Nice explanation as always, Julian!

@JulianKrause

4 жыл бұрын

Thanks Curtis! I really liked your first impression video on the F6. Hopefully I can get my hands on one soon :)

Extremely well done Julian!

@JulianKrause

5 жыл бұрын

Thanks Bandrew!

@mhmnmhmn4213

5 жыл бұрын

This is how 2 audio experts have a conversation

You are a fantastic communicator. Great video-I learned a lot!

I love these in-depth tests. You deserve way more subs...

No gain, no pain.

@JulianKrause

5 жыл бұрын

Haha, hopefully!

@YoungBlaze

5 жыл бұрын

@@JulianKrause im hungry, are you making any tacos?

Excellent video, very informative and very well explained. Nice job .

Awesome video! Thank you

Great explanation...very helpful...Thanks!

Useful video, thanks for explaining👍🏽

Kind of reminds me of how HDR works for Photographers, but simultaneously.

Great video 👍🏼

Thank you so much for this explanation. I thought it was a gimmick or something professionals wouldn't use (like auto gain/auto mix).

Brilliant video.

Thank you for the good video.

Hi Julian. Great video and excellent explanations. My only worry with this type of recording is that there's the possibility of the record engineer getting lazy and leaving everything to the mix engineer in post. Cheers, Dave.

Basically this is HDR of sound. We used this logic in negative for 100 year now. Double exposure for low and high separately.

Thanks for this explanation; a much better job of messaging this technology than Zoom themselves. I've already started using the F6 and it's a very impressive recorder (quite an upgrade from my long-time staple Tascam DR-70D).

Excellent video Julian! A very clear explanation of this system. It seems as if Zoom has developed something that could be quite useful for some scenarios where the trim may not be accessible. But as an audio pro I must admit I’m a bit skeptical of the marketing claims by Zoom. I do also think it’s important to emphasize that no matter the dynamic range of the dual ADC and benefit of 32 bit float in the digital realm, the total dynamic range of the full audio system is still very dependent on the dynamic range of analog front end. And if anything is clipping on the mic, mic pre or wireless system on the analog front end it cannot be recovered in 32 bit float file. And if the pre amp gain level is not optimized for the mic or input sensitivity of the signal unnecessary noise may be added to the signal chain. So not being able to adjust the input trim on this unit and Zoom’s claims that it is not necessary with a 32 bit dual ADC system totally negates the importance of a nominal operating level for all analog and digital systems to achieve the best results and lowest noise floor.

@JulianKrause

5 жыл бұрын

Of course, every distortion that happens to the audio signal prior to the F6 will be recorded, so setting up your wireless system correctly is still a big deal. I think the F6 shines when the mic is directly connected to the unit because this way the dynamic range should theoretically only be limited by the mic. And a mic only clips at a very high SPL. It is going to be interesting how the F6 is going to fit into existing workflows because the mentality when using the F6 is a little different to current recording solutions. "could be quite useful for some scenarios where the trim may not be accessible" Maybe Zoom developed this because the fader knobs on the F6 are so fiddely to control XD

helps me a lot!!! dankeschön

It sounds like a very interesting development for audio recording.

@JulianKrause

5 жыл бұрын

Yeah, I mean this technology has been around for quite some time. Companies like Zaxcom, Sonosax and Stagetec have been using this approach to capture a considerably higher dynamic range than what is possible with a single ADC for a few years now. But I have been waiting for someone to bring this technology into a more affordable range and even though the pricing for the F6 is not out yet, I have a feeling it is going to be much cheaper than devices with dual ADCs from other manufacturers.

32 bits = 24 bits for audio signal + 8 bits for gain information. Mixing two ADC should be pretty easy if they are always in phase. Weighted average should do the trick of smooth transition. For me it all sounds like a great idea and the future.

@JulianKrause

5 жыл бұрын

Merging the signals of two ADCs is actually quite tricky. For the right timing you could use the same clock, but for there will still be tiny variations in the signal amplitude between the two ADCs. If you average the signal in the transitional area it is neither the real signal of the first nor the second ADC. This signal deviation will result in harmonic distortion even if there is a smooth transition between the two ADCs.

