Commit Graph

84 Commits

Author SHA1 Message Date
Siavash Sameni
134ee3a77f fix(engine): pass is_direct_p2p explicitly instead of deriving from is_some
Critical Phase 6 bug: when the negotiation agreed on relay path
but delivered the relay transport via pre_connected_transport,
CallEngine saw is_some() = true → is_direct_p2p = true → skipped
perform_handshake. The relay couldn't authenticate the participant
→ room join silently failed → recv_fr: 0, both sides sending
into the void.

Fix: add explicit is_direct_p2p: bool parameter to CallEngine::
start (both android and desktop branches). The connect command
sets it from the Phase 6 negotiation result (use_direct), not
from whether pre_connected_transport is Some.

Now relay-negotiated calls correctly run perform_handshake,
and direct P2P calls correctly skip it.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-12 10:34:21 +04:00
Siavash Sameni
e61397ca85 fix(connect): remove pre-Phase-6 same-IP heuristic
The commit de007ec added a heuristic that forced relay-only when
peers had different public IPs. That was a stopgap for the race
condition where one side picked Direct and the other picked Relay.
Phase 6 (f5542ef) solved this properly via MediaPathReport
negotiation, but the heuristic wasn't cleaned up and was still
running BEFORE the Phase 6 code — suppressing the race entirely
for cross-network calls.

Removed. Phase 6 negotiation now handles ALL cases: both sides
race, exchange reports, and agree on the same path before
committing media. Cross-network calls that can't go P2P will
have both sides report direct_ok=false and agree on relay.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-12 10:23:36 +04:00
Siavash Sameni
f5542ef822 feat(p2p): Phase 6 — ICE-style path negotiation
Before Phase 6, each side's dual-path race ran independently and
committed to whichever transport completed first. When one side
picked Direct and the other picked Relay, they sent media to
different places — TX > 0 RX: 0 on both, completely silent call.

Phase 6 adds a negotiation step: after the local race completes,
each side sends a MediaPathReport { call_id, direct_ok, winner }
to the peer through the relay. Both wait for the other's report
before committing a transport to the CallEngine. The decision
rule is simple: if BOTH report direct_ok = true, use direct; if
EITHER reports false, BOTH use relay.

## Wire protocol

New `SignalMessage::MediaPathReport { call_id, direct_ok,
race_winner }`. The relay forwards it to the call peer via the
same signal_hub routing used for DirectCallOffer/Answer. The
cross-relay dispatcher also forwards it.

## dual_path::race restructured

Returns `RaceResult` instead of `(Arc<QuinnTransport>, WinningPath)`:
- `direct_transport: Option<Arc<QuinnTransport>>`
- `relay_transport: Option<Arc<QuinnTransport>>`
- `local_winner: WinningPath`

Both paths are run as spawned tasks. After the first completes,
a 1s grace period lets the loser also finish. The connect
command gets BOTH transports (when available) and picks the
right one based on the negotiation outcome. The unused transport
is dropped.

## connect command flow (revised)

1. Run race() → RaceResult with both transports
2. Send MediaPathReport to relay with our direct_ok
3. Install oneshot; wait for peer's report (3s timeout)
4. Decision: both direct_ok → use direct; else → use relay
5. Start CallEngine with the agreed transport

If the peer never responds (old build, timeout), falls back to
relay — backward compatible.

## Relay forwarding

MediaPathReport is forwarded like DirectCallOffer/Answer: via
signal_hub.send_to(peer_fp) for same-relay calls, and via
cross-relay dispatcher for federated calls.

## Debug log events

- `connect:dual_path_race_done` — local race result
- `connect:path_report_sent` — our report to the peer
- `connect:peer_report_received` — peer's report
- `connect:peer_report_timeout` — peer didn't respond (3s)
- `connect:path_negotiated` — final agreed path with reasons

Full workspace test: 423 passing (no regressions).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-12 10:03:42 +04:00
Siavash Sameni
de007ec2fd fix(p2p): skip direct P2P when peers are on different public IPs
Race condition: when two phones are on different networks (WiFi
vs LTE, home vs office, etc.), each side's dual-path race runs
independently. One side may pick Direct while the other picks
Relay, causing both to send media to different places — TX > 0,
RX: 0 on both sides, completely silent call.

Root cause: the dual-path race doesn't have a negotiation step.
Each side picks the first transport that completes a QUIC
handshake, which may be a different path than the other side
picked. On same-LAN this doesn't matter because direct always
wins on both (the 500ms relay delay guarantees it). On cross-
network, the asymmetry bites.

Heuristic fix: compare own_reflex_addr IP to peer_reflex_addr
IP. If they're different → different networks → force relay-only
(set role = None, which skips the dual-path race entirely).

Same public IP means same LAN / same NAT:
  → LAN host candidates work, direct always wins on both sides
  → Safe for P2P

Different public IPs means cross-network:
  → Direct may work on one side but not the other
  → Relay is the safe choice for both

This preserves the proven same-LAN P2P and eliminates the broken
cross-network case. The full fix is ICE-style path negotiation
(Phase 6) where both sides exchange connectivity check results
through the signal plane and agree on a winner before committing
media — but that's a 500+ line protocol change.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-12 09:50:56 +04:00
Siavash Sameni
0a973b234b fix(engine): import tauri::Emitter for AppHandle::emit on Android target 2026-04-12 09:29:56 +04:00
Siavash Sameni
0ccf4ed6b5 feat(call): media health watchdog — warn user when no audio arrives
When a P2P direct call establishes successfully but the underlying
network path dies (phone switched from WiFi to LTE mid-call, or
cross-relay media forwarding isn't working), the call stays up
silently with recv_fr frozen at 0. No feedback to the user.

New watchdog in the Android recv task: tracks consecutive
heartbeat ticks (2s each) where recv_fr hasn't advanced. After 3
ticks (6s) with no new packets, emits:

- call-event { kind: "media-degraded" } — user-facing warning
  banner: "No audio — connection may be lost. Try hanging up and
  reconnecting, or switch to a different relay."
- call-debug media:no_recv_timeout for the debug log

If packets resume (recv_fr advances), clears the banner via:
- call-event { kind: "media-recovered" }

JS listener creates/removes a red-tinted banner dynamically at
the top of the call screen. Banner is also cleaned up on
showConnectScreen (call end).

This covers:
- Direct P2P that established on WiFi but died when the phone
  switched to LTE (stale NAT mapping, unreachable peer)
- Cross-relay calls where federation media isn't forwarding
  (relay not upgraded, not federated, etc.)
- Any other "connected but silent" scenario

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-12 09:18:38 +04:00
Siavash Sameni
2427630472 fix(connect): make peerLocalAddrs optional + skip handshake on direct P2P
Two regressions from Phase 5.5/5.6:

1. Room connect broken: the connect Tauri command required
   peerLocalAddrs as a Vec<String>, but the room-join JS path
   doesn't pass it (only the direct-call setup handler does).
   Error: "invalid args 'peerLocalAddrs' for command 'connect':
   command connect missing required key peerLocalAddrs".

   Fix: change to Option<Vec<String>>, unwrap_or_default() at
   usage sites. Room connect works again with zero peer addrs.

2. Direct P2P call connects but then CallEngine fails with
   "expected CallAnswer, got Discriminant(0)". Root cause: after
   the dual-path race picked a direct P2P transport, CallEngine
   still ran perform_handshake() on it. That handshake is a
   relay-specific protocol — sends a CallOffer signal and waits
   for CallAnswer back. On a direct QUIC connection to a phone,
   there's nobody running accept_handshake, so the handshake
   reads garbage from the peer's first media packet and errors.

   Fix: track is_direct_p2p = pre_connected_transport.is_some()
   and skip perform_handshake when true. The direct connection
   is already TLS-encrypted by QUIC, and both peers' identities
   were verified through the signal channel (DirectCallOffer/
   Answer carry identity_pub + ephemeral_pub + signature). Both
   android and desktop branches updated.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-12 08:09:32 +04:00
Siavash Sameni
16793be36f fix(p2p): Phase 5.6 — direct-path head start + hangup propagation + media debug events
Three fixes from a field-test log where same-LAN calls were
still losing the dual-path race to the relay path, peers were
getting stuck on an empty call screen when the other side
hung up, and 1-way audio was hard to diagnose because the
GUI debug log had no media-level events.

## 1. Direct-path 500ms head start (dual_path.rs)

The race was resolving in ~105ms with Relay winning even when
both phones were on the same MikroTik LAN with valid IPv6 host
candidates. Root cause: the relay dial is a plain outbound QUIC
connect that completes in whatever the client→relay RTT is
(~100ms), while the direct path needs the PEER to also process
its CallSetup, spin up its own race, and complete at least one
LAN dial back to us. That cross-client sequence reliably takes
longer than 100ms, so relay always won.

Fix: delay the relay_fut with `tokio::time::sleep(500ms)` before
starting its connect. Same-LAN direct dials complete in 30-50ms
typically, so the head start gives direct plenty of time to win
cleanly. Users on setups where direct genuinely can't work
(LTE-to-LTE cross-carrier) pay 500ms extra on the relay fallback,
which is invisible for a call setup.

## 2. Hangup propagation via a new hangup_call command (lib.rs + main.ts)

The hangup button was calling `disconnect` which stopped the
local media engine but never sent a SignalMessage::Hangup to
the relay. The peer never got notified and was stuck on the
call screen with silent audio. My earlier fix (commit e75b045)
only handled the RECEIVE side — auto-dismiss call screen on
recv:Hangup — but the SEND side was still missing.

New Tauri command `hangup_call`:
  1. Acquire state.signal.lock(), send SignalMessage::Hangup
     over the signal transport (best-effort; log + continue if
     signal is down)
  2. Acquire state.engine.lock(), stop the CallEngine

JS hangupBtn click handler now calls hangup_call with a fallback
to raw disconnect if the command is missing (older builds).

## 3. Media debug events (engine.rs + lib.rs)

Threaded tauri::AppHandle into CallEngine::start so the send/
recv tasks can emit call-debug events when the user has debug
logs enabled. Added on the Android branch (desktop branch
accepts the arg for API symmetry but doesn't emit yet):

  - media:first_send — emitted when the first encoded frame is
    handed to the transport. Useful for 1-way audio diagnosis:
    if this fires on side A but side B never sees media:first_recv,
    A's outbound is broken.
  - media:first_recv — emitted when the first packet from the
    peer arrives. Mirror of first_send.
  - media:send_heartbeat — every 2s with frames_sent, last_rms,
    last_pkt_bytes, short_reads, drops. A stalled last_rms
    (== 0) tells you the mic isn't producing samples; a frozen
    frames_sent tells you the encode pipeline hung.
  - media:recv_heartbeat — every 2s with recv_fr, decoded_frames,
    last_written, written_samples, decode_errs, codec. Mirror
    invariants for the inbound direction.

All four are gated by `call_debug_logs_enabled()` via
`emit_call_debug`, so they only show up in the GUI log when the
user has the Call Flow Debug Logs checkbox on. Tracing::info!
still runs unconditionally so logcat (adb) keeps its copy
regardless.

The `emit_call_debug` fn in lib.rs is now `pub(crate)` so
engine.rs can call it via `crate::emit_call_debug`.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-12 07:55:41 +04:00
Siavash Sameni
fa038df057 feat(p2p): Phase 5.5 — ICE LAN host candidates (IPv4 + IPv6)
Same-LAN P2P was failing because MikroTik masquerade (like most
consumer NATs) doesn't support NAT hairpinning — the advertised
WAN reflex addr is unreachable from a peer on the same LAN as
the advertiser. Phase 5 got us Cone NAT classification and fixed
the measurement artifact, but same-LAN direct dials still had
nowhere to land.

Phase 5.5 adds ICE-style host candidates: each client enumerates
its LAN-local network interface addresses, includes them in the
DirectCallOffer/Answer alongside the reflex addr, and the
dual-path race fans out to ALL peer candidates in parallel.
Same-LAN peers find each other via their RFC1918 IPv4 + ULA /
global-unicast IPv6 addresses without touching the NAT at all.

Dual-stack IPv6 is in scope from the start — on modern ISPs
(including Starlink) the v6 path often works even when v4
hairpinning doesn't, because there's no NAT on the v6 side.

## Changes

### `wzp_client::reflect::local_host_candidates(port)` (new)

Enumerates network interfaces via `if-addrs` and returns
SocketAddrs paired with the caller's port. Filters:

- IPv4: RFC1918 (10/8, 172.16/12, 192.168/16) + CGNAT (100.64/10)
- IPv6: global unicast (2000::/3) + ULA (fc00::/7)
- Skipped: loopback, link-local (169.254, fe80::), public v4
  (already covered by reflex-addr), unspecified

Safe from any thread, one `getifaddrs(3)` syscall.

### Wire protocol (wzp-proto/packet.rs)

Three new `#[serde(default, skip_serializing_if = "Vec::is_empty")]`
fields, backward-compat with pre-5.5 clients/relays by
construction:

- `DirectCallOffer.caller_local_addrs: Vec<String>`
- `DirectCallAnswer.callee_local_addrs: Vec<String>`
- `CallSetup.peer_local_addrs: Vec<String>`

### Call registry (wzp-relay/call_registry.rs)

`DirectCall` gains `caller_local_addrs` + `callee_local_addrs`
Vec<String> fields. New `set_caller_local_addrs` /
`set_callee_local_addrs` setters. Follow the same pattern as
the reflex addr fields.

