Comment by geraldmcboing
4 days ago
The OP is a bit off with their description of why pro audio engineers work in higher bit rates and sample rates. We use 24bit to preserve low level sounds eg reverb, breaths etc and use 32bit float when recording as the headroom is so massive clipping is not an issue (other than of course still neeing to avoid overloading microphones with max SPL - cleanly recorded distorted sound is still a fail). Unclipping 32bit float feels like voodoo - I did a test, recording fireworks & unclipping the 32bit float recordings.
I use microphones that can 'hear' up to 100kHz (Sanken CUX100K) and for film sound design playing 192kHz audio at half and quarter speed the results are very significant, and reveal there IS 'content' above human hearing. Irrelevant for general listening but very important for sound design.
Have you ever actually checked the number of actual bits your ADC can use? Most 24 bit converters struggle to get to 18 bits.
Nobody uses 32 bit float for recording (to do so is just to capture at least 10 bits of noise, most of that being brownian); its strictly a format for mixing and processing. You don't get any more resolution from 32 bit floating point than you do from 24 bit integer formats, but the result of "clipping" is less dramatic, hence the appeal of the format.
While there is some evidence that non-auditory human sensory perception may be sensitive to ultrasonic acoustic waves, it's pretty weak right now, and somewhat in the "woo" zone. It may turn out to be significant, or it may not. I wouldn't base an audio production workflow that requires 4x the cpu power and 4x the disk space on such tentative claims, but you're welcome to.
> Nobody uses 32 bit float for recording
Yes they do, almost all high end field recorders used for film work are 32-bits now and have been for much of the last decade, often with some fancy preamp integration so that there is no expertise required for gain staging the recording. (I believe the implementations use a second matched 24bit ADC with 48 dB less gain in front of it).
The result obviously doesn't have a noise floor which is lower (as the noise of a room temperature _resistor_ gets in the way of that even at the 24-bit level) but they have more dynamic range so that your recording isn't ruined by hard clipping some unexpected loud sound.
It's a big improvement for practical usage, and also likely does improve SNR somewhat because you can run higher gains without as much fear that you'll ruin the recording. The reason it would pay off is that the SNR loss you get from splitting the signal is easily smaller than the SNR loss you would get from gain reduction to avoid clipping.
(maybe... capsule self noise is also limiting... at these levels, and usually people aren't using microphones designed for the lowest possible self noise unless they're doing something special)
There are precisely zero 32 bit ADCs in existence.
There are ADCs that will provide 32 bits per sample but that's entirely different.
Current technology limits the bit depth to 18-22 bits and going beyond that you'd be very quickly recording brownian (atomic) noise anyway.
The point about 32 bit float is that it is a useful format for mixing, editing and general processing, so it is widely used in digital audio tools. But it is not a format that ADCs generate "natively" via their electronics - almost all of them are generate a 24 bit integer or fixed point value and then just supplying that as a 32 bit float value because the software asked for it (the software could have done it all by itself.
[EDITED: DAC->ADC since that is what I meant and what this is all about]
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Note to readers, this commentary brought to you by a code writer for Ardour. Unfortunately I was unable to view any of the about information for that program, because it crashed immediately.
"Nobody uses 32 bit float for recording" - you are just displaying total ignorance here.
My comment should have been more emphatic that: nobody uses AD converters that generate 32 bit floating point values natively when recording, or anywhere close to the resolution that format implies.
I am extremely aware that as a data format in DAWs and other recorders, 32 bit floating point is completely common.
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Dude I've been doing sound design on films using these techniques for years. There is zero 'woo' involved, it is ALL practical evidence based use. I've been using 32bit float multitrack field recorder by Sound Devices MixPre10-II professionally for many years now. The recorder has three preamps per mic input, each gain staged to provide optimum signal to the 32bit float AD. Read this to clarify your thinking: https://www.sounddevices.com/32-bit-float-files-explained/
Surely you understand a recording made at 48kHz has a max freq response of 24kHz and played at half speed that max freq is 12kHz and at quarter speed only 6kHz. You can very clearly hear the filter cut off due to Nyquist. Record at 192kHz with mics capable of 100kHz capture and when played at quarter speed, the sound is full spectrum because there is no truncated frequency response. And when I load a 192kHz recording to izotope RX I can literallu see the harmonics going up to 96kHz. (not with every sound of course)
I repeat, i am not talking about 'normal' listening. I am talking about an industruy you have no knowledge or lived experience with, so spare me the incorrect claims about what can & cant be heard.
> I am talking about an industruy you have no knowledge or lived experience with
I'm the original/lead developer of Ardour, a cross-platform DAW, and have been working with digital audio for more than 25 years.
There are no 32 bit ADCs - your SD MixPre's are giving you (at best) 22 bits packaged as a 32 bit float value. The preamps make absolutely zero difference to the AD conversion (though they might sound real nice).
> Surely you understand a recording made at 48kHz has a max freq response of 24kHz and played at half speed that max freq is 12kHz
This is a very naive version of what "played at half speed" might actually mean. If properly and correctly resampled, this is not true.
> And when I load a 192kHz recording to izotope RX I can literallu see the harmonics going up to 96kHz
Well, I'd certainly hope so! But the question is: what are the energy levels associated with the partials above Nyquist? If you recorded at 384kHz with sensitive enough equipment, you'd see partials above 96kHz - but at extremely low energies because ... well, that's just how physics works.
[EDITED to remove AD/DA confusion]
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