DirectByteBuffer.clear() crashes with null pointer in ART's JIT OSR
compiled code on Android 16. Revert AudioPipeline to use the original
ShortArray writeAudio/readAudio path.
The DirectByteBuffer JNI functions remain in WzpEngine.kt and
jni_bridge.rs for future use once the OSR issue is resolved.
The original SIGBUS from ART GC is rare (~1 crash per 8 min call)
and doesn't warrant the DirectByteBuffer approach until we can
allocate the buffer as a class field outside the hot loop.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Adds nativeWriteAudioDirect / nativeReadAudioDirect JNI functions
that accept a DirectByteBuffer instead of ShortArray. The buffer's
native memory is accessed directly by Rust via pointer — no
GetShortArrayRegion / SetShortArrayRegion, no GC-managed array
copies on the audio hot path.
This fixes SIGBUS crashes on Android 16 where ART's concurrent
mark-compact GC crashes when flipping thread roots during JNI
array operations on MAX_PRIORITY audio threads.
Old ShortArray methods kept for backward compatibility.
AudioPipeline switched to use Direct variants.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Rust tracing subscriber was never initialized — all info!/warn!/error!
calls in the engine went to /dev/null. This meant our send/recv health
logging was invisible and we couldn't confirm the congestion fix was
active.
Now initializes tracing-android layer on first nativeInit(), routing
all Rust logs to logcat under tag "wzp_android". Also expanded logcat
filter in DebugReporter to capture engine-level log lines.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Root cause: send_media() returns Err(Blocked) when QUIC congestion
window is full. The send task treated ANY send error as fatal (break),
killing the entire call. Now send errors drop the packet and continue.
Also hardened recv task to survive transient errors and added health
logging (recv gap tracking, periodic stats) to both send and recv.
Relay: added comprehensive debug logging — recv gaps, lock contention,
forward latency, send errors — all per-participant with 5s stats.
Other changes:
- AEC toggle in Settings (persisted, applied on next call)
- Debug report: records call audio (WAV), RMS histogram (CSV), logcat,
stats. Emailed as zip via Android share intent after call ends.
- Replaced LinearProgressIndicator with Box (compose version compat)
- FileProvider for sharing debug zip attachments
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Mic mute: the send loop now zeros the capture buffer when muted instead
of relying on write_audio() to skip writes. Previously stale ring data
and AGC amplification of near-silence caused crackling artifacts.
AEC: attach Android's hardware AcousticEchoCanceler to the AudioRecord
session. Also attach NoiseSuppressor when available. Both are released
on capture stop.
Room UI: deduplicate participants by fingerprint so ghost entries from
stale relay state don't show duplicate names.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
UI: deduplicate room participants by fingerprint so ghost entries from
stale relay state don't show duplicates.
Engine: after select! ends, call close_now() + connection.closed() with
500ms timeout to wait for the relay to acknowledge the CONNECTION_CLOSE.
Previously the close frame was queued but the runtime died before quinn
could retransmit if the first packet was lost.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
o.optString("alias", null) returns the string "null" when the JSON value
is JSON null. Use o.isNull() check first. Also handle empty fingerprint
edge case with "unknown" fallback.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Settings now uses draft state — changes only persist on explicit Save
- Back button discards unsaved changes
- Added applyServers() for batch server updates
- Added missing alias field to CallOffer in featherchat tests
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Add SettingsScreen with identity (alias, key backup/restore), audio defaults,
server management, network prefs, and default room
- SettingsRepository persists all settings via SharedPreferences
- Auto-generate random display names on first launch (e.g. "Swift Wolf")
- Thread alias through CallOffer → relay handshake → RoomUpdate broadcast
- Derive caller fingerprint from identity key in relay handshake (fixes null
fingerprints when --auth-url is not set)
- Persist identity seed for stable fingerprints across reconnects
- Add alias field to SignalMessage::CallOffer (serde default for backward compat)
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Add RoomUpdate signal message to wzp-proto with participant count + list
- Add RoomParticipant struct (fingerprint + optional alias)
- Store fingerprint/alias in relay Participant struct
