Thank you for this video. Just three important clarifications in order any confision to be avoided. (1) These 6 knobs are not signal gain controls on either 16-bit, 24-bit or 32-bit recordings - by design these are ALWAYS audio mix faders and have nothing to do with the input signals! F6 is a professional-grade device and there are more similarities between F6 and L-12 as a digital mixer, for example, rather than F6 and H5 or H6 as a handy recorder. So on 16-bit and 24-bit the true "gain" is called "trim" (mentioned in your video), and on 32-bit this "trim" is disabled (no gain control), but still the audio mix faders are fully opeational, as expected. (2) I will highly recommend to control this device remotly via bluetooth. You will find out that you can control the gains and the audio faders with no need to touch the knobs, exactly as a digital mixer where F6's LR track works as a Master-bus. (3) And last but not least, do what you want but keep in mind that Line-out doesn't support ultra-high audio levels produced by the 32-bit recording! Fortunatly, it has a dedicated audio limiter (compressor) with soft and hard "knee" modes that can be used to avoid the audio clippings. Hope this helps...
Honestly, this is such a good tutorial that Zoom should feature it on their own channel with a proper reference to your channel. Wonderfully done. Thank you.
Thank you for explaining the track knobs in 32bit. I just picked up the F6 after using the F3 for the past year or so and was a little confused on what was going on.
Thank you for your feedback. If you'd like to chat more about this kind of stuff, feel free to join my Community where we discuss quality content creation with great gear: QualityCreators.Club
If ZOOM does not listen to you or Rebel Tech's suggestions for future firmware upgrades then I've lost all hope in ZOOM. I have no idea how you translate the Zoom menu into such a crystal clear video tutorial. Great job and THANKS.
I agree. Perhaps even if the knobs change to a fader knob when you are within the PFL settings or something. I'm not too keen on setting the trim with the buttons.
Great video bud! I love the Zoom F6, it's one my favorite pieces of gear. I was thinking about the next potential update, being 1.7, and there are some things that they should address. 1) Sending the audio signal to the app via BT (If that is even possible) 2) As you said, having the option to change the fader knob to a gain knob in all modes 3) Personally a 6 second pre roll in 96Khz It's not much, and this device it pretty much perfect, but I feel it needs some tweaking. I'll be putting some new Zoom F6 video up soon as well (I know shocker lol). I think this topic of a 1.7 update could be a good video. I also want to go into my settings I had during the short film I helped make a couple of weeks ago. Keep up the great work!
Thanks for your comment, and I agree: a new update would really be something good and there are a handful of improvements that I think should be included in that. I am not holding my breath though 🙈. For me the Bluetooth thing is not really an issue, the 6 sec in 96khz… pretty sure that's not coming as it seems to be a limit of the RAM or something, and the gain knob… jea… that'd be an obvious one. Additionally the gain of the Line Out channel (so that that can actually be sent to a camera for scratch audio even if the camera only supports mic level input). Looking forward to seeing your next video on the topic as well 👍.
Thank you so much for this explanation! I was going crazy at a shoot yesterday trying to figure out why the knob wouldn’t adjust my gain in 24 bit mode!
Having to use the trim function in a submenu with 24 bit recordings to set gain levels was a big disappointment when I purchased my F6. Highly impractical.
Thank you for this video. What about have a quick access to switching between trim-role and mix-role for these buttons ? Like : "hold the PFL button 1second to toggle between thesse modes" ? That would be a game changer, compared to the actual menu browsing !
Thanks for the great explanation! I think one of the things I am struggling with is understanding how, in 32 bit mode, there is no trim but the fader adjust the gain. If we don’t need trim because we have all of this dynamic range, why do we need a gain function which affects the recorded values?
Hi Eric, thanks for your comment. The reason you still have a way to adjusting these values is that you may not want to do this in post processing or you may want to output the signal to a line out with a certain volume. But as you may be able to guess. I am using mine in Ref Mode - aka it only changes the gain on the channel before hitting record. Because all volume control (for me) is done in post processing through loudness normalization with RX 8 by iZotopes.
@@ChrisSpiegl I tried the Izotope loudness normalization with a 32 bit recording yesterday. It was great for spoken words, but. I’m guessing it would mess with the dynamics of music and simply adjusting gain might be a better option in that scenario
@@Minus20Design that’s right, for music I am not sure how useful the normalization would be - I usually only do it for spoken things. The gain is probably your best bet (and in certain areas also doing the gain adjustments for the main part - cause a sync clap for example can be a much higher peak than the actual recording but usually you also don’t care about the clap peaking or not ☺️).
Thanks I learned a lot! What I have learned using the F6 to record a line level from a DJ speaker is that it is best to use an attenuator because there is no way to set the levels in 32 bit float. So in my old recorders I could turn down the incoming signal, with 32 bit float you really need to attenuate the signal before it even comes into the recorder, otherwise, you get clipping. So for now I will have an XLR form the DJ and have a splitter XLR cable. One channel will be normal without an attenuator, the other channel I will have the attenuator so when loud music is played I don't get a blown out recording. I don't want both channels to be attenuated because when people do normal talking it would record too low. This way I also don't need to monitor it. No matter what happens I will have the best recording.
This is not how 32bit float works and you should not need the attenuator at all. When recording the 32bit float signal you will not have clipping (at least not in the file). When you play it back it may seem like it is clipping but you can easily adjust for that when using Adobe Audition or any other audio editing program that supports 32bit float. The whole point of 32bit float is that you can recover everything that was too loud and also everything that was too silent so you can make adjustments in post processing (for example loudness normalize). The reason why it may seem that it is clipping is that when you play back the file on your computer, the output will be 24bit… but the information is still there and you can recover it with the mentioned programs.
@@ChrisSpiegl I thought that too but I have audio recordings from DJ’s that are recorded way too loud and clipping and when I just tried to bring it into audacity to lower the gain it just lowered the overall sound but the sound is still clipped. 32bit I think is great but it can’t perform miracles if the sound is recorded bad to begin with. I would love to send you the file though if you think there is something else that I am missing when it comes to recovering clipped and distorted audio.
@@WeddingDocumentary That sounds really extreme. I mean the Zoom F6 has two analog to digital converters which give you a pretty crazy dynamic range but maybe the signal is over those so that these are clipping. The 32 bit float file is unlikely to be the issue. Now, if it's the dynamic range of the AD that is being clipped, then I would look into the setup of the ports. Are they setup as "Mic" or "Line" signal inputs? That could be the issue there because it sounds like you are receiving a line level signal which is generally louder than a mic level.
