Remove Codex task-report.md from repo root
Transient implementer handoff note; its content is preserved in the handoff docs. Not repo content. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
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Claude Opus 4.8
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# Codex task report - 2026-06-16
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## W2 - Per-peer EQ
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- Added `src/audio/eq.rs`: a 3-band listener-side RBJ biquad EQ (low shelf, mid peaking, high shelf) with per-peer state and flat bypass.
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- Added local config persistence in `AppConfig.peer_eq`, keyed by peer node id string.
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- Added local `CoreCommand::SetPeerEq` and mixer-side per-peer `Eq` state. EQ is applied after local volume and before pan/mix; raw multitrack stems remain pre-volume/pre-EQ.
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- Added participant-card controls for Low/Mid/High gain sliders (-12 dB to +12 dB). Changes apply live and persist on slider release.
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- Tests added for flat identity, low/high boost energy, coefficient finiteness, clamping, and hot-signal processing.
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Unverified: subjective voice quality and zipper/noise behavior on real devices.
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## W1 - Per-listener pan / stereo playback
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- Added `src/audio/pan.rs`: constant-power `pan_gains()` with tests, plus playback gains that preserve the legacy default dual-mono center.
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- Converted playback mix to interleaved stereo in `src/core/mod.rs`.
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- Switched PipeWire playback output to 2-channel S16LE and adjusted ring target/capacity/stride accounting in `src/audio/pipewire_impl.rs`.
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- Kept capture, Opus encode/decode, jitter buffers, and network audio mono.
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- Limiter now receives the interleaved stereo bus; shared limiter gain ducks both channels consistently.
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- Mixed WAV and multitrack convenience mix fold the listener stereo mix back to mono before writing. Per-peer stems remain raw mono.
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- Updated `audio_probe` to send dual-mono stereo frames.
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- Added tests for exact center dual-mono behavior, hard-left pan contribution, and stereo fold-down.
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Decision for senior sanity-check: pure pan law is constant-power, but playback scales it by sqrt(2) so pan=0 is exactly the old mono signal in both ears. This satisfies the "default behavior unchanged" guardrail at the cost of louder hard-panned extremes, which the existing limiter catches.
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Unverified: real PipeWire stereo playback, underrun behavior on actual hardware, and recorded WAV listening checks.
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## W5 - Focused hotkeys + info popup
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- Added `src/hotkeys.rs`: serializable `KeyBinding`, `HotkeyAction`, `HotkeyMap`, parse/format/lookup, tier checks, and duplicate conflict detection.
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- Added `AppConfig.hotkeys` with defaults: F9 mute, F10 deafen, F2 Settings, Space push-to-talk, Leave unset.
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- Replaced the hard-coded PTT key capture with config-backed binding capture.
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- Added Settings hotkey editor with Set/Clear per action and live conflict warnings.
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- Added top-right hotkey info popup that lists every action and current binding, showing `unset` for unbound actions.
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- Routed focused iced key events through the map. App-wide actions can fire from any screen while focused; room-only actions require an active call. PTT press/release still uses `SetPttActive`.
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- Tests added for unset formatting, duplicate detection, room-tier lookup, defaults, and character parse/format.
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Unverified: manual keyboard interaction in the GUI. No OS-global hooks were added.
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## W3 - PipeWire pro-routing plan (not implemented)
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I stopped at design for W3. The current backend already supports simple target-node routing through PipeWire stream property `node.target`, but true "pro routing" (explicit ports / manual graph links / no-autoconnect patching) would require backend changes that are not safely verifiable offline.
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Proposed future scope:
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- Expose two advanced route targets: capture source node and playback sink node, with optional future per-port routing.
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- Enumerate available nodes with the existing `pw-cli list-objects Node` parser. For port-level routing, add a separate parser for `pw-cli list-objects Port` collecting `object.id`, `node.id`, `port.name`, direction, and channel position.
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- For node-level routing, continue using PipeWire stream property `node.target` on stream creation. This is the low-risk path and matches current backend behavior.
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- For explicit port routing, do not use `AUTOCONNECT`; instead capture the created PeerSpeak stream node/port ids from the PipeWire registry, then link with PipeWire-native APIs or `pw-link <source-port-id> <sink-port-id>`. Degrade by falling back to `node.target` autoconnect if any selected node/port is missing.
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- Offline tests should cover pure routing-plan decisions: selected node exists/missing, selected port exists/missing, capture/playback direction mismatch, and fallback choice. Real-device tests still need a PipeWire graph.
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Reason for not implementing: the current `run_playback` / `run_capture` code does not retain stream node or port ids, and changing `AUTOCONNECT` behavior plus adding manual `pw-link` calls could destabilize the working audio path. That matches the assignment's "bail if risky" instruction.
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## Backlog A21/A22 - correctness fixes
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- Fixed A21 in `src/core/jitter.rs`: implausibly large sequence discontinuities now reset the per-peer jitter stream instead of being treated as ordinary late packets or packet loss.
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- The reset threshold is `500` frames, about 10 seconds at 20 ms/frame. That covers both same-identity sender restart back to sequence 0 and a faulty/malicious jump far ahead that would otherwise force a long PLC run.
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- Added jitter regression tests for both far-behind restart and far-ahead jump cases.
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- Fixed A22 in `src/audio/recorder.rs`: `WavWriter` now tracks data bytes as `u64`, checks additions before writing, and rejects data that cannot fit both the RIFF size field and the `data` chunk size field.
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- Added a WAV overflow regression test that exercises the limit without creating a huge file.
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Unverified: the same-identity peer restart has not been exercised in a live 2-machine call; the WAV fix is counter/size-field tested, not a real >12h recording.
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## Backlog A14 - orderly window-close shutdown
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- Added `CoreCommand::Shutdown` and `UiEvent::ShutdownComplete`.
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- Window close now saves config, marks the GUI as closing, asynchronously queues `Shutdown`, and exits only after the core acknowledges completion or after a 5-second fallback timeout.
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- Core shutdown finalizes active mixed/multitrack recordings before session teardown, stops the standalone mic monitor, runs `ActiveSession::shutdown()` for active calls, clears room presence/routing, closes the persistent network stack, sends `ShutdownComplete`, and ends the core loop.
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- The shutdown command is queued with an awaited `mpsc::Sender::send` task instead of the best-effort `try_send`, so a full command queue does not immediately drop the close command.
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Unverified: actual GUI window-close behavior during a live call/recording still needs a manual run; tests/builds only prove the path compiles and existing unit coverage still passes.
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## Verification
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- `cargo check` passed.
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- `cargo test --lib` passed: 288 passed, 0 failed, 2 ignored.
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- `cargo clippy --all-targets` passed.
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- `cargo build --release` passed.
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- Formatted the touched Rust files with `rustfmt --edition 2024`; I did not run repo-wide `cargo fmt` to avoid unrelated formatting churn.
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No new dependencies were added. Runtime/manual/field verification is still pending for audio-device and 2-machine behavior.
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