- Relay selector as dropdown with green/yellow/red status dots
(green < 200ms, yellow > 200ms, red = offline, gray = unknown)
- All relays pinged on startup, RTT shown next to each
- "Manage Relays..." dialog: add/remove servers, see live status
- Clicking a relay in dropdown selects it, fills connect form
- Recent room chips auto-select matching relay
- Migrates old single-relay settings format automatically
- Prevents connecting to offline relays
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- New ping_relay Tauri command: QUIC connect with 3s timeout, returns RTT ms
- Relay status shown next to input field: "42ms" (green) or "offline" (red)
- Auto-pings on app startup and debounced on relay input change
- Fix SyncWrapper dead_code warning with #[allow(dead_code)]
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
#7 Fingerprint shown before connecting — new get_identity command reads
~/.wzp/identity at startup (generates if missing). Click to copy.
#8 Recent rooms store (relay, room) pairs — clicking a chip fills both
fields. Settings panel shows relay alongside room name. Migrates
old string[] format automatically.
#9 Auto-reconnect on unexpected disconnect — exponential backoff
(1s, 2s, 4s... max 10s), up to 5 attempts. Yellow blinking dot
shows reconnecting state. Stops if user clicks hangup.
#10 Audio handle cleanup — CPAL handles stored in SyncWrapper (no more
mem::forget), dropped properly on CallEngine::stop().
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Rust tracing subscriber was never initialized — all info!/warn!/error!
calls in the engine went to /dev/null. This meant our send/recv health
logging was invisible and we couldn't confirm the congestion fix was
active.
Now initializes tracing-android layer on first nativeInit(), routing
all Rust logs to logcat under tag "wzp_android". Also expanded logcat
filter in DebugReporter to capture engine-level log lines.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
The VPIO/CPAL audio handles were dropped at the end of start(),
killing the audio unit immediately. Audio I/O stopped working
after the first frame.
- Store audio handle in CallEngine via SyncWrapper
- Drop MutexGuard before returning from status() (Send future)
- Audio streams now live for the entire call duration
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Change status() from blocking_lock to async lock().await —
fixes "Cannot block the current thread from within a runtime" panic
that froze the call timer and broke audio
- Click fingerprint to copy to clipboard (both connect and settings screens)
- Show "Copied!" feedback on click
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
send_datagram() returns Err(Blocked) when the QUIC congestion window
is full. This is transient — the window reopens once ACKs arrive.
Previously, all send paths treated this as fatal (break/return),
which killed the send task and cascaded via tokio::select! to kill
the entire call.
Now: log warning, drop the packet, continue. Brief audio glitch
(20-100ms) instead of complete call death. FEC on the receiver
side recovers most dropped packets.
Fixed in:
- CLI run_live send task (continue + error counter)
- CLI run_file_mode send paths (2 locations)
- Desktop engine send task
Also hardened recv tasks: transient errors (non-closed/reset)
are survived instead of causing exit.
Matches the fix applied to Android client (engine.rs).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Full settings page as modal overlay (blur backdrop)
- Opens via gear icon on connect/call screens or Cmd+, (Ctrl+, on Win/Linux)
- Escape or click outside to close
- Settings: relay, room, alias, OS AEC toggle, AGC toggle
- Identity section showing fingerprint and identity file path
- Recent rooms management (remove individual, clear all)
- Save syncs back to connect form
- Gear icon on both connect and in-call screens
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Audio level meter with log-scale RMS visualization
- Call duration timer
- VPIO (OS AEC) wired through to engine with fallback to CPAL
- "You" badge on own participant entry
- Recent rooms list (click to reuse)
- Enter key to connect from form fields
- Improved dark theme with pulse animation on status dot
- Settings persistence via localStorage (relay, room, alias, AEC, recent rooms)
- Fingerprint display on connect screen
- Keyboard shortcuts skip input fields
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Root cause: send_media() returns Err(Blocked) when QUIC congestion
window is full. The send task treated ANY send error as fatal (break),
killing the entire call. Now send errors drop the packet and continue.
Also hardened recv task to survive transient errors and added health
logging (recv gap tracking, periodic stats) to both send and recv.