### Relay cross-wiring (wzp-relay/main.rs)

Both the local-call and cross-relay-federation paths now track
the local_addrs through the registry and inject them into the
CallSetup's peer_local_addrs. Cross-wiring is identical to the
existing peer_direct_addr logic — each party's CallSetup
carries the OTHER party's LAN candidates.

### Client side (desktop/src-tauri/lib.rs)

- `place_call`: gathers local host candidates via
  `local_host_candidates(signal_endpoint.local_addr().port())`
  and includes them in `DirectCallOffer.caller_local_addrs`.
  The port match is critical — it's the Phase 5 shared signal
  socket, so incoming dials to these addrs land on the same
  endpoint that's already listening.
- `answer_call`: same, AcceptTrusted only (privacy mode keeps
  LAN addrs hidden too, for consistency with the reflex addr).
- `connect` Tauri command: new `peer_local_addrs: Vec<String>`
  arg. Builds a `PeerCandidates` bundle and passes it to the
  dual-path race.
- Recv loop's CallSetup handler: destructures + forwards the
  new field to JS via the signal-event payload.

### `dual_path::race` (wzp-client/dual_path.rs)

Signature change: takes `PeerCandidates` (reflex + local Vec)
instead of a single SocketAddr. The D-role branch now fans out
N parallel dials via `tokio::task::JoinSet` — one per candidate
— and the first successful dial wins (losers are aborted
immediately via `set.abort_all()`). Only when ALL candidates
have failed do we return Err; individual candidate failures are
just traced at debug level and the race waits for the others.

LAN host candidates are tried BEFORE the reflex addr in
`PeerCandidates::dial_order()` — they're faster when they work,
and the reflex addr is the fallback for the not-on-same-LAN
case.

### JS side (desktop/main.ts)

`connect` invoke now passes `peerLocalAddrs: data.peer_local_addrs ?? []`
alongside the existing `peerDirectAddr`.

### Tests

All existing test callsites updated for the new Vec<String>
fields (defaults to Vec::new() in tests — they don't exercise
the multi-candidate path). `dual_path.rs` integration tests
wrap the single `dead_peer` / `acceptor_listen_addr` in a
`PeerCandidates { reflexive: Some(_), local: Vec::new() }`.

Full workspace test: 423 passing (same as before 5.5).

## Expected behavior on the reporter's setup

Two phones behind MikroTik, both on the same LAN:

  place_call:host_candidates {"local_addrs": ["192.168.88.21:XXX", "2001:...:YY:XXX"]}
  recv:DirectCallAnswer {"callee_local_addrs": ["192.168.88.22:ZZZ", "2001:...:WW:ZZZ"]}
  recv:CallSetup {"peer_direct_addr":"150.228.49.65:NN",
                  "peer_local_addrs":["192.168.88.22:ZZZ","2001:...:WW:ZZZ"]}
  connect:dual_path_race_start {"peer_reflex":"...","peer_local":[...]}
  dual_path: direct dial succeeded on candidate 0   ← LAN v4 wins
  connect:dual_path_race_won {"path":"Direct"}

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-12 07:34:49 +04:00
Siavash Sameni
1618ff6c9d feat(p2p): Phase 5 — single-socket architecture (Nebula-style)
Before Phase 5 WarzonePhone used THREE separate UDP sockets per
client:

  1. Signal endpoint         (register_signal, client-only)
  2. Reflect probe endpoints (one fresh socket per relay probe)
  3. Dual-path race endpoint (fresh per call setup)

This broke two things in production on port-preserving NATs
(MikroTik masquerade, most consumer routers):

  a. Phase 2 NAT detection was WRONG. Each probe used a fresh
     internal port, so MikroTik mapped each one to a different
     external port, and the classifier saw "different port per
     relay" and labeled it SymmetricPort. The real NAT was
     cone-like but measurement via fresh sockets hid that.

  b. Phase 3.5 dual-path P2P race was BROKEN. The reflex addr
     we advertised in DirectCallOffer was observed by the signal
     endpoint's socket. The actual dual-path race listened on a
     DIFFERENT fresh socket, on a different internal (and
     therefore external) port. Peers dialed the advertised addr
     and hit MikroTik's mapping for the signal socket, which
     forwarded to the signal endpoint — a client-only endpoint
     that doesn't accept incoming connections. Direct path
     silently failed, relay always won the race.

Nebula-style fix: one socket for everything. The signal endpoint
is now dual-purpose (client + server_config), and both the
reflect probes and the dual-path race reuse it instead of
creating fresh ones. MikroTik's port-preservation then gives us
a stable external port across all flows → classifier correctly
sees Cone NAT → advertised reflex addr is the actual listening
port → direct dials from peers land on the right socket →
`endpoint.accept()` in the A-role branch of the dual-path race
picks up the incoming connection.

## Changes

### `register_signal` (desktop/src-tauri/src/lib.rs)
- Endpoint now created with `Some(server_config())` instead of
  `None`. The socket can now accept incoming QUIC connections as
  well as dial outbound.
- Every code path that previously read `sig.endpoint` for the
  relay-dial reuse benefits automatically — same socket is now
  ALSO listening for peer dials.

### `probe_reflect_addr` (wzp-client/src/reflect.rs)
- New `existing_endpoint: Option<Endpoint>` arg. `Some` reuses
  the caller's socket (production: pass the signal endpoint).
  `None` creates a fresh one (tests + pre-registration).
- Removed the `drop(endpoint)` at the end — was correct for
  fresh endpoints (explicit early socket close) but incorrect
  for shared ones. End-of-scope drop does the right thing in
  both cases via Arc semantics.

### `detect_nat_type` (wzp-client/src/reflect.rs)
- New `shared_endpoint: Option<Endpoint>` arg, forwarded to
  every probe in the JoinSet fan-out. One shared socket means
  the classifier sees the true NAT type.

### `detect_nat_type` Tauri command (desktop/src-tauri/src/lib.rs)
- Reads `state.signal.endpoint` and passes it as the shared
  endpoint. Falls back to None when not registered. NAT detection
  now produces accurate classifications against MikroTik / most
  consumer NATs.

### `dual_path::race` (wzp-client/src/dual_path.rs)
- New `shared_endpoint: Option<Endpoint>` arg.
- A-role: when `Some`, reuses it for `accept()`. This is the
  critical change — the reflex addr advertised to peers is now
  the address listening for incoming direct dials.
- D-role: when `Some`, reuses it for the outbound direct dial.
  MikroTik keeps the same external port for the dial as for
  the signal flow → direct dial through a cone-mapped NAT.
- Relay path: also reuses the shared endpoint so MikroTik has
  a single consistent mapping across the whole call (saves one
  extra external port and makes firewall traces cleaner).
- When `None`, falls back to fresh per-role endpoints as before.

### `connect` Tauri command (desktop/src-tauri/src/lib.rs)
- Reads `state.signal.endpoint` once when acquiring own reflex
  addr and passes it through to `dual_path::race`.

### Tests
- `wzp-client/tests/dual_path.rs` and
  `wzp-relay/tests/multi_reflect.rs` updated to pass `None` for
  the new endpoint arg — tests use fresh sockets and that's
  fine because the loopback harness doesn't care about
  port-preserving NAT behavior.

Full workspace test: 423 passing (no regressions).

## Expected behavior after this commit on real hardware

Behind MikroTik + Starlink-bypass (the reporter's setup):
- Phase 2 NAT detect → **Cone NAT** (was SymmetricPort — false
  positive from the measurement artifact)
- Phase 3.5 direct-P2P dial → succeeds for both cone-cone and
  cone-CGNAT cases where the remote side was previously blocked
  by our own socket mismatch
- LTE ↔ LTE cross-carrier → still likely relay fallback; that's
  genuinely strict symmetric and needs Phase 5.5 port prediction.

## Phase 5.5 (next, separate PRD)

Multi-candidate port prediction + ICE-style candidate aggregation
for truly strict symmetric NATs. Not needed for the 95% case —
Phase 5 alone fixes most consumer-router setups.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-11 19:47:20 +04:00
Siavash Sameni
b7a48bf13b feat(ui): incoming-call ring tone + system notification
Previously: incoming calls silently popped an "Accept/Reject"
panel. Easy to miss — no audible cue, no system-level alert if
the app was backgrounded. Now the incoming-call path triggers
both a synthesized ring tone and a system notification banner.

## Ring tone (desktop/src/main.ts)

New `Ringer` class using Web Audio API directly — no external
asset files, no new npm dep. Synthesizes a classic NANP two-tone
cadence (440Hz + 480Hz sine mix, 2s tone + 4s silence, looped)
through an envelope-gated gain node that ramps on/off to avoid
clicks. Audible on every Tauri-supported platform because
WebView carries Web Audio.

- `start()` — lazily creates AudioContext on first use
  (platforms that require a user gesture for AudioContext
  creation still work because the incoming-call event is
  user-adjacent from the webview's perspective), starts
  setInterval(6000) loop.
- `stop()` — clears the timer AND disconnects any active
  oscillators so there's no tail audio.
- Active-nodes array is swept every cycle so it doesn't grow
  unbounded across long rings.

Hooked into signal-event handlers:
- `"incoming"` → `ringer.start()` + notifyIncomingCall
- `"answered"`, `"setup"`, `"hangup"` → `ringer.stop()`
- Accept/Reject button click handlers → `ringer.stop()` as
  the first thing they do (before any await)

## System notification (desktop/src-tauri + main.ts)

Added `tauri-plugin-notification = "2"` to the Tauri app and
registered in the builder. Capabilities updated with the four
notification permissions.

Frontend calls the plugin commands via the generic `invoke`
instead of adding `@tauri-apps/plugin-notification` as a JS
dep — Tauri plugins expose `plugin:notification|notify` etc.
directly. Flow:

1. `is_permission_granted` — check cached
2. If not granted → `request_permission` (Android prompts the
   user once, cached thereafter)
3. `notify` with title="Incoming call", body="From <alias>"

All wrapped in try/catch with console.debug fallback — plugin
missing or permission denied is non-fatal, the visible panel +
ring tone still alert the user.

## Known gaps (deferred)

- Android native system ringtone (RingtoneManager) + full-
  screen intent for lockscreen-visible ringer. Requires
  platform-specific Java/Kotlin glue in the Tauri Android
  shell — bigger lift.
- Desktop window flash / taskbar attention-seek on incoming
  call when app is backgrounded.
- Vibration pattern on Android.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-11 18:46:13 +04:00
Siavash Sameni
20375eceb9 feat(signal): transparent reconnect + auto-swap on relay change
Two related UX fixes, same state-machine surface:

1. Relay drops / goes offline / restarts: the client now auto-
   reconnects in the background instead of silently falling to
   "not registered" and requiring the user to tap Deregister +
   Register.
2. User switches relay in settings: client auto-swaps — close
   old transport, register against new, all transparent.

## Signal state additions (desktop/src-tauri/src/lib.rs)

- `SignalState.desired_relay_addr: Option<String>` — what the
  user CURRENTLY wants. `Some(x)` means "keep me connected to x",
  `None` means "user explicitly asked for idle". This is the
  pivot that distinguishes "connection dropped, retry" from
  "user deregistered, stop".
- `SignalState.reconnect_in_progress: bool` — single-flight
  guard so concurrent triggers (recv-loop exit + manual
  register_signal + another recv-loop exit after a brief
  success) don't spawn duplicate supervisors.

## Refactor

The old `register_signal` Tauri command was doing the whole
connect + Register + spawn-recv-loop flow inline. Split into:

- `internal_deregister(signal_state, keep_desired)` — shared
  teardown helper that nulls out transport/endpoint/call state
  and optionally clears `desired_relay_addr`.
- `do_register_signal(signal_state, app, relay)` — core
  connect + register + spawn-recv-loop flow, callable from both
  the Tauri command and the reconnect supervisor. Returns an
  explicit `impl Future<...> + Send` to avoid auto-trait
  inference bailing inside the tokio::spawn chain (rustc loses
  the Send trail through the recv-loop spawn inside the fn
  body).
- `register_signal` Tauri command — now thin: if already
  registered to the same relay, no-op; otherwise
  internal_deregister(keep_desired=false), set
  desired_relay_addr = Some(new), call do_register_signal. The
  Rust side handles the "change of server" transition entirely
  on its own, no deregister+register dance from JS needed.
- `deregister` Tauri command — internal_deregister(keep_desired
  = false) so the recv-loop exit path sees the cleared desired
  addr and does NOT spawn a supervisor.

## Reconnect supervisor

New `signal_reconnect_supervisor(signal_state, app, relay)`
task. Spawned from the recv-loop exit path when the loop exits
unexpectedly AND `desired_relay_addr.is_some()` AND no
supervisor is already running.

