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NeverClip... in simple pictures


soundslikejustin

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I'm pretty sure it isn't a feature you can turn off --- it is the way the preamp > A to D converters work.

Jeff you are correct - NeverClip is there for you protection and can't be turned off. Now I think there may be some confusion is with ISO attenuation on the record tracks - which can be turned on or off - depending on your preference.

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I agree, I don't see how speculating leads to better understanding.

 

From my experience, when I have an overload while using a traditional preamp I hear clipping and when I have an overload while using my Maxx I just hear the limiter in action, not preamp clipping.

 

That's the only significant difference to me.

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I agree, I don't see how speculating leads to better understanding.

From my experience, when I have an overload while using a traditional preamp I hear clipping and when I have an overload while using my Maxx I just hear the limiter in action, not preamp clipping.

That's the only significant difference to me.

that's weird, because really it should be the other way around. The idea of NC is to not have a limiter at all, only as a last resort. That's the whole point of it.
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that's weird, because really it should be the other way around. The idea of NC is to not have a limiter at all, only as a last resort. That's the whole point of it.

 

To clarify I'm referring to the output limiter which I have set more conservatively to protect wireless transmitters or camera inputs. It's very easy to have limiter-free iso tracks using the ISO attenuation feature and limiter settings that allow greater dynamic range before engaging. 

 

I'm also referring to an overload situation, for example when you have your inputs trimmed for conversational level and suddenly everyone starts laughing at full volume.

 

To get the same kind of protection with a standard preamp + iso recording rig you have to under-amplify a bit on the trim and makeup a larger amount of gain on the main fader. 

 

With a NC input you set your trim for a good overall level and mostly forget it, allowing the limiter to handle peaks or just use the main fader to attenuate during loud passages. 

 

At least that's how I've been doing it and I'm getting good results.

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Agreed, Senator. This thread is doing more harm than good. I understand how the system works as far as the gain staging and dynamic range staging goes. The number of times it's been misconstrued is making us look bad.

I do think it's important for there to be understanding of what this system does by the professionals who operate it and also who will be pitching its benefits to potential clients and post. That, however is Glenn's message to relate and I think he has done this several times. What comes from him makes total sense to me and I think he has done the best job of explaining his system.

I would highly advise anyone interested in his system to find his own explaination of Neverclip (search, it's there) and also to seek out the Trew Audio Nomad/Maxx demo video as Glenn does a good job explaining his system there too.

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  • 5 weeks later...

Jeff you are correct - NeverClip is there for you protection and can't be turned off. Now I think there may be some confusion is with ISO attenuation on the record tracks - which can be turned on or off - depending on your preference.

 

But the extra dynamic range can't be recorded unless you use ISO attenutation, correct? It says in the first post ;

 

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.

 

 

So from what I understand, if I plug in a mic, crank up the fader/gain on a very loud source, route it directly to a track and press record...that signal will be distorted unless I use ISO attenuation, correct? Or am I misunderstanding this concept?

 

Sorry to bump the thread, just saw this and wanted clarification...

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But the extra dynamic range can't be recorded unless you use ISO attenutation, correct? It says in the first post ;

So from what I understand, if I plug in a mic, crank up the fader/gain on a very loud source, route it directly to a track and press record...that signal will be distorted unless I use ISO attenuation, correct? Or am I misunderstanding this concept?

Sorry to bump the thread, just saw this and wanted clarification...

Yes. If the signal coming in from the dual A/D of Neverclip remains high enough to exceed the 24-bit container of the recording tracks or output converters, it will be distorted.

You can lower the level using the fader (your post fader tracks will be safe, anything pre-fader may still clip), ISO attenuation (your pre-fader ISO tracks will have their level reduced to fit the 24-bit container and will not clip) or a compressor (might clip, or suffer compressor/limiter distortion).

Hopefully that helps a little.

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If you attenuate the signal, after in post you need to restore this attenuation. For example you attenuate 6dB. Clip gain in post at 6dB. NeverClip works in ISO tracks. Mix track, output bus still need comp/lim settings.

 

Am I correct?

 

:)

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  • 4 months later...

