wow, that's an incredibly creative rewrite of digital audio history... but not particularly accurate. Here are some readily verifiable facts:
1. 44.1KHz sample rate was chosen well before CDs were invented. This was during the first era of consumer digital recorders, which recorded on videocasettes. That rate turns out to allow you to hold 3 stereo samples per video line, and still give you a little more than 2KHz transition band for antialiasing filters. Redbook adopted 44.1K/16b because sony won the first format war (philips wanted 44.056Khz), and they wanted to leverage all the recordings already made in that format.
2. those devices had a choice of 14 or 16 bits as there were two competing formats, from sony and philips. The philips format was 14bits, which by the way, was still significantly better than any analog tape recorder of the time (in theory).
3. before these recorders, digital audio recorders were already being used professionally. (A popular sample rate back then was 50KHz). Apogee Electronics got their start back then making high quality antialiasing filters for some of these machines.
Most of these used SAR DACs and ADCs, which was really the only way to get >14 bits. There is no way to match resistors closely enough to do R-2R DACs with that resolution,
Regardless, both types suffer from large amounts of differential nonlinearity, well over 1LSB, which is much more audible and objectionable than large amounts of integral nonlinearity.
Digital oversampling filters were introduced not to cut cost (those chips were quite expensive at the time) but to improve the whole process of antialiasing. It allowed the freedom for the analog filters to have much more desirable characteristics, like lower Q (less ringing), better headroom, less critical component matching requirements, etc.
Sigma-delta converters ICs were introduced in the 80's not to cut cost (the first ones, manufactured by dbx, were very expensive) but to improve audio quality. They had superior specs and sound, due mostly to the lack of any measurable differential nonlinearity, and very smoothly shaped integral nonlinearity.
It turned out that, once the theory of sigma delta converters was better understood by the engineering community, the process to manufacture them could be made far cheaper than high-resolution SAR converters. But part of that is just the march of technology, as it's possible to get very good, inexpensive SAR converters these days, based on some of the same technological advances that were driven in part by development of S-D converters.
So, the cost reduction followed the innovation, but wasn't the impetus for it. These advances were done by engineers dedicated to improving the audio experience (even if they didn't always succeed).
The main downside with most current SD converters is they do not have very good accuracy at DC (though there are some around that do), and they have a few mS of latency due to digital filtering which can interfere with industrial uses such as motor control applications where they are in the feedback loop.
Bottom line is, for conversion to/from analog, one needs antialiasing and anti-imaging filters and a sampler/quantizer. Almost from the beginning, the filtering process has been a combination of analog and digital filtering. S-D converters are a result of looking at the theory and coming up with a much more elegant solution, though one that because of the heavy duty math involved, isn't exactly intuitive.
I'm still scratching my head at what bitperfect is supposed to mean in this context. I can't come up with any logical explanation. Analog signals do not have any 'real bits' or 'intrinsic bits' hidden inside them...
