Zoom F6 Portable Field Recorder Review

[PeteL]

What are some real world applications for this? Recording music while camping?

The brodcasting industry, tv mainly or low budget movies on location recording. They rarely have 6 inputs tough but I can see how this could be usefull to have more than 4. Tipically, outside captations, the same soundguy would handle the boom, woud have to walk around to follow the cast, maybe a couple hidden lavaliers and some ambience mics. So it needs to be battery operated and computer less, having this in a shoulder bag.

 

[amirm]

Are you sure you had it in 24 bit or 32 bit mode? It does have a 16 bit mode which would explain the seeming 16 bit performance.

I think those modes are only for recording to SD card, not as real-time streaming. I have set it to 24 bit mode.

 

[amirm]

Hi @amirm, I have a MixPre 3 II and I live near your general area. If you'd like to review this one, let me know!

Hi there. Yes, let's do it. Please start a conversation with me and we take it from there.....

 

[AnalogSteph]

Are you sure you had it in 24 bit or 32 bit mode? It does have a 16 bit mode which would explain the seeming 16 bit performance.

Yeah, something seems very off here. I mean, I wouldn't necessarily expect a Zoom recorder to be pure canned perfection but these results are just a mess. My hunch would be some sort of weird bug(s) or setup issue.

 

[PeteL]

I think those modes are only for recording to SD card, not as real-time streaming. I have set it to 24 bit mode.

I know that there was a few weeks back a thread about quick reviews, my comment probably should be there instead but I don't find it. Is it better more review with quick measurments or not. I'll allow myself to be a bit critical on this this here, and I don't want in any way reduce all the great work you do. In this case here, a headless panther, unrecommended product. I agree that there really IS NO excuse on poor performance on any of the features, but for something out of the left fiels like that, I'm mot sure these reviews are really relevant, unless asked for by the manufacturer, The 2 interface tested here USB out and Line In, are bonus features, edge cases. I get why you wouldn't have time to test the rest, but 99% of the people on the market for this would use only mic in, and smart card recording. These are low volume products that really are of no use and little interest for the community here. On the other end, someone on the market for this will search for reviews. Probably the main features are not much better, but they haven't been tested. There won't be many reviews for someone to form it's opinion due to the specialized nature of this. My point is, it does hurt the manufacturer. And at the same time, it's not wrong, what you do is unbiased from the manufacturer and it's important to keep this, there aren't enough. This product doesn't perform well, it's ok to say it, but we don't know how it performs when used the way people will use it, that's my point. Yes you do clearly state what you are testing but humans sometime don't discriminate that well. They go to the conclusion:unrecommended product. Basically, your voice matter, and of course you don't owe anything to manufacturers, but I'm not sure about how useful this is in this particular case.

 

[amirm]

I get why you wouldn't have time to test the rest, but 99% of the people on the market for this would use only mic in, and smart card recording.

I tested Mic in and it had the same performance as line in which was surprising to me. As to recording to SD card, that better be transparent and same as streaming to a PC. I don't see how transferring bits to a PC raises low frequency noise and distortion.

 

[amirm]

My point is, it does hurt the manufacturer.

Manufacturer could show us their measurements and we can discuss. To the extent they don't show a single distortion number much less the types of graph I am showing, for an expensive professional product no less, then I am not going to feel bad for them.

Why don't you all drop them a note and ask them for feedback?

 

[PeteL]

Manufacturer could show us their measurements and we can discuss. To the extent they don't show a single distortion number much less the types of graph I am showing, for an expensive professional product no less, then I am not going to feel bad for them.

Why don't you all drop them a note and ask them for feedback?

Fair enough, hope my comment didn't come as too harsh, again you are doing important work. I missed the part about testing mic, and I've seen some pretty bad usb design. Don't take this personally, this was my feeling but other may disagree. It's ok

 

[jerryfreak]

thanks @amirm for the tests

both the zoom and the tascam tested recently are my units, FWIW

I've never used the zoom as a USB interface, only as a field recorder in 24/32 bit mode to 48K/96K files. the menu is rather complex so its possible something is goofed. I dont have the unit in front of me obviously, and with the pandemic i havent used it since january so ill try to root out anything that might be an issue.

from the manual, we see that certain USB modes are limited to 16/48.

Based on this comment

Yes, something is wrong there as this device won't even support sampling rates above 48 kHz. I am testing something else but will go back to this a bit later. Thanks for catching it.

it appears we are operating in that mode here. You can confirm this by looking at the USB mode selected in the menu, on pg 141 of the manual

first of all- what we know about the input topology:

-Amir mentioned this is a 32-bit unit masquerading as a 16-bit, but in reality, it is 32-bit float so the best we could hope for is really 23 bits of resolution in a given gain range. just to distinguish from a 32-bit fixed signal with 192dB of significant bits

-the unit uses dual ADCs tailored for different gain ranges in an effort to increase dynamic range. Perhaps the distortion that Amir is seeing is crossover distortion between the two ADCs, or some dither/LSB noise from the 16-bit operation