 

 

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.

 

 

Great diagram Justin.

 

By chance, have you found a diagram for the variation of Tx Neverclip?

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Great diagram Justin.

 

By chance, have you found a diagram for the variation of Tx Neverclip?

Neverclip in the TRX's now works more similarly to the Nomad version - there is no compressor (well, there's an optional one) - but instead of letting audio through into the internal mixer of Nomad over "0dB" (in a 24bit word), it switches to a preamp with a lower gain level. This means that normal talking, whispers, and shouting CAN all register around the same level on the TRX, as it changes preamp gain level.

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  • 2 weeks later...

Neverclip in the TRX's now works more similarly to the Nomad version - there is no compressor (well, there's an optional one) - but instead of letting audio through into the internal mixer of Nomad over "0dB" (in a 24bit word), it switches to a preamp with a lower gain level. This means that normal talking, whispers, and shouting CAN all register around the same level on the TRX, as it changes preamp gain level.

As much as I want your statement to be true it is not. 

Trx never clip is nothing like the Nomad never clip.

Compressor is not optional but required.

There is no option to disable it other then minimizing its effect.

But minimizing the compressor minimizes the never clip effectiveness.

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  • 2 months later...

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:

2 - What you are saying is a kind of compression... Also i would like to know what happens if you normalize an ISO track that has been attenuated, since you´ve technically lost the LSB which were bringing noise...

 

Also, shoudnt we be always using this attenuation, since we are losing noise?

 

Cheers!

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2 - What you are saying is a kind of compression... Also i would like to know what happens if you normalize an ISO track that has been attenuated, since you´ve technically lost the LSB which were bringing noise...

 

Also, shoudnt we be always using this attenuation, since we are losing noise?

 

Cheers!

 

it is not compression since the level is never attenuated acording a threshold level of  a control signal. It is merely a constant  attenuation to all the audio before being recorded  into a 24 bit word, since the signal can contain higher values than a 24 bit word is able to.

 

If  when recording you set the gain so the input signal never  goes over  0 dBfs, it is not necessary to use any attenuation. It works like a normal system. But If the level goes over 0 dBfs, it is necessary  to use attenuation to disk. The internal 32 bit float mixer is capable of work with higher than 0 dBfs levels, but the recorded audio file is 24 bit and cannot contain values over 0dBFS.

 

 

 

with the new available data I conclude this:

 

Never clip  in more detail

 

 0 dBfs is the maximum peak voltage  a normal system is designed to work with. When connecting "that" voltage level, the normal converter will generate a 24 bit sample word filled with "1" (so any voltage over that value will be coded with the same 24 bit word full of "1" i.e. clipping).

 

 In the never clip case, at 0dB fs voltage, the first converter wil generate a 32 bit float word that consist of :

-the same 24 bit word full of "1",

-and an 8 bit exponent that in this case doesnt count because  it represents the number "0".  

(any voltage over 0 dBfs will be coded with the same 24 bit word full of "1" i.e. clipping).

 

And here comes the "magic"

 

 The second converter is conected to a 20 db attenuated signal, so it will generate a 32 bit float word that consist of:

- a 24 bit word representig 20 dB less than the first converter,

- and an 8 bit exponent representing the number "0".

 

  Because the analog input signal is 20 dB lower, the signal /noise gets 20 dB worse than that of the first converter

 

Then, the internal digital 32 bit float mixer aply a 20 dB digital gain to this signal, so the sample becomes approximately "equal" to the sample of the first signal, restoring the attenuation of the analog input.

 

So, for a voltage input of 0 dBfs, the second converter plus the digital mixer will "finally" generate a 32 bit float word that consist of :

- a 24 bit word full of "1" and

- an 8 bit exponent representing "0"

 

( None of the never clip "converters" can generate a sample word representing a value over 24 bit,nor any of them can code voltage levels over 0dBfs , so the exponent in the 32 bit float word directly after the converters is allways zero. They are normal 24 bit converters.)

 

 

      When connecting voltages over 0 dBfs, the first converter will always clip. but the second converter analog input is 20 dB attenuated, so the second converter will start cliping (coding a sample word of 24 bit full of "1") when conecting an input voltage of  +20 dBfs.

