It occurred to me that the original 11025 sps files probably represent a higher-than-needed sampling rate. I downsampled to 8 ksps and gain back about 15 K flash, and some relaxed execution requirements.
It had bugged me for a bit that the sampling rate for the existing audio is 11025 ksps. This is quite appropriate for the SP0256 in general, which is advertised as having a 5 KHz output bandwidth, however on the -AL2 I find the signal to use less than that.
I discovered an audio processing application I use has a 'batch convert' mode, so I batch converted all the samples to an 8 KHz sampling rate (appropriate anti-aliasing filtering is done prior). The app I am using is 'Cool Edit', which to wit is long since discontinued (and it has a kind of interesting back story), but I would suspect that the free alternatives such as Audacity would similarly be useful.
This reduced the audio size by about 25%, and it seems to sound more-or-less the same. I modded the Python PoC to re-compute the ADPCM version and added that to the project. The sample rate timer, TIM4, needed to be adjusted. Everything else worked as-is.
This allowed me to gain back 15 K flash, and also reduced the responsivity requirements for preparing sample buffers (since they are now last 64 ms, up from 46). I wasn't in a crisis in either of those two areas, but it's still nice to have a little more wiggle room.
Who knows? Is it done, now?