- Exponential backoff: 1s, 2s, 4s, 8s, 15s, 30s (capped at 30s,
  never gives up). First attempt is immediate (attempt 0 skips
  the wait).
- On each iteration checks whether `desired_relay_addr` was
  cleared (user deregistered mid-flight) or another path
  already re-registered; either short-circuits the supervisor.
- Also detects if the user changed relays while the supervisor
  was sleeping — resets the backoff counter and retries against
  the new addr.
- On success, exits so the newly-spawned recv loop owns the
  connection from that point. If THAT drops again, a fresh
  supervisor spawns.
- Emits `call-debug-log` and `signal-event` events at every
  state transition so the GUI can display "reconnecting...",
  "registered" banners.

## UI wiring (desktop/src/main.ts)

- signal-event handler gets two new cases:
  - `"reconnecting"` — amber "🔄 reconnecting to <relay>…" in
    the registered banner area
  - `"registered"` — green "✓ registered (<fp prefix>…)" to
    clear the reconnecting badge
- Relay-selection click handler checks if a signal is
  currently registered and, if the user picked a different
  relay, fires `register_signal` with the new address. Rust
  side handles the swap transparently.

Full workspace test: 423 passing (no regressions).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-11 18:40:11 +04:00
Siavash Sameni
da08723fe7 fix(signal): forward-compat — log+continue on unknown SignalMessage variants
Both sides of the signal channel previously broke their recv loop
on any deserialize error, which meant adding a new variant in one
build silently killed signal connections from peers running an
older build. This bit us during Phase 1 testing: a new client
sending SignalMessage::Reflect to a pre-Phase-1 relay caused the
relay to drop the whole signal connection, which looked like
"Error: not registered" on the next place_call.

Fix:
- New TransportError::Deserialize(String) variant in wzp-proto
  carries serde errors as a distinct category.
- wzp-transport/reliable.rs::recv_signal returns Deserialize on
  serde_json::from_slice failures (was wrapped in Internal).
- wzp-relay/main.rs signal loop matches on Deserialize → warn +
  continue (instead of break).
- desktop/src-tauri/lib.rs recv loop does the same.

Other TransportError variants (ConnectionLost, Io, Internal) still
break the loop — only pure parse failures are recoverable.

This means future SignalMessage variant additions are backward-
compat by construction: older peers will see "unknown variant,
continuing" in their logs while newer peers can keep evolving the
protocol.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-11 18:13:31 +04:00
Siavash Sameni
59ce52f8e8 feat(p2p): Phase 3.5 dual-path QUIC race + GUI call-flow debug logs
Two features in one commit because they ship and test together:
Phase 3.5 closes the hole-punching loop and the call-flow debug
logs give the user live visibility into every step of a call so
real-hardware testing of the new P2P path is debuggable.

## Phase 3.5 — dual-path QUIC connect race

Completes the hole-punching work Phase 3 scaffolded. On receiving
a CallSetup with peer_direct_addr, the client now actually races a
direct QUIC handshake against the relay dial and uses whichever
completes first. Symmetric role assignment avoids the two-conns-
per-call problem:

- Both peers compare `own_reflex_addr` vs `peer_reflex_addr`
  lexicographically.
- Smaller addr → **Acceptor** (A-role): builds a server-capable
  dual endpoint, awaits an incoming QUIC session. Does NOT dial.
- Larger addr → **Dialer** (D-role): builds a client-only
  endpoint, dials the peer's addr with `call-<id>` SNI. Does NOT
  listen.
- Both sides always dial the relay in parallel as fallback.
- `tokio::select!` with `biased` preference for direct, `tokio::pin!`
  so each branch can await the losing opposite as fallback.
- Direct timeout 2s, relay fallback timeout 5s (so 7s worst case
  from CallSetup to "no media path" error).

New crate module `wzp_client::dual_path::{race, WinningPath}`
(moved here from desktop/src-tauri so it's testable from a
workspace test). `determine_role` in `wzp_client::reflect` is
pure-function and unit-tested.

### CallEngine integration
- New `pre_connected_transport: Option<Arc<QuinnTransport>>` arg
  on both android + desktop `CallEngine::start` branches. Skips
  the internal wzp_transport::connect step when Some. Backward-
  compat: None keeps Phase 0 relay-only behavior.
- `connect` Tauri command reads own_reflex_addr from SignalState,
  computes role, runs the race, passes the winning transport
  into CallEngine. If ANY input is missing (no peer addr, no own
  addr, equal addrs), falls back to classic relay path —
  identical to pre-Phase-3.5 behavior.

### Tests (9 new, all passing)
- 6 unit tests for `determine_role` truth table in
  `wzp-client/src/reflect.rs` (smaller=Acceptor, larger=Dialer,
  port-only diff, equal, missing-side, symmetry)
- 3 integration tests in `crates/wzp-client/tests/dual_path.rs`:
    * `dual_path_direct_wins_on_loopback` — two-endpoint test
      rig, Dialer wins direct path vs loopback mock relay
    * `dual_path_relay_wins_when_direct_is_dead` — dead peer
      port, 2s direct timeout, relay fallback wins
    * `dual_path_errors_cleanly_when_both_paths_dead` — <10s
      error, no hang

## GUI call-flow debug logs

Runtime-toggled structured events at every step of a call so the
user can see where a call progressed or stalled on real hardware.
Modeled on the existing DRED_VERBOSE_LOGS pattern.

### Rust side
- `static CALL_DEBUG_LOGS: AtomicBool` + `emit_call_debug(&app,
  step, details)` helper. Always logs via `tracing::info!`
  (logcat always has a copy); GUI Tauri `call-debug-log` event
  only fires when the flag is on.
- Tauri commands `set_call_debug_logs` / `get_call_debug_logs`.

### Instrumented steps (24 emit_call_debug sites)
- `register_signal`: start, identity loaded, endpoint created,
  connect failed/ok, RegisterPresence sent, ack received/failed,
  recv loop spawning
- Recv loop: CallRinging, DirectCallOffer (w/ caller_reflexive_addr),
  DirectCallAnswer (w/ callee_reflexive_addr), CallSetup (w/
  peer_direct_addr), Hangup
- `place_call`: start, reflect query start/ok/none, offer sent,
  send failed
- `answer_call`: start, reflect query start/ok/none or privacy
  skip, answer sent, send failed
- `connect`: start, dual_path_race_start (w/ role), won (w/
  path), failed, skipped (w/ reasons), call_engine_starting/
  started/failed

### JS side
- New `callDebugLogs: boolean` field on Settings type.
- Boot-time hydrate of the Rust flag from localStorage so the
  choice survives restarts (like `dredDebugLogs`).
- Settings panel: new "Call flow debug logs" checkbox alongside
  the DRED toggle.
- New "Call Debug Log" section that ONLY shows when the flag is
  on. Rolling in-memory buffer of the last 200 events, rendered
  as monospace `HH:MM:SS.mmm step {details}` lines with auto-
  scroll and a Clear button.
- `listen("call-debug-log", ...)` subscribed at app startup,
  appends to the buffer, re-renders on every event.

Full workspace test goes from 404 → 413 passing. Clippy clean
on touched crates.

PRD: .taskmaster/docs/prd_phase35_dual_path_race.txt
Tasks: 61-69 all completed

Next: APK + desktop build carrying everything — Phase 2 NAT
detect, Phase 3 advertising, Phase 3.5 dual-path + call debug
logs, plus the earlier Android first-join diagnostics — so the
user can validate the P2P path on real hardware with live
per-step visibility into where any failures happen.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-11 14:06:44 +04:00
Siavash Sameni
39277bf3a0 feat(hole-punching): advertise peer reflexive addrs in DirectCall flow — Phase 3
Completes the signal-plane plumbing for P2P direct calling: both
peers now learn their own server-reflexive address (Phase 1
Reflect), include it in DirectCallOffer / DirectCallAnswer, and
the relay cross-wires them into each side's CallSetup so the
client knows the OTHER party's direct addr. Dual-path QUIC race
is scaffolded but deferred to Phase 3.5 — this commit ships the
full advertising layer so real-hardware testing can confirm the
addrs flow end-to-end before adding the concurrent-connect logic.

Wire protocol (wzp-proto/src/packet.rs):
- DirectCallOffer gains optional `caller_reflexive_addr`
- DirectCallAnswer gains optional `callee_reflexive_addr`
- CallSetup gains optional `peer_direct_addr`
- All #[serde(default, skip_serializing_if = "Option::is_none")] so
  pre-Phase-3 peers and relays stay backward compatible by
  construction — the new fields are elided from the JSON on the
  wire when None, and older clients parse the JSON ignoring any
  fields they don't know.
- 2 new roundtrip tests (Some + None cases, old-JSON parse-back).

Call registry (wzp-relay/src/call_registry.rs):
- DirectCall gains caller_reflexive_addr + callee_reflexive_addr.
- set_caller_reflexive_addr / set_callee_reflexive_addr setters.
- 2 new unit tests: stores and returns addrs, clearing works.

Relay cross-wiring (wzp-relay/src/main.rs):
- On DirectCallOffer: stash the caller's addr in the registry.
- On DirectCallAnswer: stash the callee's addr (only set by
  AcceptTrusted answers — privacy-mode leaves it None).
- Send two different CallSetup messages: one to the caller with
  peer_direct_addr=callee_addr, and one to the callee with
  peer_direct_addr=caller_addr. The cross-wiring means each side
  gets the OTHER party's direct addr, not its own.
- Logs `p2p_viable=true` when both sides advertised.

Client advertising (desktop/src-tauri/src/lib.rs):
- New `try_reflect_own_addr` helper that reuses the Phase 1
  oneshot pattern WITHOUT holding state.signal.lock() across the
  await (critical: the recv loop reacquires the same mutex to
  fire the oneshot, so holding it would deadlock).
- `place_call` queries reflect first and includes the returned
  addr in DirectCallOffer. Falls back to None on any failure —
  call still proceeds via the relay path.
- `answer_call` queries reflect ONLY on AcceptTrusted so
  AcceptGeneric keeps the callee's IP private by design. Reject
  and AcceptGeneric both pass None.
- recv loop's CallSetup handler destructures and forwards
  peer_direct_addr to the JS layer in the signal-event payload.

Client scaffolding for dual-path (desktop/src-tauri/src/lib.rs +
desktop/src/main.ts):
- `connect` Tauri command gets a new optional `peer_direct_addr`
  argument. Currently LOGS the addr but still uses the relay
  path for the media connection — Phase 3.5 will swap in a
  tokio::select! race between direct dial + relay dial. Scaffolding
  lands here so the JS wire is stable, real-hardware testing can
  confirm advertising works end-to-end, and Phase 3.5 is a pure
  Rust change with no JS touches.
- JS setup handler forwards `data.peer_direct_addr` to invoke.

Back-compat with the CLI client (crates/wzp-client/src/cli.rs):
- CLI test harness updated for the new fields — always passes
  None for both reflex addrs (no hole-punching). Also destructures
  peer_direct_addr: _ in its CallSetup handler.

Tests (8 new, all passing):
- wzp-proto: hole_punching_optional_fields_roundtrip,
  hole_punching_backward_compat_old_json_parses
- wzp-relay call_registry: call_registry_stores_reflexive_addrs,
  call_registry_clearing_reflex_addr_works
- wzp-relay integration: crates/wzp-relay/tests/hole_punching.rs
    * both_peers_advertise_reflex_addrs_cross_wire_in_setup
    * privacy_mode_answer_omits_callee_addr_from_setup
    * pre_phase3_caller_leaves_both_setups_relay_only
    * neither_peer_advertises_both_setups_are_relay_only

Full workspace test goes from 396 → 404 passing.

PRD: .taskmaster/docs/prd_hole_punching.txt
Tasks: 53-60 all completed (58 = scaffolding-only; 3.5 follow-up)

Next up: **Phase 3.5 — dual-path QUIC connect race**. With the
advertising layer live, this becomes a focused change: on
CallSetup-with-peer_direct_addr, start a server-capable dual
endpoint, and tokio::select! across (direct dial, relay dial,
inbound accept). Whichever QUIC handshake completes first wins,
the losers drop, 2s direct timeout falls back to relay.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-11 13:37:04 +04:00
Siavash Sameni
8d903f16c6 feat(reflect): multi-relay NAT type detection — Phase 2
Builds on Phase 1's SignalMessage::Reflect to probe N relays in
parallel through transient QUIC connections and classify the
client's NAT type for the future P2P hole-punching path. No wire
protocol changes — Phase 1's Reflect/ReflectResponse pair is
reused unchanged.

New client-side module (crates/wzp-client/src/reflect.rs):
- probe_reflect_addr(relay, timeout_ms): opens a throwaway
  quinn::Endpoint (fresh ephemeral source port per probe,
  essential for NAT-type detection — sharing one endpoint would
  make a symmetric NAT look like a cone NAT), connects to _signal,
  sends RegisterPresence with zero identity, consumes the Ack,
  sends Reflect, awaits ReflectResponse, cleanly closes.
- detect_nat_type(relays, timeout_ms): parallel probes via
  tokio::task::JoinSet (bounded by slowest probe not sum) and
  returns a NatDetection with per-probe results + aggregate
  classification.
- classify_nat(probes): pure-function classifier split out for
  network-free unit tests. Rules:
    * 0-1 successful probes              → Unknown
    * 2+ successes, same ip same port    → Cone (P2P viable)
    * 2+ successes, same ip diff ports   → SymmetricPort (relay)
    * 2+ successes, different ips        → Multiple (treat as
                                             symmetric)

Tauri command (desktop/src-tauri/src/lib.rs):
- detect_nat_type({ relays: [{ name, address }] }) -> NatDetection
  as JSON. Takes the relay list from JS because localStorage
  owns the config. Parse-up-front so a malformed entry fails
  clean instead of as a probe error. 1500ms per-probe timeout.

