I'd like to point out the potentially non-obvious here. This is a test of delivery format.
There still are benefits to using 24-bit audio in the recording and processing stages. This is in large part due to most recording systems expecting 0dbVU = -18dbFS and the subsequent processing that can bring the noise floor well into the audible range (dynamic range processors are notoriously effective at this, and heavily used in modern music). We could take a simple example of a snare drum being recorded at 16bit, being EQ'd with a +6db boost anywhere on the spectrum, then compressed with a reduction peaking around 10db (not uncommon). After brickwall limiting in the final mix, this track will easily have a noise floor (via quantization error only) > -68dbFS best case. (-96db starting. -12dbFs peaking snare, +6 + 10 to noise floor, assuming limiting with no gain reduction) -68dbFS is already audible in a critical listening scenario. With dozens (sometimes hundreds) of uncorrelated signals being subjected to similar processing, this noise floor raises well into the audible range for even a modest playback system.
While I realize that delivery format is the only thing important to most people, it is important to differentiate since the article does make a point to separate out musicians, sound engineers and hardware reviewers. These are groups of people that _should_ be aware of the benefits of higher sample resolution. Since it's fairly obvious that most people in these categories are confused about their ability to discern delivery formats, it's not beneficial to confuse them even further about working formats.
To be more succinct, the difference between 16bit and 24bit is largely inaudible when the source material is worked in a higher resolution format and properly converted.
> These are groups of people that _should_ be aware of the benefits of higher sample resolution.
I don't think the author is necessarily dismissing the idea of high fidelity audio; especially for the reasons you point out. Rather, the author is claiming that if you're simply _listening_ to the playback, it won't make a bit of difference, regardless of how your ears are trained.
Edit: Also note that several of the respondents were really confident that they heard a difference. This small study demonstrates that their confidence was misplaced. This, I believe, is what the author is trying to drive home.
> There still are benefits to using 24-bit audio in the recording and processing stages.
An analogy of what you said:
It's similar to HDR for audio[1] (but not exactly like it). HDR can be used for photography that, once composed and edited, will present more realistic information to our eyes. For example, with HDR you wouldn't have an overexposed sky - however the HDR is only used in order to get to that final 16 bit image (and even with 16 bit your eyes have a hard time discerning different colors).
The same applies to audio. Listening to 24-bit is pointless, however, if you are editing something you want to retain as much information as possible until the final render so that you don't run into clamping issues as you described.
Therefore, sites that provide 192/24 downloads are valuable. If I'm a DJ getting music for my gig I do want those production quality files, as I cross-fade between two songs I don't want artefacts popping (excuse the pun) up.
On to my own opinion: 24-bit is still not good enough. DAWs should be working in floats. Audio needs to go true HDR, 24-bit is a cop-out. Why would you even used a 24-bit int when floats are there and ready to go? Imagery went floating point, what, 10 years ago? Why can't audio catch up? Being able to exceed the clip in my DAW in a channel, and then wrangle it back down in another would be awesome.
Unrelated: That Xiph video really amazes in terms of what nature does. We rarely care about it (we do in terms of e.g. intercontinental fibre cables), but nature does all of this when we send a signal to a speaker. Even normal sound does actually have a band limit and does behave (albeit, far higher dynamic range) exactly the same way automatically. Shoot a signal down a fibre cable that can't handle it, and you'll get Nyquist. Too high frequency for RTP air? Expect distortion (that we can't hear). You don't even have to include electronics to get nature to impose these limitations for you, you have to do no extra work. Completely amazing - a deeper level of logic that is mind boggling.
192khz however is not beneficial to the processing though. There is an argument that can be made for 96khz in a limited set of cases of processing as a form of implicit pre-process upsampling, but itt can actually be detrimental. (see for instance: https://www.gearslutz.com/board/mastering-forum/968641-some-...)
Since this is a discussion about bit-depth, I don't see much of a reason to clutter it with a discussion about sample rate. This subject is already difficult enough for most people to understand it seems.
I might add that it's factually impossible to determine what it means, from the listener's perspective.
