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'course you are. How many layers of committee do changes have to go through? ;-) I know, you're talking about improvements within the spec (and so am I), but all of AAC's recent amazing advance has been due to spec revision. I've been itching to get out my revision boots myself...
Couldn't agree more, actually there are two huge additions to the spec planned in the next year as well, but that's the requirement of scientific development

Lawyers are fine - no submarines so far

Now, on the other hand - you also must admit that there have been changes within the current limits of the specs - FhG is no longer the best encoder at 128 kbps, for example - and QT and Nero are being developed very rapidly and being better and better with each new version - without spec change. After all, Vorbis lost at 128 kbps tests against ... LC AAC ... audio (QT 6.3) which was standardized in 1997.
More-than-year old 64 kbps test, before HE AAC, was won by Vorbis because QT AAC, which was used in the test, didn't use lossy stereo mode - it is a general concensus among major today's implementators (Dolby, FhG, Ahead) that stereo image shouldn't be traded for better spectral fullness, to say in that way - however this is not related to any other coding system, for example Ogg does much better job with lossy stereo at 64 kbps, of course.
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Seriously, You list all these things that Vorbis doesn't have as reasons why it's not as advanced. We evaluated and discarded most of the techniques you listed as 'unneccessary gimmickry' in the past five years... before they appeared in the AAC kitchen sink. "Perfection is reached not when there's nothing left to add... but rather when there's nothing extraneous left to take away". Our poor little minimalist encoder seems to hold its own very well, and Vorbis II will likely be considerably smaller a spec than I.
Of course, each codec has its planned field and industry... AAC, within complete multimedia framework (like MPEG-4) must have error robustness and concealment because one of its applications might be error-prone streaming over air. I must say that implementing codec-sensitive "smart" error protection scheme might give much better error correction possibility if the error rate is very big, like with cellular or digital AM transmissions, compared to "dumb" (this is a wrong name, I know) error protections implemented on a hardware link level. The best way is to provide one and standardized way to do that, instead of making tons of non-compatible error-protection schemes and that's why ER AAC exists.
Low Delay also has applications, maybe not so widespread, but it is nice to have ability to integrate your codec in 20ms frame-switching systems, like g.7xx speech codecs etc - you might ask do we really need 48 kHz 64 kbps 2-way communication - but I dunno.. maybe hi-fi teleconferencing would require that.. or some kind of tele-music with one member sitting in CA, other in WA and third in one NJ

Anyway... I think we all have valid points - so.. may research the development continue and best system win