Great video, just what I needed! I've only had the F6 for a few days now and this video really helped me to make sense of this somewhat confusing part. Coming from an H4n Pro, the F6 is a major step up and there's alot to get my head around.
The Zoom F6 claims to support 32-bit Float format by using two analog-to-Digital convertors together. It DOES NOT claim to support actual 32-bit audio, which would consist of a sign bit followed by 31 bits of actual resolution. The distinction is important. The Zoom F6 MIGHT be based on the Asahi Kasei AK5736 chip, a 6-channel, 24-bit ADC designed specifically for audio applications in the Mic to Line input voltage range at sample rates up to 192kHz. Supporting evidence for this is an announcement by Zoom of a fire at a supplier’s factory (Asahi Kasei Microdevices) in 2021 that caused some replacement components to be used in production. Asahi Kasei Microdevices happens to be the maker of the AK5736 chip, as well as some others in the design. While 24-bit audio file format actually uses a sign bit along with data bits, the 32-bit Float audio format does not mean a sign bit along with 31 bits of integer data. In fact, packed within the 32 bits is a sign bit (+/-), an 8-bit exponent, and only 23 bits for the number’s mantissa. The implication here is that the extended dynamic range of the 32-bit Float format and therefore the F6 is not the result of more fine resolution than any common 24-bit convertor has, but the ability to start to populate the 8-bit exponent field in the data file. The exponent field in 32-bit Float actually provides a theoretical potential of more than 1582dB in dynamic range, as compared to just 144dB for conventional 24-bit. (Never mind that the surrounding hardware in the F6 limits the actual dynamic range to more like 124dB at best.) From Wikipedia: “The trade-off between floating point and integers is that the space between large floating-point values is greater than the space between large integer values of the same bit depth. Rounding a large floating-point number results in a greater error than rounding a small floating-point number whereas rounding an integer number will always result in the same level of error. In other words, integers have round-off that is uniform, always rounding the LSB to 0 or 1, and floating point has SNR that is uniform, the quantization noise level is always of a certain proportion to the signal level. A floating-point noise floor will rise as the signal rises and fall as the signal falls, resulting in audible variance if the bit depth is low enough.” Zoom has said publicly that the F6’s inputs are connected directly to the A-to-D convertor(s). The AK5736 chip supports this as a default. Each of the 6 channels has a differential input (i.e., pins 2 and 3 from the XLR connector) through the chip’s onboard differential PGA (programmable gain amplifier), one per channel. Each PGA is programable in the range of 0 to 20dB gain. There are no configurable “attenuation pads” in the design as have been speculated by others. It would be logical to assume at first that what the Zoom F6 would call a “Mic” input would be programmed for 20dB of gain, while a “Line” input would be programmed for 0dB of gain. But this may not actually be the case, because Zoom would have then needed only a single 24-ADC to populate the 32-bit Float file format. They say they actually used two chips, in search of greater dynamic range than is available with just 24 bits. So how did they do that? The AK5736 chip explicitly supports connecting multiples of itself to expand the total channel count (6, 12, 18…), with all conversions happening in different chips under the supervision of a common timing scheme. One logical way to run a two-ADC design that would provide 20dB more dynamic range would therefore be to (1) connect all the inputs in parallel to both ADCs, and (2) always program one ADC’s 6 channels at 0db gain while programming the other’s 6 channels at 20dB gain. This would cause both ADCs to provide results to the CPU (albeit amplified to different levels), along with any corresponding “conversion overflow” warning flags. The CPU would obviously have to ignore data coming an overflowing convertor and use only data from the other. In the case where both convertors produced a valid result, 20dB could be added to one, with that result then being averaged with the other in order to allow calculation of an averaged result supported by both convertors. In this way, up to 5 exponent bits could be populated over the entire dynamic range to support the additional 20dB of dynamic range in the final data file. This is a full strategic use of the 32-bit Float WAV audio format. In this scenario, however, it would not have been necessary at all for the user to specify “Mic” or “Line” input at all for each input, so this may not be exact scheme that Zoom used. (Another scheme would involve setting each channel independently to 0dB or +20dB in the first ADC as specified by the user, but then to reverse each of those settings in the other ADC. The final results would be the same as above, but the user might feel better that he/she had some input to the process!) The advantages of using 32-bit Float format in field or live music recording is obvious: far less attention can be paid to recording level changes up front; it is possible to “set it and forget it” in many circumstances. That said, Zoom did not do itself any favors with how it programmed level settings in 32-bit Float format. In this mode, the F6 does not allow the menu-based adjustment of input “Trim” like it does in other formats, which seems to discourage the idea of optimum level settings during recording of 32-bit Float audio. But it DOES actually allow recording level changes by another method, the front-panel dials. Confusingly, the file recording level can in fact be set in 32-bit mode for armed channels as long as recording has been activated but not started. Once recording, however, adjustment of the same dials does not change the recording level, it just changes the mix of that channel’s level to the LR mix balance. The channel meters change as expected in EITHER mode and it’s quite easy to forget that you are not actually making recording level changes when you think you are. Such are the pains of a tiny user interface like the one on the F6. A final note on the Asahi Kasei AK5736 chip: The device includes programmable “bias” (aka phantom power) voltage per channel for connected microphones, which sounds perfect for the F6. However, the programmable voltage range is limited to 9VDC. So it seems the phantom power in the F6 does not originate from this chip. In fact, the F6’s 24 or 48VDC phantom power has to be blocked from the chip’s inputs by a capacitor.
Ich bin mir nicht sicher ob im float-Modus wirklich die Aussteuerung beeinflusst wird, oder nur die Zuordnung wie hoch der Pegel interpretiert wird. Dazu müsste man die genaue Schaltungstechnik und die Umsetzung kennen. Das bei 24bit die Aussteuerung nur so umständlich eingestellt werden kann war mir nicht bewusst, weil ich nur 32bit aufnehme. Werde wohl demnächst mal etwas experimentieren. Gruß Klaus
Wie Alex bereits geschrieben hat, es wird der Aufnahme Pegel verändert und dies wird dann entsprechend in die 32bit Datei geschrieben. Nun kann man diese natürlich auch wieder hoch ziehen wenn man in Audition oder ähnlichem ist. @Alexan Der: Jap, ich habe mir das so genau angesehen da ich total verwirrt war dass die Regeler in 24 eben nicht wie erwartet den Gain eingestellt haben sondern nur das Fading in den L/R Kanal. War ganz schön nervig am Anfang 🙈.