Relay: added comprehensive debug logging — recv gaps, lock contention,
forward latency, send errors — all per-participant with 5s stats.
Other changes:
- AEC toggle in Settings (persisted, applied on next call)
- Debug report: records call audio (WAV), RMS histogram (CSV), logcat,
stats. Emailed as zip via Android share intent after call ends.
- Replaced LinearProgressIndicator with Box (compose version compat)
- FileProvider for sharing debug zip attachments
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- New desktop/ directory with Tauri v2 + Vite + TypeScript
- Rust backend: CallEngine wrapping wzp-client audio + transport
- Web frontend: connect screen, in-call screen with participants,
mic/speaker mute, keyboard shortcuts (m/s/q)
- Dark theme UI, settings persistence via localStorage
- Platform-aware --os-aec: warns on Windows/Linux (not yet implemented)
- Workspace updated to include desktop/src-tauri
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Mic mute: the send loop now zeros the capture buffer when muted instead
of relying on write_audio() to skip writes. Previously stale ring data
and AGC amplification of near-silence caused crackling artifacts.
AEC: attach Android's hardware AcousticEchoCanceler to the AudioRecord
session. Also attach NoiseSuppressor when available. Both are released
on capture stop.
Room UI: deduplicate participants by fingerprint so ghost entries from
stale relay state don't show duplicate names.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
UI: deduplicate room participants by fingerprint so ghost entries from
stale relay state don't show duplicates.
Engine: after select! ends, call close_now() + connection.closed() with
500ms timeout to wait for the relay to acknowledge the CONNECTION_CLOSE.
Previously the close frame was queued but the runtime died before quinn
could retransmit if the first packet was lost.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
shutdown_background() killed the tokio runtime before quinn could send the
CONNECTION_CLOSE frame on the wire, so the relay never knew the client left.
Now use shutdown_timeout(500ms) to give quinn time to flush the close frame,
matching the desktop client pattern (which uses 2s timeout).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Replace Mutex-based CPAL callbacks with atomic SPSC ring buffers
- Proper async send/recv loops (no block_on), 20ms playout tick
- Add signal task for RoomUpdate presence display
- Add --alias, --raw-room flags and key persistence (~/.wzp/identity)
- Add SetAlias signal variant and relay-side handling
- Graceful Ctrl+C shutdown with force-quit on second press
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
o.optString("alias", null) returns the string "null" when the JSON value
is JSON null. Use o.isNull() check first. Also handle empty fingerprint
edge case with "unknown" fallback.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
stop_call() now calls close_now() on the stored transport handle before
killing the tokio runtime. This sends a QUIC CONNECTION_CLOSE frame so
the relay's recv loop breaks immediately, triggering leave() + RoomUpdate
broadcast. Previously the runtime was killed first, so transport.close()
never ran and the relay kept stale participants until idle timeout.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Settings now uses draft state — changes only persist on explicit Save
- Back button discards unsaved changes
- Added applyServers() for batch server updates
- Added missing alias field to CallOffer in featherchat tests
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Add SettingsScreen with identity (alias, key backup/restore), audio defaults,
server management, network prefs, and default room
- SettingsRepository persists all settings via SharedPreferences
- Auto-generate random display names on first launch (e.g. "Swift Wolf")
- Thread alias through CallOffer → relay handshake → RoomUpdate broadcast
- Derive caller fingerprint from identity key in relay handshake (fixes null
fingerprints when --auth-url is not set)
- Persist identity seed for stable fingerprints across reconnects
- Add alias field to SignalMessage::CallOffer (serde default for backward compat)
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Forgejo doesn't support @actions/artifact v4. Package the tarball
and print sizes instead. Binaries can be grabbed from the runner
workspace or deployed directly.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
The featherchat submodule uses SSH URL which doesn't work in CI.
Convert to HTTPS via git insteadOf before submodule init.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
The Forgejo runner needs Node.js for actions/checkout@v4.
catthehacker/ubuntu:act-latest has Node.js pre-installed.
Also install Rust in the workflow since the base image doesn't have it.
