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The Wired magazine online portal has a nice +++ ARTICLE +++ explaining what 32 bit floating point audio is as well as why you might, or might not, want to use it.
Jim Fogle - 2024 BiaB (1111) RB (5) Ultra+ PAK DAWs: Cakewalk by BandLab (CbB) - Standalone: Zoom MRS-8 Laptop: i3 Win 10, 8GB ram 500GB HDD Desktop: i7 Win 11, 12GB ram 256GB SSD, 4 TB HDD Music at: https://fogle622.wix.com/fogle622-audio-home
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Pretty good article. It's refreshing to read something realistic on the subject instead of the marketing and fanboy crap that accompanies every piece of gear that can handle it.
I've been doing 32 bit float a long time now except for Export/Bounce to Disk. Clients want to see 24 bit.
The only problem I've had is that Adobe Premiere Elements 2022 does not like 32 bit audio. Interestingly, PE 2021 handles it fine. The workarounds are obvious, use 24 bit or 2021 which works fine on my Intel Mac over Monterey 12.4.
BIAB 2023 Audiophile, 24/60 Core M2 Mac Studio Ultra/8TB/192GB Sonoma, M1 MBAir, 2012 MBP
Digital Performer 11, LogicPro
Finale27.4, Dorico5, Encore5, SmartScorePro64, Notion6, Overture5
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The weakest links in the chain with floating-point like this are the pre-amps+ADC at the front and the DAC+PA+Speakers at the back, and the article glosses over that a bit.. Both of those still need attention to getting the levels sensible and getting the signal into its initial digital format (invariably, I think, non-floating). A quick look at Cirrus and Burr Brown ADCs shows none better that 127dB SNR. Even achieving that with a mic pre-amp as well is a real challenge (e.g., Focusrite show ~110dB dynamic range on their 18i20). It's possible to do better if money is no object. So whether integer, fixed point, or floating point, one definitely still needs to set the front-end gain/level sensibly. Once into the digital domain and converted to floating point, one can thenceforth essentially ignore signal levels and gain, right up to the point where she signal goes back to non-floating-point, or to analogue at the amplifiers/speakers. The significant-digits part of 32bit floating-point is 24-bits, so exactly the same resolution/precision as 24-bit integer data, but it adds also an 8-bit exponent (10^+/-127), which is huge for audio purposes. If one changes the level many times, by arbitrary amounts, the signal will eventually degrade, due to rounding errors in the calculations, but in practice it's by such a small amount that it's likely irrelevant. I'm not confident of the maths here, but I suspect that would be around 32,000 manipulations to degrade the signal to the equivalent of 16-bit CD quality. I can live with that 
Jazz relative beginner, starting at a much older age than was helpful.
Kawai MP6, Korg M50, Ui24R, Saffire Pro 40.
AVL:MXE Linux; Windows 11; Win8.1: Scarletts
BIAB2022 UltraPAK, Reaper, a bunch of stuff.
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The weakest links in the chain with floating-point like this are the pre-amps+ADC at the front and the DAC+PA+Speakers at the back, and the article glosses over that a bit.. Both of those still need attention to getting the levels sensible and getting the signal into its initial digital format (invariably, I think, non-floating). A quick look at Cirrus and Burr Brown ADCs shows none better that 127dB SNR. Even achieving that with a mic pre-amp as well is a real challenge (e.g., Focusrite show ~110dB dynamic range on their 18i20). It's possible to do better if money is no object. So whether integer, fixed point, or floating point, one definitely still needs to set the front-end gain/level sensibly. Once into the digital domain and converted to floating point, one can thenceforth essentially ignore signal levels and gain, right up to the point where she signal goes back to non-floating-point, or to analogue at the amplifiers/speakers. The significant-digits part of 32bit floating-point is 24-bits, so exactly the same resolution/precision as 24-bit integer data, but it adds also an 8-bit exponent (10^+/-127), which is huge for audio purposes. If one changes the level many times, by arbitrary amounts, the signal will eventually degrade, due to rounding errors in the calculations, but in practice it's by such a small amount that it's likely irrelevant. I'm not confident of the maths here, but I suspect that would be around 32,000 manipulations to degrade the signal to the equivalent of 16-bit CD quality. I can live with that I'm expecting the next generation of inexpensive interfaces to use a chip that supports a 384MHz sampling rate with 32bit float. Although such units exist already (some with 64 bit float), they're pretty expensive. When that happens, just watch: 32 bit float will be sold to us as the only thing we can use. As you've pointed out, it depends. The current inexpensive chip is 192/24 (aka Gen 3 since so many are using it in their 3rd generation units) and found in 3 dozen or more current interfaces. The eight I/O version with two 'phone amps costs $6.73 ea with a minimum quantity of 1,000. The 4th Gen doesn't have to be that inexpensive—if it can be delivered under $20 per, that should suffice. Going 32 bit float in my DAW increased my per-track headroom noticeably but I'm putting together projects with 50-100+ tracks of vocals recorded in home studios. I need all the help I can get. Not doing 64 bit float, however, though I could—even Logic Pro X supports that. Certainly don't need anything greater than 24 bit bwav when I bounce.
