Bluetooth: wire existing AudioRouteManager SCO support through both app
variants. Replace binary speaker toggle with 3-way route cycling
(Earpiece → Speaker → Bluetooth). Tauri side adds JNI bridge functions
(start/stop/query SCO, device availability) and Oboe stream restart.
Network awareness: integrate Android ConnectivityManager to detect
WiFi/cellular transitions and feed them to AdaptiveQualityController
via lock-free AtomicU8 signaling. Enables proactive quality downgrade
and FEC boost on network handoffs.
Build: add --arch flag to build-tauri-android.sh supporting arm64,
armv7, or all (separate per-arch APKs for smaller tester binaries).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
The CallSetup enum gained peer_direct_addr and peer_local_addrs
in Phase 5.5 but the wzp-android signal recv match arm was never
updated, breaking cargo ndk builds.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Phase 3c mirrors Phase 3b on the Android receive path. With Phase 0-3b
landed on desktop + Android encoder, this commit completes codec-layer
loss recovery on the Android decoder side.
Architectural difference vs desktop: engine.rs has NO jitter buffer.
The recv task reads packets directly from the transport via
recv_media().await and writes decoded audio straight into the playout
ring. There is no PlayoutResult::Missing equivalent. Gap detection
therefore has to be done via sequence-number tracking — when a packet
arrives with seq > expected_seq, the frames in between are missing and
we attempt to reconstruct them via DRED before decoding the newly-
arrived packet.
Implementation:
Imports & types:
- Added wzp_codec::AdaptiveDecoder, wzp_codec::dred_ffi::{
DredDecoderHandle, DredState} imports.
- Changed the `decoder` local from Box<dyn AudioDecoder> (via
wzp_codec::create_decoder) to concrete AdaptiveDecoder::new(profile).
Same reasoning as Phase 3b: reconstruct_from_dred is an inherent
method, not a trait method, so we need the concrete type.
Recv task state (all task-local, no new struct fields):
- dred_decoder: DredDecoderHandle
- dred_parse_scratch: DredState (reused, overwritten per parse)
- last_good_dred: DredState (cached most-recent valid state)
- last_good_dred_seq: Option<u16>
- expected_seq: Option<u16> (for gap detection)
- dred_reconstructions: u64 (telemetry)
- classical_plc_invocations: u64 (telemetry)
Recv loop body (Opus source packets only):
1. Parse DRED from the new packet first so last_good_dred reflects
the freshest state available for gap recovery.
2. Detect a gap: gap = pkt.seq.wrapping_sub(expected_seq). Cap at
MAX_GAP_FRAMES = 16 (320 ms) to avoid huge wraparound scenarios.
3. For each missing seq in the gap:
offset = (last_good_dred_seq - missing_seq) * frame_samples
if 0 < offset <= last_good_dred.samples_available():
reconstruct_from_dred + write to playout ring
bump dred_reconstructions
else:
decoder.decode_lost (classical PLC) + write + bump plc counter
4. Decode the current packet normally and write to playout ring
(unchanged from Phase 2).
5. Update expected_seq = pkt.seq.wrapping_add(1).
Profile-switch handling: when the incoming codec changes (triggering
decoder.set_profile), reset last_good_dred_seq and expected_seq to
None. The cached DRED state is tied to the old profile's frame rate
and would produce wrong offsets after the switch; starting fresh is
correct.
Decode-error fallback: the existing `Err(e) => decode_lost` branch
now also increments classical_plc_invocations so the counter
accurately reflects all PLC invocations (gap-detected AND decode-
error-triggered).
Telemetry (CallStats additions):
- stats.dred_reconstructions: u64
- stats.classical_plc_invocations: u64
Both updated on every packet arrival in the existing stats.lock()
block alongside frames_decoded/fec_recovered, so the Android UI and
JNI bridge already have these values without any further plumbing.
The periodic recv stats log now includes both counters.
Ordering note: DRED gap reconstruction happens BEFORE decoding the new
packet's audio because the playout ring is FIFO. Gap samples must be
written before the new packet's samples so temporal order is preserved.
Out-of-order late arrivals (seq < expected_seq) are naturally dropped
as stale by the gap detection (gap would be a large wraparound value
exceeding MAX_GAP_FRAMES).