Thank you for this excellent tutorial on how to manage gain when recording with a Zoom F6. This was very helpful. I’m not familiar with the term PFL, have you made any tutorials on this topic. It is not explained in the manual?
You're very welcome! I have not specifically talked about the PFL settings but Justin from @Rebel Tech has a video about his production settings and talks about the PFL specifically: czcams.com/video/fDVVPBtaRWA/video.html
@@ChrisSpiegl Thanks so much for the link. Sorry to be a pest, but what DAW do you recommend for processing the gain of F6 Floating 32 bit reocrding in post? I'm using Logic Pro which unfortunately converts 32 bit files to 24 bit and would cause a great deal of clipping.
@@Minus20Design well - in some way all DAWs are going to display and playback the 32bit float files as 24bit. That’s quite normal. However, when you use a normalization, gain change, or loudness normalization effect on your audio it should bring it back in full quality. At least that works in iZotope RX8 as well as Adobe Audition which I am using. Even Premiere Pro does it the same. So my guess is: logic does support the 32bit float files, you just have to adjust the gain / normalize your files.
I thought about that… decided against it last minute. Maybe next time I am talking about this. Generally though: if you are specifically asking about the 32bit file. In Reference Level the mic level will be one level throughout and with that the wave form would be equal in hight as long as the sound source stays at the same volume. In Rec Level: the wave form would be smaller or bigger depending on how the fader knob is set. You can then of course still recover it all - it's 32bit float after all.
great video! thanks a lot! I have a general question: I work a lot on "run and gun" kind of video-projects, and I can hardly think of a scenario, where I (or anyone) would prefer to have control over the channel's faders rather than the channel's gain trim. So why are the knobs faders and not gain-trim anyways? To me that seems so strange... By the way, same thing happens with the Sound Devices Mixpre 6-ii, where the knobs are set to fade by default, rather than gain trim (you can change that with the mixpre 6-ii, but, then you loose the possibility to use limiters...)
I think the reason why they chose to make the knobs faders rather than gain is that with 32bit float you do not "need" trim/gain. Depending on how you set up the faders, they actually don't affect the individual 32bit float tracks at all and only the mixdown track. I really like that because it means that I can change how loud a mic goes into the mix (for live-streaming / live mixing) but at the same time have one constant 32bit float per mic channel which I then can auto-mix in post-processing manually.
I would like to ask as about what we can see on the display: in your example, you have one mic, so we can see it on channel 4. Now we can see also four more levels, LR and LINE. Would you mind explaining why is that so, there is one mic (mono) but we see LR and yet LINE? Why is that so, and what those levels are? That's a beginner question, so I hope my question is clear enough. Thanks
Hay friend. What exactly is happening when 1 presses in the volume nob? I can haar a hicup which suggests I guess the audio changes and I do kind of hear a difference. But as I'm blind when doing this what's showing on screen? Also which of these profiles or waht have you is the out of box 1?
Hi Chris, thanks again for doing these videos! What is the purpose of the Line out/Line level when you only needs a stereo signal into the SD-card. I just find it confusing/annoying to have these 2 bars clipping for no reason? Or am I totally wrong? Stein
Hi Stein, maybe it would be interesting to be able to turn the line out off but I think it's good to have it available. There are multiple good uses for this: 1) the line out can be used to send the signal to a PA system. 2) the line out can be sent to a camera for additional recording a backup track there 3) the line out can be sent to a live streaming setup (ATEM Mini for example - this is actually what I use almost daily). So it's not something you will need in your use case, but for others it is very helpful.
@@ChrisSpiegl Hi again Yes, found it in the (not so intuitive) routing that I actually could turn it off, learned a lot form you and the guys at F6 Facebook-group so now I turn on my machine, goes out and have a lot of fun with my F6 :) I can see that many will have great use of the Line out, so good it's there! I've used F4 for several years and Mixpre3 II for a year, but this F6 was a real nut to crack to get going.. Again thanks a lot!
As Espen already pointed out, you have to go to the PFL settings and there go to Input and set it to "Mic" and not "Mic(PH)". And in terms of the potentiometer: that's the topic I covered in this video? Is it not?
So if I have the gain on the channel fader where I'm getting basically no signal in the L/R mix, but I have my trim set so the channel is registering proper peaks around -12, if I record, I will have sound in a track just not in the mix file? Also what do you normally run your trim at? It's dependent on the sensitivity of the mic I assume?
Hi there! thanks for your great videos! Quick question, I need to use an f6 as a audio interface to record a live performance. Do you know if its possible to use headphone output to speakers and line out as a 3-4 output to send click to the musicians? Thank you!
Quick answer - F6 operates as USB interface in two modes (stereo and multitrack). You can select up to 4 USB input channels from Mac/PC and route these to 1-6 digital mix channels back to your Mac/PC for recording/monitoring or to the stereo channels (line-out and headphones) of F6. You can deside how to commence this routing using the prefader or postfader of F6. The second part of your question is not related to the USB interface at all. The F6's stereo line-out output impedance is up to 100 ohms (max load of 10 kiloohms), and the headphone speakers output inpedance is up to 15 ohms with max level of 100mW per channel (max load of 32 ohms). Obviously, your approach is not appropriate, and dangerous, btw. You should do the opposite - you should connect the line-out jack to your aux-in/line-in of the active speaker system or to line-in of an amplifier with passive speakers, and the headphones out jack should be connected to an external pre-amp active splitter with two or more headphones for monitoring. Btw, some external headphone amplifiers, like Samson QH4, support aux-in and aux-out, so you can use only F6's line-out for both monitoring and playback to active speakers. Be careful and good luck.
You need an audio editing program that can handle 32bit float audio. But I think even Audacity (open source audio editor - crsp.li/audacity ) can do that. Another alternative would be Reaper: crsp.li/reaper
@@ChrisSpiegl Audacity can do it, but it's difficult to pull parts of the recording down to 0 as you can with Audition (works fantastic!) AND it seems there is only a certain amount you can reduce in Audacity, so I do Audition to normalize>save and then into Audacity in 32-bit mode for High-pass-filter and 4-5db noise if necessary. Then save a clean file as wav or mp3.
Hi @@snipolar , great input but I am wondering why you'd go to Audacity for the High-Pass? there are effects like that in Adobe Audacity as well? Why switch between programs?
