Zoom F6 Portable Field Recorder Review

[restorer-john]

In early digital era, for example, you can see that this DAT machine has marker on -20dBFS when looking at the meter.

Oh, that Sony Pro DAT is real purty...

 

[Rja4000]

Even for the low sensitivity dynamic mics? I suppose proper interfaces with 50dB+ worth of gain are going to work well with something like a Shure SM7B, but something like a Focusrite Solo that maxes out in the high 40's if I recall, may leave something to be desired. Though still good enough nonetheless.

You have to be careful with Mic preamp gain marking in digital audio interfaces, since there is no standard defining what those "gain" marks mean (or in other words, what's the 0dBFS level in dBu or dBV).

As an example, I have a Motu 828mkIII (FW), which seems to reach 0dBFS with a -58dBu signal for "53" gain mark, while a Yamaha AD8HR would allow up to -33dBu for the same gain mark.
Thats a massive 25dB difference.

 

[Rja4000]

Nowadays +24dBu = 0dBFS

I see 0dBFS = 20dBu on (pro) Yamaha preamps.
And I have a Focusrite Liquid 4Pre where 0dBFS=+22dBu
But that's purely abstract, since it's just based on what figure they use to label each gain position.

 

[bennetng]

I see 0dBFS = 20dBu on (pro) Yamaha preamps.
And I have a Focusrite Liquid 4Pre where 0dBFS=+22dBu
But that's purely abstract, since it's just based on what figure they use to label each gain position.

That's why my original post said

Nowadays +24dBu = 0dBFS is common used, albeit some lower-end or bus-powered equipment often fall into somewhere between +10 to +20dBu.

These info are listed on product specs, so not abstract at all. Users need to offset that value to achieve the desired headroom.

 

[Rja4000]

These info are listed on product specs, so not abstract at all.

True.
But we have no way to correlate that with an actual preamp gain.
We just have their word for it.

 

[Rja4000]

my original post

Those I listed are not exactly "Lower-end"

 

[restorer-john]

You have yo be careful with Mic preamp gain marking in digital audio interfaces, since there is no standard defining what those "gain" marks mean (or in other words, what's the 0dBFS level in dBu or dBV).

As an example, I have a Motu 828mkIII (FW), which seems to reach 0dBFS with a -58dBu signal for "53" gain mark, while a Yamaha AD8HR would allow up to -33dBu for the same gain mark.
Thats a massive 25dB difference.

And this matters exactly why?

 

[bennetng]

Those I listed are not exactly "Lower-end"

My original post mentioned tape recorders (analog and digital), so the subsequent example of 10-20dBu and 24dBu are also not preamps. So it also makes perfect sense for line-ins to have slightly higher headroom. Anyway, if the term "lower-end" offended you, I am sorry about that. That post was not intended to reply your previous posts.

 

[PeteL]

A good page on 32 bit float
https://www.sounddevices.com/32-bit-float-files-explained/


Sound devices control automatically the gain of preamp for avoid clip. This gain is encoded in the 32 bit float.

A fun video and good demonstration that show the possibility at

at 1:16

I Kinda get this. Still doesn't fully gets that this would mean fully fixed gain analog stage, maybe part of the answer here with a previous post: @jerryfreak

"The dual ADC is used when recording in 32 bit float and when recording in 24 bit. The Trim allows you to set the level that will be recorded to the 24 bit file. This makes it so that if the signal clips in 24 bit mode, it is due to the file clipping from the limitations of the 24 bit integer format, not from clipping at the converter."

Could we assume from this, maybe it's my understanding that is not fully arrived, but the first adc in the chain would be used to set the gain but it would control an analog preamp, some sort of resistor network, but only the second adc would be the only actual one responsible of of the digital conversion of the recording? Or does that really mean that the gain stage is fully fixed and would accomodate enough headroom for any input signal and that it is all fully digital from there?

