It's the same rationale for doing calculations in full 80-bit or 64-bit FP even if you don't need the full bits of precision -- if you start your calculations at your target precision, your result will have insufficient precision due to truncation, roundoff, overflow, underflow, clipping, etc. in intermediate calculations.
In theory, if you knew what your editing pipeline is, one could mathematically derive the starting headroom required to end up with acceptable error, but in practice that's probably a very risky proposition because 1) you don't know every intermediate calculation going on in every piece of software and 2) errors grow in highly non-linear ways, so even a small change in your pipeline might cause a large change in required headroom [1].
[1] https://en.wikipedia.org/wiki/Numerical_analysis#Generation_...
Also, you don't cause errors when you compress and decompress using a lossless compression format (as the name sort of implies).
That's not how it works. You can totally hear clipping or noise even as an amateur. Just as an amateur photographer first thing first learns to look out for overlit skies and hilights that aren't really recoverable at all unless they perhaps have access to the camera's RAW files. But unlike in photography, in sound work you typically need to process as well as sum multiple recordings, all while not blowing out the peaks that might just happen to line up.
If anything, dealing with the large dynamic range of 24-bit is easier for the amateur. An experienced pro would probably have a good bunch of tricks in his bag should he really have to mix music in 16 bits, enough to produce something that doesn't sound horrible. The amateur would be more likely to struggle.
In practice, fixed-point or floating-point is better depends on the operation you plan to use and whether you have a good idea whether you'll be able to stay in a fixed range ahead of time [1].
[1] The loss of 'footroom bits' corresponds to the extra resolution lost whenever the FP mantissa increases - http://www.tvtechnology.com/audio/0014/fixedpoint-vs-floatin...
For mixed and mastered music, the ~120dB perceived dynamic range of properly dithered 16-bit audio is more than sufficient. If you're listening at sufficient volume for the dither noise floor to be higher than the ambient noise floor of the room, you'll deafen yourself in minutes.
For production use, the extra dynamic range of 24 bit recording is invaluable. You're dealing with unpredictable signal levels and often applying large amounts of gain, so you'll run into the limits of 16-bit recording fairly often. In a professional context, noise is unacceptable and clipping is completely disastrous. Most DAW software mixes signals at 64 bits - the computational overhead is marginal, it minimises rounding errors and it frees the operator from having to worry about gain staging.
You can probably get away with 16 bit recordings for a sample library, but it's completely unnecessary. Modern sampling engines use direct-from-disk streaming, so only a tiny fraction of each sample is stored in RAM to account for disk latency. The software used by OP (Hauptwerk) is inexplicably inefficient, because it loads all samples into RAM when an instrument is loaded.
Digital samples can and will generate intermodulation distortion, quantization and other audio artifacts when mixed. Using 24-bit lessens or eliminates the effect versus 16-bit.
Experiments and results on this software are here: http://www.sonusparadisi.cz/en/blog/do-not-load-in-16-bit/
While people here like to quote that infamous looks-like-science-but-isn't Xiph video, the reality is that a lot of professional engineers absolutely can hear the difference.
If you have good studio monitors and you know what to listen for the difference between a 24-bit master and a downsampled and dithered 16-bit CD master is very obvious indeed, and there are peer reviewed papers that explain why.
Dither was developed precisely because naively interpolated or truncated 16-bit audio is audibly inferior to a 24-bit source.
Many people certainly can hear the effect of dither on half-decent hardware, even though it's usually only applied to the LSB of 16-bit data. From a naive understanding of audio a single bit difference should be completely inaudible, because it's really, really quiet.
From a more informed view it isn't hard to hear at all - and there are good technical reasons for that.
For synthesis, you don't want dither. You want as much bit depth as possible so you can choose between dynamic range and low-level detail. So 24-bit data is the industry standard for orchestral samples.
Dither was developed because it's technically better. Audibly inferior though? Perhaps, with utterly exceptional source material (e.g. test tones) and when the output gain is high enough for peaks levels to be uncomfortably loud.
In reality, many recording studios have a higher ambient noise level than the dither, making it redundant in practice — the lower bits are already noise, so audible quantisation artefacts weren't going to happen anyway. That said, dithering is pretty much never worse than not dithering, and almost all tools offer it, so everyone does it.
24 bits is important because it gives the recording engineer ample headroom, and it gives the mixing and mastering engineers confidence that every last significant bit caught by the microphone will survive numerous transformation stages intact. Once the final master is decimated to 16 bits per sample, you know that your 16 bits will be as good as they could have been.
Have you actually heard the difference between simple dither, noise-shaped dither, and non-dithered 16-bit audio? A test tone is the worst possible way to hear what they do.
24-bit audio is used because you want as clean a source as possible.
This also applies to mastering for output at the other end. With the exception of CD and MP3, most audio is delivered as 24-bit WAV at either 44.1, 48, 88, or 96k.
Even vinyl is usually cut from a 24-bit master. Here's a nice overview of what mastering engineers deliver in practice:
https://theproaudiofiles.com/audio-mastering-format-and-deli...
Dither is noise. Well chosen noise, very quiet noise, but noise nonetheless. Whether the signal is noisy for one reason or another, the consequences at the point of decimation/quantization are the same. Either way the least significant bits are filled with stochastic values and the desired signal isn't plagued with quantization noise artifacts.
> Have you actually heard the difference between simple dither, noise-shaped dither, and non-dithered 16-bit audio? A test tone is the worst possible way to hear what they do.
...is the sort of thing someone who hasn't done a blinded listening test would say. Stop assuming the commercially successful "experts" are also technical experts, because few are. I doubt more than a tiny fraction could describe what a least significant digit is.
(Cutting vinyl, a hilariously lossy process that requires compression and EQ to avoid the needle jumping the groove, doesn't need a 24 bit master. Barely needs a 14 bit master. But since an extra hundred megabytes of really accurate noise floor doesn't make anything worse, they do it anyway.)
Their words mean very little.
I totally get going for 24 bit over 16 for the noise floor, though.
Such audible differences constitute a testable claim. So far, the number of claims that have withstood reasonable test conditions is zero.
Nobody is arguing that 16 bits is enough during the recording and mixing process. If they are claiming that 16 bits is insufficient for consumer distributed music, they need to go back to remedial science class.
I'm all in favor of keeping sample sizes better.. but raising the price for everyone by 10-20%, so that less than 1% can take advantage of it is a bit ridiculous.
Do you really need this much headroom, e.g. for producing music professionally? (If so, why are you arguing over the $50-100 extra for a Xeon?)
Because for listening, it's proven fact that even professional musicians and mixing engineers using equipment costing >$100k can't tell 24 bit from 16 bit in a proper double-blind A/B test. Presumably you use a high sample rate as well (maybe 192 kHz), which is equally useless. Drop both of those, and you're within 64GB easily with no noticable SQ loss.