@@ChrisSpiegl Hi again, Audition creates a too sharp border to my ears when I reduce the lowest part, actually (the free) Audacity creates a very smooth transition, I often reduce the lowest 500Hz for most birdsounds. And the noise-filter in Audition creates box-like appearance in nature recordings, sounds very unnatural in my ears. Probably much more to learn as it is a great software.
@@snipolar I see, probably there would be some options you can tweak in both to get the desired result. Especially when using a EQ to cut the lower frequencies. Happy learning 👍.
Excellent tutorial. ZOOM chosen to reinvent the wheel with those knobs. It's not intuitive at all. Yet, if I got all correctly, you can't change level on the fly when recording in 24 bits. Zoom forcing us t 32 bits :) well this is probably what 99% of people are attracted anyway in this recorder. ZOOM could leave the decision to the users instead of an arbitrary choice. Nonetheless, ZOOM F6 is a fantastic piece of equipment.
Hi Chris. Great info here! After transferring the files over usb, the F6 seems to forget all the levels I’ve set every time, so I have to remember to reset before recording again. Am I missing something? Thanks.
I have not experienced this. Maybe try this: turn on, set the settings, turn off (to lock in the settings). Then try again with the file transfer. Additionally: you can save settings to the SD Card and load those again if all else fails.
@@ChrisSpiegl Yes, if you turn the F6 completely off and back on the settings are there as you say, however this issue occurs if you don’t power off after file transfer. I record a lot of long form content so every 30 mins or so I like to stop and transfer the file to free up space before getting straight back into it. This is when the issue occurs. Sure, I can power it on and off, but that’s an extra step from a workflow point of view. PS. We spoke a while ago about using a 3.5mm splitter for timecode to multiple cameras. I can confirm that this works with no issues that I’ve encountered which is great, especially because I’ve been waiting for a pair of tentacle sync E mkii to arrive for over 6 months now! Hope you’re well. Joel.
@@JoelPurnell Maybe let Zoom know and send an email to their support team. Personally, I mostly use the Zoom in some standard settings and do not transfer the files via USB-C Card Reader Mode but instead pop the SD Card and transfer that way. And right, the Timecode with a splitter cable. I since also used that feature occasionally. It's really helpful when all the cameras are close by and easy to reach.
As I mention in the video, I prefer Reference Level mode because that way the audio levels in my 32bit recording are never changed during the recording and I just send it through loudness normalization once I am back at my desk.
@@ChrisSpiegl Hi, I do recordings of (mostly) birds with stereo parabolic mic on my F6, what setting would you suggest for a stereo in / stereo out to SD-card with maximum gain. I do the normalization in Audition so no problem with overload. Hope you have some thoughts ab this! Stein
Hi @@snipolar/videos , I have little experience with recording birds. The one thing I would consider important is to link the channels together so that they are recorded as a Stereo Pair and not individually. I would then also consider recording simply in 32bit float and the highest sample rate desirable for your work. Cheers
@@ChrisSpiegl Thank again, the F6 facebookgroup helped me out, I've now linked 1 & 2 as Stereo AND with both set with Phantom Power: Fader 1 adjusts both channels and I get a nice signal to my SD-card. BUT there are lots of other stuff to be found within this machine..tnx
That bit about how convoluted it is to change recording levels on the individual channels... Having to open the menu to adjust the "trim" when the actual knobs are right there, ready to be used if they could... uhm... I am not sure I would buy the recorder anymore...
Not sure what you mean. When you record the audio in 32bit float and you get back to your editing station, you can lower the gain and it will be recovered without quality issues.
@@ChrisSpiegl Yes , I found it. The only thing always go throught the menu takes time, But it is just in the beguining of the main file to share with post audio and calibrate.
@@mariorivas7886/videos I absolutely agree and would also prefer if I would be able to change the behavior of the faders to actually work as Trim Knobs instead. But sadly the Zoom F6 has no setting for that.
Why? It must be programmable to set the trim via the knobs. And they must know that this is what is desired. And it can't be because they don't want to cannibalize their top level line because it doesn't get any better than this for them. Do they have a deal with other companies?
Thank you for this video. Just three important clarifications in order any confision to be avoided. (1) These 6 knobs are not signal gain controls on either 16-bit, 24-bit or 32-bit recordings - by design these are ALWAYS audio mix faders and have nothing to do with the input signals! F6 is a professional-grade device and there are more similarities between F6 and L-12 as a digital mixer, for example, rather than F6 and H5 or H6 as a handy recorder. So on 16-bit and 24-bit the true "gain" is called "trim" (mentioned in your video), and on 32-bit this "trim" is disabled (no gain control), but still the audio mix faders are fully opeational, as expected. (2) I will highly recommend to control this device remotly via bluetooth. You will find out that you can control the gains and the audio faders with no need to touch the knobs, exactly as a digital mixer where F6's LR track works as a Master-bus. (3) And last but not least, do what you want but keep in mind that Line-out doesn't support ultra-high audio levels produced by the 32-bit recording! Fortunatly, it has a dedicated audio limiter (compressor) with soft and hard "knee" modes that can be used to avoid the audio clippings. Hope this helps...
Thanks for this helpful information. I appreciate it.
Honestly, this is such a good tutorial that Zoom should feature it on their own channel with a proper reference to your channel. Wonderfully done. Thank you.
Thank you, I really appreciate your comment 🌸.
Thank you for explaining the track knobs in 32bit. I just picked up the F6 after using the F3 for the past year or so and was a little confused on what was going on.
Thank you for your feedback. If you'd like to chat more about this kind of stuff, feel free to join my Community where we discuss quality content creation with great gear: QualityCreators.Club
If ZOOM does not listen to you or Rebel Tech's suggestions for future firmware upgrades then I've lost all hope in ZOOM. I have no idea how you translate the Zoom menu into such a crystal clear video tutorial. Great job and THANKS.
Thank you, I appreciate that a lot and I am keeping my 🤞 for an update on this. Thank you for the support.
I agree. Perhaps even if the knobs change to a fader knob when you are within the PFL settings or something. I'm not too keen on setting the trim with the buttons.
This is exactly the tutorial I’ve been looking for. I had some confusion between 24 and 32, but you just solved my problems. Thank you sir!
Great to hear!
Great video bud! I love the Zoom F6, it's one my favorite pieces of gear. I was thinking about the next potential update, being 1.7, and there are some things that they should address.
