That's why residual vector quantization is a useful technique - using multiple dictionaries to quantize a single timeslice, each conditioned on the previous residual level. You can also quantize a signal at different frequencies.
There are samples towards the end of the post of their LLM trained on their Mimi audio codec.
I read the article and confess some of the modeling parts were above my comprehension. But I would like to add that as an audio engineer, the "key question" you describe is solved, just not applied to transformer models (?).
An experienced engineer can look at a waveform in a DAW and identify specific consonants, vowels, specific words, etc quite fluently. And with tools like Melodyne - which already quantize audio semantically - they can identify (and manipulate) pitch and formants as well, turning an O vowel into an E vowel, or changing the inflection of a phrase (up-speak vs down-speak, for example).
I don't know how to apply this to a neural codec, but it seems like it shouldn't be that hard (that's my naivete coming through)
As an experienced DAW author, I very, very much doubt this.
What can be done relatively easy is to "see" or rather "follow along" in the waveform when listening to the audio. But I read your claim as being that someone could look at the waveform (which is already decimated from the original) and identify words or phonemes without hearing the associated audio. I am extremely skeptical that there is anyone anywhere in the world who can do this.
I feel like there should be a model that can do much of this for me but I haven't really looked into it, ironically due to laziness, but also because I edit across multiple tracks at this stage, and I'm afraid to feed the model an already mixed stereo track. I'm curious why you still do it manually, if you still do and if you've looked into alternatives.