BIAB 2023 Audiophile, 24/60 Core M2 Mac Studio Ultra/8TB/192GB Sonoma, M1 MBAir, 2012 MBP
Digital Performer 11, LogicPro
Finale27.4, Dorico5, Encore5, SmartScorePro64, Notion6, Overture5
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I'm presuming the 384MHz was a typo for 384kHz.
$20 I presume is the BoM cost of the ADC and is probably OK ... $100 of the list price?
A question, though, has to be whether we really need it. 384kHz sampling rate is good, though the data rate doesn't really need to be that high; but 64bit is probably totally pointless. I think that's a resolution way beyond what physics will allow. It's a huge amount of data and for what purpose?
Jazz relative beginner, starting at a much older age than was helpful.
Kawai MP6, Korg M50, Ui24R, Saffire Pro 40.
AVL:MXE Linux; Windows 11; Win8.1: Scarletts
BIAB2022 UltraPAK, Reaper, a bunch of stuff.
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I'm presuming the 384MHz was a typo for 384kHz.
$20 I presume is the BoM cost of the ADC and is probably OK ... $100 of the list price?
A question, though, has to be whether we really need it. 384kHz sampling rate is good, though the data rate doesn't really need to be that high; but 64bit is probably totally pointless. I think that's a resolution way beyond what physics will allow. It's a huge amount of data and for what purpose? Thee is no list price when the minimum quantity of the chip is 1,000. Based on the current $6.73 of the one being used now, I expect the initial run to be under $20 when introduced and $10–12 within a year. As for 64 bit float being excessive, I’ll let you argue with ProTools users who think it’s important. I stay away from those conversations.
BIAB 2023 Audiophile, 24/60 Core M2 Mac Studio Ultra/8TB/192GB Sonoma, M1 MBAir, 2012 MBP
Digital Performer 11, LogicPro
Finale27.4, Dorico5, Encore5, SmartScorePro64, Notion6, Overture5
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There is no list price when the minimum quantity of the chip is 1,000. Maybe a slightly crossed line ... I meant a $20 chip means around $100-worth of the final product's list price. ... 64 bit float ... I stay away from those conversations. Very sensible. I think it's in the same territory directional cables. :-)
Jazz relative beginner, starting at a much older age than was helpful.
Kawai MP6, Korg M50, Ui24R, Saffire Pro 40.
AVL:MXE Linux; Windows 11; Win8.1: Scarletts
BIAB2022 UltraPAK, Reaper, a bunch of stuff.
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I just started using a zoom f3 for some live recording. I captures in 32bit float.
Kind of weird not setting any volume levels.
I've only recorded some test stuff at the house here but it seems to work.
It also looks like I need the latest version of Logic 10.7.5.1 to support it natively
Could be very exciting for live recording.
biab2023(Mac)
Logic Pro X
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Kind of weird not setting any volume levels. I wondered how they were able to do that as the pre-amps themselves presumably are not floating point (that seems feasible but unlikely). It has a pretty good noise floor and good dynamic range, which would cover most circumstances OK, but not all circumstances. Do they soft-switch a pad in the preamp? Then I saw the clue ... line input has an option of phantom power. OK, so it actually has a manually applied pad loosely disguised as line-input. Both inputs are described as having input impedance "3k or greater", which tends to support that ... line input is more conventionally 10k. Neat thinking on their part. It isn't so long ago that 127dB was fairly exceptional. It's becoming common. I wonder if it switches in that line-level pad for itself if it sees an overload.
Jazz relative beginner, starting at a much older age than was helpful.
Kawai MP6, Korg M50, Ui24R, Saffire Pro 40.
AVL:MXE Linux; Windows 11; Win8.1: Scarletts
BIAB2022 UltraPAK, Reaper, a bunch of stuff.
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Just out of interest my sound card is the sound blaster AE-9. The AE-9 info states Ultimate 32-bit / 384 kHz PCI-e Sound Card and DAC with Xamp Discrete Headphone Bi-Amp and Audio Control Module.
Seems ok for my hearing aid needed ears
Keith
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