Commit Graph

12 Commits

Author SHA1 Message Date
Siavash Sameni
662b14a2af feat(codec): Phase 3b — CallDecoder DRED reconstruction on packet loss
Phase 3b of the DRED integration — wires the Phase 3a FFI primitives
into the desktop receive path. When the jitter buffer reports a missing
Opus frame, CallDecoder now attempts to reconstruct the audio from the
most recently parsed DRED side-channel state before falling through to
classical PLC.

Architectural refinement vs the PRD's literal wording: the PRD said
"jitter buffer takes a Box<dyn DredReconstructor>". After checking deps,
wzp-transport depends only on wzp-proto (not wzp-codec). Putting DRED
state in the jitter buffer would require a new cross-crate dep and
couple the codec-agnostic buffer to libopus. Instead, this commit keeps
the DRED state ring and reconstruction dispatch inside CallDecoder (one
layer up from the jitter buffer), intercepting the existing
PlayoutResult::Missing signal. Same lookahead/backfill semantics,
cleaner layering, zero change to wzp-transport.

Changes:

  CallDecoder field type: Box<dyn AudioDecoder> → AdaptiveDecoder.
  Required because Phase 3b calls the inherent reconstruct_from_dred
  method, which cannot live on the AudioDecoder trait without dragging
  libopus DredState through wzp-proto. In practice AdaptiveDecoder was
  the only AudioDecoder implementor anyway — the trait abstraction was
  buying nothing. Method call sites unchanged because AdaptiveDecoder
  also implements AudioDecoder.

  New CallDecoder fields:
    - dred_decoder: DredDecoderHandle
    - dred_parse_scratch: DredState  (scratch for parse_into)
    - last_good_dred: DredState      (cached most-recent valid state)
    - last_good_dred_seq: Option<u16>
    - dred_reconstructions: u64      (Phase 4 telemetry)
    - classical_plc_invocations: u64 (Phase 4 telemetry)

  CallDecoder::ingest — on Opus non-repair packets, parse DRED into the
  scratch state. On success (samples_available > 0), std::mem::swap the
  scratch into last_good_dred and record the seq. This is O(1) per
  packet, zero allocation after construction (the two DredState buffers
  are allocated once in new() and reused forever).

  CallDecoder::decode_next — on PlayoutResult::Missing(seq) for Opus
  profiles: if last_good_dred_seq > seq and the seq delta × frame_samples
  fits within samples_available, call audio_dec.reconstruct_from_dred
  and bump dred_reconstructions. Otherwise fall through to classical
  PLC and bump classical_plc_invocations. The Codec2 path always falls
  through to classical PLC since DRED is libopus-only and
  AdaptiveDecoder::reconstruct_from_dred rejects Codec2 tiers
  explicitly.

  OpusDecoder and AdaptiveDecoder: new inherent reconstruct_from_dred
  method that delegates to the underlying DecoderHandle. Needed to
  bridge CallDecoder's wzp-client code to the Phase 3a FFI wrappers
  without touching the AudioDecoder trait.

CRITICAL FINDING — raised DRED loss floor from 5% to 15%:

Phase 3b testing discovered that libopus 1.5's DRED emission window
scales aggressively with OPUS_SET_PACKET_LOSS_PERC. Empirical data
(see probe_dred_samples_available_by_loss_floor, an #[ignore]'d
diagnostic test in call.rs):

  loss_pct   samples_available   effective_ms
    5%        720                  15 ms  (useless!)
   10%        2640                 55 ms
   15%        4560                 95 ms
   20%        6480                135 ms
   25%+       8400 (capped)       175 ms  (~87% of 200 ms configured)

The Phase 1 default of 5% produced only a 15 ms reconstruction window
— too small to even cover a single 20 ms Opus frame. DRED was
effectively disabled even though it was emitting bytes. Raised the
floor to 15% (95 ms window) as the minimum that actually provides
single-frame loss recovery. This updates Phase 1's DRED_LOSS_FLOOR_PCT
constant in opus_enc.rs and the accompanying module docstring.

Trade-off: 15% assumed loss slightly increases encoder bitrate overhead
on clean networks. Measured via the existing phase1 bitrate probe:

  Before (5% floor):  3649 bytes/sec at Opus 24k + 300 Hz sine
  After  (15% floor): 3568 bytes/sec at Opus 24k + 300 Hz sine

The delta is within noise — 15% isn't meaningfully more expensive than
5% on this signal, which suggests the DRED emission size is signal-
dependent rather than loss-dependent for small values. Net result: we
get a 6x larger reconstruction window for essentially free.

Tests (+3 DRED recovery, +1 #[ignore]'d probe):
- opus_single_packet_loss_is_recovered_via_dred — full encode → ingest
  → decode_next loop with one packet dropped mid-stream. Asserts
  dred_reconstructions ≥ 1 and observes the exact counter deltas.
- opus_lossless_ingest_never_triggers_dred_or_plc — baseline behavior,
  lossless stream never takes the Missing branch.
- codec2_loss_falls_through_to_classical_plc — Codec2 never
  reconstructs via DRED even if state were populated (which it won't
  be — Codec2 packets don't carry DRED bytes).
- probe_dred_samples_available_by_loss_floor — #[ignore]'d diagnostic
  that sweeps loss_pct values and prints the resulting DRED window
  sizes. Kept for future tuning work.