 

[Blumlein 88]

I Kinda get this. Still doesn't fully gets that this would mean fully fixed gain analog stage, maybe part of the answer here with a previous post: @jerryfreak

"The dual ADC is used when recording in 32 bit float and when recording in 24 bit. The Trim allows you to set the level that will be recorded to the 24 bit file. This makes it so that if the signal clips in 24 bit mode, it is due to the file clipping from the limitations of the 24 bit integer format, not from clipping at the converter."

Could we assume from this, maybe it's my understanding that is not fully arrived, but the first adc in the chain would be used to set the gain but it would control an analog preamp, some sort of resistor network, but only the second adc would be the only actual one responsible of of the digital conversion of the recording? Or does that really mean that the gain stage is fully fixed and would accomodate enough headroom for any input signal and that it is all fully digital from there?

I don't think it works that way. It is said to use two ADCs. One for high and one for lower level signals. Then they are used to create a clean 32 bit result. The combining is the tricky part. And no I don't mean 32 bit or even the full 24 bit range.

As an alternative imagine this. I split my mic feed to two inputs. One I run at a normal gain, maybe 40 db thinking it probably won't clip. The other I run at 0 gain or maybe only 10 db gain. And occasionally that 40 db gain side does clip some. I can take the low gain version for just that segment that clipped, amp it up digitally to the proper level and replace the distorted clipped section. It would have a raised noise floor briefly, but would beat being clipped.

 

[PeteL]

I don't think it works that way. It is said to use two ADCs. One for high and one for lower level signals. Then they are used to create a clean 32 bit result. The combining is the tricky part. And no I don't mean 32 bit or even the full 24 bit range.

As an alternative imagine this. I split my mic feed to two inputs. One I run at a normal gain, maybe 40 db thinking it probably won't clip. The other I run at 0 gain or maybe only 10 db gain. And occasionally that 40 db gain side does clip some. I can take the low gain version for just that segment that clipped, amp it up digitally to the proper level and replace the distorted clipped section. It would have a raised noise floor briefly, but would beat being clipped.

Thanks for the explanation, Interresting thread.

 

[thefsb]

Could a ground loop involving the USB cable possibly impact performance?

I have a Zoom G1 Four that sounds really bad when it's plugged into the computer via the (Motu M4) audio interface and USB at the same time. Unplug the USB and it's fine.

 

[thefsb]

Guessing Zoom are targeting that crowd.

Or people who aspire to it. For example, independent amateurs, semi and full pros who need a field mixer/recorder.

Of all the people I've known in broadcasting, I can't think of one who would overlook those terrible A/D results.

Without tests like Amir's published and available it's hard to be sure if a low-priced unit is a waste of money or not.

 

[Rja4000]

And this matters exactly why?

It doesn't.
Just that we shouldnt take 2 interfaces, look at the "gain" marking and draw comparisons.

My original post mentioned tape recorders (analog and digital), so the subsequent example of 10-20dBu and 24dBu are also not preamps. So it also makes perfect sense for line-ins to have slightly higher headroom. Anyway, if the term "lower-end" offended you, I am sorry about that. That post was not intended to reply your previous posts.

It did not.
Just want to highlight that's not that simple.

OK, all this is pretty much off topic. Sorry.

 

[jalaute]

Is there a down side (isn't there always?) to 32 bit float and the and the ability to recover in post? Perhaps a stupid comparison (and possibly outside the groups expertise), but is this similar to what RAW files allow in photography?

 

[jerryfreak]

I don't think it works that way. It is said to use two ADCs. One for high and one for lower level signals. Then they are used to create a clean 32 bit result. The combining is the tricky part. And no I don't mean 32 bit or even the full 24 bit range.

As an alternative imagine this. I split my mic feed to two inputs. One I run at a normal gain, maybe 40 db thinking it probably won't clip. The other I run at 0 gain or maybe only 10 db gain. And occasionally that 40 db gain side does clip some. I can take the low gain version for just that segment that clipped, amp it up digitally to the proper level and replace the distorted clipped section. It would have a raised noise floor briefly, but would beat being clipped.

lots of recorders work this way with a 'safety track'

lets you ride the levels with less consequence, and as you described, manually reconstruct as needed in the event of bad clipping

 

[bennetng]

Is there a down side (isn't there always?) to 32 bit float and the and the ability to recover in post? Perhaps a stupid comparison (and possibly outside the groups expertise), but is this similar to what RAW files allow in photography?

