7 ms·

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 re

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.

PaulDavisThe1st 3mo ago
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.
- nullc 3mo 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)
  - PaulDavisThe1st 3mo ago
    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]
    - nok22kon 3mo ago
      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
      - PaulDavisThe1st 3mo ago
        > 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).
        nok22kon 3mo ago
- geraldmcboing 3mo ago
  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/ 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.
  - PaulDavisThe1st 3mo ago
    > 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]
    - 3mo ago
- geraldmcboing 3mo ago
  "Nobody uses 32 bit float for recording" - you are just displaying total ignorance here.
  - PaulDavisThe1st 3mo ago
    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.
    - adrian_b 3mo ago
      While the best that ADCs can provide is linear 24-bit audio samples, the following audio processing is better done after converting the samples to FP32, and keeping this format until the final 16-bit encoded audio suitable for listening is generated. For the same reason, video processing is preferably done on FP16 samples of the color components even if both the input ADCs and the output video signal may use only 10-bit or 12-bit per sample, at most. Moreover, most high-resolution audio ADCs do not really sample the input audio at a 24-bit resolution, but they use only a sigma-delta method where the actual samples have only a few bits, possibly only even 1 bit. Then DSP techniques are used to convert the audio stream with a high sampling frequency and a low resolution per sample into an audio stream with a low sampling frequency and a high resolution per sample, which is the external output of the ADC. If you had access to the raw audio bit stream as actually captured by the ADC, you could modify the decimation algorithm to really output FP32 samples, though no existent ADC could actually have a so high dynamic range (except if the output bandwidth would be reduced a lot, to filter the input noise).
      - Archit3ch 3mo ago
        > the following audio processing is better done after converting the samples to FP32 Or, in some cases, FP64.
- The_President 3mo ago
  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.

PaulDavisThe1st 3mo ago

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.

nullc 3mo 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)
- PaulDavisThe1st 3mo ago
  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]
  - nok22kon 3mo ago
    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
    - PaulDavisThe1st 3mo ago
      > 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).
      - nok22kon 3mo ago
geraldmcboing 3mo ago
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/ 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.
- PaulDavisThe1st 3mo ago
  > 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]
  - 3mo ago
geraldmcboing 3mo ago
"Nobody uses 32 bit float for recording" - you are just displaying total ignorance here.
- PaulDavisThe1st 3mo ago
  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.
  - adrian_b 3mo ago
    While the best that ADCs can provide is linear 24-bit audio samples, the following audio processing is better done after converting the samples to FP32, and keeping this format until the final 16-bit encoded audio suitable for listening is generated. For the same reason, video processing is preferably done on FP16 samples of the color components even if both the input ADCs and the output video signal may use only 10-bit or 12-bit per sample, at most. Moreover, most high-resolution audio ADCs do not really sample the input audio at a 24-bit resolution, but they use only a sigma-delta method where the actual samples have only a few bits, possibly only even 1 bit. Then DSP techniques are used to convert the audio stream with a high sampling frequency and a low resolution per sample into an audio stream with a low sampling frequency and a high resolution per sample, which is the external output of the ADC. If you had access to the raw audio bit stream as actually captured by the ADC, you could modify the decimation algorithm to really output FP32 samples, though no existent ADC could actually have a so high dynamic range (except if the output bandwidth would be reduced a lot, to filter the input noise).
    - Archit3ch 3mo ago
      > the following audio processing is better done after converting the samples to FP32 Or, in some cases, FP64.
The_President 3mo ago
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.