     Also, because of the digital gain applied after the second converter ,when connecting input voltages over 0 dBfs, there wil be  digital signal levels representing voltage levels higher than 0 dBfs, and the internal 32 bit float mixer, will start to  use the exponent on the second signal, because when using the exponent it can represent values higher than is possible with the 24 bit word.  For coding voltages over 0 dBfs it is necesary to use the exponent, and then the system is really working in 32 bit float. ( I think this is the difference with never clip in wireless, since it  should use a 24 bit processor)

 

 

 in the end, the system have 2 nearly identical signals:

- the first with the better quality but allways with a 32 bit float sample word with an exponent equal to zero.   If the input voltage               exceeds   0 dBfs   the signal will clip.

- the second with 20 dB more noise, and his 32 bit float sample word could include an exponent different from zero if the input voltage       exceeds 0 dBfs.

 

Switching:  (see my next post for the real switching system taken from the patent )

 

The system is normaly operating with the first signal.

the switching probably uses some crossfade .

 

switching from the first signal to the second:

 

When conecting a voltage over 0 dBfs to a never clip input, the first signal will clip , but when the internal mixer detects a 24 bit sample word very close to full of "1"   (signal clipping), it switches to the second unclipped signal,( this signal will be unclipped up to + 20 dBfs).

 

Since the recording format is 24 bit type, when exceeding 0 dBfs at the input voltage, the 32 bit float signal must be attenuated  before going to disk so the exponent becomes zero, then , when sending to disk, the zero exponent is ignored.

 Remenber that this is attenuation, it is diferent than truncation. The less significant bits are there,  the noise go down, but te signal does it to. If you truncate (replace whit zeros the LSBs, so loosing them) you lose presicion  and increse the noise.

The reduced noise level for the quiet sounds with never clip is because the system uses the first (cleaner) signal to record them, but this lower noise is relevant Only if there is a big dinamic range in the acustic sound being captured , i.e. there must be loud sounds too, so the second signal is used, and so then there is a need for the atenuation . The loud sounds are recorded from the second noisier signal, with the extra noise "covered " by the loud sound itself. The quiet sounds are recorded from the first cleaner signal. 

 If the input voltage didnt go over 0 dBfs, then neverclip was not used, and atenuation was not needed, Atenuatting the signal will not improve the recording if never clip was not activated

 If digital attenuation is used, there is  an increse in noise due to imprecise calulation of the attenuation but this is absolutely inaudible in this case in a  32 bit float system, so aplying digital gain or normalizing a digitaly attenuated signal doesnt have an audible effect .

 

If the input voltage doesnt go over 0 dBfs, then it is not necesary to attenuate the signal, the exponent of the 32 bit float samples will allways be zero and the system will  just ignore it before sending to disk.

 

switching from the second signal to the first:

 

to perfectly do this, it is necesary to use a long buffer, but that would  introduce to much latency, so it is  not possible.

 

  The switching from the first to the second signal can be immediate, but switching from the second to the first needs a release time (using the full of "1" 24 bit sample word system for this  will lead in a switch about every 90 degree fase change in the input signal, so it is useless).

 

The system ideally needs to analize the second signal for a time enough to cover the longest ( higher "period" ) signal  ( 20 Hz signal= 50 miliseconds, this is the minimum but probably the product implemented time is longuer) to detect a real decrease in level under 0 dBfs before switching back. But using such a  time can  in some scenarios lead to a noticeable residual noise as described in another thread,  when recording pulsating or fast decaying  sounds.

Remember that the second signal has 20 db higher noise floor, so, when recording signal voltages over 0 dBfs and there is a fast decay of the sound, the 20 db extra noise floor remains for at least 50 miliseconds before the system switch back to the first cleaner signal.  This is more  noticeable when recording very quiet sounds, because normaly you would set the gain very high, and with this you raise the noise floor, and, plus the extra 20 db noise of the second signal, the noise floor becomes audible when there is no sound to cover it (after the fast decay)

  The release time have been adjusted by zaxcom to minimise this residual noise at least in the 742 plug on tx, I supose this is done by Hipassing the signal to 80 hz (in order to excluding the longer period signals), With this, they could lower the release time to nearly  a minimum of 12.5 to 20 miliseconds, 

 

 

 So, when recording  quiet and pulsating sounds, If you are sure there wil not be loud sounds, it is better to set the gain so never clip does not engages.

