whisper.cpp's encoder always runs over a full, padded 30s mel window
(1500 encoder positions) unless audio_ctx is explicitly reduced. The
~4s live-transcription streaming windows were never setting it, so
every window paid the full 30s-equivalent encode cost on one serial
worker thread -- the actual cause of the reported 30-65s lag between
speaking and a segment appearing, not "CPU is just slow".
Scales audio_ctx proportionally to the window's real sample count
instead (audio_ctx_for_window), floored and capped to whisper.cpp's
own valid range. Verified with a controlled A/B against the real
base.en-q5_1 model: ~30% faster even under heavy concurrent CPU
contention on this machine, which is masking most of the improvement
--the fix targets the encoder's O(n^2)-ish attention cost specifically
(1500 -> ~201 positions for a 4s window), so the gain on an otherwise-
idle machine should be substantially larger.
iamdoubz