I'm curious if sites like Qobuz will suddenly replace some of their existing files with higher-bit copies if they already had them and just had no way to encode them as FLAC.
The kind of equipment I'd expect to see in order to record that kind of range is where you'd have something like multiple microphones, and switch back and forth between them based on the level of the material you're recording (which would have to be something like actual gunfire or explosions).
I don't know what kind of equipment you'd need to reproduce this dynamic range, but I don't think it's audio equipment.
My guess is that the main real application of this is going to be compression of non-audio data that happens to compress well with FLAC.
It's not that hard to beat 16-bit, which is 96 dB, using easy-to-find, off-the-shelf equipment. One example scenario is that you are recording something but you don't have a precise idea of how loud it will be ahead of time, so you record at low levels and rely on 24-bit capture to give you headroom above and noise floor below. Trying to capture at 16-bit can, in practice, be annoying and difficult because it is more likely that you will ruin takes by setting the gain wrong.
However, there is a legitimate purpose behind having higher dynamic range for production purposes and sample sources. There are some recording sources that can actually produce 32-bit audio. Plus, you might want to do some processing on the sound that would end up affecting the dynamic range, or otherwise benefit from the increased resolution. One example is nonlinear processing that generates new musical information from the original signal - you can of course just reduce the gain after processing, but you are then sacrificing some of the resolution of the new combined signal, which itself could otherwise be used by further downstream processes. This all happens post-recording, but can still be musically important before getting to the finished product.
This is why DAWs work in 32-bit or 64-bit processing internally, and why many high-quality sample libraries will come in 32-bit, especially smaller one-shots. I often convert samples to .flac for space reasons, and have to either skip 32-bit .wavs or downsample them to 24.
This is the justification for 24-bit audio... is there a reason why 24 bits is not enough here?
If you're capturing audio sources directly, you'd use something like a 24-bit ADC, which you can find easily enough. The "raw" output of the ADC is 24 bits.
If you're doing intermediate processing in your DAW, then the DAW is using single-precision floats (or possibly double), which cannot be losslessly converted either to 32 bit or 24 bit integers, so how would you choose the right format to store? It seems to me that you'd either store the original floating-point data, or you'd perform some kind of lossy conversion to a high-quality archival format... but if you do that, isn't 24 bits good enough? You're quantizing either way, and at 24 bits, you can have plenty of headroom and noise floor at the same time. Loads, even.
Also, once inside the digital world, there are many processes you can perform that add new musical information to the original sound that might be higher in gain but that you want to preserve for downstream processing until you're ready to actually "print" and quantize the final product, at which point, yes, 24-bit will be more than enough.
Physically, you are talking about signals measured in the tens of nano-volts (or nano-amps, depending on the mechanism of the microphone).
Also, it's less about the absolute resolution, and more about the ability to boost the gain, often by a lot, while still having a wide and useful dynamic range after the fact.
The stacked ADC approach does allow for a wider overall dynamic range, but I would be skeptical of the physical transducers capabilities in using it all, but the 32-but float format here appears to be largely about lossless transfer to a DAW such that further processing is lossless, as we are not converting from int to float and back.
This format could then be said to be useful as a master media format.
The article in the OP is about a flac containing a 32-bit int audio stream, however, which appears less useful and not at all related to 32-bit float
Some of your ADC/DAC chain can reasonably claim 125dB range. Some amplifiers can claim 17 or even 18 bits above their noise floor - 108dB.
No full-spectrum microphones, headphones or speakers can reasonably claim 125dB without distortion, but if they did you would still want to limit your exposure to "never". Long-term damage begins with long-term exposure under 96dB.
My thought is that it would be very hard to get that 130 dB range all the way from an audio source to your ADC, and it would be very hard to get it back out all the way to speakers again.
Neumann makes no such claims. They do claim max spl of 144 dB. But the TLM and is 10 dB less and this is prior to ADC of variable ability. Distortion isn't a ceteris paribus value for signal to noise calculations.
> Due to its enormous dynamic range of 132 dB [...]
Not really interested in dissecting what this means, or how to reconcile it with the SNR numbers. Just saying that Neumann does, in fact, make this claim.
Not sure what point you're making about distortion.
https://en-de.neumann.com/d-01
156 dB max spl 86 dB snr 130 dB dynamic range
Neumann patented a dual ADC preamp to make this possible.
Additionally from the product page:
"Unprecedented fidelity and detail, 130 dB dynamic range"
I've just realized that the product page I linked is extremely unusual for stating a dynamic range value.
However this makes sense if you read the papers covering Neumann's dual ADC and pre design of which they were justifiably proud. System D was a mid nineties introduction.
In principle, every time you convert from the 32 bit float back to 24 bit integer, there's an opportunity for a careless human to screw up the scaling, and throw away some of the available integer range. Rinse and repeat until you have an audible problem.
So this forms the basis for an argument in favour of keeping the audio in a float representation for as much of the production process as can reasonably be achieved.
I don't see any benefit in 32 bit representation for delivery of the finished content, but I suppose that if, as part of the production process, you're transferring the audio online between different sites/people/whatever, then having lossless compression that works without having to convert back to integer might be useful.
EDIT: Just read the thing more carefully, and realised that it's specifically talking about 32 bit int, NOT float. So right now, I can't see much practical use for this.
As far as human hearing goes, 24-bit is already overkill. Even 16-bit should be an imperceptible downgrade for anything with a naturally low dynamic range (i.e. most music).