OK, I see the problem. I have to say, your misunderstanding of how digital audio works is not uncommon. This subject has been covered numerous times in the past here and on other forums like gearslutz but I'll try to give you as simple an explanation as I can, without going too much into the math and scientific theory which details how it all works.
The misunderstanding of digital audio exists mainly because of two incorrect assumptions: The first is the assumption that more data means higher quality and the second is diagrams or waveform views in programs like ProTools. If you look at a waveform view in ProTools at say 24bit / 96kHz and compare it with a waveform view of say 16bit / 44.1kHz. The 24bit version will look less blocky and more like the smooth, continuously varying waveform we would see in an analog recording, so it's a natural assumption that when this 24bit waveform is sent to the speakers it's going to sound smoother and be more accurate. Unfortunately, this assumption misses some vital information: When you look at a waveform view in PT (or any other digital audio software or diagram) you are not actually looking at a waveform, you are just looking at a graphical representation of the digital data stored on your disk. This graphical representation is
NOT what will be sent to your speakers! First of all, this digital data will have to pass through a Digital to Analog Converter (DAC) and one of the processes in the DAC will be to apply what is called a dithering quantiser. The Dithering Quantiser converts all those blocks of digital data into a smooth, virtually perfect, linear recreation of the original recorded signal. Without going into the math, it's difficult to explain how this happens but I'll give it a go :)
If we record a signal with say 2 bits of data, we have 4 available values, the points at which we measure the signal's amplitude (the sample point) is unlikely to match any of our 4 available values so our digital system will be forced into assigning the nearest available value. The difference between the assigned digital value and actual value is called the quantisation error. What a dithering quantiser does is use some clever math to randomise these quantisation errors and this randomisation is perceived as white noise. So with 2bits (4 values) available we can in fact achieve a virtually perfect reconstruction of our original signal, unfortunately there would be so much quantisation error that this perfect original signal would be almost completely masked by the resultant white noise. This is the basis of the
Nyquist theorum. With me so far? OK, if we add another bit (3bit) we double the number of values available (8 values), which proportionately reduces the amount of quantisation error (noise), by half. Half the noise is -6dB, so each time we add a another bit of data, we double the accuracy and therefore reduce the noise by 6dB. So with 16bits, 16 x 6 = 96, 96dB is approximately the dynamic range of CD, 24 x 6 = 144, 144dB is roughly the dynamic range of 24bit.
In other words, 24bit is of course more accurate than say 1bit (or 16bit) but that added accuracy results
ONLY in less noise (and therefore a larger dynamic range), the waveform coming out of the DAC is just as virtually perfect at 1bit as it is at 24bit. The obvious proof of this is the existence of SACD, which is a 1bit format. Obviously with 1bit SACD there is a huge amount of noise from all the quantisation errors but this is dealt with by moving ("noise shaping") the noise into ultra sonic freqs so it can't be heard. Using the theory that more bits = higher quality, a CD should sound roughly 32,000 times better than an SACD! The fact that if anything SACDs often sound higher quality than CDs proves that sound quality cannot be directly related to bit depth.
With all this in mind, you can see that even a 16bit CD is a bit of an overkill, as it has about 10 times more dynamic range than a vinyl disk and therefore 24bit is unbelievable overkill (about 1,000 times more dynamic range than a vinyl disk), so what's the point of 24bit? Basically, when recording and mixing, 24bit allows us to leave 18dB (3bits if you want to think of it like that) of headroom and not even have to think about it. Remember, headroom is only required to give us some leeway for processing and summing signals together (when mixing) or to accommodate unexpected peaks when recording. None of these considerations exist for the consumer and therefore headroom is not required on a CD.
Your last post, about CDs: Again, the CD is just a storage device for digital data, that digital data has to pass through a DAC before being sent to the amp or speakers. However, the lower you record to the CD the closer you are getting to the -96dB noise floor, not a problem until you get so low that the consumer can hear the digital noise floor as they crank up their amp to hear the music. We still have to consider signal to noise ratio (SNR) when going to CD, even though we have a fairly wide margin of error.
Hope this helps, G
PS. Although Formfunction was a little intolerant, I think you were maybe too harsh in your response. ProTools, as the name implies is a Professional Tool, for recording engineers, mix engineers, mastering engineers and other audio professionals. With the word "engineer" in the job description you really need to have at least a basic understanding of the underlying engineering. While myself and many others here will be happy to help with specific questions, it's just not possible to provide an audio engineering course on this forum or to write in a few posts what has taken us many years to learn. Maybe your "extensive reading" could do with being a little more extensive?