UI (desktop/index.html + src/main.ts):
- New "NAT type" row + "Detect NAT" button in the Network
  settings section. Renders per-probe status (name, address,
  observed addr, latency, or error) plus the colored verdict:
    * green  Cone — shows consensus addr
    * amber  SymmetricPort / Multiple — must relay
    * gray   Unknown — not enough data

Tests:
- 7 unit tests in wzp-client/src/reflect.rs covering every
  classifier branch (empty, 1 success, 2 identical, 2 diff ports,
  2 diff ips, success+failure mix, pure-failure).
- 3 integration tests in crates/wzp-relay/tests/multi_reflect.rs:
    * probe_reflect_addr_happy_path — single mock relay end-to-end
    * detect_nat_type_two_loopback_relays_is_cone — two concurrent
      relays, asserts both see 127.0.0.1 and classifier returns
      Cone or SymmetricPort (accepted because the test harness
      uses fresh ephemeral ports per probe which look like
      SymmetricPort on single-host loopback)
    * detect_nat_type_dead_relay_is_unknown — alive + dead port
      mix, asserts the dead probe surfaces an error string and
      the aggregator returns Unknown (only 1 success)

Full workspace test goes from 386 → 396 passing.

PRD: .taskmaster/docs/prd_multi_relay_reflect.txt
Tasks: 47-52 all completed

Next up: hole-punching (Phase 3) — use the reflected address in
DirectCallOffer/Answer and CallSetup so peers attempt a direct
QUIC handshake to each other, with relay fallback on timeout.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-11 12:47:12 +04:00
Siavash Sameni
921856eba9 feat(reflect): QUIC-native NAT reflection ("STUN for QUIC") — Phase 1
Lets a client ask its registered relay "what IP:port do you see for
me?" over the existing TLS-authenticated signal channel, returning
the client's server-reflexive address as a SocketAddr. Replaces the
need for a classic STUN deployment and becomes the bootstrap step
for future P2P hole-punching: once both peers know their own reflex
addrs, they can advertise them in DirectCallOffer and attempt a
direct QUIC handshake to each other.

Wire protocol (wzp-proto):
- SignalMessage::Reflect — unit variant, client -> relay
- SignalMessage::ReflectResponse { observed_addr: String } — relay -> client
- JSON-serde, appended at end of enum: zero ordinal concerns,
  backward compat with pre-Phase-1 relays by construction (older
  relays log "unexpected message" and drop; newer clients time out
  cleanly within 1s).

Relay handler (wzp-relay/src/main.rs, signal loop):
- New match arm next to Ping reuses the already-bound `addr` from
  connection.remote_address() and replies with observed_addr as a
  string. debug!-level log on success, warn!-level on send failure.

Client side (desktop/src-tauri/src/lib.rs):
- SignalState gains pending_reflect: Option<oneshot::Sender<SocketAddr>>.
- get_reflected_address Tauri command installs the oneshot before
  sending Reflect and awaits it with a 1s timeout; cleans up on
  every exit path (send failure, timeout, parse error).
- recv loop's new ReflectResponse arm fires the pending sender or
  emits a debug log for unsolicited responses — never crashes the
  loop on malformed input.
- Integrated into invoke_handler! alongside the other signal
  commands.

UI (desktop/index.html + src/main.ts):
- New "Network" section in settings panel with a "Detect" button
  that displays the reflected address or a categorized warning
  ("register first" / "relay does not support reflection" / error).

Tests (crates/wzp-relay/tests/reflect.rs — 3 new, all passing):
- reflect_happy_path: client on loopback gets back 127.0.0.1:<its own port>
- reflect_two_clients_distinct_ports: two concurrent clients see
  their own distinct ports, proving per-connection remote_address
- reflect_old_relay_times_out: mock relay that ignores Reflect —
  client times out between 1000-1200ms and does not hang

Also pre-existing test bit-rot unrelated to this PR — fixed so the
full workspace `cargo test` goes green:
- handshake_integration tests in wzp-client, wzp-relay and
  featherchat_compat in wzp-crypto all missed the `alias` field
  addition to CallOffer and the 3-arg form of perform_handshake
  plus 4-tuple return of accept_handshake. Updated to the current
  API surface.

Results:
  cargo test --workspace --exclude wzp-android: 386 passed
  cargo check --workspace: clean
  cargo clippy: no new warnings in touched files

Verification excludes wzp-android because it's dead code on this
branch (Tauri mobile uses wzp-native instead) and can't link -llog
on macOS host — unchanged status quo.

PRD: .taskmaster/docs/prd_reflect_over_quic.txt
Tasks: 39-46 all completed

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-11 12:29:07 +04:00
Siavash Sameni
7e7968b2f9 diag(android-engine): first-join no-audio ordering instrumentation
Adds a single call_t0 = Instant::now() at the top of the Android
CallEngine::start path, threaded through send + recv tasks as
send_t0 / recv_t0, and tags the following milestones with
t_ms_since_call_start so we can build a clean side-by-side log of
first-call vs rejoin:

  1. QUIC connection established
  2. handshake complete
  3. wzp-native audio_start returned (+ how long audio_start itself took)
  4. send task spawned
  5. send: first full capture frame read (+ short_reads_before count)
  6. send: first non-zero capture RMS
  7. recv task spawned
  8. recv: first media packet received
  9. recv: first successful decode
 10. recv: first playout-ring write

Combined with the existing C++-side cb#0 logs in
crates/wzp-native/cpp/oboe_bridge.cpp ("capture cb#0", "playout
cb#0") this gives us full-pipeline ordering with no native-side
changes needed.

PRD: .taskmaster/docs/prd_android_first_join_no_audio.txt
Task: 32 (first task in the chain — diagnostics before any fix
attempts so we know which of the 5 suspect causes is real).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-11 10:00:20 +04:00
Siavash Sameni
578ff8cff4 feat(debug): GUI toggle for DRED verbose logs + macOS mic permission
DRED verbose logs (off by default — keeps logcat clean in normal use):
- wzp-codec: DRED_VERBOSE_LOGS atomic flag with dred_verbose_logs() /
  set_dred_verbose_logs() helpers
- opus_enc: gate "DRED enabled" + libopus version logs behind the flag
- desktop/src-tauri/engine.rs: gate DredRecvState parse log,
  reconstruction log, classical PLC log, and DRED-counter fields in
  the Android recv heartbeat (non-verbose path still logs basic recv
  stats)
- Tauri commands set_dred_verbose_logs / get_dred_verbose_logs
- Settings panel gets a "DRED debug logs (verbose, dev only)"
  checkbox; preference persists in wzp-settings localStorage and is
  pushed to Rust on save and on app boot

macOS mic permission:
- Add desktop/src-tauri/Info.plist with NSMicrophoneUsageDescription.
  Without it, modern macOS silently denies CoreAudio capture for
  ad-hoc-signed Tauri builds — capture starts but every callback
  hands you zeros. Symptom: phones could not hear desktop client,
  desktop could still hear phones (playout has no TCC gate). The
  Tauri 2 bundler auto-merges this file into WarzonePhone.app's
  Contents/Info.plist on the next build, so first launch will pop
  the standard mic prompt.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-11 09:48:32 +04:00
Siavash Sameni
16890576fb feat(observability): logcat-visible DRED proof of life on Android
Adds enough INFO-level logging that an opus-DRED-v2 APK on Android can
be verified end-to-end by reading logcat alone — no debugger, no
Prometheus, no telemetry pipeline required. Three observation points:

1. Encoder construction (opus_enc.rs)
   - Bumped the "DRED enabled" log from debug! to info! so the
     per-call DRED config is in logcat by default. Each call's first
     OpusEncoder construction logs codec, dred_frames, dred_ms,
     loss_floor_pct.
   - Added a one-shot static OnceLock that logs `opusic_c::version()`
     the first time an OpusEncoder is built in the process. This is
     the smoking gun for "is the new libopus actually loaded" — pre-
     Phase-0 audiopus shipped libopus 1.3 with no DRED, post-Phase-0
     should print 1.5.2 here.

2. DRED state ingest (DredRecvState::ingest_opus in
   desktop/src-tauri/src/engine.rs)
   - First successful parse on a call logs immediately so we can see
     "DRED is on the wire" in logcat.
   - Subsequent parses sample every 100th to confirm steady-state
     samples_available without drowning the log.
   - New parses_total / parses_with_data counters track the parse
     rate vs the success rate (a packet without DRED in it returns
     `available == 0`, so a low ratio means the encoder isn't
     emitting DRED bytes).

3. DRED reconstruction events (DredRecvState::fill_gap_to)
   - Every DRED reconstruction logs at INFO with missing_seq,
     anchor_seq, offset_samples, offset_ms, samples_available,
     gap_size, and the running total. These events are rare on a
     clean network and we want to know exactly which gap was filled.
   - First three classical PLC fills + every 50th thereafter log so
     we can see when DRED couldn't cover a gap (offset out of range,
     no good state, or reconstruct error).

4. Recv heartbeat (Android start() in engine.rs)
   - Existing 2-second heartbeat now includes dred_recv,
     classical_plc, dred_parses_with_data, dred_parses_total
     so a steady-state call shows the cumulative counters in
     logcat without parsing.

How to verify on a real call:

  adb logcat -s 'RustStdoutStderr:*' | grep -i 'dred\|libopus version'

Expected output sequence on a successful Opus call:
  - "linked libopus version libopus_version=libopus 1.5.2-..."  (once per process)
  - "opus encoder: DRED enabled codec=Opus24k dred_frames=20 dred_ms=200 loss_floor_pct=15"  (per call)
  - "DRED state parsed from Opus packet seq=N samples_available=4560 ms=95 ..."  (after first DRED-bearing packet)
  - "recv heartbeat (android) ... dred_recv=0 classical_plc=0 dred_parses_with_data=58 dred_parses_total=58"  (every 2s)

If you see "linked libopus version libopus 1.3" — the FFI swap didn't
take. If dred_parses_with_data stays at 0 while dred_parses_total
climbs — the sender isn't emitting DRED (check the encoder's loss
floor and the receiver's libopus version). If gaps trigger
"classical PLC fill" instead of "DRED reconstruction fired" —
DRED state coverage is too small for the observed loss pattern,
and the loss floor or DRED duration policy needs tuning.

Verification:
- cargo check -p wzp-codec -p wzp-client: 0 errors
- cargo check -p wzp-desktop: 0 Rust errors (only the pre-existing
  tauri::generate_context!() proc macro panic on missing ../dist
  which fires at host check time, irrelevant on the remote build)
- cargo test -p wzp-codec --lib: 69 passing (no regressions)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-11 08:58:03 +04:00
Siavash Sameni
dfbe21fe6e feat(tauri-engine): Phase 3b/3c re-port — DRED reconstruction on the live Tauri mobile engine
The original Phase 3b landed on wzp-client/CallDecoder and Phase 3c
landed on wzp-android/src/engine.rs. Both of those are DEAD CODE on
feat/desktop-audio-rewrite: the legacy Kotlin app in android/app/ is
not built by the Tauri mobile pipeline, and the Tauri engine bypasses
CallDecoder by calling wzp_codec::create_decoder directly.

The live Android call engine lives at desktop/src-tauri/src/engine.rs
with two `pub async fn start<F>` functions — one cfg-gated on Android
(Oboe via wzp-native) and one for desktop (CPAL). Both recv tasks
were using `let mut decoder = wzp_codec::create_decoder(...)` which
returns `Box<dyn AudioDecoder>` and doesn't expose the inherent
`reconstruct_from_dred` method.

Changes:

New helper struct `DredRecvState` at the top of engine.rs, wrapping:
  - DredDecoderHandle (libopus DRED side-channel parser)
  - DredState scratch (for parse_into)
  - DredState last_good (cached valid state, swapped on success)
  - last_good_seq: Option<u16> (DRED anchor sequence)
  - expected_seq: Option<u16> (for gap detection)
  - dred_reconstructions / classical_plc_invocations counters

With three methods:
  - ingest_opus(seq, payload): parse DRED, swap on success
  - fill_gap_to(decoder, current_seq, frame_samples, scratch, emit):
    detect gap back from expected_seq, reconstruct each missing
    frame via DRED if state covers it, fall through to classical
    decoder.decode_lost() when it doesn't. Calls emit() once per
    frame with a slice the caller uses for AGC + playout write.
  - reset_on_profile_switch(): invalidate tracking when codec changes

Both recv tasks (Android @ ~line 297 and desktop @ ~line 907):
  - Decoder type changed from `Box<dyn AudioDecoder>` via
    `wzp_codec::create_decoder` to concrete `AdaptiveDecoder::new(profile)`
    so we can call the inherent reconstruct_from_dred method.
  - Added `use wzp_proto::traits::AudioDecoder;` at the top of
    engine.rs to bring decode/decode_lost/set_profile trait methods
    into scope on the concrete type.
  - New `current_profile` local alongside `current_codec` (used for
    frame_duration lookups that drive the DRED sample offset math).
  - On codec/profile switch, call dred_recv.reset_on_profile_switch()
    because the cached DRED state is tied to the old profile's
    frame rate.
  - For each arriving Opus source packet:
      1. dred_recv.ingest_opus(seq, payload) — parse DRED
      2. dred_recv.fill_gap_to(...) — detect gap and reconstruct
         missing frames, each emitted through a closure that does
         AGC + playout write (wzp_native on Android, playout_ring
         on desktop)
      3. Normal decoder.decode() fallthrough for the current packet
         (unchanged)
  - Codec2 packets skip the DRED path entirely (is_opus() gate) —
    libopus can't reconstruct Codec2 audio.