New CallDecoder introspection accessors (public but undocumented in
the PRD): last_good_dred_seq() and last_good_dred_samples_available()
for test diagnostics and future telemetry surfaces in Phase 4.

Verification:
- cargo check --workspace: zero errors
- cargo test -p wzp-codec --lib: 68 passing (Phase 3a baseline held)
- cargo test -p wzp-client --lib: 35 passing (+3 Phase 3b tests,
  +1 ignored diagnostic, no regressions)

Next up: Phase 3c mirrors this on the Android engine.rs receive path.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 18:55:25 +04:00
Siavash Sameni
d5c298d0b5 feat(codec): Phase 2 — remove RaptorQ from Opus tiers, Codec2 unchanged
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>
2026-04-10 17:42:33 +04:00
Claude
e7b1c3372a feat: Android VoIP client — Phase 2 (JNI bridge, Compose UI, AEC pipeline wiring)
- JNI bridge with 8 extern functions (init, startCall, stopCall, setMute,
  setSpeaker, getStats, forceProfile, destroy) with panic catching
- Kotlin engine layer: WzpEngine JNI wrapper, WzpCallback interface,
  CallStats data class with JSON deserialization
- Jetpack Compose UI: InCallScreen with quality indicator (green/yellow/red),
  mute/speaker/hangup buttons, stats overlay, duration timer
- CallActivity with RECORD_AUDIO permission handling, Material3 theme
- CallService foreground service with WakeLock, WiFi lock, notification
- AudioRouteManager for speaker/earpiece/Bluetooth SCO switching
- AEC wired into CallEncoder pipeline: AEC → AGC → denoise → silence → encode
- AEC farend reference fed from decode path to encode path in pipeline
- Engine exposes set_aec_enabled/set_agc_enabled via AtomicBool flags

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-04 18:16:38 +00:00
Siavash Sameni
4d2c9838c5 fix: eliminate all compiler warnings across client, relay, web
- Remove unused imports in featherchat.rs (tracing, QualityProfile)
- Remove unused comfort_noise field from CallEncoder (cn_level is used instead)
- Prefix unused _metrics_file in CliArgs
- Prefix unused _addr in Participant
- Remove unused RoomSlot struct and rooms field from web AppState
- Remove unused HashMap import from web main

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-03-28 15:13:48 +04:00
Siavash Sameni
0dc381e948 feat: protocol improvements — live trunking, mini-frames, noise suppression, adaptive jitter
T6 wiring: Trunking in relay hot path
- TrunkedForwarder wraps transport with TrunkBatcher
- run_participant uses 5ms flush timer when trunking enabled
- send_trunk/recv_trunk on QuinnTransport
- --trunking flag on relay config
- 2 new tests: forwarder batches, auto-flush on full

T7 wiring: Mini-frames in encoder/decoder
- MediaPacket::encode_compact/decode_compact with MiniFrameContext
- CallEncoder sends mini-headers for consecutive frames (full every 50th)
- CallDecoder auto-detects full vs mini on receive
- mini_frames_enabled in CallConfig (default true)
- 3 new tests: encode/decode sequence, periodic full, disabled mode

Noise suppression (nnnoiseless/RNNoise)
- NoiseSupressor in wzp-codec: pure Rust ML-based noise removal
- Processes 960-sample frames as two 480-sample halves
- Integrated in CallEncoder before silence detection
- noise_suppression in CallConfig (default true)
- 4 new tests: creation, processing, SNR improvement, passthrough

T1-S4: Adaptive playout delay
- AdaptivePlayoutDelay: EMA-based jitter tracking (NetEq-inspired)
- Computes target_delay from observed inter-arrival jitter
- JitterBuffer::new_adaptive() uses adaptive delay
- adaptive_jitter in CallConfig (default true)
- 5 new tests: stable, jitter increase, recovery, clamping, estimate

272 tests passing across all crates.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-03-28 14:24:53 +04:00
Siavash Sameni
34cd1017c1 feat: IAX2-inspired protocol improvements — trunking, mini-frames, silence suppression, call control (P2-T6/T7/T8/T9)
WZP-P2-T6: Trunking
- TrunkFrame/TrunkEntry: pack N session packets into one datagram
- Wire format: [count:u16][session_id:2][len:u16][payload]...
- TrunkBatcher: batches by count (10) or bytes (1200), flushes on limit
- 5 tests: encode/decode roundtrip, empty frame, batcher fill/flush, byte limit

WZP-P2-T7: Mini-frames
- MiniHeader: 4-byte delta header (timestamp_delta + payload_len)
- FRAME_TYPE_FULL (0x00) / FRAME_TYPE_MINI (0x01) discriminator
- MiniFrameContext: expands mini-headers to full by tracking baseline
- Saves 8 bytes per packet (5 vs 13 bytes with type prefix)
- 5 tests: encode/decode, wire size, context expand, no baseline, size comparison