Build triggers on main + feat/* branches now.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Add RoomUpdate signal message to wzp-proto with participant count + list
- Add RoomParticipant struct (fingerprint + optional alias)
- Store fingerprint/alias in relay Participant struct
- Broadcast RoomUpdate to all room members on join and leave
- Add signal recv task in Android engine to handle RoomUpdate
- Surface room_participant_count + room_participants in CallStats JSON
- Show "X in room" with participant names in Android in-call UI
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Wire CallService foreground service for background calls (microphone type)
- Add Voice Volume + Mic Gain sliders (-20 to +20 dB) applied in Kotlin
- Connect AudioRouteManager for real speaker toggle via AudioManager
- Feed quinn QUIC RTT into PathMonitor, display Loss/RTT/Jitter from live data
- Nuclear teardown between calls — recreate engine + audio pipeline each call
- Fix re-entrant teardown loop from CallService notification callback
- Park audio threads as daemons to avoid libcrypto TLS destructor crash on exit
- Remove duplicate wakelocks from Activity (service owns them now)
- Strip AEC + denoise from capture path, keep AGC only (incremental approach)
- Fix .so copy target: libwzp_android.so not libwzp.so
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Wire AutoGainControl on both capture (mic → encode) and playout
(decode → speaker) paths to normalize volume levels
- Add server list with add/remove custom server dialog
- Add IPv4/IPv6 preference toggle for DNS resolution
- Resolve DNS hostnames to IP in Kotlin before passing to Rust engine
- Revert to IP addresses for default servers (DNS still broken on QUIC)
AGC confirmed working — voice levels noticeably improved in testing.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Partial wake lock + WiFi high-perf lock during calls — audio
continues when screen is off / phone is locked
- Server selector: toggle between LAN (172.16.81.175) and Pangolin
(pangolin.manko.yoga) before connecting
- Room name editable in idle screen
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- AudioPipeline: Kotlin AudioRecord/AudioTrack on JVM threads, PCM
shuttled to Rust via lock-free ring buffers + JNI
- FEC: RaptorQ fountain codes on encode (5 frames/block, 20% repair
ratio for GOOD profile), decoder feeds repair symbols for recovery
- Real audio level meter from mic RMS (replaces fake animation)
- Room name editable in UI (default: "android")
- Relay changed to pangolin.manko.yoga:4433
- Stats overlay shows FEC recovered count
- CallState now synced from polled stats (fixes "Connecting" stuck bug)
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
JS-based upload-artifact action doesn't work with act runner.
Use curl to create a pre-release and attach the tarball instead.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
SSH has no keys in the container. Use exact URL remap to
https://<token>@git.tbs.amn.gg/manawenuz/featherChat.git
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Use ssh://git@git.tbs.amn.gg:2222/ instead of HTTPS token auth
which gets 403 on cross-repo access.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
wzp-crypto depends on deps/featherchat (git submodule). Remap the
origin SSH URL to the Forgejo HTTPS mirror with github.token auth.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
The act runner can't clone a private repo over HTTPS without credentials.
Inject the auto-provided github.token into the clone URL.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
act runner uses bare ubuntu:24.04 without Node.js — actions/checkout,
actions/upload-artifact, etc. all fail. Replace with plain git clone
and shell commands.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
The act runner doesn't have Node.js in the rust:1-bookworm container,
breaking JS-based actions (checkout, cache, upload-artifact).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
pthread_create crashes on Android due to static bionic __init_tcb stubs
in the Rust std prebuilt rlibs. This is unfixable without rebuilding std.
Solution: run the entire call (QUIC connect, handshake, media send/recv)
on a single tokio current_thread runtime. The JNI startCall() now blocks,
so Kotlin dispatches it to Dispatchers.IO (JVM thread, not pthread).
Audio pipeline temporarily simplified to silence frames — will restore
once threading is solved (either via Java Thread or rebuilding std).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
The getauxval override (dlsym wrapper) fixes SIGSEGV in
init_have_lse_atomics at library load time. The current_thread
tokio runtime avoids SEGV_ACCERR in pthread_create/__init_tcb.
Both fixes are required together.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