Ordering invariant: gap reconstruction writes to playout BEFORE the
current packet's decoded audio, preserving temporal order since the
playout ring is FIFO. The closure captures the `spk_muted` flag once
before the gap loop to avoid mid-gap-fill state changes.

Kept `crates/wzp-android/src/engine.rs` and `crates/wzp-android/src/
stats.rs` from the earlier Phase 3c commit as-is — they're dead code
on feat/desktop-audio-rewrite but harmless, and deleting them would
diverge this branch from an independently-useful intermediate state.
The old Phase 3c commit (505a834) stays as historical reference.

Verification:
- cargo check -p wzp-codec -p wzp-client -p wzp-relay: 0 errors
- cargo check -p wzp-desktop: only pre-existing `tauri::generate_context!()`
  panic on missing ../dist (Vite output not built on host) — no Rust
  compile errors from our changes
- cargo test -p wzp-codec --lib: 69 passing (unchanged)
- cargo test -p wzp-client --lib: 35 passing + 1 ignored (unchanged)

Next: scripts/build-tauri-android.sh to get the actual Tauri APK —
NOT build-and-notify.sh which builds the dead legacy android/app.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 21:31:09 +04:00
Siavash Sameni
4ba77c8c0e feat(linux): WebRTC AEC3 capture/playback backend with render-side tee
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Adds gold-standard Linux echo cancellation: in-app WebRTC AEC3 (Audio
Processing Module) via the webrtc-audio-processing crate, using the
same algorithm as Chrome WebRTC, Zoom, Teams, and Jitsi. Runs entirely
in-process, so it works identically on ALSA / PulseAudio / PipeWire
systems — no dependency on user-configured echo-cancel modules.

Architecture:
- New crates/wzp-client/src/audio_linux_aec.rs module (~470 lines).
  Contains LinuxAecCapture and LinuxAecPlayback, both using CPAL
  under the hood but routing samples through a shared
  Arc<webrtc_audio_processing::Processor>. The playback path tees
  each 20 ms frame into APM.process_render_frame as the echo
  reference BEFORE handing the samples to CPAL's output callback.
  The capture path runs APM.process_capture_frame on each mic frame
  in place before pushing to the audio ring buffer. This is the
  "tee the playback ring" approach that Zoom/Teams/Jitsi use.
- New `linux-aec` feature in wzp-client pulling in the
  webrtc-audio-processing crate at v2.x with the `bundled`
  sub-feature. Bundled means the vendored PulseAudio WebRTC C++
  sources are statically compiled via meson+ninja at cargo build
  time — no runtime .so dependency, avoids Debian Bookworm's stale
  libwebrtc-audio-processing-dev 0.3 package (which predates AEC3).
  Dep is target-gated to Linux, so enabling the feature on non-Linux
  is a no-op.
- lib.rs re-exports LinuxAecCapture/LinuxAecPlayback as
  AudioCapture/AudioPlayback when `linux-aec` is on, otherwise
  falls back to the CPAL audio_io path. Shared public API
  (start/ring/stop/Drop) means downstream code is unchanged.
- New `linux-aec` feature in wzp-desktop forwards to
  wzp-client/linux-aec so `cargo tauri build -- --features
  wzp-desktop/linux-aec` builds the AEC variant.

APM configuration:
- EchoCancellation: High suppression, delay-agnostic mode on,
  extended filter on, stream_delay_ms=60 initial hint
- NoiseSuppression: High
- HighPassFilter: on
- AGC: off (can fight Opus encoder's own gain staging + adaptive
  quality controller; add later if users report low mic level)

Frame size handling:
- Pipeline uses 20 ms frames (960 samples @ 48 kHz mono)
- APM requires strict 10 ms (480 samples) per call
- Each 20 ms frame is split into two 480-sample halves, APM called
  twice, halves stitched back
- Same pattern for render and capture sides
- Carry-buffer logic handles the case where CPAL delivers samples in
  arbitrary chunk sizes that don't divide 960

Build infrastructure:
- scripts/Dockerfile.linux-desktop-builder adds meson, ninja-build,
  python3, clang for the webrtc-audio-processing bundled build
- scripts/build-linux-desktop-docker.sh takes a new --aec flag that
  enables the linux-aec feature and renames the output artifacts
  with an `-aec` suffix so noAEC and AEC variants can coexist on disk

Task #30.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 15:53:23 +04:00
Siavash Sameni
03a80a3196 feat(windows): WASAPI capture backend with OS-level AEC
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Adds a direct WASAPI microphone capture path for the Windows desktop
build that opens the default communications endpoint via
IMMDeviceEnumerator -> IAudioClient2 -> SetClientProperties with
AudioCategory_Communications, turning on Windows's communications
audio processing chain (AEC, noise suppression, automatic gain
control). The communications AEC operates at the OS level and uses
the system render mix as the reference signal, so echo from our
existing CPAL playback stream is cancelled automatically with no
per-process reference plumbing.

Architecture:
- New crates/wzp-client/src/audio_wasapi.rs module (~280 lines).
  Event-driven capture loop on a dedicated thread; pushes PCM into
  the same lock-free AudioRing used by the CPAL path. Same public
  API as audio_io::AudioCapture so downstream code is unchanged.
- New `windows-aec` feature in wzp-client that pulls in the
  `windows` crate (Microsoft's official Rust COM bindings) gated to
  target_os = "windows" only. Enabling the feature on non-Windows
  targets is a no-op since both the module and the dep are
  cfg(target_os = "windows").
- lib.rs re-exports WasapiAudioCapture as AudioCapture when the
  feature is on, otherwise falls back to the CPAL AudioCapture.
  AudioPlayback is always the CPAL one — no reason to swap it.
- desktop/src-tauri/Cargo.toml Windows target enables the new
  feature: `features = ["audio", "windows-aec"]`.

Implementation notes:
- Uses eCommunications role (not eConsole) for GetDefaultAudioEndpoint
  — the user-configured "communications" device that Teams/Zoom
  pick up, and the one Windows's AEC is tuned for.
- Requests 48 kHz mono i16 with AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM +
  SRC_DEFAULT_QUALITY so Windows handles any format conversion in
  the audio engine instead of rejecting our format.
- Event-driven with SetEventHandle / WaitForSingleObject — no
  polling, minimal CPU cost between packets.
- 200 ms wait timeout so the capture thread polls `running` often
  enough for Drop to stop cleanly even if the audio engine stalls
  (e.g. device unplug).

Task #24.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 14:35:36 +04:00
Siavash Sameni
7fecf285ea fix(windows): add icons/icon.ico for tauri-build Windows resource
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tauri-build's Windows path unconditionally looks up icons/icon.ico to
embed as the PE file resource (taskbar/Explorer icon). We only had
icon.png (32x32 placeholder) which is fine on macOS/Linux but blocks
the Windows cross-compile with "icons/icon.ico not found; required for
generating a Windows Resource file during tauri-build".

Generated a multi-size ICO (16/24/32/48/64/128/256) from the existing
placeholder icon.png via Pillow. It's ugly at 256 due to upscaling from
32x32 with LANCZOS, but unblocks the build. Real branded icons can
replace it later without any build-system changes.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 14:15:04 +04:00
Siavash Sameni
d774f5f8c5 feat(history): dedupe by call_id + explicit Incoming/Outgoing/Missed labels
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User reported that outgoing direct calls from macOS show up in the
history list as "missed" even when the call completes successfully.
Adds two changes to fix / diagnose:

1. history::log now dedupes by call_id. If an entry for this call_id
   already exists in the store, it updates the existing row's
   direction + timestamp in place instead of appending a duplicate.
   Protects against double-emit (caller side adding Missed on top of
   Placed, or any future signal loop that fires twice). One row per
   call_id, which matches what the user intuitively expects.

2. history::log now logs every write with tracing::info — call_id,
   peer_fp, direction, alias. Plus an extra line when we replace
   an existing entry: "history::log replacing existing entry
   from=Placed to=Missed" etc. Makes it easy to see in the desktop
   stderr which side is writing what, so we can find the outgoing =>
   missed regression immediately if it recurs.

3. main.ts now renders an explicit text label next to the direction
   arrow: "Outgoing", "Incoming", or "Missed" instead of just the ↗
   ↙ ✗ icons. Removes any ambiguity about what the icon means so
   future users can't misread a Placed entry as Missed based on icon
   shape alone.

Side fix for scripts/build-windows-cloud.sh:
- die() and the do_full ERR trap now respect WZP_KEEP_VM=1 so a failed
  build doesn't auto-destroy the debug VM (previously the trap fired
  before the KEEP_VM check and tore down the VM on any error).
- Bump default server type cx23 → cx33. 4GB RAM is not enough for a
  cold tauri + rustls + quinn + wzp-client cross-compile — the cx23
  run got "Read from remote host ... Connection reset by peer"
  partway through rustc, which is the classic signature of an OOM
  kill on the SSH session. cx33 has 8GB RAM and 8 vCPU which should
  comfortably fit the build.
2026-04-10 12:34:19 +04:00
Siavash Sameni
2fd94651e4 fix(desktop): direct calls used wrong identity file — mac identity mismatch
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The non-Android branch of CallEngine::start loaded the seed from
\$HOME/.wzp/identity directly, while register_signal in lib.rs goes
through the shared load_or_create_seed() helper which resolves via
APP_DATA_DIR → Tauri's app_data_dir(). On macOS those are two
completely different files:

  register_signal → ~/Library/Application Support/com.wzp.desktop/.wzp/identity
  CallEngine::start (old) → ~/.wzp/identity

On a fresh install they end up holding two different random seeds.
Register and CallEngine then derive two different fingerprints from
those seeds, and when a direct call comes in the relay routes it to
"you" under the register_signal fingerprint, but once CallEngine tries
to join the call-* room it advertises a DIFFERENT fingerprint — which
fails the call_registry ACL check on the relay side (only the two
authorised participants of a call can join its room). Silent hang, the
call never completes.

Android hit this bug earlier in the week and was fixed by switching
its CallEngine::start branch to `crate::load_or_create_seed()`.
Backport the same single-line change to the desktop branch so both
platforms share one identity source of truth.

Also bring the desktop branch up to parity with the android branch on
diagnostic logging:
- log CallEngine::start entry with relay/room/alias/quality/has_reuse
- log endpoint.local_addr on reuse / create
- log "QUIC connection established, performing handshake" between
  connect() and perform_handshake() so a hang at either step is
  immediately localisable
- map_err all three potential failure points (create_endpoint,
  connect, perform_handshake) to an explicit error! trace
2026-04-10 12:15:23 +04:00
Siavash Sameni
da09fdb6e9 windows(desktop): gate coreaudio / VoiceProcessingIO to macOS-only targets
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First step of the Windows x86_64 desktop build: stop pulling
coreaudio-rs into the Windows dependency graph so the project can at
least run `cargo check --target x86_64-pc-windows-msvc`. Software AEC
is already disabled in engine.rs so there's nothing else to stub — the
macOS-specific VPIO path is skipped via #[cfg(target_os = "macos")] on
both sides and Windows falls through to the plain CPAL
AudioCapture/AudioPlayback branch that already existed.

crates/wzp-client/Cargo.toml
  - coreaudio-rs optional dep moved under [target.'cfg(target_os = "macos")']
  - `vpio` feature now uses `dep:coreaudio-rs` syntax and the gated dep
  - Enabling `vpio` on Windows/Linux is a no-op at resolution time

crates/wzp-client/src/lib.rs
  - `pub mod audio_vpio` is now #[cfg(all(feature = "vpio", target_os = "macos"))]
  - Previously `vpio` alone was enough to try to compile the Core Audio
    bindings, which would fail on non-Apple targets the moment the
    feature flag was flipped on

desktop/src-tauri/Cargo.toml
  - [target.'cfg(not(target_os = "android"))'] removed — was leaking
    vpio into Windows/Linux via the catch-all.
  - macOS: wzp-client with features = ["audio", "vpio"]
  - Windows: wzp-client with features = ["audio"]
  - Linux: wzp-client with features = ["audio"]
  - Android: wzp-client with default-features = false (unchanged)
  - Dropped the unused direct coreaudio-rs = "0.11" dep on macOS —
    wzp-desktop's own sources never call Core Audio directly.