Verification:
- cargo check --workspace: zero errors
- cargo test -p wzp-codec --lib: 68 passing (unchanged from Phase 3b)
- cargo test -p wzp-client --lib: 35 passing (unchanged from Phase 3b)
- cargo check -p wzp-android --lib: zero errors
- cargo test -p wzp-android cannot run on macOS host (pre-existing
-llog linker dep, unrelated). Real end-to-end verification happens
via the Android APK build on the remote Docker builder
(scripts/build-and-notify.sh).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Phase 2 of the DRED integration (docs/PRD-dred-integration.md). With
Phase 1 having enabled DRED on every Opus profile, the app-level RaptorQ
layer is now redundant overhead on those tiers: +20% bitrate, +40–100 ms
receive-side latency (block wait), +CPU for stats we never used. This
phase removes RaptorQ from the Opus encode and decode paths on both the
desktop (wzp-client/call.rs) and Android (wzp-android/engine.rs) sides.
Codec2 tiers keep RaptorQ with their current ratios unchanged — DRED is
libopus-only and Codec2 has no neural equivalent.
Encoder changes (the real bandwidth / CPU win):
- CallEncoder::encode_frame and engine.rs encode loop now gate the
RaptorQ path on !codec.is_opus():
- Opus source packets emit fec_block=0, fec_symbol=0,
fec_ratio_encoded=0 in the MediaHeader
- fec_enc.add_source_symbol is skipped on Opus
- generate_repair + repair packet emission is skipped on Opus
- block_id and frame_in_block counters stay frozen at 0 for Opus
- Codec2 path is byte-for-byte identical to pre-Phase-2 behavior.
Decoder changes (mostly cleanup, since both live decoder paths were
already reading audio directly from source packets and only using the
RaptorQ decoder output for stats):
- CallDecoder::ingest skips fec_dec.add_symbol on Opus packets. Source
packets still flow to the jitter buffer; Opus repair packets from old
senders are dropped cleanly (repair packets never hit the jitter
buffer either).
- engine.rs recv loop skips fec_dec.add_symbol, fec_dec.try_decode, and
fec_dec.expire_before on Opus packets. The `fec_recovered` stat
counter becomes Codec2-only (a separate DRED reconstruction counter
lands in Phase 4).
Wire-format backward compat verified at pre-flight:
- Old receiver + new sender: engine.rs pipeline.rs path gates on
non-zero fec_block/fec_symbol which now never fire for Opus, so the
RaptorQ decoder simply isn't fed. Audio flows normally. Desktop
CallDecoder's old path accumulated packets into the stale-eviction
HashMap, which cleans up after 2s — harmless.
- New receiver + old sender: new receiver skips RaptorQ on Opus so
old-sender repair packets are ignored entirely (no crash, no double-
decode). Loses the (previously vestigial) RaptorQ recovery benefit,
which was never actually active in the audio path. Source packets
still decode normally.
- No wire format version bump required. MediaHeader is unchanged; we
just zero the FEC fields on Opus packets.
Test changes:
- Removed `encoder_generates_repair_on_full_block` — asserted the old
(pre-Phase-2) RaptorQ-on-Opus behavior and is now incorrect. Replaced
with two symmetric tests:
- `opus_source_packets_have_zero_fec_header_fields` — verifies
Phase 2 invariants on Opus packets
- `opus_encoder_never_emits_repair_packets` — runs 20 frames of
non-silent sine wave through a GOOD-profile encoder, asserts
exactly 20 output packets, zero repair
- `codec2_encoder_generates_repair_on_full_block` — same shape as
the old test but on CATASTROPHIC profile (Codec2 1200, 8
frames/block, ratio 1.0) to verify Codec2 path still emits
repairs as before
Verification:
- cargo check --workspace: zero errors
- cargo test -p wzp-codec --lib: 61 passing (Phase 1 baseline held)
- cargo test -p wzp-client --lib: 32 passing (+3 new Phase 2 tests,
-1 old test removed)
- cargo check -p wzp-android --lib: zero errors (host link of
wzp-android tests fails on -llog per pre-existing Android-only
build.rs, unrelated to this work; integration build via
build-and-notify.sh will validate Android end-to-end)
- Pre-existing broken integration test in
crates/wzp-client/tests/handshake_integration.rs (SignalMessage
schema drift) is NOT caused by this commit — baseline had the same
3 compile errors before Phase 2. Flagged as a separate cleanup task.
Expected observable effects on a real call:
- Opus 24k outgoing bitrate drops from ~28.8 kbps (ratio 0.2 RaptorQ)
to ~25 kbps (base 24 kbps + DRED ~1–10 kbps signal-dependent)
- Opus receive-side latency drops ~40 ms on clean network (no more
block wait — jitter buffer emits as soon as a source packet arrives)
- Codec2 calls show no latency or bitrate change
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Three fixes for 3-relay chain (R1→R2→R3):
1. Room lookup in handle_datagram: hub relay (R2) has no local
participants, so active_rooms() was empty and datagrams were
silently dropped. Now also checks global_rooms config directly,
allowing hub relays to forward without local clients.
