whew quite a bit to catch up here, excuse the long combined post
On the other hand, if what you are recording has a more limited dynamic range, then the advantages in 32 bit are less evident. This I will agree with. However, the blanket statement that there is zero advantage to 32 bit is incorrect.
we are actually in agreement, that is *exactly* what i said. Based on typical use case of most tapers here, very limited (more like no) advantage for most. I never said zero advantage for all applications and i trust in Voltronics extensive use in his application.
I'm not sure i quite understand the applicability of dumping a 24-bit USB signal into a 32 bit container on the computer to discussion of this gear. Nothing is gained in that scenario other than limiting quantization errors on downstream processing, which is basically how every wave editor already works fundamentally (regardless of native bitrate, all calcs done in 32bitFP, and a final render/dither to bitrate of choice).
Its fundamentally different than the application we are discussing here, the architecture of the 32bit FP recorders which use multiple ADCs to overlap in proprietary ways to provide a larger ADC dynamic range
It may be true that you can record in 24 bit with the Peak amplitude not going over -20dB
in a live concert environment with a recorder with a similar 120dB+ EIN, probably more like -40dB or more. Ill do some tests and well see if people can pick them out
However, let's say that you are recording an artist that has a much wider range in dynamics (SPL as presented to the mic, for instance). If you recording in 24 bit, you can absolutely clip the signal at the recorder before the mic reaches its' maximum SPL limit.
of course you can. You can also clip the same input at the same SPL at 24 bit. The zoom has relatively modest max input level 0f +24 dBu in line-in mode (relative to some other gear that can take signals approaching +30 dBu)
Whether you record at 24 or 32FP, that max input level does not change
In regard to recording at +24 dBu line in, vs +4dBu mic in, all the zoom is doing in the former case is using a 20 dB pad.
Ive tested it myself that recording with this pad (aka "line-in) raises the very real noise floor of the analog input substantially At the end of the day there is no free lunch. The analog input is still constrained by its max input level and EIN. You cant entirely scrap the analog gain stage like in this design, and rely on wide ranging ADs and not have compromises. This is why real preamps that can offer 60-70 dB of clean gain are so valued. Im sure that in theory, that with much more expensive/complex gear, there could be an input stage that offers more substantial use of the 32-bit FP but youd be hard pressed to find an application to take advantage of it (recording bird calls next to a rocket launchpad, perhaps, with some advanced imaginary transducer with over 130 dB dynamic range)
If you recording in 24 bit, you can absolutely clip the signal at the recorder before the mic reaches its' maximum SPL limit. This is not the case with 32 bit recording with a properly implemented input gain stage. In this scenario the mic will clip before the recording stage clips. Of course, this can be prevented in 24 bit by riding the recorder gain and tweeking up or down
stepping back a bit this is demonstrably false. again its the same analog input regardless of bitrate/ even in 24-bit fixed the dynamic range of the input does not change. And you dont need to be anywhere close to 0dB with 24 bit, its still relatively easy to leave yourself, 12,20, possibly 36 or more dB of headroom. certainly no need to 'ride the levels'. Again ive successully abused good 24bit ADCs, eschewing a preamp, and made concert recordings peaking near -30dB with the noise floor inaudible beyond ambient noise.
2) Can the raw recorded 32-bit files be FLAC'd for storage without having to do any processing other than renaming the files if desired and doing the FLAC compression?
i record in polywav, and end up saving my raw 32-bit polywavs as large .w64 files from soundforge
FLAC can do 32FP stereo files fine, I think fundamentally the standard allows FLAC to handle polywavs, but my standard workflow for making FLAC files chokes on it somewhere, so i just do as above. the .w64 files are relatively enormous, >10GB per set at 6 ch of 32/96
3) Can the raw recorded files (in their native file format and/or FLAC compressed format) be played easily via standard computer media players without having to undergo file conversion?
foobar plays back 2-ch .wav and .flac fine
I'll also note that none of the current 32-bit FP capable recorders require recording in that format, so the mere presence of this capability needn't influence one's choice of recorder if one is not planning to use it for whatever reason.
cheers to this. even in absence of having the 32-bit feature, i still think the Zoom F6 is an outstanding piece of hardware for the price. In fact, some of its better features (like fader adjustment and simultaneous USB output) are crippled in 32FP mode
There are also differences in implementation between the SD and Zoom flavors. I have no idea to what extent that influences sound quality, but SD obviously thought it was worth a lot of R&D and a patent application. As I understand the patent, it is a little bit hardware but mostly it is software to deal with the issue DSatz has described (admittedly based on a cursory reading and when it was first published). Perhaps that offsets the inconvenience of the polywave format (although, personally, I don't find that to be a problem).
i think their trick is using 3 ADCs instead of 2, and interpolating the handoff between them somehow. Again I dont think that weve seen that in either design that handing off between multiple ADCs is causing distortion in either device
I'm not sure that their design makes any better use of the real-world constrained dynamic range of analog input in portable gear
i think we all win in the end when manufacturers take their own path to achieve their means, healthy competition breeds innovation.