 

finaly

 

  Never clip is for high dinamic range acoustic sound material.

Is for saving you of the sudden scream, the unscripted performance changue, sudden laughs.

It replaces the limiter, but is not intended to replace a proper gain setting.

it is not for you to constantly  peaking over 0 dBfs on purpose.

Edited by Victor Rubilar
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Doug that already exists. All you need to do is create a HP matrix and call it label it as "PLAY". Then when you hit play Nomad will automatically switch the headphones to the "PLAY" matrix - when you hit stop nomad will then automatically jump back to the HP matrax you were at previously.

Is this a thing in Maxx as well?

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VR: " with the new available data I conclude this: "

you could look up the patent...

 

thanks  Senator, I didnt know the patent had been just published.

 All my general conclusions were correct since the begining.   ::)

If people know how it works,  then they can  make better choices on how to use it to get the best audio.

 

 

resume of main points in the oficial patent (if I understand it correctly):

 

http://patents.justia.com/patent/8878708

 

 

- diference of aproximately 20 dB of input level of the 2 converters through a gain stage or an atenuation stage.

 

-  " due to uncompensated differences between the two preamplifier circuits" and the use of the user selectable gain, to match  the 2 signals into the digital domain, the system applyes a "pre" digital gain  called CCG factor to make a stable diference between both signal levels, This factor is constantly calculated because in analog system the gain varies whith temperature,etc,

 

-  the switching can or can not be crossfaded.

 

- the switching from the first to the second signal is at -3 dBfs

 

- there is a small but important  error in the text I found. Maybe it is not in the originaly submitted document by zaxcom.

 

  This paragraph describes the normal operation.

"At 324, process 300 determines whether the value of the Timer variable is greater than zero. If no, process 300 proceeds to 328, at which the ADCComp [ first signal multiplyed by the CCG factor] audio sample calculated at step 318 is output to an outgoing audio circuit... "

 

When a sample over -3 dBfs is detected, the "Timer" is set from zero to 1000 and the second signal must engage...

"If, at 324, the value of the Timer variable is not greater than zero, process 300 proceeds to 326, at which the ADC2 [second signal] audio sample is output to an outgoing audio circuit. That is, the outgoing audio is switched to use the sample received from ADC2 rather than the calculated ADCComp sample..."

    It says " is not " and must  say IS  for the system to switch the signal.   ;D 

 

 

- the release time for switch-back from the second to the first signal is  controlled by the  "Timer".  It is set to "aprox". (?) 1000 every time there is a switch to the second converter, and decremented by "1" every time there is new "sample"  , It says that it takes aprox. 10 miliseconds to reach 0 (release time), so 1 Timer unit is not equivalent to a sample?!!!  :blink: , (if it was as they say, the release time would be aprox 20 miliseconds [see below] . I believe this part is not so well drafted). But then it says that the 10 ms. release time can be substituted. So the Tmer can be set higher in the real implementation.

  

   When in the second signal, the Timer is set to 1000 (again and again) every time there is a sample over -6 dBfs. So, the signal must be at least 1000 continuous samples under  -6 dBfs for the Timer to reach down to zero, When the Timer reach zero, the system switch back to the first signal. Acording to a fast calculation, defining the switch back to -6 dBfs with this system allows for a release time of 18.75 ms (3/8 of a wave cycle) in order to suport a 20 hz signal input.at 48khzSF, and 22.6 ms. at 44.1khzSF.

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yes

I must be missing something.

In Maxx I created a Headphone setting and called it "PLAY" (sans quotes).

When I play back it simply plays from whatever headphone setting I'm currently on. I can push the HP button to switch between different settings, including "PLAY", but it does not automatically switch to or from it.

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