@KarolMurawski

5 жыл бұрын

@@JulianKrause Weighted average should do the trick of smooth transition with no harmonic distortion. Algorithm may be as follow. Abbreviations for conciseness: HG-ADC = High Gain ADC, LG-ADC = Low Gain ADC. For each sample decide if signal high, low or in the overlap region. I mean absolute value high for simplicity. For high signal take data from LG-ADC, for low signal take data from HG-ADC. For overlap region calculate weighted average. (P.S. overlap should not be too big - it should be specified in a way that each ADC is still quite good in this region.) Step 1. Calculate auxiliary_amplitude = 50% HG-ADC + 50% LG-ADC. Step 2. Use auxiliary amplitude to define weights. If auxiliary_amplitude is at the upper end of overlap region weight=1, if at the lower weight=0, if in the middle weight=0.5, etc. Step 3. Calculate final amplitude. final_amplitude = (1-weight)*HG-ADC+weight*LG-ADC.

@davidgriffin79

4 жыл бұрын

@@KarolMurawski This is too simplistic; a simple weighted average will not take into account any differences in waveform from the DACs in the transition region. As Julian has pointed out, if you just combine a weighted average between the DACs the resultant superposed waveform will be distorted due to any small differences between the two DACs - that is the harmonic distortion which would be seen on a Fourier analysis of an example waveform. One way to remove the anomalies between the DACs might be to subtract the two DAC waveforms to obtain the difference signal which would be the error; this could then be subtracted from the final waveform - just a guess.

can't wait for your review of Mixpre-3 ii

Great video, thank you. Will you be making a follow-up video of the Zoom F6?

Good video! Thanks. Does this only work if you record with the Zoom F6 or also when I connect the zoom to my Ninja V and record Video and Audio there?

Great video, Julian! Where can I find your actual review of the F6?

When I lower the gain in either Audacity 3.0.2 (confirmed can edit 32bit float) or AudioDirector 11, it just flattens all the peaks to one equal level. Using a Rode VideoMic Pro+ with a Zoom F2 BT. I specifically got this combo since I record very loud concerts. Any suggestions? I see in your video you just adjust the gain and it works as expected. Thank you!

I hope this works as well as Zoom are saying, but I'm holding on to my skepticism until there's real-world testing/reviews. When they first introduced it, my main concern was that this dual ADC approach would put a lot of pressure on the preamp quality (I assumed both ADCs would be taking their signal from the preamp) - and Zoom does not have the best preamps. But the possibility that the lower-intensity range might be handled by bypassing the preamp entirely (with or without attenuation) is very interesting, as it may be able to bypass the preamp's noise floor... But I am 100% enthusiastic about recording in 32 bit float with no reservations! It's akin to log for video, but even better.

Sort of like HDR photography. Reminds me of the old 3m Dynatrack analog tape recorder.

I wonder which is more expensive for manufacturers to implement. High quality analog limiters, or dual ADCs. I have a feeling the market will really gravitate toward this kind of dual ADC and 32-bit float setup, because it will make clipping and gain staging mistakes a thing of the past. Even pros will eventually appreciate having one less potential headache to deal with every day. I found to my great disappointment that it was very easy to end up with numerous clipped samples even on a MixPre if you set the gain too high, especially on the stereo mix since it aggregates the total merged energy of the isolated tracks. It will be so interesting to see how Sound Devices and others will respond to the F6 in the future. I think it will be very popular.

@JulianKrause

5 жыл бұрын

I think the dual ADC approach is superior to an analog limiter and the cost wouldn't be too much higher. If done correctly you capture the whole dynamic range with dual ADCs, so you can use a digital limiter in post. A digital limiter can work much better, because it can essentially look into the future and change settings like attack and release times to appropriately. And yes, it is going to be interesting to see how much use it gets in the future.

@lamenamethefirst

5 жыл бұрын

I had the same issue with the Mix Pre 6 recently. I wasn't too impressed by the preamp quality. It has a low noise floor for sure but it can't tolerate high gain. Clips easily like my H6. Not something I expected from Sound Devices. The preamps on older SD mixers like the 302, 552 are way better in my opinion.

I feel like a student, thanks !

Do you know if the F3 also have this feature where you don't have to adjust your gain again?