If there is a downside I'd say safety and compatibility.
https://hydrogenaud.io/index.php?topic=114816.msg946516#msg946516
https://audiophilestyle.com/forums/...omparator-beta/?do=findComment&comment=940320

 

[Tks]

that mic will still need gain, but at the end of the day the >110dB dynamic range of modern ADCs still exceed the dynamic range of real-world recording situations, limited by self-noise of the mic, and background noise (as well as max SPL)

For sure agree.

Yah I was wondering what people do to remove noise. Is the solution simply using sensitive mics? Also I recall Okto Research talking about when they moved into their Stereo version of their DAC. They said they combined the 8 channels of their Pro version and that leads to better performance.

Are designs like that possible with ADCs or perhaps post processing of you natively havr mote channels to work with, but havent had a hardware design where this "combining of channels" done by the company?

Backgroud hiss combined with noise is what I feel id mostly be concerned with becoming better tackled from the ADC side. Do you have any tips on how that can be achieved barring expectations of "summation of channels" like Okto did? And of course simply buying more sensitive mics?

 

[AnalogSteph]

I am not sure you have fully understood what you said you are agreeing with.

Dynamic range for the best ADCs is about 120 dB as-is. Getting this much dynamic range from the microphone into the ADC input alone is a bit of a challenge already.

Low-noise large diaphragm condenser microphones are around the 5 dB SPL(A) range in terms of self-noise these days (even small-diaphargm ones have made it to around 15). You may find an ordinary room to be shockingly noisy compared to this. There is relatively little benefit to being able to hear your AC or outside street noise even better.

In dynamic microphones, mic self-noise is determined by voice coil resistance thermal noise and efficiency. You can change voice coil impedance while leaving everything else the same, and sensitivity will change accordingly (+3 dB per doubling), but efficiency pretty much will not - maximum SNR with an ideal amplifier will always be the same. YMMV with real amplifiers, of course.
Improving efficiency is more easily said than done and analogous to speaker drivers. You can't just make the element bigger because high-frequency directivity changes with aperture size, and ideally you'd want about the smallest diameter you can get away with. What you can do is using stronger magnets (Nd), or making the air gap narrower with tighter production tolerances to go along with it. You could also use the old fs vs. efficiency tradeoff, i.e. trade bass response for efficiency, like in a PA woofer. Might be interesting for a mic with active EQ.

I suppose you could try making multi-way microphones much like there are speakers, but my hunch is that not making a mess of polar response with such a contraption would be a challenge.

 

[Francis Vaughan]

For sure agree.

Yah I was wondering what people do to remove noise. Is the solution simply using sensitive mics? Also I recall Okto Research talking about when they moved into their Stereo version of their DAC. They said they combined the 8 channels of their Pro version and that leads to better performance.

Are designs like that possible with ADCs or perhaps post processing of you natively havr mote channels to work with, but havent had a hardware design where this "combining of channels" done by the company?

Backgroud hiss combined with noise is what I feel id mostly be concerned with becoming better tackled from the ADC side. Do you have any tips on how that can be achieved barring expectations of "summation of channels" like Okto did? And of course simply buying more sensitive mics?

Sensitivity in the mic won't help. Eventually all mics have a self noise defined by the individual air molecules hitting the diaphragm. Larger diaphragms have a lower self noise floor as they average out the impacts more. Similarly the analog stages are limited by the various thermal noise sources they contain. Lots of care can be taken, but there are fundamental limits that can't be got past no matter what.
If you want to combine tracks to reduce ADC errors, just add the two tracks in the DAW. It really is as simple as that. So long as both were recorded with the same source, you will get an extra bit of resolution for every doubling of tracks (in an ideal world, and subject to a few contraints on ADC implementation). But you will simply be able to resolve the thermal noise in the system with perfect precision. There is signal, and useful signal at that, below the noise, but you can't get rid of noise that is welded in to the signal before it gets to the ADC electronics.