As far as "limited cases", I work at an ISV, so I am preconditioned to not accepting limited cases - as demonstrated by excessive overtime just this very week. Our customers do some really crazy shit with our software.
Typical DAW already run in floating point, as do most digital mixing consoles. It's only input/output to physical audio cards that's quantized to 24bit.
If you want to "render", i.e. produce a final .wav file from your input tracks, with applied effects, filters, gain-changes,... most DAW let you output a .wav-file with floating point data (not that it would make sense).
> there are people already discussing the merits of 64bit floating point over 32bit floating point
In my very humble opinion (haven't had the extremely fortunate fortune to be formally educated on DAC and in general the math involved there), I think that discussion is greatly valuable. When you are compounding information into information you can never become accurate enough. Error accumulates - that is the woe if digital composition.
I would love to accelerate all this stuff on a GPU - given workable knowledge on how to specifically turn all those crazy mathematical formulas into code; which given only a rip-off degree is practically impossible.
Even if you add 1 unit of interference at each processing stage (and since rounding tends not to be malicious, you may well do better than that), you'd need 128 poorly-implemented processing stages for a 32-bit float to be reduced to mere 16-bit integer precision - but in practice, likely more.
When it comes to clipping or loss of data on the lower end, well, 32-bit floats have an 8 bit exponent (254 reasonable values); that means that the loudest full-precision unclipped signal is 765 dB (!) louder than the softest un-quantized signal. Even with mediocre centering, that's more than enough.
I don't think 64-bit audio is likely to be noticable, even for processing purposes, outside of really specialist kind of niches.
Really, you don't understand how much of a marketing scheme my degree was. An absolute disgrace in terms of what education should be, I'm an idiot for falling for it.
I learnt convolution in my own time, so if you have resources (that don't include integration) I would absolutely love to read them. DSP is something that seems hard and I'd love to get into it.
Every major DAW already does this- 32 bit float is standard for internal processing, and quite a few of them support import/export of 32 bit files as well. Some even work at 64 bit precision internally (Studio One and Reaper both come to mind).
Sadly, I think that mine (FL Studio) doesn't, or at least it doesn't between channels when mixing into channels. It definitely goes above clip, but horrible things are be to expected when mixing it with other channels.
I assumed it was the norm as FL Studio is gaining immense amounts of traction with the more recent updates to advanced competence. I might have been wrong in that assertion.
Either way, my DAW does not represent it in a way that makes sense to a software engineer.
I would certainly love to actually work in HDR, as opposed to it being a ghost in the machine.
I would guess it's because even really good DACs have trouble reaching noise as low as 24 bits, so capture is in 24 bits. I think most DAWs support floats as an intermediate format though.
That's true. But I understood this as talking about playback, not about mastering. The article linked below makes this distinction more clearly: https://news.ycombinator.com/item?id=8727591
There still are benefits to using 24-bit audio in the recording and processing stages. This is in large part due to most recording systems expecting 0dbVU = -18dbFS and the subsequent processing that can bring the noise floor well into the audible range (dynamic range processors are notoriously effective at this, and heavily used in modern music). We could take a simple example of a snare drum being recorded at 16bit, being EQ'd with a +6db boost anywhere on the spectrum, then compressed with a reduction peaking around 10db (not uncommon). After brickwall limiting in the final mix, this track will easily have a noise floor (via quantization error only) > -68dbFS best case. (-96db starting. -12dbFs peaking snare, +6 + 10 to noise floor, assuming limiting with no gain reduction) -68dbFS is already audible in a critical listening scenario. With dozens (sometimes hundreds) of uncorrelated signals being subjected to similar processing, this noise floor raises well into the audible range for even a modest playback system.
While I realize that delivery format is the only thing important to most people, it is important to differentiate since the article does make a point to separate out musicians, sound engineers and hardware reviewers. These are groups of people that _should_ be aware of the benefits of higher sample resolution. Since it's fairly obvious that most people in these categories are confused about their ability to discern delivery formats, it's not beneficial to confuse them even further about working formats.
To be more succinct, the difference between 16bit and 24bit is largely inaudible when the source material is worked in a higher resolution format and properly converted.