Can it be used as an audio interface replacing a motu m2 for example ?

Can you check out the channel noise with the limiters like you did with the F8? I did some simple listening tests and I don't hear a difference where on my F8n it is very clear. Maybe even in 24 bit mode it is all 32 bit in the F6 so it adds 10db of noise but the noise floor is hundreds of dB down and isn't close to the 24 bit range. Or get an F8n Pro and check that. That is why I actually am interested.

Hi Julian, I watched several videos from you about the zoom f6 32bit float production. I'm currently working on a long feature movie shot with this new technique. I'm having troubles inside Premiere Pro : it reads the file as a 32bit float and the project is in 32bit float, but I think the software misinterprets the file : most of my premixes are over-modulating and go higher than the 0dB, so I get saturation. If I try to low the gain, the waveform gets lower but the saturation keeps being audible. In Audition it works the same as your video. So I'm a bit confused about what I can do. Some files need something like -6dB gain to be not saturating, some others -10dB. Would you share with me some advices on how I can manage this ? It is a lot of material so if you have any idea about how I can convert the files with some automations and keep the metadata so that audio postproduction can relink on the 32bit float files ? Thanks a lot for your videos and explanations it already helped a lot !

My zoom f6 sounds too brittle, metallic, harsh and edgy with every mics. Especially when I use a brighter mic like sennheiser e935. It is hard for me to get a good smooth result with equalizer. Some certain high frequencies spot, like 4khz and or beyond seems too emphasized. Is this normal? I will send you a file sample if you dont mind. Thank you

Does anyone know which ADC chips used in Zoom F6? I would really appreciate if you could share with some technical info. Thx

great video! quick question, which camera, lens and fps did you record at/with? - thank you!

@JulianKrause

5 жыл бұрын

Hey, the video was recorded with the original Sony A7s and an adapted Canon 35mm f2 USM IS lens. I filmed at 25fps a shutter speed of 1/50th and ISO 800.

Will you do a test of the actual device? I found some bad reviews on a site but it seems they were not as professionally made as you do. Thanks!

what montiors are thes in ur video??

Can you capture 32 bit float on your PC, instead of the F6, with the F6 used as an audio interface?

Would you recommend this over the F8? I normally just do 2 mics

Would love to see a pre-amp shootout between this and the Mix-Pre series.

@JulianKrause

5 жыл бұрын

Accoding to Zoom the F6 has the same preamps as the F4 and F8. I did already make a comparison of the F4/8 and the MixPre series. You can find the video here: kzread.info/dash/bejne/p4V2msSrhbzenM4.html Of course, if I get my hand on the F6 I will test its preamp noise performance.

@BushMasterThermal

5 жыл бұрын

Mix Pre maximum input level XLR Mic: +14dBu Line +28dBu Where as for this "new release" Zoom F6 Maximum input level XLR Mic: +4dBu Line +24dBu

@JulianKrause

5 жыл бұрын

@@BushMasterThermal Well, yes the mic input has a max handling capability of 4dBu. But in most cases this is going to be sufficient. Worst case scenario: You got a mic with a high sensitivity, let's say 30dBV/PA. It produces roughly 4dBu when there is 125 dBSPL at the mic capsule. That's a very loud sound pressure level which you will rarely encounter.

@BushMasterThermal

5 жыл бұрын

​@@JulianKrause Well let's say a High SPL handling mic. With a sensitivity of around -34 dBu Roughly 32 dBV/PA. Maximum SPL handling 1% THD at 140 dBSPL at the mic capsule Maximum Output Level the mic produces +12 to +14 dBu. I have got into contact with Zoom and they did confirm the XLR Line Input Source setting is capable of Phantom Powering with Line Source selected. So in the case of high sensitivity mics and that are also capable of handling High SPL's we seem to be quite safe.

Thank you for quantifying various behaviors/aspects/parameters of the USB interfaces. This is largely missing in the reviews outside of your channel and hence highly appreciated. It also comes with the beauty and the convenience of being able to easily/objectively compare the interfaces, which I could not find anywhere else. [To the topic] Since the newest generation of dual-ADC field recorders (from Zoom, MixPre, Tascam, etc) use fundamentally new ways to convert analog signal into digital realm, I wonder if you've already pondered about and/or discovered any downsides of this technique (i.e. dual ADC with "seamless" ADC output merging), which would potentially translate to poor performance in some of your measurements? Since the dual-ADC technique is not employed in the traditional (even the newly released) USB interfaces, I assume there is a reason behind that?

