's going to distort because you're clipping the 24bit output converters.

However if you record those 6 tracks through a stereo buss, and then lower the gain of the entire recorded file so that it's back below 0dBFS, you've not lost any dynamic range, just moved it down, and it won't clip the outputs anymore.

I hope that makes sense.

This video deals with the concepts I'm talking about:

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even if signal is not distorted in the 32-bit recording because of Neverclip, when i mirror it to 24-bit audio there is a chance it may distort or clip bcos of the conversion?

Nomad does not record in 32 bit. If you send audio to an output or MARF card track that is clipping, your mirrored audio will be clipped to.

This is why you need to somehow reduce the level - the 'extra' audio can easily travel through the 32bit mixer of Nomad, but can't immediately be recorded.

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I'm still trying to wrap my head around the iso tracks. Are those now X dB lower than they would otherwise be, to avoid peak distortion? Wouldn't it be the same to essentially record all isos 10dB down?

The benefit to the ISO attenuation is the removal (or replacement) of the LSB (or 'noise bits') with audio. You can have an audio file with a greater dynamic range than is typically possible.

If you just turned your input gain down, the noise bits would remain. It's weird, because it appears to be the same thing, but it's not.

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The benefit to the ISO attenuation is the removal (or replacement) of the LSB (or 'noise bits') with audio

 

OK, thanks for your explanations.
It seems to me that your noise floor is anyway limited by the analog preamps, and that they can't be as good as 24 bits dynamic range ?
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OK, thanks for your explanations.

It seems to me that your noise floor is anyway limited by the analog preamps, and that they can't be as good as 24 bits dynamic range ?

The Zaxcom preamp is very low noise, very low distortion. The limiting factor in the past has been the range of the A-D chip. Now, it's not. I'd assume they've got the absolute end of the preamp aligned with the absolute ends of the a-d system as to maximize signal and minimize noise, but I don't know the specifics, so that might be a question for Glenn if you're curious.
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So if I have ISO Attenuation enabled, the outputs will sound the same when I'm monitoring, and if I monitor the inputs, they will sound attenuated?  I'm not really monitoring the inputs as I go, so no real difference as far as monitoring goes, correct?  I understand everything else I think but I'm a little fuzzy when it comes to monitoring the attenuated/unattenuated signal. 

 

I'll stop before I start typing a foreign language, I'm starting to confuse myself again.

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" Justin,

You are doing a great job explaining never clip. "

 I'd assume they've got "

" the internal mixer of the Nomad is 32 bit - it has room for this 'extra' digital signal. "

where is Glenn to give this interpretation his blessing  ?

 

" this graph should be a sticky, "

why?

Edited by studiomprd
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" Justin,

You are doing a great job explaining never clip. "

" this graph should be a sticky, "

" I'd assume they've got "

" the internal mixer of the Nomad is 32 bit - it has room for this 'extra' digital signal. "

where is Glenn to give this interpretation his blessing  ?

Agreed, thanks for the trying to help Justin, I'll still be skeptical until I hear it from the source..  And then I'll still probably be skeptical!  ;)

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