1) Sending the audio signal to the app via BT (If that is even possible)
2) As you said, having the option to change the fader knob to a gain knob in all modes
3) Personally a 6 second pre roll in 96Khz
It's not much, and this device it pretty much perfect, but I feel it needs some tweaking. I'll be putting some new Zoom F6 video up soon as well (I know shocker lol). I think this topic of a 1.7 update could be a good video. I also want to go into my settings I had during the short film I helped make a couple of weeks ago.
Keep up the great work!
Thanks for your comment, and I agree: a new update would really be something good and there are a handful of improvements that I think should be included in that. I am not holding my breath though 🙈.
For me the Bluetooth thing is not really an issue, the 6 sec in 96khz… pretty sure that's not coming as it seems to be a limit of the RAM or something, and the gain knob… jea… that'd be an obvious one.
Additionally the gain of the Line Out channel (so that that can actually be sent to a camera for scratch audio even if the camera only supports mic level input).
Looking forward to seeing your next video on the topic as well 👍.
Well done, Chris! Thumbs up!
Thank you 🌸.
Thank you so much for this explanation! I was going crazy at a shoot yesterday trying to figure out why the knob wouldn’t adjust my gain in 24 bit mode!
Glad my video was helpful. 🌸
Having to use the trim function in a submenu with 24 bit recordings to set gain levels was a big disappointment when I purchased my F6. Highly impractical.
@@Minus20Design absolutely - if only they would let us choose to set the knobs to trim in 24bit mode 😭.
@@Minus20Design that’s why I’m just going to use 32 bit every time now lol
Thank you for this video. What about have a quick access to switching between trim-role and mix-role for these buttons ? Like : "hold the PFL button 1second to toggle between thesse modes" ?
That would be a game changer, compared to the actual menu browsing !
I agree. Sending that feedback directly to Zoom may help.
Thanks for the great explanation! I think one of the things I am struggling with is understanding how, in 32 bit mode, there is no trim but the fader adjust the gain. If we don’t need trim because we have all of this dynamic range, why do we need a gain function which affects the recorded values?
Hi Eric, thanks for your comment. The reason you still have a way to adjusting these values is that you may not want to do this in post processing or you may want to output the signal to a line out with a certain volume.
But as you may be able to guess. I am using mine in Ref Mode - aka it only changes the gain on the channel before hitting record. Because all volume control (for me) is done in post processing through loudness normalization with RX 8 by iZotopes.
@@ChrisSpiegl Thanks Chris for the reply - that helps clear things up!
@@ChrisSpiegl I tried the Izotope loudness normalization with a 32 bit recording yesterday. It was great for spoken words, but. I’m guessing it would mess with the dynamics of music and simply adjusting gain might be a better option in that scenario
@@Minus20Design that’s right, for music I am not sure how useful the normalization would be - I usually only do it for spoken things. The gain is probably your best bet (and in certain areas also doing the gain adjustments for the main part - cause a sync clap for example can be a much higher peak than the actual recording but usually you also don’t care about the clap peaking or not ☺️).
Thanks I learned a lot! What I have learned using the F6 to record a line level from a DJ speaker is that it is best to use an attenuator because there is no way to set the levels in 32 bit float. So in my old recorders I could turn down the incoming signal, with 32 bit float you really need to attenuate the signal before it even comes into the recorder, otherwise, you get clipping. So for now I will have an XLR form the DJ and have a splitter XLR cable. One channel will be normal without an attenuator, the other channel I will have the attenuator so when loud music is played I don't get a blown out recording. I don't want both channels to be attenuated because when people do normal talking it would record too low. This way I also don't need to monitor it. No matter what happens I will have the best recording.
This is not how 32bit float works and you should not need the attenuator at all.
When recording the 32bit float signal you will not have clipping (at least not in the file). When you play it back it may seem like it is clipping but you can easily adjust for that when using Adobe Audition or any other audio editing program that supports 32bit float.
The whole point of 32bit float is that you can recover everything that was too loud and also everything that was too silent so you can make adjustments in post processing (for example loudness normalize).
The reason why it may seem that it is clipping is that when you play back the file on your computer, the output will be 24bit… but the information is still there and you can recover it with the mentioned programs.
@@ChrisSpiegl I thought that too but I have audio recordings from DJ’s that are recorded way too loud and clipping and when I just tried to bring it into audacity to lower the gain it just lowered the overall sound but the sound is still clipped. 32bit I think is great but it can’t perform miracles if the sound is recorded bad to begin with. I would love to send you the file though if you think there is something else that I am missing when it comes to recovering clipped and distorted audio.
@@WeddingDocumentary That sounds really extreme. I mean the Zoom F6 has two analog to digital converters which give you a pretty crazy dynamic range but maybe the signal is over those so that these are clipping. The 32 bit float file is unlikely to be the issue.
Now, if it's the dynamic range of the AD that is being clipped, then I would look into the setup of the ports. Are they setup as "Mic" or "Line" signal inputs? That could be the issue there because it sounds like you are receiving a line level signal which is generally louder than a mic level.
Great video, just what I needed! I've only had the F6 for a few days now and this video really helped me to make sense of this somewhat confusing part. Coming from an H4n Pro, the F6 is a major step up and there's alot to get my head around.
Absolutely, I am glad you got to do this upgrade. 32float is incredible ❤.
This is exactly what I needed. Great video!
Thank you, I appreciate that 👍.
Big thanks - just get F6 and your video was big help:)
Thank you, I appreciate that 🌸.
Thanks so much - this has cleared up a lot of confusion
Glad it was helpful!
The Zoom F6 claims to support 32-bit Float format by using two analog-to-Digital convertors together. It DOES NOT claim to support actual 32-bit audio, which would consist of a sign bit followed by 31 bits of actual resolution. The distinction is important.
The Zoom F6 MIGHT be based on the Asahi Kasei AK5736 chip, a 6-channel, 24-bit ADC designed specifically for audio applications in the Mic to Line input voltage range at sample rates up to 192kHz. Supporting evidence for this is an announcement by Zoom of a fire at a supplier’s factory (Asahi Kasei Microdevices) in 2021 that caused some replacement components to be used in production. Asahi Kasei Microdevices happens to be the maker of the AK5736 chip, as well as some others in the design.
