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.
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
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
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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>
Build #20 failed to compile on Android because I over-gated the
wzp_proto imports to non-Android. resolve_quality() is compiled on
every platform (it's outside the CallEngine impl) and references
QualityProfile + CodecId — both platform-independent types from
wzp_proto. Move those back to an unconditional import. tracing stays
gated (only the desktop start() body logs; the Android stub is silent).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Third incremental variable. Previously the engine module was cfg-gated
out of the Android build entirely (`#[cfg(not(target_os = "android"))]
mod engine;` in lib.rs). Now it's always compiled, so any link-time
effect of having engine.rs in the compilation unit can be measured
against the working baseline from build #19.
Changes kept deliberately small:
- lib.rs: drop the cfg gate on `mod engine;`. `use engine::CallEngine`
stays gated because the Android-specific connect/disconnect/... stubs
in lib.rs don't reference the type.
- engine.rs: the `wzp_client::{audio_io, call}` imports + CodecId +
QualityProfile are gated to non-Android (they require the `audio`
feature on wzp-client which Android doesn't pull in). On Android we
keep only the MediaTransport import for transport.close(). The impl
block now has two `start()` methods: the full CPAL-backed one for
desktop, and a 6-line Android stub that returns `Err("audio engine
not yet wired on Android")` so attempts to `connect` from the UI
fail cleanly.
Goal: verify that linking in the compiled engine module (plus the
types it references) on Android doesn't regress the working baseline.
Home screen should still render and register_signal should still work.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Spent 10+ builds chasing a __init_tcb+4 / pthread_create SIGSEGV after
adding the oboe audio backend. Every "fix" made things worse. Reverting
all Android-specific files to the state at 35642d1 (build #6), which
was the last commit where the Tauri Android app actually launched,
rendered the home screen, and successfully registered on a relay.
Reverted files (all back to their 35642d1 content):
- desktop/src-tauri/Cargo.toml (no build-dep cc, no tracing-android)
- desktop/src-tauri/build.rs (git hash only, no Oboe / cc build)
- desktop/src-tauri/src/lib.rs (engine cfg-gated on non-android)
- desktop/src-tauri/src/main.rs (two-line desktop entry)
- desktop/src-tauri/src/engine.rs (desktop-only audio setup)
- scripts/Dockerfile.android-builder (no android24→26 clang shim)
- scripts/build-tauri-android.sh (no linker env vars / manifest patch)
Deleted (were added between b314138 and e2e023d):
- desktop/src-tauri/cpp/getauxval_fix.c
- desktop/src-tauri/cpp/oboe_bridge.{h,cpp}
- desktop/src-tauri/cpp/oboe_stub.cpp
- desktop/src-tauri/src/oboe_audio.rs
Next: rebuild image on remote (to drop the baked-in clang shim), build
an APK, install on Pixel 6, verify the UI renders the same way build #6
did. From there we add features back ONE at a time so we can actually
bisect which one triggers the tao::ndk_glue crash. User's rule:
"if you want to change stack, change incrementally, so we can debug".
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
This is the big one — the Tauri Android app now has a real audio stack
capable of full-duplex VoIP, reusing the proven C++ Oboe bridge from the
legacy wzp-android crate.
Architecture:
- desktop/src-tauri/cpp/ — copies of oboe_bridge.{h,cpp}, oboe_stub.cpp,
and getauxval_fix.c from crates/wzp-android/cpp/. build.rs clones
google/oboe@1.8.1 into OUT_DIR and compiles the bridge + all Oboe
sources as "oboe_bridge" static lib, linking against shared libc++
(static would pull broken libc stubs that SIGSEGV in .so libraries).
- src/oboe_audio.rs — Rust side: an SPSC ring buffer matching the C++
bridge's AtomicI32 layout, plus OboeHandle::start() which returns
(capture_ring, playout_ring, owning_handle). The ring exposes the same
(available / read / write) methods as wzp_client::audio_ring::AudioRing
so CallEngine treats both backends interchangeably.
- src/engine.rs — compiled on every platform now. A cfg-switched type
alias picks wzp_client::audio_ring::AudioRing on desktop and
crate::oboe_audio::AudioRing on Android. The audio setup block has
three branches: VPIO/CPAL on macOS, CPAL on Linux/Windows, Oboe on
Android. Send/recv tasks are identical across platforms.
- src/lib.rs — removes all the "step 3 not done" Android stubs. The
engine module is no longer cfg-gated; connect / disconnect / toggle_mic
/ toggle_speaker / get_status are single implementations used by both
desktop and Android. Identity path resolves via app.path().app_data_dir()
from the Tauri setup() callback (already wired in step 1).
