Comment by nullc
4 days ago
> 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]
The ADCs that do direct sampling of the input signal (i.e. by successive approximation or by the pipelined algorithm) become very expensive at high resolutions and they are limited to 18 bits per sample or at most 20 bits per sample.
Due to their high cost such ADCs have no longer been used in audio for many decades. They may still be encountered in some expensive measurement instruments that need high resolutions at significantly higher sampling frequencies than needed for audio.
All audio ADCs have a very low resolution per sample, e.g. 4 bits or even lower, but they sample at a very high frequency, of many MHz. Then the bit stream is digitally processed to generate whatever format is desired for output, at a lower sampling frequency and a higher resolution, e.g. 24 bits @ 192 kHz.
There is a difference between the actual resolution at the output and the effective resolution, which is limited by noise, e.g. the 24 bit samples may have an effective resolution of 20 bits or 21 bits or 23 bits, etc., i.e. they contain noise with an amplitude corresponding to those effective resolutions.
The digital algorithm that converts the low resolution input samples (e.g. 4 bits @ 5 MHz) inside the ADC can easily be modified to generate a different numeric output format, e.g. FP32.
Neither FP32 nor 24-bit is the native format of the A/D conversion. If the ADC outputs FP32, that is even more convenient for further audio processing. Obviously, the quality of the ADC is independent of whether it outputs FP32, and the FP32 samples will have a different effective resolution on each ADC, which seldom would be as high as 24 bits, due to the noise.
> There are precisely zero 32 bit ADCs in existence.
> There are ADCs that will provide 32 bits per sample but that's entirely different.
Now that requires elaboration.
There is e.g. AD's LTC2500 (https://www.analog.com/en/products/ltc2500-32.html). Not meant for audio (too slow at 32b) and not noise free, but it's a bona-fide 32b ADC.
Now there might be no ADC which provides 32b wide noise-free samples at sample rates needed for audio and given the absurdly low level of a LSB signal that might be as infeasible as it would be pointless, but that's a bit of a different statement.
I didn't say anything about DACs! I'm correcting a specific claim you made
> 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);
This is not true and not true for a good and important reason! One which has no bearing on the kind of DACs that exist.
Modern field recorders allow gains set a 'reasonable' level that maximizes SNR for recordings but still won't clip when there are much louder peaks. Not so dissimilar to how a 6-digit multimeter can achieve its advertised performance both on a 0-5v range and a 0-300v range but cannot give more than 6 digits at the higher range.
When I said "nobody uses 32 bit float for recording", I am referring to the result of the DA process that generates samples values used by a recorder.
Obviously, everyone and their mother uses 32 bit float as an internal sample format because of its fitness for purpose (except the folks who think they need 64 or 80 bit floating point, of course). But they are not using "32 bit floating point samples" - the samples come from an (at best) 18-22 bit integer conversion.
Rode NT1-A 5th gen microphone claims 32-bit float output, insisting it will not clip peaks
so maybe they do sample at 24 bit at a well chosen gain level and then convert to 32 bit float, with the max 24 bit value being above 1.0 float
or as GP said, use two separate ADCs at two different gains and combine their output
> Rode NT1-A 5th gen microphone claims 32-bit float output, insisting it will not clip peaks
Of course it does! And that's what it does, of course. But that has absolutely nothing to do with the AD process itself, which is chip-limited to 24 bits and likely physics-limited to somewhat less than that.
You can't beat the physical limit of a DA circuit by doubling them up at different gains.
And .. you don't want to. Going beyond 22 bits gets you into brownian noise pretty quickly, which is completely pointless.
The best you can do (or could do) is get a very, very, very good DA that can really do 22 bits (likely not commercially available because of the expense), and then get the samples from it in whatever format works best for your purpose (24 bit integer, some fixed point value, or 32 bit floating point).
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> use two separate ADCs at two different gains and combine their output
That's what could be done if ADCs were perfectly linear and noise free and limited only by their bit-width. Sadly, they are not. The non-linearity one can in theory measure and correct for, but the noise can be corrected for only by oversampling. And then you might as well use a single ADC of lesser bit width and higher sampling rate.
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Why are you obsessed with DAC? Its the ADC that is WHY we capture 32/192.
If I said DAC, it was a mistyping. I am (in this context) always talking about the ADC.