I once got a ~30 page .pdf from a professor, but it was because he had literally written the book we were using for the course. The book store was charging somewhere above $100 for the brand new edition so he told us to just buy the old one (no more than ~$20, brand new) and sent us all the pages that had been added/changed. :D
edit: this was many years ago so this isn't the actual edition in question, but shoutout to Ken Pohlmann anyways.
Jitter is not a problem that produces any perceptible impairment, at least not since 1985 when internal re-clocking of the signal and longer sample & hold buffering were implemented in pretty much every DAC on the market. The same is true of quantization noise, and other artifacts brought up by hi-fi woo woo "articles" (read: advertisements), which is below the noise floor of 16-bit LPCM.
Please read the more academic and objective primer (regarded by many engineers as the "Bible" of digital audio): Principles of Digital Audio by AES Society Fellow Ken Pohlmann.
True. Continuous version improvement of the Apple/Fraunhofer IIS algo's has led to more efficient perceptual coding within the framework... My entire library is in 256kbps AAC with only one exception: Any content that I have original 24/96 or 24/192 masters for.... there's a much larger leap between 16-bit dithered LPCM and 24-bit LPCM than there is between 256kbps AAC and 16-bit dithered LPCM.
One of my favorite resources on the fundamentals of audio encoding is Pohlmann's <em>Principles of Digital Audio</em>. Since 1985, it's been the definitive read on the subject.
The problem with the argument about Mp3 bitrates vs. AAC vs. other formats is that there are three concepts at work:
AAC's perceptual coding, which is based on Dolby's original AC-3/A52 codec, removes a lot of data that results in imperceptible "information" for the most part. So it gains efficiencies differently from other perceptual coding schema like Mp3 which rely on older developments in perceptual coding that keep gradually improving as different consortiums work to find new ways of reducing data requirements.
Just to give an example of how lossless formats work: ADPCM throttles the word length up and down, instead of keeping it at a fixed 16 bit length, and how it does this is by storing the relative change in amplitude as opposed to the absolute value, storing the fewest bits necessary per quantization interval. DTS theatrical encoding was originally a form of ADPCM.
Because Mp3 and AAC fall into the arena of perceptual coding and rely on different techniques to throw out unnecessary data, a 128kbps AAC file can have greater fidelity than a 320kbps Mp3. (please, no layperson arguments, take this up with NIST or AES, write a paper... etc.)
An excellent reference, probably THE definitive reference on all things digital audio related is Ken Pohlmann's Principles of Digital Audio. Since 1985 it has been the most thorough engineers "bible" on the technical aspects of digital audio encoding/transcoding, recording, mastering, etc.
This one is fantastic... It's like a user friendly version of Pohlmann's <em>Principles of Digital Audio</em>. First edition was published in 1985.