JayKay

I see, that wasn't my intention. I'm with you and think it is very likely that they will use dual 24bit ADCs and stitch their data together in a 32bit float file. Initially 0dBFS of this file would line up with the clipping point of the more insensitive ADC. At this point 0dBFS is of course still the maximum signal level you can record. It is only after the initial AD conversion that you could push values above 0dBFS with the digital gain. But this would then be recoverable because of the 32bit float.

That's true for when you are capturing audio. In the process of analog to digital conversion, 0dBFS is really the maximum signal level you can record without clipping. But once you have your signal in the digital domain and in 32bit float you can go above 0dBFS. This has to do with how 32bit float is encoding the data. You can try it yourself: Take any sound file in your DAW. Your DAW has to work in 32 bit float. Then apply a lot of digital gain to your audio, so that it goes above 0dBFS. Export this sound file as 24bit int and also as 32bit float. Import these two files again and use digtal gain to lower their amplitude. You will see that the audio in the 24bit file is clipped at 0dBFS and with the 32bit float file you can recover the audio above 0dBFS just fine.

I think one of the biggest reasons why Zoom is using 32bit floating-point is because of the digital gain knobs on the F6. If you turn up the digital gain it is possible that the signal would go above 0dBFS. With 24bit fixed-point 0dBFS is the maximum value and everything above that simply clips. 32bit float on the other hand allows the signal to go above 0dBFS and can still capture it just fine. You could then turn down the audio in post and save all the audio above 0dBFS with the 32bit float file, which wouldn't be possible with 24bit file. So, the 32bit float ensures that even when the digital gain on the device wasn't set properly, the whole signal is captured without clipping.

This interview from Curtis Judd just confirmed that the F6 has dual ADCs. This means that the F6 should record such a high dynamic range that you can capture the whole dynamic range of a microphone. This would completely eliminate the need to set your gain. You can set your gain digitally in post. It's a bit like RAW but for audio. Exciting stuff!

Sorry, I did read that wrong... So yes, the difference between the two mics is about 6,5mV which in this case comes out to be 2dB. But 9,7dB difference sounds a bit excessive. @Adam White You have to be 100% sure that the gain of both channels are set to an equal amount for your test to work. You could do the following test to check if your system is setup properly: Record the same audio (e.g. short music track) with the same mic but first on channel one and then on channel two with the gain set to an equal amount on both channels. Both recorded tracks should then have the same amplitude. So, your gain plugin should then measure zero difference between the two recorded tracks. If that's not the case then there is something wrong with your test setup.

@Rick Reineke how did you calculate the sensitivity difference? I also did the math and I got a 2dB difference in sensitivity. NTG3: 31.60mV/Pa = -30dB re 1 VPa MKH416: 25mV/Pa = -32dB re 1 VPa -30dB re 1 VPa - (-32dB re 1 VPa) = 2dB diffrence in sensitivity between the two mics according to manufacturers specs @Adam White how did you test the sensitivity difference?

In the past I tested the Limiter of the F4 (which should be the same as on the F8) and the dynamic range did not increase when the limiter was engaged. As expected, it merely shifts the dynamic range upwards a bit. Depending on the gain setting you do get at least 10db more headroom but of course the noise floor of the ADC also rises by 10dB. So, overall the dynamic range stays the same. At 2:55 you can see the noise measurements if you are interested: That said, an ADC with 120dB dynamic range is already very good and will be sufficient for the most recording situations.

@mikewest I did a few more tests and found out that this problem only occures if phantom power is enabled. My guess would be, that phantom power is leaking into the preamp on channel 1. Sound Devices set me up with an RA and I´m currently sending them my devices.

You are right, but this problem is not affected by the source impedance. I also tested it with a 150 ohms resistor... Same result. Btw. the noise ch 1 is only more noisy in the lower frequencies. 500 hertz and below which is unusual. At 20 Hz ch1 has 14db!!! more noise than ch 2 or 3. This device was already sent to Sound Devices by my dealer and they said, that everything is fine So if anybody else is experiencing this, it might even be a design flaw...

Hi together, would someone with a MixPre-3 do me a favor? Could you connect nothing to the XLR-Inputs at all, turn up the gain on all channels all the way and turn the faders to unity. Then take a photo of the display, where I can see the levels of all tracks. My MixPre-3 seems to have a problem with extensive noise in channel 1 compared to channel 2 and 3. And when I do the test above, the level of channel 1 is quite a bit higher than channel 2 and 3. I want to know, if I´m the only one with this problem or do other MixPre-3 users experience this as well. Thanks in advance!