Hi Julian. Are you planning to review F6 anytime soon? I'm very interested to see measurements. Many thanks for your time and efforts you put into these contents.

@JulianKrause

4 жыл бұрын

I'm really interested in the measurements too but I currently don't have any access to a F6 so don't expect a review soon. Sorry!

Does anyone know if you use this with a Shure SM7B for podcasting, would you still need some kind of mic activator like a Cloudlifter? thanks

@JulianKrause

5 жыл бұрын

Hey Gary, according to Zoom the F6 is using the same preamps as the F4 and F8. These preamps have a very low noise floor and that's why I think you won't need a Cloudlifter with the F6. Even if you use it with a low sensitive mic, like the SM7B.

i just received it today, is amazing but i noticed a problem :( we all know that when we zoom i the wave form it affect the line out and the recording, BUT if you set the Zoom and start recording then any changes on the zoom will affect the Line out only, not the recording, in my case i need this feature as i record everything coming out of the DJ, so i would set different zoom level when people dance than the level at speeches time and i don't want to keep stop and starting the recording!! anyone knows any thing about that??

@JulianKrause

2 жыл бұрын

Hey, as far as I know this is done deliberately, so the level doesn't change while recording. But you can simply boost the audio in the speech parts in post. I would probably just set a decent level, leave the F6 running the whole time and do any volume adjustements in post. That's the whole idea behind the 32bit float and dual ADC recording. Capture everything, adjust gani in post.

It would be better if microphones had chips in them that the recorder can read & set the gain automatic to stop clipping

Hi Julian. Great explanation; thank you for that. I have a Zoom F6 but it continues to clip when I use the dual 32bit Float and linear formats. I essentially can't hear any difference between the two recording formats. Any idea what I might be doing wrong?

@JulianKrause

3 жыл бұрын

Hey, which software do you use to monitor the audio? I have the feeling that the software might clip the audio.

@redharemedia1034

3 жыл бұрын

@@JulianKrause I import the files into Pro Tools, if that's what you mean.

7:54 WOOOOO

Very well explained how this new audio recorder works, but for me personally, I do not like the looks of it and the way how the knobs are placed at the front, much too close to each other if you ask me.

@JulianKrause

5 жыл бұрын

Yeah, I think the design could have been better. It is going to be tricky to turn the faders.

I enjoy the thoughtful balance of theory and practice in your videos, but you feed my desire for more knowledge. OK, so here's a hypothetical question: let's say someone wishes to save his pennies to purchase the F6, but in the interim still wants a nice, big clean sound image; however, he only has a pair of H1n recorders. Could he theoretically use the low cut on one and the limiter on the other with the gain up on both, then merge the recordings in the DAW? Afterwards, could you theoretically trim off the top and bottom for a nicer image, kind of like the F6? What do you think?

@JulianKrause

4 жыл бұрын

Interesting question! I don't think it would work with two separate H1ns because their clocks are not synced. This would result in some kind of cross-over distortion when the signal jumps from one H1n to the other one. What should be possible though is to use the stereo channels one H1n. The left channel gets the normal signal from the mic and the right one an attenuated version. This could then be spliced together in post. Though I don't know any program that would be able to do this, so you would have to program this on your own.

I am hoping for Julian's EIN comparison of a Zoom F6 vs. comparable Mix Pre!

@JulianKrause

5 жыл бұрын

Me too ;)

Nice!!! What DAW do you use? Thanks

@JulianKrause

3 жыл бұрын

Hey, I use Adobe Audition.

@Liquidfusionvideo

3 жыл бұрын

@@JulianKrause - ever try Reaper? It's great software!!!

Are you going to review the Zoom F6 ?

