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I applaud a new/better FFMPEG AAC encoder, but there are two pretty massive caveats that are mentioned in the specifics that need to be called out:

- CBR only

- Only optimized for 48khz sampling

Not being able to do quality-based variable bitrate encoding is a major gap, and since all of the CD audio in the world is at 44.1k sampling, that seems like a huge miss too.

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You can use -q:a, for "true" VBR, but its metrics are a few percent (imperceptable, we still win) less. "The benchmarks I posted were done mainly on 44.1Khz. I tuned by ear on 48Khz data though, so some of the windowing/transient logic is tied to 48Khz. It translated to 44.1Khz well enough that I left it as-is, since the timing difference isn't that large."

Why do you need VBR for audio encoding? VBR audio encoding sounds horrible and it can't save much of bitrate anyway.

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