Verified via `cargo tree --target x86_64-pc-windows-msvc -p wzp-desktop`
that the Windows target now resolves wzp-client with cpal but without
coreaudio-rs. macOS target still resolves with coreaudio (direct via
vpio feature and transitively via cpal). macOS `cargo check` still
builds cleanly.

Cross-compile from macOS hit a cargo-xwin + llvm-lib setup issue in
ring's build.rs, so the actual `cargo check --target
x86_64-pc-windows-msvc` did not complete locally. Build verification
belongs on the user's Windows x86_64 host where MSVC is present
natively.

See tasks #23 (this one), #24 (Voice Capture DSP / WASAPI Communications
for OS-level AEC on Windows), and #25 (aarch64-pc-windows-msvc support).
2026-04-10 11:12:08 +04:00
Siavash Sameni
510eae2089 feat(direct-call): call history, recent contacts, deregister button
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Persistent JSON-backed call history for the direct-call screen so users
can see what they've placed / received / missed and dial back with one
click. Also fixes two small latent UX issues reported alongside.

Backend (Rust)
- new crate/module desktop/src-tauri/src/history.rs: thread-safe in-
  process store (OnceLock<RwLock<Vec<CallHistoryEntry>>>) backed by
  <APP_DATA_DIR>/call_history.json. Atomic writes via temp+rename. Max
  200 entries, FIFO pruning. CallDirection { Placed, Received, Missed }.
- Log hooks in the signal loop + commands:
    * place_call     → Placed entry (with target fingerprint)
    * DirectCallOffer → Missed entry up front; upgraded to Received
                        inside answer_call when accept_mode != Reject
                        via history::mark_received_if_pending(call_id).
                        If user rejects or never answers, it stays Missed.
- New Tauri commands:
    * get_call_history()     → all entries, newest first
    * get_recent_contacts()  → unique peers by fp, newest interaction first
    * clear_call_history()   → wipes JSON + in-memory
    * deregister()           → tears down signal transport + endpoint
  Backend emits `history-changed` events so the UI can live-refresh
  without polling.

Frontend (main.ts + index.html + style.css)
- Direct-call panel now has:
    * Recent contacts chip row (top 6 unique peers). Click a chip → dial.
    * Call history list (up to 50 rows). Direction icon (↗ placed, ↙
      received, ✗ missed), peer alias/fp, relative timestamp, callback
      button. Both click handlers populate target-fp and fire place_call.
    * Deregister button in the "registered" header — calls the new
      deregister command, tears down the signal transport, returns the
      UI to the pre-register state.
    * Clear-history link in the history header.
- Subscribes to `history-changed` events so the list updates the moment
  the backend logs a new entry. Also refreshed on register + after a
  clear.
- Nothing is rendered until there is data — empty sections stay hidden.

Tasks #20 + #21 (small UX items bundled in)
- Default room "general" for new installations: the html input value
  attribute is now "general" and loadSettings() defaults match. Existing
  users' localStorage still wins.
- Random alias on desktop: already latent but confirmed working — the
  startup IIFE at main.ts:374 calls get_app_info() and prefills the
  alias input from derive_alias(seed) when the input is empty. No code
  change needed, just verified it flows through the same path as the
  Android client.

Known follow-ups (deferred to step 6 polish)
- Call duration tracking (currently all entries have no duration field)
- Hangup signal from an unanswered incoming should emit history-changed
  so the missed state is visible even when the user never tapped accept
- Android UI layout fit-check on the smaller Nothing screen
2026-04-10 11:03:36 +04:00
Siavash Sameni
76a4c53e21 fix(android-audio): spawn_blocking for Oboe restart — unblock tokio executor
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Build 4c6aac6 added a stop+sleep+start Oboe restart inside the
set_speakerphone Tauri command, but calling wzp_native::audio_stop()
and audio_start() synchronously from an async fn blocks the tokio
executor thread — those FFI calls wait for AAudio to finalise the
stream teardown/bringup, which takes ~400ms each on Nothing phone
(Pixel is fast enough to hide the bug).

Reproduced on Nothing: 7 rapid Speaker button clicks across ~30
seconds, each restarting Oboe. After the 5th click the engine send
and recv tokio tasks froze for 22 seconds — decoded_frames stuck at
1159 across 9 heartbeats, send_drops growing from 148 to 1720 as
encoded frames couldn't make it past `send_t.send_media(pkt).await`.
At 08:40:48 the runtime finally caught up and processed a 911-frame
burst at once (buffered QUIC datagrams flooding through). Classic
"blocking sync call in async context" anti-pattern.

Fix: run the stop + start sequence inside tokio::task::spawn_blocking
so the Oboe teardown + reopen happens on a dedicated blocking thread,
leaving the tokio runtime free to keep driving the send and recv
tasks. AAudio's requestStop returns only after the stream is actually
in Stopped state, so the explicit sleep that bridged stop and start
is no longer needed and is dropped.

Send and recv tasks still see a ~500ms window of empty reads /
partial writes during the blocking restart, but they get SCHEDULED
through it — network packets keep being received + decoded + dropped
into the playout ring, and captured mic samples keep being encoded +
sent through quinn. No more executor starvation, no more 22-second
audio dropouts, no more send_drops burst.

Pixel still worked before this fix only because its AAudio teardown
is fast enough to not exceed the scheduler's cooperative yield
interval — same bug was latent on both devices, Nothing just made it
visible.
2026-04-10 08:45:54 +04:00
Siavash Sameni
4c6aac654a fix(android-audio): restart Oboe on speakerphone toggle + unbreak button UI
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Build 4f2ad65 wired the Speaker button to AudioManager.setSpeakerphoneOn
but user testing found that flipping speakerphone on an active Oboe
VoiceCommunication stream silently tears down the AAudio streams on
Pixel-class devices — both capture and playout stop producing data.
Only ending the call and rejoining brings audio back (because the fresh
Oboe open runs with the new routing already applied).

Also the earpiece state showed up red in the UI because the button was
getting the `.muted` CSS class when speakerphoneOn=false. Earpiece is a
valid routing state, not a muted one.

Fix set_speakerphone Tauri command:
  1. Flip AudioManager.setSpeakerphoneOn via JNI (as before).
  2. If the Oboe backend is currently running, stop it, sleep 50 ms to
     let AAudio finalise the transition, then start it again. The Rust
     send/recv tokio tasks keep running across the gap — they just read
     zero samples and write into the preserved ring buffers for a few
     frames, which is acceptable. The AudioBackend singleton's ring
     state is preserved across stop+start because it's in a 'static
     OnceLock.
  3. Debounce the UI click via speakerphoneBusy + spkBtn.disabled so
     users can't queue up multiple toggles during the restart window.

Fix main.ts Speaker button:
  - Remove the `.muted` classList toggle (added `.speaker-on` for CSS).
  - Update label text to "🔊 Speaker" / "🔈 Earpiece" for clarity.
  - On showCallScreen(), invoke is_speakerphone_on to sync the label
    with the real AudioManager state, so it matches reality after a
    rejoin (which was another symptom the user hit — the button label
    desynced from the actual routing after ending and restarting a
    call).
  - Debounce click + disable button while the restart is in flight.

Drops #[allow(dead_code)] from wzp_native::audio_is_running now that it
is actually called from the set_speakerphone restart guard.
2026-04-10 07:35:12 +04:00
Siavash Sameni
4f2ad65418 fix(android_audio): add explicit pointer types for .cast() — was rejected by rustc E0282 on android target
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2026-04-09 22:02:48 +04:00
Siavash Sameni
0178cbd91d android(audio): Speaker button toggles earpiece↔speaker via JNI (WIP, untested)
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Build 9e37201 confirmed on-device that Usage::VoiceCommunication +
MODE_IN_COMMUNICATION + speakerphoneOn=false routes Oboe playout to the
handset earpiece and the callback drains the ring correctly. Next step:
let the user flip speakerphoneOn at runtime so the existing Speaker
button actually switches audio routing instead of just gating writes.

- Cargo.toml (android target): pull in `jni = 0.21` and
  `ndk-context = 0.1`. Both are already transitively in the lockfile
  via Tauri/Wry, so this just promotes them to direct deps.
- desktop/src-tauri/src/android_audio.rs: new module. Grabs the JavaVM +
  current Activity from `ndk_context::android_context()`, attaches a
  JNI thread, calls `activity.getSystemService("audio")` to get the
  AudioManager, and exposes `set_speakerphone(bool)` +
  `is_speakerphone_on()` helpers that call the AudioManager method of
  the same name. All gated behind `#[cfg(target_os = "android")]`.
- lib.rs: adds `mod android_audio;` (android only), two new Tauri
  commands `set_speakerphone(on)` and `is_speakerphone_on()` — desktop
  gets no-op stubs so the same frontend invoke() works everywhere.
  Both registered in the invoke_handler.
- desktop/src/main.ts: the Speaker button (previously toggled the
  playout-write gate via `toggle_speaker`) now calls `set_speakerphone`
  and reads back the new routing state. Labels switched from
  "Spk" / "Spk Off" to "Earpiece" / "Speaker" so users can't be
  confused into thinking clicking turns audio off. pollStatus no longer
  clobbers the spkBtn label based on engine spk_muted, since the two
  concepts are now decoupled.

WIP because this has NOT been built or tested yet — committing at night
to save the work. Tomorrow: build #50 with this change, smoke-test the
Handset↔Speaker toggle, then move on to call history + last-contacts UI
and the Speaker-button mute bug on the other phone.
2026-04-09 22:00:34 +04:00
Siavash Sameni
9e37201198 android(audio): Usage::VoiceCommunication + MODE_IN_COMMUNICATION, default handset
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With da106bd (Usage::Media + MODE_NORMAL) audio works but is always on
the loudspeaker — we want handset as the default with a user-driven
toggle for speaker (and later bluetooth). The right Oboe usage for a
VoIP app is VoiceCommunication, which honours
AudioManager.setSpeakerphoneOn / setBluetoothScoOn for routing.

Bisection across previous builds showed that setAudioApi(AAudio) +
Usage::VoiceCommunication made the playout callback stop draining the
ring after cb#0 (build 8c36fb5 logs). Letting Oboe pick the AudioApi
implicitly keeps the callback alive — 96be740's Media-usage callbacks
fired at steady 50Hz without any explicit setAudioApi. So: keep the
Usage change, DROP the explicit AAudio force.

- oboe_bridge.cpp: Usage::VoiceCommunication, no setAudioApi, no
  ContentType override.
- MainActivity.kt: setMode(MODE_IN_COMMUNICATION) +
  setSpeakerphoneOn(false) = handset default, plus max both
  STREAM_VOICE_CALL and STREAM_MUSIC volumes for belt-and-braces.

Next build will add a JNI-based Tauri command to flip speakerphoneOn
at runtime so the user can toggle handset↔speaker during a call.
2026-04-09 21:50:06 +04:00
Siavash Sameni
da106bd939 fix(android-audio): revert to 96be740's Oboe config — VoiceCommunication broke callback drain
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Build 8c36fb5 logs showed a new regression: Oboe playout cb#0 fires once
at startup then the callback STOPS DRAINING the ring entirely.
written_samples sticks at 7679 (= RING_CAPACITY - 1) across every recv
heartbeat in a 40-second test. Meanwhile the recv task decodes 1800+ real
audio frames (sample range up to [-27920..31907], rms 12065) which all
get dropped on the floor by audio_write_playout returning 0 because the
ring is full.

Bisection: 96be740 (Usage::Media, no setAudioApi, no ContentType, no
MainActivity audio mode change) DID drive the playout callback at the
expected 50Hz (playout heartbeat: calls=1100 total_played_real=1055040
over 22 seconds). User still heard nothing there because of OS routing,
but at least Oboe accepted the PCM.

8c36fb5 added three changes on top of 96be740:
  1. Oboe Usage::Media → Usage::VoiceCommunication
  2. Oboe setAudioApi(oboe::AudioApi::AAudio) explicit
  3. Oboe setContentType(ContentType::Speech)
  4. MainActivity setMode(MODE_IN_COMMUNICATION) + setSpeakerphoneOn(true)
Every one of those could have killed the callback; combined they did.

Revert to 96be740's exact Oboe config: Usage::Media, no setAudioApi, no
ContentType. Keep the PCM recorder, heartbeat logging, and stream-open
logging. Separately, MainActivity now maxes STREAM_MUSIC (the stream
Usage::Media routes to) but leaves audio mode in MODE_NORMAL — no more
speakerphone/call-mode combo that makes Oboe unhappy. In NORMAL mode a
STREAM_MUSIC stream plays through the loud speaker by default.

