If your mastering is done competently, you really aren't going to be able to hear it in a realistic scenario. Which is why:

  "Talking about dynamic range completely misses the point."

isn't really sensible. It's the use of dynamic range that decides how much useful resolution you have when quantizing a signal. This is really why higher bit depths on record and mixing are useful - they let you be sloppier with the inputs without losing much information before you've had a chance to work with it. It still doesn't gain you anything fundamental but it does mean if you got the levels a bit wrong you can salvage it. Higher bit rates here are excellent.

There is nothing magic about 24 bits here. Record something with 48 bits but set up your equipment all screwy so your only actually using the first 8 bits... and you've got an effectively 8 bit recording.

In real world applications the codec is giving you trouble with the low amplitude stuff, not the quantizer. Not that in realistic situations your equipment is likely to be able to generate this cleanly anyway.

   "24-bit doesn't need tricks like dither - because it does the job properly in the first place."

No. Dither isn't a trick, it is a fundamental approach to quantization error at any depth.

On playback, the issue goes the other way around. If you've mastered things correctly you'll be using the available dynamic range of the output in such a way that the information content of your signal is well represented. This is sufficient at CD rates for all practical listening scenarios.

"This is why using more bits does not mean you can "reproduce music with a bigger dynamic range" - not without turning the volume up, anyway. What actually happens is that the maximum possible volume of a playback system stays the same, but quieter sounds are reproduced with more or less accuracy."

You are slightly confused. (It may help to remember that a decibel always refers to a ratio, so the setting of your volume knob is not important.) Greater bit depth does allow for greater dynamic range, this stems directly from the definition of dynamic range. 16-bit audio has a theoretical dynamic range of:

  10 * log10 (2^16)^2 ~ 96dB

24-bit audio has a theoretical dynamic range of:

  10 * log10 (2^24)^2 ~ 144dB

For reference, a quiet recording room has a noise floor of ~40dB SPL and the loudest amplified music rarely exceeds 115dB SPL. This gives a dynamic range of 75db, which indicates that a well-recorded 16-bit master should be more than adequate.

The idea that having bits in excess of this amount will somehow result in the perception of a smoother or more accurate sound is fallacious. Even at maximum playback volume, this information will exist well below the noise floor and will simply not be perceived. In fact, this information will likely exist well below the noise floor of the recording studio and thus, in some sense, will not even be recorded.

> No they're not. And no matter how many times this gets linked to on the Internet, it's still wrong.

The article is still correct, just like it always was.

Ironically, most of your analysis is also correct. Somewhere in your understanding though, you're leaping sideways to an incorrect conclusion.

>The basic problem: the quieter a sound or detail gets, the fewer bits of resolution are used to represent it.

So far so good, but you're about to go wrong again once you start thinking in terms of stairsteps and boxes and looking instead of hearing.

Back to the bits.

What lower amplitude (and fewer used bits) means is that the sound, as represented, is not as far above the noise floor as a full-amplitude sound. The digital noise floor is completely analogous to, eg, the noise floor of analog tape. If you use a dithered digital representation, you get something that behaves exactly as analog does. You hear and perceive both the same way.

>In 16-bit recording, there simply aren't enough bits to represent very low level details without distorting them with a subtle but audible crunchy digital halo of quantisation noise.

On an audio tape, the magnetic grains are just too large to represent very low level details without distorting them with a subtle but audible crunchy halo of analogue distortion and hiss.

In a 24 bit recording, the noise you mention is still there! It's just shifted down [theoretically] 8 bits or -48dB. That's the only difference. The noise floor is lower.

[In reality, 24 bit isn't. Most recordings don't even hit a full 16 bits, and no recordings, unless they're mathematically rendered, can get deeper than about 21 bits. There is no such thing as a 24-bit audio ADC/DAC that delivers 24 bits. The very best available today are about 21 bits of signal + 3 bits of noise.]

So the difference in playback between 16 bits and 24 bits is about 5 actual bits. If you're complaining about soft sounds in a 16-bit recording 'not having enough detail' because they're down at, say, 3 bits of resolution, are you saying it's all fixed by using 8? Aren't 8 bits woefully too few for any kind of quality sound?

(I hope at this point, you realize you're barking up an incorrect tree)

If you're following me so far, we can continue, but I expect even this much is going to require more conversation.

For a properly dithered recording, bit depth is exactly equivalent to noise floor. If low-volume details are lost in 16 bit recordings it is because their volume falls near or below the noise floor imposed by the 16 bit recording. 16 bits is good enough because the noise floor is low enough to be imperceptible to a human, if the gain of the recording is such that the full dynamic range available is used

""" in a studio, 0dB full-scale meter redline is calibrated to a standard voltage reference, and both consumer and professional audio has equivalent standard levels for the loudest level possible"""

This is only minor nitpicking, but the standard 0dB levels for professional audio (0dB reference at +4dBu == 1.23Vrms) and consumer audio (0dB reference at -10dBV == 0.32Vrms) are not meant to indicate the maximum ("loudest level possible") but just serve at a reference point, for e.g. the 1kHz sine you inject when setting up your gain throughout the signal chain. On most studio gear, you'll easily have +15dB headroom left.

AD/DA converters haven't really standardized on a full-scale level and there are quite a few different definitions in use: https://en.wikipedia.org/wiki/DBFS. Most "line level" ADC/DACs will have switches or jumpers to select between two or so settings. You'll choose them so that you are not likely to clip your ADC, and will only playback on your DAC with an appropriate level trimmed to not clip your analog gear.

Are you an expert?

In a 16-bit master, a noise shaping function is applied during down-conversion, by which quantization noise will be re-distributed so that most of the noise energy goes to the high frequencies (>15k) where it is completely inaudible.

For a good example of such a recording, see Ahn-Plugged (Ahn Trio, 2000, Sony BMG Masterworks). Fire up a good spectrum analyzer. You'll find the noise floor is well below <110 dB throughout most of the spectrum, even though it's 'just' a 16 bit CD.