After reading "How Music Works" by David Byrne, it seem like FLAC is solving the wrong problem. No one needs over 4 billion discrete levels.
For an example, music artists/producers can't afford to lose information when they "bounce" a track. FLAC solves that. "People" don't need lossless, but for those that do, it's great.
You can't find 32 bit PCM music anywhere, even hifi DVD audio is 24bit/96khz.
The real reason is that there's no 32-bit floating point digital audio converters, so distributing the audio in a format which needs to be quantized to be played anyway has no purpose (outside of deceptive marketing, of course).
Even integer 32 bit makes no sense for a final format - the available noise floor is just beyond reasonable, 24 bit is more than enough. In practice, 16 bit is too.
If your input clips, you usually don't need to lower it too much to protect your ADC, and if you recording the sounds so quiet they end up below 24bit noise floor after this, well they probably didn't clip in the first place and you most probably already facing bigger issues with mic/preamp/ambient noise and dynamic range.
Also, you mention hilo can take 32 bit input. Means, the signal is already digitized, and we're talking about recording?
Yes you can (not that I think it’s useful): https://ototoy.jp/find/?q=32bit
Of course you need the right (exceptionally good!) preamps to even make a difference here, but when you are manipulating audio 32 bits and upwards (for summing) are the default (for a reason).
Having something to store those files with a little bit of lossless compression is a welcome addition
why?
https://www.mojo-audio.com/blog/the-24bit-delusion/
32bit or more makes sense for dsp and summing, and non-musical data storage. But your I/O is most likely not giving 32 bits
The big benefit from 32bit float is you can go "above" 0db, so your digital operations can't "clip" in the analog sense. I definitely see the benefits of 32 bit float hardware, especially for field recordings (zoom aren't lying, it does work). But it's a different format and the internals are relying on the integer based format and don't exceed 24 bit integers.
32 bits are less critical to set the gain for (more headroom) and audio files are tiny anyways (compared to say, video). So in a practical sense I need to be less precise with setting the gain, or can record a bit more dynamic events. I am very happy with the results.
Sounddevices has a good explaination if you are curious: https://www.sounddevices.com/32-bit-float-files-explained/
There's probably no point since your audio interface doesn't produce 32 bit audio anyway.
For transient formats it's another story: if a bit of audio gets re-printed multiple times with some processing between renders, then it's beneficial to store it in a DAW-native format (32/64 fp) to avoid quantization or multiple additions of dither noise. For recording you just waste space.
> Having something to store those files with a little bit of lossless compression is a welcome addition
You'd probably want to store 32/64 bit floating point, and WavPack works really well. On comparable formats it achieves ratios really on par with Flac.
In the practical sense what this means is that I have to be less careful with setting the gain. Typically when setting recording gain you want to go as high as possible so preamp/mic noise doesn't become an issue, but not so high that you are going into the limiter when something unexpectedly loud happens.
I often record in environments where unexpectedly loud things happen, so having a recorder that allows me to record at a lower level while still not gaining noise issues or loosing detail in the silent parts is a very welcome thing.
In my experience as a mix engineer working with music, recordings, when made proper, rarely use the full available dynamic range of 24 bits. Often I get tracks recorded too low due to incorrect gain staging. But maybe even more common is just a noisy recording. Poor mains wiring, the length of a recording chain, self-noise of various gear used all contribute to what usually amounts to noise levels much higher than -144dB. And, to be honest, in my line of work it rarely matters until it crosses the 16bit noise floor of -96dB.
What I suppose I mean by all of this and my previous posts is if you're not careful enough to fit into the higher portion of the available dynamic range of 144dB you're just adding more noise regardless of the format used, as in most cases the level of cumulative noise is higher than -144dB. YMMV, of course.
Since real-world DACs don't have infinite taps, either increasing the number of samples per second in the original audio or the number of steps done by the filter will improve how close it gets to the original waveform.
Would that apply to the number of bits used to represent the level as well? I thought that was mainly useful to give some additional headroom when editing?
(Obviously there's diminishing returns either way)
The bit depth, OTOH, is your signal-noise ratio (noise floor) when sampling/quantizing said audio. It's unrelated to Nyquist-Shannon. More bits gives you less noise when reproducing it. For CD audio, 16 bits was chosen to give an acceptable SNR of 96 dB.
Monty Montgomery (of xipf.org) has a nice YouTube video related to this: https://www.youtube.com/watch?v=cIQ9IXSUzuM
That's what I'm saying. Increase the number of samples in the audio itself, and the DAC can get closer to the original.
Actual audio equipment has a noise floor. The encoding depth you use also has a noise floor. If the noise floor of the encoding is far below the noise floor of the signal, then it won't be perceptible.
So you choose to put your noise floor at some amount below the existing noise floor. Not infinitely below, because that would require infinite bits. This is the reasoning you'd use for 24-bit audio, which has a very comfortable noise floor of -144 dB, which leaves a large margin even for extreme low-noise professional equipment (you might see -120 dB ish for extremely good equipment).
At 24-bit, even passive components like transformers and resistors are contributing measurable amounts of noise.
The reason you might pick 32-bit in practice is so that you can have lots of headroom for some DSP algorithm, or do lots of sums of different signals without accumulating quantization error. The final "archive" file will still have worse than 24-bit precision.