2. Multi-hop forwarding: removed active_rooms filter — forward to
ALL connected peers except source. The receiving peer decides
whether to deliver or forward further.
3. Android relay_label: native RoomMember now includes relay_label
from RoomUpdate signal. Kotlin UI reads it for relay grouping.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Wire AdaptiveQualityController into Android engine for auto codec
switching based on network quality reports. Add color-coded TX/RX
codec badges to the in-call screen showing active codecs and Auto mode.
- Recv task: ingest QualityReports, feed to controller, signal profile
changes via AtomicU8 to send task
- Send task: check for pending profile switch at frame boundaries,
update encoder/FEC/frame size
- Track peer codec from incoming packet headers
- Kotlin UI: codec badges (blue=studio, green=good, amber=degraded,
red=catastrophic) with Auto tag
- Add .taskmaster to .gitignore
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
1. Auto codec: new "Auto" position on quality slider (JNI index 7).
When selected, the engine uses the relay's chosen_profile from
CallAnswer instead of the local preference. Slider now has 8
positions: Studio 64k → Auto → Codec2 1.2k.
2. Force ping: added refresh button (↻) in Manage Relays dialog
header. Calls pingAllServers() to re-check all relays on demand.
3. Delete relay fix: the X button was inside a Surface(onClick=...)
which swallowed the touch event. Replaced with a separate Surface
that properly intercepts the click.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
The CallOffer only advertised GOOD/DEGRADED/CATASTROPHIC. When a
client uses a studio profile, the relay's choose_profile couldn't
pick it. Now advertises all 6 profiles (studio 64k/48k/32k + good +
degraded + catastrophic) in both Android engine and shared handshake.
Also: the relay MUST be rebuilt with the new CodecId variants,
otherwise it will fail to deserialize CallOffer messages containing
studio QualityProfiles in supported_profiles.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
1. Wire protocol: add Opus 32k/48k/64k (CodecId 6/7/8) + STUDIO
profiles with is_opus() helper. Opus enc/dec accept all Opus variants.
2. JNI bridge: expand profile_from_int to 7 levels (0-6) mapping to
GOOD, DEGRADED, CATASTROPHIC, Codec2_3200, STUDIO_32K/48K/64K.
3. Settings UI: replace radio buttons with Material3 Slider — 7 stops
from Studio 64k (green) to Codec2 1.2k (dark red), matching desktop.
4. Key-change warning: AlertDialog on connect when server fingerprint
has changed. Shows old vs new fingerprint, Accept New Key or Cancel.
Accepting saves the new fingerprint and proceeds with the call.
5. Engine recv: handle studio codec IDs in auto-switch path.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
The send loop was hardcoded to 960 samples (20ms/Opus24k), causing
DEGRADED (Opus 6k, 40ms) and CATASTROPHIC (Codec2 1200, 40ms) to
fail — the encoder needed 1920 samples but only got 960.
Changes:
- capture_buf, ring read threshold, and timestamp increment are now
computed from profile.frame_duration_ms (960 for 20ms, 1920 for 40ms)
- decode_buf sized to MAX_FRAME_SAMPLES (1920) to handle any incoming codec
- recv codec switch now uses correct QualityProfile per codec (was
inheriting original profile's frame_duration_ms, breaking cross-codec)
- added ComfortNoise guard on recv path
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Ping was a static JNI method that loaded the .so before nativeInit,
crashing jemalloc. Now ping is an instance method on WzpEngine:
- Engine is created once (nativeInit), reused for both ping and call
- pingRelay() uses same tokio runtime pattern as startCall()
- Auto-pings all servers on app launch (after engine init)
- No process restart needed
- TOFU fingerprints saved on first successful ping
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Adds average microsecond timings for each encode step:
- avg_agc_us: AGC processing
- avg_opus_us: Opus encoding
- avg_fec_us: FEC encode + repair generation
- avg_send_us: QUIC send_media
- avg_total_us: sum of above
Logged every 5 seconds in send stats. Resets each interval.
Use to identify which step is bottlenecking the encode loop
on devices where fps drops below 50.
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>
shutdown_background() killed the tokio runtime before quinn could send the
CONNECTION_CLOSE frame on the wire, so the relay never knew the client left.
Now use shutdown_timeout(500ms) to give quinn time to flush the close frame,
matching the desktop client pattern (which uses 2s timeout).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
stop_call() now calls close_now() on the stored transport handle before
killing the tokio runtime. This sends a QUIC CONNECTION_CLOSE frame so
the relay's recv loop breaks immediately, triggering leave() + RoomUpdate
broadcast. Previously the runtime was killed first, so transport.close()
never ran and the relay kept stale participants until idle timeout.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- 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>
- 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>
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>
- 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>