Runtime mic permission:
- scripts/build-tauri-android.sh now injects RECORD_AUDIO + MODIFY_AUDIO_
SETTINGS into gen/android/app/src/main/AndroidManifest.xml after init,
and overwrites MainActivity.kt with a version that calls
ActivityCompat.requestPermissions in onCreate. This is idempotent:
every build re-applies the patches so tauri re-init can't regress them.
Cargo.toml:
- cc is now an unconditional build-dep (build.rs runs on the host, so
target-gating build-deps doesn't work).
- wzp-client is now a dep on every platform. On Android it gets default
features only (no "audio"/"vpio") so CPAL isn't dragged in — oboe_audio
provides the capture/playout rings instead.
- tracing-android is added on Android so tracing events flow into logcat.
build.rs also gained embedded git hash (WZP_GIT_HASH) capture, which is
shown under the fingerprint on the home screen — already committed in
7639aaf, reinstated here alongside the Oboe build logic.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Three home-screen issues from the first Tauri Android APK:
1. Alias was empty (no seed-derived name).
Port the adjective+noun word lists from the old Kotlin SettingsRepository
into a `derive_alias()` helper that maps the first 4 bytes of the seed to
indices in those lists. Same seed → same alias forever, different seeds →
effectively random aliases — so reinstalls keep the user's identity AND
the friendly name they're used to.
2. Build identity was invisible — couldn't tell which APK was actually
installed (this caused us a lot of grief on the Kotlin app).
build.rs now captures `git rev-parse --short HEAD` and emits it as
`WZP_GIT_HASH`, exposed via a new `get_app_info` command. The frontend
stamps `build <hash> • <alias>` under the fingerprint on the home screen.
3. Register on relay failed with `Permission denied (os error 13)`.
Root cause: I hardcoded `/data/data/com.wzp.phone/files/.wzp` as the
identity dir, but the Tauri Android package id is `com.wzp.desktop` —
so the app was trying to write into another app's data directory and
getting EACCES at the filesystem layer. Fix: resolve the data dir from
Tauri's `path().app_data_dir()` API in the `setup()` callback and stash
it in a `OnceLock<PathBuf>`. Works on Android, macOS, Linux, Windows
without any cfg gymnastics.
Also: `get_app_info` returns the resolved `data_dir` so we can debug
storage issues from the UI (it's set as the build-hash element's title).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Tauri 2.x Mobile links the app as a cdylib loaded from a Java Activity, so
all of the Builder/command code has to live in a library crate. Move the
existing logic verbatim into src/lib.rs::run() and reduce src/main.rs to a
two-line desktop entry point that calls into it.
Cargo.toml gets a [lib] section (crate-types: staticlib + cdylib + rlib,
named wzp_desktop_lib) and the wzp-client dependency — which pulls CPAL +
VoiceProcessingIO — is moved behind cfg(not(target_os = "android")) so the
Android cdylib doesn't need an audio backend yet. Engine-backed Tauri
commands (connect/disconnect/toggle_mic/toggle_speaker/get_status) get
Android stubs that return clear "not yet wired" errors. The signaling
commands (register_signal/place_call/answer_call/get_signal_status/
ping_relay/get_identity) are platform-independent and unchanged.
Also: get_identity / register_signal now auto-create the seed if missing
instead of erroring with "connect to a room first", and the identity dir
resolves to /data/data/com.wzp.phone/files/.wzp on Android (proper
app-internal storage) vs \$HOME/.wzp on desktop.
Side note: src/main.rs was previously untracked — desktop builds were
working only because it existed in the local worktree. This commit fixes
that too.
Step 1 of the Android rewrite plan (tauri-mobile scaffold). No audio yet.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Desktop now shows codec badges like Android:
- Green TX badge: e.g. "Opus64k"
- Blue RX badge: e.g. "Opus24k"
Displayed in the stats line below the call controls.
Engine tracks tx_codec (set on encoder init) and rx_codec (updated
from incoming packet headers). Passed through EngineStatus → CallStatus
→ frontend.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
RoomParticipant now has optional relay_label field. Desktop client
groups participants by relay: "This Relay" (green dot) for local,
peer label (blue dot) for federated. Shows all relays in the chain
including intermediate ones.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Adds three new codec IDs (Opus32k=6, Opus48k=7, Opus64k=8) and
corresponding STUDIO_32K, STUDIO_48K, STUDIO_64K quality profiles.
All use 20ms frames with minimal FEC (10%) for maximum quality on
good networks.