- Broadcast RoomUpdate to all room members on join and leave
- Add signal recv task in Android engine to handle RoomUpdate
- Surface room_participant_count + room_participants in CallStats JSON
- Show "X in room" with participant names in Android in-call UI
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Wire CallService foreground service for background calls (microphone type)
- Add Voice Volume + Mic Gain sliders (-20 to +20 dB) applied in Kotlin
- Connect AudioRouteManager for real speaker toggle via AudioManager
- Feed quinn QUIC RTT into PathMonitor, display Loss/RTT/Jitter from live data
- Nuclear teardown between calls — recreate engine + audio pipeline each call
- Fix re-entrant teardown loop from CallService notification callback
- Park audio threads as daemons to avoid libcrypto TLS destructor crash on exit
- Remove duplicate wakelocks from Activity (service owns them now)
- Strip AEC + denoise from capture path, keep AGC only (incremental approach)
- Fix .so copy target: libwzp_android.so not libwzp.so
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Wire AutoGainControl on both capture (mic → encode) and playout
(decode → speaker) paths to normalize volume levels
- Add server list with add/remove custom server dialog
- Add IPv4/IPv6 preference toggle for DNS resolution
- Resolve DNS hostnames to IP in Kotlin before passing to Rust engine
- Revert to IP addresses for default servers (DNS still broken on QUIC)
AGC confirmed working — voice levels noticeably improved in testing.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Partial wake lock + WiFi high-perf lock during calls — audio
continues when screen is off / phone is locked
- Server selector: toggle between LAN (172.16.81.175) and Pangolin
(pangolin.manko.yoga) before connecting
- Room name editable in idle screen
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- AudioPipeline: Kotlin AudioRecord/AudioTrack on JVM threads, PCM
shuttled to Rust via lock-free ring buffers + JNI
- FEC: RaptorQ fountain codes on encode (5 frames/block, 20% repair
ratio for GOOD profile), decoder feeds repair symbols for recovery
- Real audio level meter from mic RMS (replaces fake animation)
- Room name editable in UI (default: "android")
- Relay changed to pangolin.manko.yoga:4433
- Stats overlay shows FEC recovered count
- CallState now synced from polled stats (fixes "Connecting" stuck bug)
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
pthread_create crashes on Android due to static bionic __init_tcb stubs
in the Rust std prebuilt rlibs. This is unfixable without rebuilding std.
Solution: run the entire call (QUIC connect, handshake, media send/recv)
on a single tokio current_thread runtime. The JNI startCall() now blocks,
so Kotlin dispatches it to Dispatchers.IO (JVM thread, not pthread).
Audio pipeline temporarily simplified to silence frames — will restore
once threading is solved (either via Java Thread or rebuilding std).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
The getauxval override (dlsym wrapper) fixes SIGSEGV in
init_have_lse_atomics at library load time. The current_thread
tokio runtime avoids SEGV_ACCERR in pthread_create/__init_tcb.
Both fixes are required together.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Multi-thread tokio runtime crashes with SEGV_ACCERR in __init_tcb
during pthread_create on Android (static bionic stubs from CRT).
Switch to current_thread runtime which runs network I/O on the
calling thread without spawning additional OS threads.
Also: clean up build.rs — use only libc++_shared.so (dynamic),
remove getauxval_fix.c hack, remove static c++/c++abi linking.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- CallActivity no longer auto-starts a call on launch
- CallViewModel lazily inits engine only when startCall() is called
- nativeGetStats nullable return handled safely in Kotlin
- Removed tracing_subscriber::fmt() which panics on Android (no stdout)
- All JNI calls wrapped in try/catch on Kotlin side
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- New wzp-android crate with Oboe C++ backend, lock-free SPSC ring buffers,
engine orchestrator, codec pipeline, and Android Gradle project structure
- AEC (NLMS adaptive filter), AGC (two-stage with fast attack/slow release),
windowed-sinc FIR resampler replacing linear interpolation (wzp-codec)
- Opus encoder tuning: complexity 7 default, set_expected_loss support
- Mobile jitter buffer: asymmetric EMA (fast up/slow down), handoff spike
detection with 2s cooldown, configurable safety margin
- Network-aware quality control: cellular-specific thresholds, faster
downgrade on cellular, proactive tier drop on WiFi→cellular handoff,
FEC ratio boost during network transitions
- Handoff detection in PathMonitor via RTT jitter spike analysis
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
