forked from unom/punktfunk
enricobuehlerandClaude Fable 5
a7d4213778
Latency, three ways, all inside mic.rs: - 48 kHz stereo 20 ms becomes mono 10 ms: speech gains nothing from a second channel, the shorter frame shaves a buffering interval off the uplink, and the host already decodes any Opus frame <= 120 ms with its stereo decoder (mono packets upmix) — no protocol change. The encoder follows: 48 kbps, complexity 5, in-band FEC at an assumed 10% loss so a dropped datagram reconstructs from its successor instead of a hole. - The latency ratchet is gone: the capture callback drops the NEWEST chunk when the hand-off channel fills, so an encode-side stall used to convert into standing mic delay that never drained. The encode loop now drains the whole backlog in one lump and, past ~60 ms, jumps to the newest ~20 ms (one audible blip, live again), counting what it shed in the periodic log line. The realtime callback stays exactly as allocation-free as it was. - The encode thread registers with the client's hot-thread set, so the ADPF session keeps mic encode on a fast core alongside audio decode. No .frames_per_data_callback() pin: AAudio's own docs say leaving it unset is the lowest-latency path (the callback runs at the device's optimal burst), and the encode side re-chunks to 10 ms frames anyway. Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>