Updated across: wire protocol (codec_id.rs), encoder/decoder
(opus_enc/dec.rs), adaptive codec switch (call.rs), CLI
(--profile studio-64k), desktop engine + UI slider (8 quality
levels from Studio 64k green to Codec2 1.2k red).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Adds a Quality dropdown (Auto / Opus 24k / Opus 6k / Codec2 3.2k /
Codec2 1.2k) to both the connect screen and settings panel. The
selected profile is passed through to the engine which configures
the encoder and decoder accordingly.
The desktop engine recv path now auto-switches the decoder codec
when incoming packets use a different codec than expected, enabling
cross-codec interop between clients on different quality settings.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Identicon generator:
- Deterministic 5x5 symmetric pattern from fingerprint hash
- HSL-derived colors, rendered as inline SVG
- Click any identicon to copy its fingerprint to clipboard
- Used for participants, user identity, and relay servers
Server identity (TOFU — Trust On First Use):
- Ping returns server fingerprint (QUIC peer certificate hash)
- First contact: auto-saved as known fingerprint
- Subsequent pings: compared against known fingerprint
- Lock icons: locked (verified), unlocked (new), warning (changed), red (offline)
- Fingerprint mismatch shows confirmation dialog before connecting
UI updates:
- Participants show identicons instead of letter avatars
- User identity shows identicon + fingerprint on connect screen
- Manage Relays shows identicon per server with lock status
- Relay button shows lock icon instead of colored dot
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- New ping_relay Tauri command: QUIC connect with 3s timeout, returns RTT ms
- Relay status shown next to input field: "42ms" (green) or "offline" (red)
- Auto-pings on app startup and debounced on relay input change
- Fix SyncWrapper dead_code warning with #[allow(dead_code)]
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
#7 Fingerprint shown before connecting — new get_identity command reads
~/.wzp/identity at startup (generates if missing). Click to copy.
#8 Recent rooms store (relay, room) pairs — clicking a chip fills both
fields. Settings panel shows relay alongside room name. Migrates
old string[] format automatically.
#9 Auto-reconnect on unexpected disconnect — exponential backoff
(1s, 2s, 4s... max 10s), up to 5 attempts. Yellow blinking dot
shows reconnecting state. Stops if user clicks hangup.
#10 Audio handle cleanup — CPAL handles stored in SyncWrapper (no more
mem::forget), dropped properly on CallEngine::stop().
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
The VPIO/CPAL audio handles were dropped at the end of start(),
killing the audio unit immediately. Audio I/O stopped working
after the first frame.
- Store audio handle in CallEngine via SyncWrapper
- Drop MutexGuard before returning from status() (Send future)
- Audio streams now live for the entire call duration
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Change status() from blocking_lock to async lock().await —
fixes "Cannot block the current thread from within a runtime" panic
that froze the call timer and broke audio
- Click fingerprint to copy to clipboard (both connect and settings screens)
- Show "Copied!" feedback on click
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
send_datagram() returns Err(Blocked) when the QUIC congestion window
is full. This is transient — the window reopens once ACKs arrive.
Previously, all send paths treated this as fatal (break/return),
which killed the send task and cascaded via tokio::select! to kill
the entire call.
Now: log warning, drop the packet, continue. Brief audio glitch
(20-100ms) instead of complete call death. FEC on the receiver
side recovers most dropped packets.
Fixed in:
- CLI run_live send task (continue + error counter)
- CLI run_file_mode send paths (2 locations)
- Desktop engine send task
Also hardened recv tasks: transient errors (non-closed/reset)
are survived instead of causing exit.
Matches the fix applied to Android client (engine.rs).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Audio level meter with log-scale RMS visualization
- Call duration timer
- VPIO (OS AEC) wired through to engine with fallback to CPAL
- "You" badge on own participant entry
- Recent rooms list (click to reuse)
- Enter key to connect from form fields
- Improved dark theme with pulse animation on status dot
- Settings persistence via localStorage (relay, room, alias, AEC, recent rooms)
- Fingerprint display on connect screen
- Keyboard shortcuts skip input fields
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- New desktop/ directory with Tauri v2 + Vite + TypeScript
- Rust backend: CallEngine wrapping wzp-client audio + transport
- Web frontend: connect screen, in-call screen with participants,
mic/speaker mute, keyboard shortcuts (m/s/q)
- Dark theme UI, settings persistence via localStorage
- Platform-aware --os-aec: warns on Windows/Linux (not yet implemented)
- Workspace updated to include desktop/src-tauri
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