-the least-sensitive ADC (or the input stage directly in front of it) is fixed to a max input level of +4dBU. As Amir has noted, in my tests clipping was severe above this level. When 'line in' is selected a 20 dB pad is applied to the front end that raised the noisefloor by about the same 20dB in my tests. switching the line-in pad also raises the input impedance slightly from 3K to 5K

-with its relatively low +4dBU max input, dynamic range is constrained between that and the unit's EIN, which is stated as -127dBU 'A-weighted', so likely a few dB higher unweighted. So no matter which mode you use you are going to be limited to a practical 20 bits at best

this was the summary of a series of discussions i had with zoom support

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. Since linear formats cannot accommodate the wide dynamic range of the dual ADC on the F6, a software Trim is necessary to set the level of the recorded file.

in terms of what you wrote i'm not sure i understand the difference between clipping because of integer file format vs clipping the converter. wouldn't both fundamentally describe going over FSD (unless you're talking about overloading the analog front end of the converter)

Yes, we are referring to overloading the analog front end.

i've done some tests recording relatively quiet source material , both line in, and mic in, and i really can't hear a difference in the integer vs float files when normalized. it seems like the EIN is the limiting factor.

As you point out here, the theoretical implications of 32-bit float are limited by hardware. With a quiet signal, I wouldn't expect you to hear a difference in the integer vs float files.

considering that in either integer or float the max input level is +4dbu, and the EIN is fixed, i don't quite understand how the float files increase dynamic range between these two endpoints. i understand how the two ADCs can have a wider range of 200 db or more, but i still can't wrap my head around how that increases the 120ish db dynamic range of the input

The preamp and ADC are going to act the same whether in float or integer. Floating point allows the final recorded file to have an extended dynamic range. So if recording in integer, the bit depth itself will limit the dynamic range meaning that somewhere in the full dynamic range of the preamps and ADC's it will have to cut it, meaning there is a ceiling and a floor which will either clip or not record. When recording float files, you can capture the entire dynamic range possible.

one more question: since the trim is acting as an attenuator, when recording in 24-bit mode would the best performance be achieved with trim set to 0dB?

Not necessarily. Because both AD converters are working all the time, where the trim is set doesn't have as much effect on the performance as what the AD converters decide to do

...or set to its maximum? would the latter be adding additional digital gain?. What trim setting would be considered 'unity gain' in 24 bit? 0dB isnt available as a setting in mic-in, of course

The trim adjustment is really just setting where within the available dynamic range you want to capture. The usual rules of setting gain in 24-bit should be followed because you still want to capture loud enough signal that it is high quality.

im assuming in 32 bit float, (where the trim is disabled), that the input range is already optimally matched to the input of the ADC

Correct

@amirm, id recommend feeding it a +4dBU sine wave in standalone mode at 32 bit float, 96K. none of the gain settings really do anything to the sound other than peg the level within the 768-dB range of the 32-bit container it is written in.

or perhaps you can do a multitone and analyze that?

ive been using Rightmark audio analyzer for my basic tests, recording 24/96 test .wavs on handhelds (rightmark>USB>mytek brooklyn dac+>unbalanced>recorder), with the DAC set to output levels matching the input of each recorder, as long as the wav is recorded at 24/96 with levels close to -1dBFS, Rightmark can parse it and run measurements. Perhaps your AP software can do more advanced tests on imported .wavs of known source.

I've done quite a few measurements, and i find it difficult to swallow that all of these handhelds which predate the zoom by several years are consistently out performing it. I would expect the zoom to perform closer to the Tascam DR100 you reviewed, which in my casual tests is a class above all the handhelds. Yes Zoom has a reputation for its consumer-bent, but the F-series should have slightly better specs than their H series products, in theory.

Anyway heres what Rightmark can do with its 'generate wave'/'analyze wave' feature at the bottom of its main window. Note that if you record a test .wav in 32bit float you would probably need to adjust peak level to about -1dBFS and render it to a 24-bit wav for Rightmark to analyze it

 

[617]

I used to have a lower end zoom (with XLR and built in mics) and traded it in for a Tascam DR70d pictured above. I think I was looking at a Sony unit as well, but Sound Design is the main player for field recording.

The Zoom unit was one of the very few audio devices I got rid of due to bad performance; the noise levels were just too high for quality recordings of relatively quiet things (speech). A lot of this is mic related, and I'm far from an expert, but the Tascam will make very clean recordings with small condenser microphones (https://naiant.com/ for interesting small microphones). It even has enough gain for dynamic microphones, with a SM58 I can get acceptable quality.

With field recording you are typically recording in bad acoustic environments - windy outdoors, reverberant rooms, table surface reflections, etc. As a result, your ability to choose a microphone with the right pickup pattern is essential. Oftentimes the best solution is a dynamic omni microphone, which requires a lot of gain to work well, and noise becomes an issue.

Anyway, when I was trying to do all this (I had a bunch of interviews to conduct and wanted to get the best quality) the resources online were quite dire. I couldn't afford a Soundesign, which I assumed was the best, so I got the Zoom, which didn't cut it (I think it would be fine for recording a band) and so I got the Tascam. Data would have been preferable.

