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Alex Weinberg

Zoom F6 vs Sound devices MixPre II series Dynamic Range

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The Zoom F6 has a dynamic range of 131 DB (I got that number from Zoom) and the MixPre II series has a dynamic range of 142 DB.

 

Since the whole point of the dual AD converters and 32 bit audio is that you cannot clip and no limiters are needed.

 

My question is do you guys think 131DB in the F6 is enough or the extra 11DB on the MixPre II series has any real advantage?

 

-Alex

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You cannot clip the 32-bit float converter, regardless of the dynamic range. At any given time the range from the loudest to the quietest sound will be either 131 or 142dB. I think both will be plenty. 

However, you may be able to clip the preamp or find it too noisy for really quiet parts. So to that end you may get better results with the better preamp. 

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Someone please explain to me how you "cannot not clip the converter". According to what I know, 0dBFS means the maximum "full scale", and anything that crosses above that line at the converter stage is indeed clipped (regardless of how many bits are being written after the fact). Is there a second circuit recording at an artificially lower level, and somehow when the clipping is detected, there's some automatic replacement with the lower recording, brought back up to match levels with the original recording...? To my understanding, only once you are in the digital domain, the extra bits are usable for enormous headroom... But I am probably wrong, and everything I have ever learned on this topic is no longer valid LOL.

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Clipping is a thing of the past! Unless of course if you, like everyone else, are using wireless systems with inferior dynamic range. Sure, you'll be able to get that whole range, but it'll still be less than what your recorder is capable of. It's only if you plug a mic straight into the recorder that you're safe. This will have a lot of newbies confused. Or is it magic!?

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7 minutes ago, Olle Sjostrom said:

Clipping is a thing of the past! Unless of course if you, like everyone else, are using wireless systems with inferior dynamic range. Sure, you'll be able to get that whole range, but it'll still be less than what your recorder is capable of. It's only if you plug a mic straight into the recorder that you're safe. This will have a lot of newbies confused. Or is it magic!?

 

Actually, there is no such mic, nor a pre-amp in existence, that has that kind of dynamic range.

Perhaps I'm living in the past, but surely not a newbie.

More bits / more dynamic range = lower noise floor.

So if you record, let's say at -40dBFS, just to take advantage of the headroom, then bring it back up to -8dBFS, you also brought up the noise floor by 32dB. Granted the converter noise floor will be extremely low, but anything else in the room, including you mics, pres, traffic outside etc are also brought up 32dB.

Again, I could be wrong, so please explain to me what I am missing.

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I remember taking to Glenn Sanders about field recorders moving to 32bit over ten years ago. I’m surprised it took this long to be honest. That said, a good engineer doesn’t need his/her tools to do their work for them. However I’m sure that 32bit recording will be happily implemented in the fx recording world. 

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3 hours ago, Johnny Karlsson said:

 

Actually, there is no such mic, nor a pre-amp in existence, that has that kind of dynamic range.

Perhaps I'm living in the past, but surely not a newbie.

More bits / more dynamic range = lower noise floor.

So if you record, let's say at -40dBFS, just to take advantage of the headroom, then bring it back up to -8dBFS, you also brought up the noise floor by 32dB. Granted the converter noise floor will be extremely low, but anything else in the room, including you mics, pres, traffic outside etc are also brought up 32dB.

Again, I could be wrong, so please explain to me what I am missing.

Your not missing anything, excellent point about relative levels. Most of the noise floor issues on a film set are all purely acoustic, so once again gain staging and signal to noise is important and it always will be.

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5 hours ago, Johnny Karlsson said:

 

Actually, there is no such mic, nor a pre-amp in existence, that has that kind of dynamic range.

Perhaps I'm living in the past, but surely not a newbie.

More bits / more dynamic range = lower noise floor.

So if you record, let's say at -40dBFS, just to take advantage of the headroom, then bring it back up to -8dBFS, you also brought up the noise floor by 32dB. Granted the converter noise floor will be extremely low, but anything else in the room, including you mics, pres, traffic outside etc are also brought up 32dB.

Again, I could be wrong, so please explain to me what I am missing.

No you're not missing anything... I was being sarcastic in the Swedish way. Sure, 32 bit is great and all, but it's still not magic, and it's still not going to solve everything for everyone, which some of the advertising suggests. And the newbie crowd, like the videographers who don't have time for learning audio at all, will still not be able to get good audio. Neither will aspiring photographers be able to get great images from a camera with a full sensor and Lightroom if they don't learn the basics and hone their skills. 

 

Don't get me wrong, I appreciate the benefits of 32 bit, I just think the marketing is a bit weird...