When I asked Zoom "What is the dynamic range of your dual A/D converters?" I have got the answer that "The dynamic range of inputs of the F6 is 131dB". The overlap of ADC's must be huge. 131dB is substantially more that 100dB and even 120dB, but I must say it's disappointing, comparing to Sound Devices MixPre II 142dB. There are many microphones with dynamic range higher (sometimes way higher) than 131dB (like the Rode NT1-A, not to mention Rode K2). Saying that you clip a mic before you clip a ADC is simply not true. Even Sound Devices don't make such a claim, despite the fact that with 142dB its very hard to find a mic that wouldn't clip before their ADC saturate.

@JulianKrause

4 жыл бұрын

I agree, the statement "the ADC is unclippable" is definitely not correct. On the other hand I have to say that a dynamic range of 131dB is still huge and there are hardly any szenarios where you need such a dynamic range, let alone 142dB.

@przybylskipawel

4 жыл бұрын

@@JulianKrause There is one scenario. The one when you record, say, a scene when you record someone wispering while setting an ambush that turs into a gunfight... :) But orchestral recording can also be a chalenge.

If this does work like that, it makes this device pretty much the perfect recording solution. Of course, that is without other issues, the device might have. I am curious, if you can send a live 32 bit floating point feed to the PC, when using the F6 as an audio interface. If so, this device would be VERY interesting to me, despite its steep, estimated price point.

@JulianKrause

4 жыл бұрын

I agree. Btw, SoundDevice just had the same idea :www.sounddevices.com/product/mixpre-3-ii/

Thanks for your video. I think I cought what you explained and, if it's so, I don't understand why someone could set the gain the mic at whatever level without permanent consecuences under some scenarios. If you set it TOO low you could have problems with "digital noise floor" (I don't know if it's called that way) bacause even for a 24bit file there is a limit (-144dBFS), so if you drop a pin and try to record it three meters away with the gain to the lowest position you might not get anything. By the other side, if you set the gain all the way up, for a 32bit-float file there's a limit (+48dBFS if I'm not mistaken) so you could eventually clip if you recorded an interview with your "famous jet" if you set the gain to the top (assuming the mic puts up with such a sound preasure). I don't know if I could explain everything correctly or if there's anything I'm missing. Maybe it's because my native language is not English haha. Thank you for your videos.

@JulianKrause

5 жыл бұрын

Hey, the reason you can set your fader very low and don't run into problems is because the "digital noise floor" is very low in 32bit float. It is at -700dBFS, so even though your digital signal is very low, it won't drop below -700dBFS. With the 32bit float you can go quite a bit above 0dBFS. +144dBFS are totally usable and you can go even higher if you are willing to sacrifice a bit of signal accuracy. So, it is nearly impossible to clip your signal in 32bit float. Hope that answers your question!

@markolinostyle

5 жыл бұрын

@@JulianKrause Thank you so much for your answer. After reading your answer I started reading some articker and experimenting with my daw and now I allways thoght that a 32bit float was what I now Know it is a 32bit fix (I imagined that the term "float" had something to do with the ability to go beyond 0dBFS without distortion an ammount of 8 more bits). Whay I still don't know is how to calculate its exact posibilities in terms of dynamic range. If you know it it would be a good idea that you could publish a video explaining it to us. Now, talking about the Zoom F6, I imagine that, if the recorder has 2 ADCs that work at 24bit fix, its dynamic range will be 288dB, is that correct? Again, thank you for your anwer.

Why haven't you reviewed this yet?

Interesting, the theoretical dynamic range is 192.7 dB, assuming two 16bit DACs. However, the transition region would give an overlap so maybe giving an equivalent 28-30 bit DAC ~169dB - 181dB of dynamic range; that's still pretty huge.

@JulianKrause

4 жыл бұрын

Hey, in practice it is not that big. The dynamic range of the Zoom F6 is allegedly about 131dB(A). Sound Devices just announced the MixPre II series which has around 142dB(A) dynamic range. This is still way more dynamic range you usually need.

@davidgriffin79

4 жыл бұрын

@@JulianKrause 131 dB is well within 24 bits so why not apply some some form of normalisation with 1 24 bit DAC? It's all interesting stuff..