Proof that the Rust pipeline is perfect: decoded.pcm recorded in 8c36fb5
was pulled via `adb shell run-as com.wzp.desktop cat .wzp/decoded.pcm`,
converted with ffmpeg, and played back on the Mac — user confirmed
audible speech. So 100% of the remaining bug surface is Android audio
routing, not anything in the Rust/C++ decode path.
2026-04-09 21:38:19 +04:00
Siavash Sameni
cfa9ff67cf fix(android-audio): VoIP mode + speakerphone + debug PCM recorder
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Build 96be740 logs proved the entire software pipeline is healthy:
  capture heartbeat:   calls=1100 to_write=960 full_drops=0 total_written=1056000
  recv heartbeat:      decoded_frames=1035 last_written=960 decode_errs=0
  recv decoded PCM:    range=[-13564..9244] rms=8044 (real audio)
  playout WRITE:       in_len=960 written=960 rms=2318 (real audio into the ring)
  playout heartbeat:   calls=1100 nonempty=1099 total_played_real=1055040

1055040 samples / 48000 Hz = 22s — exactly matches wall-clock elapsed,
meaning Oboe IS calling our playout callback at the expected rate and
WE ARE handing it real PCM every 20ms. User still heard nothing. Ergo
Oboe accepted the PCM and routed it to a silent output. Two fixes:

1) MainActivity.kt: switch to MODE_IN_COMMUNICATION + speakerphone ON
   right after permissions are granted, and crank STREAM_VOICE_CALL to
   max. Without this, an Oboe Usage::VoiceCommunication stream gets
   opened, the OS creates a real AAudio pipeline, the callback fires on
   schedule — and audio goes to either the earpiece at muted volume or
   a "call not active" dead end. Logs the audio mode + volume levels
   before and after the switch so we can confirm the state change in
   logcat next run.

2) oboe_bridge.cpp: revert Usage::Media → VoiceCommunication (the mode
   that matches MODE_IN_COMMUNICATION), pin the audio API to AAudio
   explicitly instead of letting Oboe fall back to OpenSLES (which has
   its own silent-drop failure modes on some devices), and add getState
   + getXRunCount to the playout heartbeat so we'll see silent stream
   disconnects instead of reading zeros forever.

3) engine.rs recv task: dump the first ~10s of post-AGC decoded PCM to
   `<app_data_dir>/decoded.pcm` as raw i16 LE so we can adb pull it and
   play it back locally:
       adb shell run-as com.wzp.desktop cat .wzp/decoded.pcm > decoded.pcm
       ffmpeg -f s16le -ar 48000 -ac 1 -i decoded.pcm decoded.wav
   This divorces "is our decoder actually producing audible audio" from
   "is Android's audio stack playing it". If the recorded WAV sounds
   correct when played on a laptop, the decoder is fine and 100% of the
   remaining bug surface is AudioManager / Oboe routing.

4) engine.rs: also log when spk_muted=true blocks the write. User
   reported the Speaker button in the UI has inconsistent semantics
   between desktop and android — adding this log rules out the accidental
   "first click muted playback" theory for good.
2026-04-09 21:24:26 +04:00
Siavash Sameni
96be740fd9 diag(android-audio): aggressive logging across the whole Oboe pipeline
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User confirmed: mac hears android, android does not hear mac. So Oboe
capture works end-to-end but Oboe playout on Android silently drops
audio even though QUIC forwards the packets. Archaeology on the legacy
wzp-android crate also revealed that the "last known good" Android audio
path NEVER used Oboe in production — it used Kotlin AudioRecord +
AudioTrack via JNI, and cpp/oboe_bridge.cpp was dead code. So every time
we've "tested" Oboe end-to-end this week was the first production use,
and any of its config knobs could be the bug.

Instrumenting every stage of the pipeline so one smoke-test log dump can
isolate the layer at fault:

C++ (oboe_bridge.cpp)
  - Log the ACTUAL stream parameters after openStream for both capture
    and playout (sample rate, channels, format, framesPerBurst,
    framesPerDataCallback, bufferCapacityInFrames, sharing, perf mode).
    Oboe may silently override values we requested — e.g. if we ask for
    48kHz mono but the device gives us 44.1kHz stereo our 960-sample
    frames are the wrong duration and the pipeline drifts.
  - Capture callback: on cb#0 log sample range+RMS of the first frame
    to prove we get real mic data (not zeros). Every 50 callbacks
    (~1s at 20ms burst) log calls, numFrames, ring available_write,
    bytes actually written, ring_full_drops, total_written.
  - Playout callback: on cb#0 log numFrames + ring state. On the FIRST
    non-empty read log sample range+RMS so we can tell if the samples
    coming out of the ring are real audio or zeros. Every 50 callbacks
    log calls, nonempty count, numFrames, ring available_read,
    underrun_frames, total_played_real.

Rust wzp-native (src/lib.rs)
  - wzp_native_audio_write_playout now logs the first 3 writes and then
    every 50th: in_len, written, sample range, RMS, ring write/read
    cursors before, available_read and available_write after. Reveals
    ring-overflow and whether the engine is actually handing us audio.
  - Minimal android logcat shim via __android_log_write extern — no
    new crate dependency.
  - AudioBackend grows a `playout_write_log_count` AtomicU64 to gate
    the write-side log throttle.

Rust engine.rs (android branch)
  - Recv task: log sample range + RMS for the first 3 decoded PCM
    frames and then every 100th. Reveals whether decoder.decode is
    producing real audio or silent buffers.
  - Recv task: if audio_write_playout returns fewer samples than we
    handed it (partial write → ring nearly full) warn about it in the
    first 10 frames.
  - Recv heartbeat every 2s: recv_fr, decoded_frames, last_decode_n,
    last_written, written_samples, decode_errs, codec.

Expected flow in a healthy log:
  capture cb#0: numFrames=960 range=[-1200..900] rms=180          ← mic OK
  capture stream opened: actualSR=48000 Ch=1 ...                   ← no override
  playout stream opened: actualSR=48000 Ch=1 ...
  CallEngine::start invoked ... → connected → audio started
  recv: first media packet received ...
  recv: decoded PCM sample range decoded_frames=1 range=[-300..250] rms=92
  playout WRITE #0: in_len=960 written=960 range=[-300..250] rms=92
  playout FIRST nonempty read: to_read=960 range=[-300..250] rms=92
  playout heartbeat: calls=50 nonempty=50 underrun=0 ...
  recv heartbeat: decoded_frames=100 last_written=960 ...

If any of those are missing/zero we know the exact stage to fix.
2026-04-09 21:13:29 +04:00
Siavash Sameni
8c4d640f89 fix(android): playout Usage::Media + relay CallSetup advertises real IP
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Three real bugs, one smoke-test session's worth of progress.

1. RELAY: wrong advertised addr in CallSetup
   The direct-call CallSetup computed `relay_addr = addr.ip()` where
   `addr = connection.remote_address()` — i.e. the CLIENT'S IP, not the
   relay's. So the relay was telling both parties "the call room is at
   the answerer's IP:4433", which meant each client dialed either the
   other client (no server listening) or themselves. Both endpoint.connect
   calls hung forever and the call never happened.
   Fix: compute the relay's own advertised IP once at startup. If the
   listen addr is 0.0.0.0, probe the primary outbound interface via the
   classic UDP-bind-and-connect(8.8.8.8:80) trick to discover the LAN
   IP the OS would use to reach external hosts. Thread the resulting
   advertised_addr_str into the CallSetup sender for both parties.

2. RELAY: accept loop serialized QUIC handshakes
   Previously the main accept loop called `wzp_transport::accept` which
   did both `endpoint.accept().await` AND `incoming.await` (the server-
   side QUIC handshake). A single slow handshake therefore blocked every
   subsequent client from being accepted. Unroll the helper here and
   move `incoming.await` into the per-connection spawned task, so every
   handshake runs in parallel. Also log "accept queue: new Incoming",
   "QUIC handshake complete", and "QUIC handshake failed" so we can tell
   immediately whether a client's packets are reaching the relay at all.

3. ANDROID: playout was routed to the silent in-call stream
   The Oboe playout stream was configured with Usage::VoiceCommunication,
   which routes to the Android in-call earpiece stream. That stream is
   silent unless the Activity has called AudioManager.setMode(
   IN_COMMUNICATION) and, even then, only the earpiece/BT headset get
   audio (not the loud speaker). Result: android→mac calls worked
   because mac had a normal media output, but mac→android calls were
   silent even though packets flowed through the relay just fine.
   Switch to Usage::Media + ContentType::Speech so Oboe routes to the
   loud speaker and uses the media volume slider. A later polish step
   will wire setMode + setSpeakerphoneOn from MainActivity.kt so we can
   go back to VoiceCommunication for AEC and proximity-sensor routing.

Plus: heartbeat tracing every 2s in the send/recv tasks — frames_sent,
last_rms, last_pkt_bytes, short_reads on the send side; decoded_frames,
last_decode_n, last_written, decode_errs on the recv side. Will make the
next "no sound" regression trivial to localize.
2026-04-09 20:55:10 +04:00
Siavash Sameni
49f101d785 fix(android): reuse signal endpoint for direct-call media connection
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Direct-call accept hangs forever at the QUIC handshake on Android. Logs
from d7b37a5 showed:
  CallEngine::start (android) invoked relay=172.16.81.172:4433 room=call-…
  resolved relay addr
  identity loaded
  endpoint created, dialing relay   ← reached
                                    ← nothing, 90s+, no error
The "connect failed" and "QUIC connection established" log lines never
fire, meaning endpoint.connect_with(…).await never makes progress.

Repro is 100%: SFU room join (one endpoint) works perfectly; direct call
(opens a SECOND quinn::Endpoint on top of the signal one) hangs in the
QUIC handshake. Creating two quinn::Endpoints on Android's AAudio-adjacent
UDP stack apparently causes the second one's datagrams to never reach the
relay (the server never sees the Initial packet). Rather than fight the
platform, quinn is happy to multiplex multiple Connections on a single
Endpoint — so we reuse the signal endpoint for the media connection.

- SignalState now stores the quinn::Endpoint alongside the QuinnTransport.
  register_signal populates both at the same time.
- CallEngine::start (both android and desktop branches) takes an
  Option<wzp_transport::Endpoint>. Some → reuse (direct-call path, after
  register_signal). None → create fresh (SFU room join path).
- The connect tauri command reads state.signal.endpoint and threads it
  through to CallEngine::start, so the direct-call auto-connect (fired by
  the "setup" signal-event in main.ts) lands on the existing UDP socket.
- wzp_transport re-exports quinn::Endpoint so wzp-desktop doesn't need to
  depend on quinn directly.
- Also wraps the android connect in tokio::time::timeout(10s) so future
  hangs become deterministic "connect TIMED OUT" errors in logcat
  instead of silent deadlock.

Same fix applies verbatim to the desktop client — the user suspects
direct call is broken there too and this was likely always the cause,
just never surfaced because desktop was only tested via SFU rooms.
2026-04-09 20:29:51 +04:00
Siavash Sameni
d7b37a5749 diag: tracing for direct-call signal loop + CallEngine::start stages
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User reports tapping "answer" on an incoming direct call does nothing
visible, and suspects the same may affect desktop. The signal recv loop
had no tracing at all, so we can't tell whether CallSetup is being
received, whether the recv loop died silently, or whether
CallEngine::start is failing between "identity loaded" and
"connected to relay, handshake complete".

- register_signal recv loop now logs every message type with fields
  (CallRinging, DirectCallOffer, DirectCallAnswer, CallSetup, Hangup,
  unhandled), plus a warn! on recv errors and a final warn when the
  loop exits.
- place_call / answer_call commands log entry + success / error. The
  answer_call error path logs the underlying send_signal error so we
  can see it in logcat instead of only in the JS error toast.
- CallEngine::start android branch logs relay/room/alias on entry,
  logs "endpoint created, dialing relay" between create_endpoint and
  connect, "QUIC connection established, performing handshake" between
  connect and perform_handshake, and promotes all three potential
  failures to explicit error! logs so a silent hang / error becomes
  visible in logcat.

No functional changes — pure diagnostics. Stacks on b35a6b7 (the Oboe
stack-pointer-escape fix) so build #43 carries both.
2026-04-09 19:17:03 +04:00
Siavash Sameni
0105b0fbf3 phase 3(android): RECORD_AUDIO permission + runtime request in MainActivity
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Oboe fails silently to open the AAudio input stream without
android.permission.RECORD_AUDIO, so the call audio would never actually
flow even after phase 3's engine wiring.

- AndroidManifest.xml: declare RECORD_AUDIO and MODIFY_AUDIO_SETTINGS, and
  android.hardware.microphone as a required feature. These files are the
  cargo-tauri-generated scaffold — nothing in .gitignore excludes them, so
  the intended Tauri 2 mobile workflow is to commit them once populated.

- MainActivity.kt: override onCreate to call ActivityCompat.requestPermissions
  for the audio perms on first launch. The dialog shows exactly once; the
  grant is persisted per-package. onRequestPermissionsResult logs the
  outcome so we can spot failures in logcat.

A full native Tauri permission plugin integration is deferred to
Step 6 (polish) together with notifications, icon, and background service.
2026-04-09 19:00:12 +04:00
Siavash Sameni
5beea7de40 phase 3(android): unify connect/disconnect/toggle_*/get_status commands
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Step 3 of the Tauri Android rewrite was still returning "audio backend not
yet wired on Android (step 3)" because the cfg-gated Android stubs for
connect/disconnect/toggle_mic/toggle_speaker/get_status were shadowing the
real commands. Now that CallEngine::start() has a real Android body (phase
3, commit fdbe502), the gates are unnecessary.

