WebRTC client bandwidth estimate now drives both encoder bitrate and
resolution tier selection, replacing the previous static-target encoder.
- webrtc.rs: enable str0m BWE (seeded at 5 Mbps), surface
EgressBitrateEstimate + KeyframeRequest events, expose
get_bwe_estimate() / set_need_keyframe()
- state_portal.rs: wire bitrate/resolution channels between the WebRTC
thread and the encode thread; tier ladder [1440p, 1080p, 720p] with
downscale at 60% budget and upscale hysteresis (120% sustained 10s)
- avhw.rs: SwEncImport::poll_resolution_commands() rebuilds the import
filter graph on UpdateResolution; SwEncEncode::recreate_encoder()
rebuilds sws/enc_video/yuv_frame atomically; hash_sampled_y_plane()
skips duplicate frames; VBV x264opts cap IDR bursts; H.264 level 4.0
(muxer) / 4.2 (WebRTC)
- state.rs: sync wlr-screencopy GOP to fps*2 max 20 for parity
- fix: drain bitrate_rx + resolution_rx BEFORE the stride check in
encode_cpu_frame() so the new (smaller-stride) frame produced after
a resolution change does not hit the stale (larger) enc_width and
crash the encode thread
- WebRTC GOP widened to fps*2 max 20 (was fps/2 max 10)
Add lightweight per-second pipeline statistics for stutter diagnosis:
- --stats CLI flag enables structured stats logging
- PipelineStats tracks capture/encode/send timing with p95/pmax
- FrameTimings records import/scale/transfer/sws/encode per-frame
- StatsSnapshot produces one structured log line per second
- Add src/webrtc.rs: HTTP signaling server + str0m Sans-IO WebRTC transport
with H.264 Annex-B → RTP packetization and key-frame request handling
- avhw: introduce FrameOutput enum (Muxer | Channel) so SwEncState can
output to either MP4 muxer or crossbeam channel for WebRTC
- cap_portal: support portal session restore tokens (PersistMode::ExplicitlyRevoked)
to skip re-authorization dialog; add --no-persist flag to force fresh dialog
- args: make --output optional when --port is used for WebRTC mode
- state_portal: integrate WebRTC pipeline (encoder channel → RTP forwarding)
with shorter GOP for WebRTC (fps/2, min 10)
- main: redirect tracing to stderr; validate --output or --port required
- Add dependencies: str0m 0.20, serde_json 1, dirs 6
dailz