While 24-bit audio file format actually uses a sign bit along with data bits, the 32-bit Float audio format does not mean a sign bit along with 31 bits of integer data. In fact, packed within the 32 bits is a sign bit (+/-), an 8-bit exponent, and only 23 bits for the number’s mantissa. The implication here is that the extended dynamic range of the 32-bit Float format and therefore the F6 is not the result of more fine resolution than any common 24-bit convertor has, but the ability to start to populate the 8-bit exponent field in the data file. The exponent field in 32-bit Float actually provides a theoretical potential of more than 1582dB in dynamic range, as compared to just 144dB for conventional 24-bit. (Never mind that the surrounding hardware in the F6 limits the actual dynamic range to more like 124dB at best.)
From Wikipedia: “The trade-off between floating point and integers is that the space between large floating-point values is greater than the space between large integer values of the same bit depth. Rounding a large floating-point number results in a greater error than rounding a small floating-point number whereas rounding an integer number will always result in the same level of error. In other words, integers have round-off that is uniform, always rounding the LSB to 0 or 1, and floating point has SNR that is uniform, the quantization noise level is always of a certain proportion to the signal level. A floating-point noise floor will rise as the signal rises and fall as the signal falls, resulting in audible variance if the bit depth is low enough.”
Zoom has said publicly that the F6’s inputs are connected directly to the A-to-D convertor(s). The AK5736 chip supports this as a default. Each of the 6 channels has a differential input (i.e., pins 2 and 3 from the XLR connector) through the chip’s onboard differential PGA (programmable gain amplifier), one per channel. Each PGA is programable in the range of 0 to 20dB gain. There are no configurable “attenuation pads” in the design as have been speculated by others.
It would be logical to assume at first that what the Zoom F6 would call a “Mic” input would be programmed for 20dB of gain, while a “Line” input would be programmed for 0dB of gain. But this may not actually be the case, because Zoom would have then needed only a single 24-ADC to populate the 32-bit Float file format. They say they actually used two chips, in search of greater dynamic range than is available with just 24 bits. So how did they do that?
The AK5736 chip explicitly supports connecting multiples of itself to expand the total channel count (6, 12, 18…), with all conversions happening in different chips under the supervision of a common timing scheme.
One logical way to run a two-ADC design that would provide 20dB more dynamic range would therefore be to (1) connect all the inputs in parallel to both ADCs, and (2) always program one ADC’s 6 channels at 0db gain while programming the other’s 6 channels at 20dB gain. This would cause both ADCs to provide results to the CPU (albeit amplified to different levels), along with any corresponding “conversion overflow” warning flags. The CPU would obviously have to ignore data coming an overflowing convertor and use only data from the other. In the case where both convertors produced a valid result, 20dB could be added to one, with that result then being averaged with the other in order to allow calculation of an averaged result supported by both convertors. In this way, up to 5 exponent bits could be populated over the entire dynamic range to support the additional 20dB of dynamic range in the final data file. This is a full strategic use of the 32-bit Float WAV audio format. In this scenario, however, it would not have been necessary at all for the user to specify “Mic” or “Line” input at all for each input, so this may not be exact scheme that Zoom used. (Another scheme would involve setting each channel independently to 0dB or +20dB in the first ADC as specified by the user, but then to reverse each of those settings in the other ADC. The final results would be the same as above, but the user might feel better that he/she had some input to the process!)
The advantages of using 32-bit Float format in field or live music recording is obvious: far less attention can be paid to recording level changes up front; it is possible to “set it and forget it” in many circumstances.
That said, Zoom did not do itself any favors with how it programmed level settings in 32-bit Float format. In this mode, the F6 does not allow the menu-based adjustment of input “Trim” like it does in other formats, which seems to discourage the idea of optimum level settings during recording of 32-bit Float audio. But it DOES actually allow recording level changes by another method, the front-panel dials. Confusingly, the file recording level can in fact be set in 32-bit mode for armed channels as long as recording has been activated but not started. Once recording, however, adjustment of the same dials does not change the recording level, it just changes the mix of that channel’s level to the LR mix balance. The channel meters change as expected in EITHER mode and it’s quite easy to forget that you are not actually making recording level changes when you think you are. Such are the pains of a tiny user interface like the one on the F6.
A final note on the Asahi Kasei AK5736 chip: The device includes programmable “bias” (aka phantom power) voltage per channel for connected microphones, which sounds perfect for the F6. However, the programmable voltage range is limited to 9VDC. So it seems the phantom power in the F6 does not originate from this chip. In fact, the F6’s 24 or 48VDC phantom power has to be blocked from the chip’s inputs by a capacitor.
That was very informative, thank you
Sehr gutes Video 👍
Um ehrlich zu sein wusste ich gar nicht genau, dass man die Funktion der Regler umstellen kann 😇
Ich bin mir nicht sicher ob im float-Modus wirklich die Aussteuerung beeinflusst wird, oder nur die Zuordnung wie hoch der Pegel interpretiert wird. Dazu müsste man die genaue Schaltungstechnik und die Umsetzung kennen.
Das bei 24bit die Aussteuerung nur so umständlich eingestellt werden kann war mir nicht bewusst, weil ich nur 32bit aufnehme.
Werde wohl demnächst mal etwas experimentieren.
Gruß Klaus
@@klauskroe4254 Der Aufnahmepegel wird dadurch verändert, Handbuch Seite 194. Im 24bit-Modus ist es wirklich etwas umständlich.
Wie Alex bereits geschrieben hat, es wird der Aufnahme Pegel verändert und dies wird dann entsprechend in die 32bit Datei geschrieben. Nun kann man diese natürlich auch wieder hoch ziehen wenn man in Audition oder ähnlichem ist.
@Alexan Der: Jap, ich habe mir das so genau angesehen da ich total verwirrt war dass die Regeler in 24 eben nicht wie erwartet den Gain eingestellt haben sondern nur das Fading in den L/R Kanal. War ganz schön nervig am Anfang 🙈.
@@AlexanDer2007
Handbuch lesen ist nicht meine Stärke 😎
Very helpful, thank you!
You're welcome!
Thank you for this.
You’re welcome. I’m glad you found it helpful 🌸.
Very informative - thank you.
Glad it was helpful!
wow your light is really bright
Thank you for this excellent tutorial on how to manage gain when recording with a Zoom F6. This was very helpful. I’m not familiar with the term PFL, have you made any tutorials on this topic. It is not explained in the manual?
You're very welcome! I have not specifically talked about the PFL settings but Justin from @Rebel Tech has a video about his production settings and talks about the PFL specifically: czcams.com/video/fDVVPBtaRWA/video.html
@@ChrisSpiegl Thanks so much for the link. Sorry to be a pest, but what DAW do you recommend for processing the gain of F6 Floating 32 bit reocrding in post? I'm using Logic Pro which unfortunately converts 32 bit files to 24 bit and would cause a great deal of clipping.