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6 hours ago, Johnny Karlsson said:

Someone please explain to me how you "cannot not clip the converter". According to what I know, 0dBFS means the maximum "full scale", and anything that crosses above that line at the converter stage is indeed clipped.

 

If the maximum possible incoming microphone level (in the case of MixPre-II, +12dBV) is mapped in such a way that the resulting output of the ADC does not exceed 0dBFS, then you will not clip it. Your next question is probably, 'yeah but what about if you boost input gain pushing that mic signal even higher beyond 0dBFS'. You would have a point if the input gain is applied pre ADC, but what if the gain is applied digitally post ADC? With 32-bit float processing you have a vast headroom above and below 0dBFS. As long as you don't boost digital gain by > 700 dB, you should be fine, although your ears may get a little sore;)

 

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21 minutes ago, Olle Sjostrom said:

No you're not missing anything... I was being sarcastic in the Swedish way. Sure, 32 bit is great and all, but it's still not magic, and it's still not going to solve everything for everyone, which some of the advertising suggests. And the newbie crowd, like the videographers who don't have time for learning audio at all, will still not be able to get good audio. Neither will aspiring photographers be able to get great images from a camera with a full sensor and Lightroom if they don't learn the basics and hone their skills. 

 

Don't get me wrong, I appreciate the benefits of 32 bit, I just think the marketing is a bit weird...

Hahaha, okej. My Swedish sarcastic humor must have worn thin after 25 years in Los Angeles.... and reading something in writing, it’s not always easy to get the “tone”.

 

Anyway, back when music was my main thing, I remember Steinberg introducing 32-bit float in both Cubase and Nuendo (early 2000’s). Once you were in the digital realm, you could basically push channels way above zero and add compression, boost eq and FX without clipping the tracks, nor the plugins. But again, that’s after being converted to digital, so it needed to come in clean before all that happened... and even so - imho, mixing with proper gain staging always seemed “better” to me anyway...

 

And btw, DAWs implemented 64-bit mixbus summing engines years ago... Logic gives you the option of “standard precision” (32-bit) or “high precision” (64-bit).

 

 

 

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18 minutes ago, Paul Isaacs said:

 

If the maximum possible incoming microphone level (in the case of MixPre-II, +12dBV) is mapped in such a way

 

SNIP

 

 if the gain is applied digitally post ADC

 

Thanks, Paul. That's basically my point. 

 

And sorry LOL, I am not trying to rain on your parade here. These new machines look great. Nice to see onboard TC in the 3 and 6 ! I may actually pick one up...

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Thanks all for the answers let me ask you this as far as input noise the Zoom F6 is -127 dBu vs Sound Devices which is -129 dBu.

In real world do you think there is any real noticeable difference in the noise floor when using ray a ribbon microphone?

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3 hours ago, Johnny Karlsson said:

 

Thanks, Paul. That's basically my point. 

 

And sorry LOL, I am not trying to rain on your parade here. 

 

I‘m not sure there’s actually any rain here. You said that if the input level would be pushed beyond 0dBFS then there would be clipping. However, as Paul Isaacs more or less said above (or that is how I understood it), the maximum output of the mic preamp is mapped to 0dBFS. You cannot apply any (more) gain in the analog realm to exceed that. You can only apply gain in the digital realm and this you cannot clip the converter or the processor. 

I think the idea is that the converter converts the entire range of the preamp and if you want to boost the signal you can only do so once it’s been digitized. 

With this 32-bit float business I think we have to sort of say goodbye to the concept of 0dBFS being the absolute max value in a digital circuit, even though it contradicts the very definition of 0dBFS. But I think that’s why Paul said if you align the max output of the preamp to 0dBFS on the scale, that would relate to 0dBFS in a 24-bit system. So you still have more than 700dB of headroom left and thus the claim: „you cannot clip the converter“ is still true. 

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12 hours ago, Johnny Karlsson said:

 

So if you record, let's say at -40dBFS, just to take advantage of the headroom, then bring it back up to -8dBFS, you also brought up the noise floor by 32dB. 

 

Yes, I think that’s true. However, to lower the overall system noise cannot be done in the MixPre anymore, because it will digitize the entire output from its preamp, and if I understood Paul correctly you can only apply digital gain, and that won’t affect the SNR anymore. Now the gain staging needs to happen outside the recorder, from transmitter to receiver into the recorder. That’s where you create a healthy signal and create enough level to keep away from any preamp or (possibly) converter noise. 

So I would set my digital gain much higher than -40. get it much closer to 0, because you have all that headroom above 0

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Maybe a new Topic about ''gain structure with 32pf recording'' would be appropriate, because the more I read about it, the more I'm trying to figure out how I'm gonna set the whole thing according to all that nice theory. For sure, I'll keep my Lectro rxs to max levels from this previous info just over here… but what about a phantom powered mic ? ISOs vs Mix tracks levels ? will it ''appear'' the same way on my meters ? Will I use the same scales/usual wokflow ?