@JulianKrause

4 жыл бұрын

@@davidgriffin79 You are right, 131dB or even 142 dB of dynamic range could be encoded with 24 bits. The problem is that when you map the maximum input of such a high dynamic range to 0dBFS, in a normal recording situation the waveform would be very small. And want to have a stronger digital signal to be able to monitor it easily. This can be done with digital amplification. But now you got the problem again that with the added digital gain the signal could go over 0dBFS which results in clipping. With 32bit floating point you have the benefit that your signal can go above "0dBFS". So, you can apply digital gain to your signal to be able to monitor it nicely and simultaneously you are still safe from clipping.

Plz review it

Why has this only just been released in 2019? They should have released this technology decades ago.

No gain no pain🙂

32 Float Dual ADC ought to have been on the H8!!

@JulianKrause

3 жыл бұрын

Hey, I don't think so. From the specs it look like a normal 24 Bit ADC implementation.

@Liquidfusionvideo

3 жыл бұрын

@@JulianKrause - ZOOM should have put 32 float on Zoom H8 along with Dual ADC

@JulianKrause

3 жыл бұрын

@@Liquidfusionvideo Yeah, I agree!

@pkb9499

2 жыл бұрын

Zoom h8 is ugly 🤣

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.

Hallo, Leider funktioniert 32bit float nur als Fieldrecorder, als Interface funktioniert nur der 24bit Modus. Also auch mit den schlimmen Übersteuerungen, wenn man nicht aufpasst. Ein Vergleich zu den von dir getesteten Interfaces wäre interessant. Die nicht deutschsprachigen Leute mögen mir verzeihen. Gruß Klaus

@JulianKrause

3 жыл бұрын

Wenn mich nicht alles täuscht kann man bei dem 24 bit modus einen digitalen Limiter einstellen. Der AD-Wander läuft immer noch mit 32 bit und profitiert von dem großen Dynamikumfang und rechnet es auf 24 bit herunter. Sollte bei 24 bit etwas übersteuern würde der Limiter das komplett einfangen, weil das Signal bei 32 bit noch nicht übersteuert ist. VG Julian

@klauskroe4254

3 жыл бұрын

@@JulianKrause ich hatte es mal ausprobiert, habe aber wohl nicht die richtigen Einstellungen gefunden. Da hatte es nicht funktionier. Werde es aber in eine ruhigen Minute noch mal versuchen, oder mal sehen ob ich im Netz was finde. Wenn es klappt werde ich es mitteilen. Dein gutes Wissen um die Audiotechnik und deine beneidenswerten Möglichkeiten alles zu testen sind einmalig hier auf KZread. Du hast mir schon oft weitergeholfen. Gruß Klaus

@JulianKrause

3 жыл бұрын

@@klauskroe4254 Danke für die netten Worte! Habe leider gerade keinen F6 zur Hand, sonst würde ich den Limiter mal testen. Vielleicht hat jemand im Internet darüber schon was geschrieben. VG Julian

There is gain setting haha 😄 😆 😅 😂 🤣 I can prove it too

From a software standpoint, merging the two signals of the A/Ds is trivial. You have a threshold which informs the decision of which input to pass along, and they are normalized to each other.

@JulianKrause

4 жыл бұрын

Yes, but from a hardware standpoint it is not that trivial. Both ADCs have to be completely in sync when sampling the audio. The time they sample the audio wave has to be identical and at the cutover point the amplitude of the sampling points has to be aligned. Otherwise you introduce distortion.

You didn t talk about the quality of preamps.....ohhhhhhh

SuperRAW Audio....

if you can just speculate on how it works, don't call your video "how it works"

Why not make sure that you are truly saved by Jesus Christ and practice this way. Remorsefully confess with your heart your sins to Jesus Christ who is God and tell Him that you right now are repenting of your sins and you want to be born again of the Spirit from above. Tell Jesus that you are remorsefully sorry for breaking His commandments and that you are begging for forgiveness from Him. Allow His blood from the cross to wash away your sins. After this is done with your heart successfully the Holy Spirit will come to live within you and He will rebuild you from the inside out. Look for signs that you are living righteously. Things like spreading the good news from Jesus, getting other people saved, a craving for the word of God, reading the Bible, etc… These things are known as a calling and fruit bearing. If you're not bearing fruit then keep doing it. Sometimes it takes time to get saved. Read Matthew chapter 13 from the King James Bible. God bless!!!