@@Minus20Design well - in some way all DAWs are going to display and playback the 32bit float files as 24bit. That’s quite normal. However, when you use a normalization, gain change, or loudness normalization effect on your audio it should bring it back in full quality. At least that works in iZotope RX8 as well as Adobe Audition which I am using. Even Premiere Pro does it the same.
So my guess is: logic does support the 32bit float files, you just have to adjust the gain / normalize your files.
@@ChrisSpiegl Brilliant. Thanks so much. I will give it a try.
this is great. thanks for the breakdown!
You're welcome .🌸
thanks for clarification!
Thanks for your comment 🌸. Always appreciate those.
Great ! Thanks!
Thank you, I appreciate that 🌸.
Thank you so much for this video
You're so welcome!
Hey man great Tutorial 💪🏻 Can you show is the different Files in Audition of another Software??
Thx
I thought about that… decided against it last minute. Maybe next time I am talking about this.
Generally though: if you are specifically asking about the 32bit file.
In Reference Level the mic level will be one level throughout and with that the wave form would be equal in hight as long as the sound source stays at the same volume.
In Rec Level: the wave form would be smaller or bigger depending on how the fader knob is set.
You can then of course still recover it all - it's 32bit float after all.
great video! thanks a lot!
I have a general question: I work a lot on "run and gun" kind of video-projects, and I can hardly think of a scenario, where I (or anyone) would prefer to have control over the channel's faders rather than the channel's gain trim. So why are the knobs faders and not gain-trim anyways? To me that seems so strange...
By the way, same thing happens with the Sound Devices Mixpre 6-ii, where the knobs are set to fade by default, rather than gain trim (you can change that with the mixpre 6-ii, but, then you loose the possibility to use limiters...)
I think the reason why they chose to make the knobs faders rather than gain is that with 32bit float you do not "need" trim/gain.
Depending on how you set up the faders, they actually don't affect the individual 32bit float tracks at all and only the mixdown track. I really like that because it means that I can change how loud a mic goes into the mix (for live-streaming / live mixing) but at the same time have one constant 32bit float per mic channel which I then can auto-mix in post-processing manually.
my zoom f6 sounds harsh and iritating at high frequency.
is that normal?
I could send you some audio samples.
Thank you
Excellent.
Thanks, I appreciate that.
I would like to ask as about what we can see on the display: in your example, you have one mic, so we can see it on channel 4. Now we can see also four more levels, LR and LINE. Would you mind explaining why is that so, there is one mic (mono) but we see LR and yet LINE? Why is that so, and what those levels are? That's a beginner question, so I hope my question is clear enough. Thanks
Hay friend.
What exactly is happening when 1 presses in the volume nob? I can haar a hicup which suggests I guess the audio changes and I do kind of hear a difference. But as I'm blind when doing this what's showing on screen? Also which of these profiles or waht have you is the out of box 1?
The volume knobs on the Zoom F6 cannot be “pressed” and so I do not know what you are talking about.
@@ChrisSpiegl Hi Criss.
Ah sorry if I wasn't clear. I ment the headphon volume nob.
The radio dile looking 1.
Hi Chris, thanks again for doing these videos!
What is the purpose of the Line out/Line level when you only needs a stereo signal into the SD-card. I just find it confusing/annoying to have these 2 bars clipping for no reason? Or am I totally wrong?
Stein
Hi Stein, maybe it would be interesting to be able to turn the line out off but I think it's good to have it available. There are multiple good uses for this:
1) the line out can be used to send the signal to a PA system.
2) the line out can be sent to a camera for additional recording a backup track there
3) the line out can be sent to a live streaming setup (ATEM Mini for example - this is actually what I use almost daily).
So it's not something you will need in your use case, but for others it is very helpful.
@@ChrisSpiegl Hi again
Yes, found it in the (not so intuitive) routing that I actually could turn it off, learned a lot form you and the guys at F6 Facebook-group so now I turn on my machine, goes out and have a lot of fun with my F6 :) I can see that many will have great use of the Line out, so good it's there! I've used F4 for several years and Mixpre3 II for a year, but this F6 was a real nut to crack to get going.. Again thanks a lot!
@@snipolar Thanks for checking in again. I appreciate that and hope you get some incredible recordings with the F6!
So , thats mean every time i record 24 bit i should go to the submenu and then trim to control the gain level ? ... correct if i am wrong...
Pretty much
Hi Chris, Q..
How to set Phamton off?
It is possible to set ext potenciometer to control mic volume snd use the trim for line out and LR?
In the "INPUT" setting you have to put it to "Mic" and NOT Mic(PH). Mic is without phantom, Mic(PH) is with Phantom power
As Espen already pointed out, you have to go to the PFL settings and there go to Input and set it to "Mic" and not "Mic(PH)".
And in terms of the potentiometer: that's the topic I covered in this video? Is it not?
So if I have the gain on the channel fader where I'm getting basically no signal in the L/R mix, but I have my trim set so the channel is registering proper peaks around -12, if I record, I will have sound in a track just not in the mix file? Also what do you normally run your trim at? It's dependent on the sensitivity of the mic I assume?
Is it possible to live stream through the x/y mics?
Hi there! thanks for your great videos!
Quick question, I need to use an f6 as a audio interface to record a live performance. Do you know if its possible to use headphone output to speakers and line out as a 3-4 output to send click to the musicians?
Thank you!
Quick answer - F6 operates as USB interface in two modes (stereo and multitrack). You can select up to 4 USB input channels from Mac/PC and route these to 1-6 digital mix channels back to your Mac/PC for recording/monitoring or to the stereo channels (line-out and headphones) of F6. You can deside how to commence this routing using the prefader or postfader of F6. The second part of your question is not related to the USB interface at all. The F6's stereo line-out output impedance is up to 100 ohms (max load of 10 kiloohms), and the headphone speakers output inpedance is up to 15 ohms with max level of 100mW per channel (max load of 32 ohms). Obviously, your approach is not appropriate, and dangerous, btw. You should do the opposite - you should connect the line-out jack to your aux-in/line-in of the active speaker system or to line-in of an amplifier with passive speakers, and the headphones out jack should be connected to an external pre-amp active splitter with two or more headphones for monitoring. Btw, some external headphone amplifiers, like Samson QH4, support aux-in and aux-out, so you can use only F6's line-out for both monitoring and playback to active speakers. Be careful and good luck.