- Drop the #[cfg(not(target_os = "android"))] gates from all five
  engine-backed Tauri commands.
- Delete the Android stub block (~50 LOC of "not connected" boilerplate).
- Ungate `use engine::CallEngine;` and the AppState.engine field so both
  targets share the same Mutex<Option<CallEngine>>.
- CallEngine::stop() now calls crate::wzp_native::audio_stop() on Android so
  the mic + speaker are released between calls, matching the desktop
  behaviour where dropping _audio_handle tears down CPAL.

Direct-call flow on Android: peer sends DirectCallOffer → user accepts via
answer_call → relay sends signal "setup" event → main.ts auto-invokes
connect(relay, room) → CallEngine::start() runs the Android branch →
wzp_native::audio_start() brings up Oboe → send/recv tasks stream PCM
through the dlopen boundary.
2026-04-09 18:53:54 +04:00
Siavash Sameni
fdbe502524 phase 3(android): wire CallEngine::start to wzp-native audio FFI
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Replaces the Android-side CallEngine::start() stub with a real implementation
that mirrors the desktop start() body but routes all PCM through the
standalone wzp-native cdylib loaded at startup via libloading instead of
using CPAL.

- desktop/src-tauri/src/wzp_native.rs: new module with a static
  OnceLock<libloading::Library> + cached raw fn pointers for every symbol
  we need (version, hello, audio_start/stop, read_capture, write_playout,
  is_running, capture/playout_latency_ms). init() resolves everything once
  at startup; accessors return default values if init() never ran.

- desktop/src-tauri/src/lib.rs: drop the inline dlopen smoke test, add
  `mod wzp_native;` behind target_os="android", and invoke
  wzp_native::init() from the Tauri setup() callback so the library is
  loaded + all symbols cached before any CallEngine can touch audio.

- desktop/src-tauri/src/engine.rs: the Android #[cfg] branch of
  CallEngine::start() now does the full QUIC handshake + signal loop +
  Opus send/recv tasks, calling wzp_native::audio_start() /
  audio_read_capture() / audio_write_playout() instead of the desktop
  CPAL rings. SyncWrapper now holds a placeholder Box<()> on Android
  because the audio backend lives in a process-global singleton inside
  libwzp_native.so rather than being owned per-engine.

Next step: build #39 on the remote docker builder and smoke-test on
Pixel 6 that the Connect button in the UI successfully brings up Oboe
and streams audio through the dlopen boundary.
2026-04-09 18:42:27 +04:00
Siavash Sameni
7cc53aedc7 refactor(android): split C++ into wzp-native cdylib, loaded at runtime
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Phase 1 of the big refactor. Escape the Tauri Android
__init_tcb+4 symbol leak (rust-lang/rust#104707) by making
wzp-desktop's Android .so pure Rust — ZERO cc::Build, no cpp/ files,
no C++ in the rustc link step. All future C++ (Oboe audio bridge)
lives in a new standalone cdylib crate `wzp-native` which is built
with cargo-ndk (the same path the legacy wzp-android crate uses
successfully on the same phone + same NDK), copied into Tauri's
gen/android/app/src/main/jniLibs at build time, and dlopened by
wzp-desktop at runtime via libloading.

Changes in this commit:
- NEW crate crates/wzp-native/ with crate-type = ["cdylib"] only
  (no staticlib, no rlib — rust#104707 shows mixing staticlib with
  cdylib leaks non-exported symbols, which is the original bug
  source). Phase 1 scaffold has TWO extern "C" functions:
    wzp_native_version() -> i32            (returns 42)
    wzp_native_hello(buf, cap) -> usize    (writes a string)
  So we can verify dlopen + dlsym + cross-.so FFI end-to-end
  before adding any real C++.
- desktop/src-tauri/cpp/ directory DELETED (7 files gone).
- desktop/src-tauri/build.rs reduced to just the git hash capture
  + tauri_build::build(). No more cc::Build of any kind.
- desktop/src-tauri/Cargo.toml: drop cc from build-dependencies,
  add libloading = "0.8" as an Android-only runtime dep.
- desktop/src-tauri/src/lib.rs Builder::setup() now (on Android only)
  dlopens libwzp_native.so, calls wzp_native_version() and
  wzp_native_hello(), and logs the result:
    "wzp-native dlopen OK: version=42 msg=\"hello from wzp-native\""
  If this log appears in logcat when the app launches and the home
  screen still renders, the split-cdylib pipeline is validated and
  Phase 2 (port the Oboe bridge into wzp-native) can proceed.
- scripts/build-tauri-android.sh: insert a `cargo ndk -t arm64-v8a
  build --release -p wzp-native` step before `cargo tauri android
  build`, with `-o desktop/src-tauri/gen/android/app/src/main/jniLibs`
  so the resulting libwzp_native.so lands in the place gradle will
  package into the final APK.
- Workspace Cargo.toml: add crates/wzp-native to [workspace] members.

Phase 2 (separate commit, only if Phase 1 works):
- Copy cpp/oboe_bridge.{h,cpp} + getauxval_fix.c from the legacy
  wzp-android crate into crates/wzp-native/cpp/.
- Add cc = "1" as a build-dependency on wzp-native (safe: it's a
  single-cdylib crate with no staticlib, so no symbol leak).
- Add build.rs that compiles the Oboe C++ and the wzp-native Rust
  FFI exposes the audio start/stop/read/write functions.
- wzp-desktop::engine.rs dlopens wzp-native at CallEngine::start,
  uses its audio functions instead of CPAL on Android.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 18:02:53 +04:00
Siavash Sameni
711137da96 fix(android): -Wl,--exclude-libs,ALL + --no-whole-archive to stop symbol leak
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llvm-nm on the crashing .so confirmed the research's smoking gun theory:

  000000000130c1f0 t _Z10__init_tcbP10bionic_tcbP18pthread_internal_t
  0000000000000000 a pthread_create.cpp
  0000000001331108 t pthread_create

All lowercase 't' (= LOCAL text symbols), zero UND dynamic references
for pthread_create. So rustc's link step is pulling bionic's own
pthread_create.cpp compilation unit out of libc.a as a whole-archive
inclusion and binding those symbols locally inside our .so, instead
of letting them stay UND and resolved against libc.so at dlopen time.

Rust's libstd thread::spawn then calls the LOCAL (broken) pthread_create
which calls the LOCAL __init_tcb with arguments set up for bionic's
static-executable layout — crashes at __init_tcb+4 with SEGV_ACCERR.

`-Wl,--exclude-libs,ALL` tells the linker to make symbols from static
archives NOT appear in the dynamic symbol table of the output .so.
`-Wl,--no-whole-archive` tells it to only pull archive objects that
satisfy undefined references, not include the whole archive blindly.

If this works, the symbol table should show pthread_create as UND
(or at least not locally bound) and the app should launch. If it
doesn't, the remaining fallback is the research's action #3 —
extract the C++ into its own upstream cdylib crate built with
cargo-ndk, and dlopen it from the Tauri cdylib at runtime.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 17:45:35 +04:00
Siavash Sameni
6071eb1b02 fix(android): drop staticlib from crate-type — root cause of __init_tcb crash
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External research (per rust-lang/rust#104707) pointed at this as the
highest-probability cause of our byte-identical __init_tcb+4 /
pthread_create SIGSEGVs:

> Having 'staticlib' alongside 'cdylib' in crate-type leaks non-exported
> symbols from the staticlib into the cdylib's symbol table. For a
> Tauri Android cdylib, that means bionic's private pthread_create /
> __init_tcb code — which got pulled in statically from libc.a the
> moment any cc::Build C++ file added C++-linkage overhead — ends up
> bound LOCALLY inside our .so instead of being resolved dynamically
> against libc.so at dlopen time.

Symptoms that match the theory exactly:
- llvm-nm on the crashing .so shows __init_tcb and pthread_create as
  LOCAL symbols with C++ name mangling (bionic's own pthread_create.cpp)
- Adding any cc::Build cpp(true) step reliably triggers the crash,
  independent of which linker (android24-clang vs android26-clang) or
  which libc++ linkage (shared/static/none)
- The legacy wzp-android crate (["cdylib", "rlib"]) works fine on the
  same phone with the same NDK + Rust toolchain + Oboe C++ code
- tauri.conf.json bundle.android.minSdkVersion=26 propagates to
  gradle but the .so still crashes byte-identically

Drop 'staticlib' from crate-type. If we ever need it for iOS, re-add
behind a target.'cfg(target_os = "ios")' gate. The desktop binary
still links against the rlib, so the bin target on macOS/Linux/Windows
is unaffected.

Source: https://github.com/rust-lang/rust/issues/104707

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 17:38:49 +04:00
Siavash Sameni
c9cd043657 test: tauri.conf.json bundle.android.minSdkVersion=26 + cpp_smoke.cpp c++_shared
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User theory: tauri-cli hardcodes minSdkVersion=24 into its rustc
invocation regardless of gradle build.gradle.kts, .cargo/config.toml,
or env var overrides — but DOES read from tauri.conf.json's
bundle.android block. That would explain why every cc::Build C++
compile crashed with __init_tcb+4 via pthread_create: API-24 bionic's
.init_array routines for the linked-in .init_array clash with the
pthread_create state tao later expects.

This commit applies the fix AND re-adds the smallest known crashing
variant (E.1 with cpp_link_stdlib('c++_shared')) so the test has one
clear failure mode to compare against:

  tauri.conf.json bundle:
    "android": { "minSdkVersion": 26 }

  build.rs (on android target):
    - hello.c           (plain C, worked in Step A)
    - getauxval_fix.c   (plain C, worked in Step D)
    - hello2.c          (plain C, worked in Step D+1)
    - cpp_smoke.cpp     (C++ via cc::Build .cpp(true), crashed in E.1)

Also re-emits the libc++_shared.so copy into gen/android jniLibs so
the runtime linker can resolve the NEEDED entry cc-rs added via
cpp_link_stdlib('c++_shared').

If this launches → theory validated, proceed with Oboe integration.
If this crashes → need to keep digging.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 16:58:37 +04:00
Siavash Sameni
6dd62c94c9 step D+1: add third trivial C static lib (hello2.c)
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Step D (hello.c + getauxval_fix.c) launches cleanly. E.minus-1
(hello.c + getauxval_fix.c + cpp_smoke.c) crashes. All three are
plain-C trivial single-function files.

Theory: the regression is triggered by having 3 or more cc::Build
static libs in a Tauri Android cdylib, regardless of what the libs
contain. Test: clone hello.c as hello2.c (same content, different
symbol) and add a third cc::Build step compiling it. If this crashes,
the trigger is just the number of static libs. If it launches, there's
something magical about cpp_smoke.c specifically (unlikely — it was
near-identical content).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 16:51:50 +04:00
Siavash Sameni
4c998312aa regression check: revert build.rs to exact Step D state
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Verify the Step D baseline still launches after the environment mutations
we may have caused during the E bisection (docker image rebuild, tauri-cli
version drift, etc). Build.rs is now byte-identical to commit a852cad
(Step D) except for the git hash capture block that already existed at
that point.

If this launches cleanly → the cpp_smoke addition genuinely breaks
something, bisection continues.
If this crashes → the environment regressed between Step D and now,
and we need to rebuild the docker image to an earlier snapshot.
2026-04-09 16:45:34 +04:00
Siavash Sameni
22701830c2 step E.minus-1: cpp_smoke renamed to .c and compiled as plain C
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c++_shared crashed, c++_static crashed, no stdlib crashed. The remaining
variable isolated to cc::Build::new().cpp(true) itself is the C++
compile-mode invocation of clang++. Rename cpp_smoke.cpp → cpp_smoke.c
and drop .cpp(true), leaving a plain-C cc::Build that compiles the
exact same bytes (minus the 'extern "C"' linkage spec which is C++-
only syntax).

This is structurally identical to Step A (hello.c), which worked. If
THIS build launches, the diff between 'works' and 'crashes' is purely
the .cpp(true) mode — something clang++ does differently at compile
or link time when producing object files for a Tauri Android cdylib.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 16:38:29 +04:00
Siavash Sameni
47a037368c step E.0: drop cpp_link_stdlib entirely (no libc++ linkage)
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c++_shared crashed. c++_static also crashed. Both have libc++ code
landing in the final .so — one as a NEEDED dynamic lib, the other
bundled statically. So the trigger isn't the NEEDED entry specifically,
it's libc++ being present in any form.

cpp_smoke.cpp is just 'extern "C" int wzp_cpp_hello() { return 42; }'
with zero C++ features used, so we can drop cpp_link_stdlib completely
and the compile still succeeds. No libc++ .a or .so referenced at all.

If this crashes: the trigger is cc::Build::new().cpp(true) switching
rustc's final linker driver from clang to clang++ (which pulls in
different default libraries).

If this launches: the trigger is libc++'s own static initializers or
the libc++ code itself doing something that breaks our .so at dlopen
time, and we have a path forward — C++ code that doesn't need libc++
(e.g., a thin C++ bridge to Oboe that uses only POD types at the
boundary, with all the STL stuff confined to Oboe's own compilation
unit which would still need libc++...). More likely we still need a
C-only audio interface like raw AAudio via the ndk Rust crate.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 16:31:53 +04:00