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2 hours ago, Constantin said:

 

I‘m not sure there’s actually any rain here. You said that if the input level would be pushed beyond 0dBFS then there would be clipping. However, as Paul Isaacs more or less said above (or that is how I understood it), the maximum output of the mic preamp is mapped to 0dBFS. You cannot apply any (more) gain in the analog realm to exceed that. You can only apply gain in the digital realm and this you cannot clip the converter or the processor. 

I think the idea is that the converter converts the entire range of the preamp and if you want to boost the signal you can only do so once it’s been digitized. 

With this 32-bit float business I think we have to sort of say goodbye to the concept of 0dBFS being the absolute max value in a digital circuit, even though it contradicts the very definition of 0dBFS. But I think that’s why Paul said if you align the max output of the preamp to 0dBFS on the scale, that would relate to 0dBFS in a 24-bit system. So you still have more than 700dB of headroom left and thus the claim: „you cannot clip the converter“ is still true. 

 

You've got it Constantin! 

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23 hours ago, Alex Weinberg said:

Since the whole point of the dual AD converters and 32 bit audio is that you cannot clip and no limiters are needed.

 

 

PS: Sounddevices has a patent for a triple AD converter. Love how they overpassed the Zaxcom patent.

Curious to see how an US patent will fight another US patent in a US court.

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What do you mean "overpassed" the Zaxcom patent? We should be pleased if Sound Devices developed their own system and was granted a patent  ---  this is so much better than just stealing some other company's patented system. Patrick, do you have the patent number for the Sound Devices patent? I would love to see what they're doing.

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9 hours ago, Patrick Tresch said:

 

PS: Sounddevices has a patent for a triple AD converter. Love how they overpassed the Zaxcom patent.

Curious to see how an US patent will fight another US patent in a US court.

 

That is really great. I love innovation. What is the patent number? 

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2 hours ago, Jeff Wexler said:

What do you mean "overpassed" the Zaxcom patent? We should be pleased if Sound Devices developed their own system and was granted a patent  ---  this is so much better than just stealing some other company's patented system. Patrick, do you have the patent number for the Sound Devices patent? I would love to see what they're doing.

 

Hi Jeff

 

Here is the patent and all the stages explained. 

https://patentimages.storage.googleapis.com/a7/6b/f5/77e31e68cca8b7/US9654134.pdf

 

 

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A technology that dates back from the 90’s télécinéma machine (made by Kodak) that has been first patented by Zaxcom.

Copy pasted by Sounddevices for another patent. The next company could put 4 A/D and also patent it.

Is this innovation? 

 

Good the evolution made this tech easily available for a variety of end users. 

 

Still curious to see how Zaxcom will respond to that patent.

 

 

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The "technology" generally that you refer to is the use of multiple A to D converters to extend dynamic range, this technology has not and probably cannot be patented. Exactly how the use of multiple A to Ds is utilized, specifically, is and has been patented by several companies including Zaxcom. As long as Sound Devices has devised their own system and they are not utilizing the routines and procedures that are protected by the other patents, there is no problem. Sound Devices should be praised for developing their own system. If, however, they are actually just doing the exact same thing as is covered by Zaxcom's patent, that could be a problem.

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As I understand the Zaxcom patent for two gain stage recording is that it they do it in a very rudimentary way. Pretty much do it at one level, then do it another level and depending on if one exceeds the other or not perform and switch between stages. 

 

The SD solution is to perform at least two and in their implementation three gain stages and ADC in parallel. Use all three by themselves, merge into two or scrap one if corrupt. 

 

And then do process their vectors to create a final stream in 32-bit float. 

 

I really would like to see how Zoom is doing it in the F6. 

Or if the sudden manufacturing problem is somehow related to someones patent. 

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4 hours ago, Display Name said:

 

I really would like to see how Zoom is doing it in the F6. 

Or if the sudden manufacturing problem is somehow related to someones patent. 

That would be quite the plot twist!
 

6 hours ago, Patrick Tresch said:

A technology that dates back from the 90’s télécinéma machine (made by Kodak) that has been first patented by Zaxcom.

Copy pasted by Sounddevices for another patent. The next company could put 4 A/D and also patent it.

Is this innovation? 


The entire patent system is utterly messed up and should be totally chucked out, then started over again from scratch. 
 

 

This book is worth a read: https://mises.org/library/against-intellectual-property-0
( Or a much more concise read: https://mises.org/library/case-against-ip-concise-guide )

 

 

 

 

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