How can you recover the clipped recording in 32-Bit Float mode with other software then Adobe Audition?
You need an audio editing program that can handle 32bit float audio. But I think even Audacity (open source audio editor - crsp.li/audacity ) can do that. Another alternative would be Reaper: crsp.li/reaper
@@ChrisSpiegl Audacity can do it, but it's difficult to pull parts of the recording down to 0 as you can with Audition (works fantastic!) AND it seems there is only a certain amount you can reduce in Audacity, so I do Audition to normalize>save and then into Audacity in 32-bit mode for High-pass-filter and 4-5db noise if necessary. Then save a clean file as wav or mp3.
Hi @@snipolar , great input but I am wondering why you'd go to Audacity for the High-Pass? there are effects like that in Adobe Audacity as well? Why switch between programs?
@@ChrisSpiegl Hi again, Audition creates a too sharp border to my ears when I reduce the lowest part, actually (the free) Audacity creates a very smooth transition, I often reduce the lowest 500Hz for most birdsounds. And the noise-filter in Audition creates box-like appearance in nature recordings, sounds very unnatural in my ears. Probably much more to learn as it is a great software.
@@snipolar I see, probably there would be some options you can tweak in both to get the desired result. Especially when using a EQ to cut the lower frequencies. Happy learning 👍.
It have an internal mic like the F8? To id. Notes
The Zoom F6 does not have an internal microphone.
just one question : is 32bit float can safe the wireless mic clipping ?
No, if the mic clipped on the transmitter, it's clipped and 32bit float will not change that.
Excellent tutorial. ZOOM chosen to reinvent the wheel with those knobs. It's not intuitive at all. Yet, if I got all correctly, you can't change level on the fly when recording in 24 bits. Zoom forcing us t 32 bits :) well this is probably what 99% of people are attracted anyway in this recorder. ZOOM could leave the decision to the users instead of an arbitrary choice. Nonetheless, ZOOM F6 is a fantastic piece of equipment.
Absolutely. User choice would have been nice. Thanks for your comment 👍.
Does Zooms Automix damage the raw footage?
does rec level works in usb multitrack mode?
Hi Chris. Great info here! After transferring the files over usb, the F6 seems to forget all the levels I’ve set every time, so I have to remember to reset before recording again. Am I missing something? Thanks.
I have not experienced this. Maybe try this: turn on, set the settings, turn off (to lock in the settings). Then try again with the file transfer.
Additionally: you can save settings to the SD Card and load those again if all else fails.
@@ChrisSpiegl Yes, if you turn the F6 completely off and back on the settings are there as you say, however this issue occurs if you don’t power off after file transfer. I record a lot of long form content so every 30 mins or so I like to stop and transfer the file to free up space before getting straight back into it. This is when the issue occurs. Sure, I can power it on and off, but that’s an extra step from a workflow point of view. PS. We spoke a while ago about using a 3.5mm splitter for timecode to multiple cameras. I can confirm that this works with no issues that I’ve encountered which is great, especially because I’ve been waiting for a pair of tentacle sync E mkii to arrive for over 6 months now! Hope you’re well. Joel.
@@JoelPurnell Maybe let Zoom know and send an email to their support team. Personally, I mostly use the Zoom in some standard settings and do not transfer the files via USB-C Card Reader Mode but instead pop the SD Card and transfer that way.
And right, the Timecode with a splitter cable. I since also used that feature occasionally. It's really helpful when all the cameras are close by and easy to reach.
great video!!!! what is your favourite settings?
As I mention in the video, I prefer Reference Level mode because that way the audio levels in my 32bit recording are never changed during the recording and I just send it through loudness normalization once I am back at my desk.
@@ChrisSpiegl Hi, I do recordings of (mostly) birds with stereo parabolic mic on my F6, what setting would you suggest for a stereo in / stereo out to SD-card with maximum gain. I do the normalization in Audition so no problem with overload. Hope you have some thoughts ab this! Stein
Hi @@snipolar/videos , I have little experience with recording birds. The one thing I would consider important is to link the channels together so that they are recorded as a Stereo Pair and not individually.
I would then also consider recording simply in 32bit float and the highest sample rate desirable for your work.
Cheers
@@ChrisSpiegl Thank again, the F6 facebookgroup helped me out, I've now linked 1 & 2 as Stereo AND with both set with Phantom Power:
Fader 1 adjusts both channels and I get a nice signal to my SD-card. BUT there are lots of other stuff to be found within this machine..tnx
@@snipolar Great, that sounds pretty much exactly like what I would have done 👍. Happy recording.
That bit about how convoluted it is to change recording levels on the individual channels... Having to open the menu to adjust the "trim" when the actual knobs are right there, ready to be used if they could... uhm... I am not sure I would buy the recorder anymore...
There is a good reason they did it this way but it's a bit sad that it is not customizable for those that want to use it differently.
So in 32bit mode If the line is clipped and didn't notice on set, it's completely gone? How do I do with out line out not on?
Not sure what you mean. When you record the audio in 32bit float and you get back to your editing station, you can lower the gain and it will be recovered without quality issues.
@@ChrisSpiegl some reason it's clipped with no fix and don't understand why, it's possible the line out went in the red
@@darkechoproductionsllc9559 Not sure why you mention the line out? Are you sending the signal somewhere?
Exist a 1000 hz tone to calibrate volume in headphones, like the F8?
@@ChrisSpiegl Yes , I found it. The only thing always go throught the menu takes time, But it is just in the beguining of the main file to share with post audio and calibrate.
@@mariorivas7886/videos I absolutely agree and would also prefer if I would be able to change the behavior of the faders to actually work as Trim Knobs instead.
But sadly the Zoom F6 has no setting for that.
A bit more complex than the Zoom F3 :)
Yes, the F3 does it differently which is great cause the F6 is years old at this point. (still great though)
czcams.com/video/eEIl7LL-Zyk/video.html
problem with f6
i have myself a ticking sound since i update the firmware with 1.90E
Why? It must be programmable to set the trim via the knobs. And they must know that this is what is desired. And it can't be because they don't want to cannibalize their top level line because it doesn't get any better than this for them. Do they have a deal with other companies?
Don't know, we only can try to make our voice heard by sending messages to the support team.
first coment from Portugal :)
I appreciate it 🌸 hope you are doing well.