Look forward to reviews of the DR100 and possible a MixPre.

I will say that the UI, design and UI of the Zoom units is very good. The Tascam doesn't feel as nice in the hand, and is harder to operate.

 

[jerryfreak]

here is the review of the DR100mkIII

 

[Rja4000]

I tested Mic in and it had the same performance as line in which was surprising to me. As to recording to SD card, that better be transparent and same as streaming to a PC. I don't see how transferring bits to a PC raises low frequency noise and distortion.

He's right that main use is Mic input, IMO.
So if you test those, that's probably the measurement to show.
But then you have another variable, which is the mic preamp gain. Which you may have no control on.

By the way, there is probably still some limiter or leveler active, which may expmain the IMD saturation.

 

[jerryfreak]

He's right that main use is Mic input, IMO.
So if you test those, that's probably the measurement to show.
But then you have another variable, which is the mic preamp gain. Which you may have no control on.

By the way, there is probably still some limiter or leveler active, which may expmain the IMD saturation.

as far as we can gather, there is no real preamp gain in this unit, its all digital
and yes, its primary use (for me) is for mic in, but it also sometimes gets used with some line-in channels simultaneously.

there is both a high pass filter, and limiter which are able to be disabled in the menu, at least in standalone mode. i cant speak to menu operation in usb interface mode. In any case their operation is described on pg 85-88 in the manual

 

[Rja4000]

it is 32-bit float so the best we could hope for is really 23 bits of resolution

That confuses me. What does that mean in practice ?
For low level signal, 32 bits float is way more resolving than 24 bits integrr.

 

[jerryfreak]

That confuses me. What does that mean in practice ?
For low level signal, 32 bits float is way more resolving than 24 bits integrr.

a 32-bit float can resolve a maximum of 23 bits of data resolution, the rest of the bits are the sign and mantissa. read this thread starting at this post:

https://www.audiosciencereview.com/.../new-member-field-recordist.10858/post-303744

in any case, for the practical limits of audio interfaces and hearing, at any recording levels close to correct, a 24-bit fixed, 32-bit fixed, and 32-bit float all have ample dynamic range to capture signals. 32-bit is used on all workstations to prevent rounding errors, primarily

 

[Tks]

A combination ADC/DAC/Headphone amp is the realm of more "pro audio" devices, IME. And considering a Scarlett Solo is about $100 and (other than the headphone amp) performs pretty well, for consumers there's little point for anything better.

I really can't see how, especially with the advent of Youtube (as much as this sentence would have you assume Youtube is something new, it's been around for over a decade). Likewise streaming platforms like Twitch (livestreaming music creation and such). Having decently performing "prosumer" ADC's seems relevant today especially with so much video content being made out and about. Not sure why we can have so many DAC's that perform so well, but ADC's that are like the one tested at such a price. Online communications have also proliferated, and you can tell all the people using pretty nice mics over the internet.

Idk, I just find it a bit weird there are quite a few DACs that perform so well they're bottlenecking measurement devices. While decent ADC's are quite few and far between. Then again, what do I know - sure not privy to the market cap for people searching for ADC's of good quality.

 

[jerryfreak]

ADCS have been exceeding the resolution of the signals they are used to record, for decades now. they are out there.

 

[PeteL]

as far as we can gather, there is no real preamp gain in this unit, its all digital
and yes, its primary use (for me) is for mic in, but it also sometimes gets used with some line-in channels simultaneously.

there is both a high pass filter, and limiter which are able to be disabled in the menu, at least in standalone mode. i cant speak to menu operation in usb interface mode. In any case their operation is described on pg 85-88 in the manual

I'm confused by this, I can't think of an adc chip that would operate properly at mic level voltages, you have to gain it up somehow no?

 

[restorer-john]

I'm confused by this, I can't think of an adc chip that would operate properly at mic level voltages, you have to gain it up somehow no?

There will of course be a microphone preamplifier stage in the unit.

 

[jerryfreak]

Can you do some testing to see how it handles clipping or not. That is the big benefit to this device when recording wav to 32 bit float. You more or less cannot clip it.

it will clip like any other device if the input level is exceeded (+4 mic +24 line), but yes in 32bit float, level control really does nothing and 'Full Scale' is pretty arbitrary. you can record something consistently over 0dBFS and the data is all there if you reduce the volume in software

Hi @amirm, I have a MixPre 3 II and I live near your general area. If you'd like to review this one, let me know!

this would be ideal to see these tested as stand-alone recorders in the same manner, as they are really same market segment

particularly if they were tested with mic level signals in the -40 dBV range typical of condenser mics

the mixpre has discrete class A preamps and I would expect it to shine in this application.

Both the mixpre and the zoom f6 have similar EINs but the mixpre can take a much hotter signal (+10dB on mic and +16dB on line) so perhaps it would make better utilization of the 32-bit float.

Sound devices also uses a completely different architecture for its multi-ADC implementation, utilizing 3 converters:

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