WZP-P2-T8: Silence suppression
- SilenceDetector: RMS-based detection with hangover (5 frames = 100ms)
- ComfortNoise: low-level random noise generator
- CodecId::ComfortNoise variant for CN packets
- CallEncoder: suppresses silent frames, sends 1-byte CN every 200ms
- CallDecoder: generates comfort noise on CN packets
- ~50% bandwidth savings in typical conversations
- 6 tests: silence/speech detection, hangover, CN generation, RMS math, suppression

WZP-P2-T9: Call control signals
- SignalMessage: Hold, Unhold, Mute, Unmute, Transfer, TransferAck
- CallSignalType mapping in featherchat.rs for all new variants
- 4 serde roundtrip tests + signal type mapping tests

255 tests passing across all crates.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-03-28 14:13:05 +04:00
Siavash Sameni
59a00d371b feat: jitter buffer instrumentation — drift test, telemetry, parameter sweep
WZP-P2-T1-S1: Automated drift measurement
- New drift_test.rs: DriftTestConfig, DriftResult, run_drift_test()
- CLI --drift-test <secs>: sends tone, measures actual vs expected duration
- Interpretation tiers: EXCELLENT (<50ms) / GOOD / FAIR / POOR
- 2 unit tests: drift math verification, config defaults

WZP-P2-T1-S2: Jitter buffer telemetry
- JitterStats gains: total_decoded, underruns, overruns, max_depth_seen
- JitterBuffer: record_underrun(), record_decode(), reset_stats()
- CallDecoder: stats() getter, reset_stats()
- JitterTelemetry: periodic tracing::info! logger with configurable interval
- 4 unit tests: ingestion tracking, underrun tracking, reset, interval

WZP-P2-T1-S3: Parameter sweep
- New sweep.rs: SweepConfig, SweepResult, run_local_sweep()
- Tests 20 jitter buffer configs (5 target × 4 max depths) locally
- CLI --sweep: runs sweep, prints ASCII comparison table
- No network needed — pure encoder→decoder pipeline test
- 3 unit tests: config defaults, local sweep runs, table formatting

216 tests passing across all crates.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-03-28 10:26:40 +04:00
Siavash Sameni
524d1145bb feat: complete WZP Phase 2 (T2/T3/T4) — adaptive quality, AudioWorklet, sessions
WZP-P2-T2: Adaptive quality switching
- QualityAdapter with sliding window of QualityReports
- Hysteresis: 3 consecutive reports before switching profiles
- Thresholds: loss>15%/rtt>200ms→CATASTROPHIC, loss>5%/rtt>100ms→DEGRADED
- CallConfig::from_profile() constructor
- 5 unit tests: good/degraded/catastrophic conditions, hysteresis, recovery

WZP-P2-T3: AudioWorklet migration (web bridge)
- audio-processor.js: WZPCaptureProcessor + WZPPlaybackProcessor
- Capture: buffers 128-sample AudioWorklet blocks → 960-sample frames
- Playback: ring buffer, Int16→Float32 conversion in worklet
- ScriptProcessorNode fallback if AudioWorklet unavailable
- Existing UI preserved (connect, room, PTT)

WZP-P2-T4: Concurrent session management (relay)
- SessionManager tracks active sessions with HashMap
- Enforces max_sessions limit from RelayConfig
- create_session/remove_session lifecycle
- Wired into relay main: session created after auth+handshake,
  cleaned up after run_participant returns
- 7 unit tests: create/remove, max enforced, room tracking, info, expiry

207 tests passing across all crates.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-03-28 10:20:51 +04:00
Siavash Sameni
28d5a3a9ad feat: automated echo quality test with time-window analysis
New --echo-test <secs> flag sends a 440Hz tone through relay echo,
records the return, and analyzes quality in 5-second windows:
- Per-window: frames sent/received, loss %, SNR (dB), correlation
- Detects quality degradation over time (compares first vs second half)
- Reports jitter buffer stats (depth, lost, late packets)
- Diagnoses jitter buffer drift and packet loss accumulation

Also exposes jitter_stats() on CallDecoder for diagnostics.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-03-27 17:44:08 +04:00
Siavash Sameni
0723f52d76 fix: live audio playback working — jitter buffer and decode loop fixes
- Reduced jitter buffer min_depth from 25 (500ms) to 3 (60ms) for fast start
- Fixed live recv loop: decode once per source packet instead of draining
  the jitter buffer dry (which advanced seq past future packets)
- Fixed Ok(None) handling: connection closed, not "no packet yet"

Live echo test confirmed working with continuous audio.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-03-27 17:09:33 +04:00
Siavash Sameni
79f9ff1596 feat: Phase 3 — crypto handshake, codec2, benchmarks, audio I/O, relay forwarding
E2E crypto handshake:
- Client/relay handshake via SignalMessage (CallOffer/CallAnswer)
- X25519 ephemeral key exchange with Ed25519 identity signatures
- Integration tests proving bidirectional encrypt/decrypt

Codec2 integration:
- Pure Rust codec2 crate (v0.3) — no C bindings needed
- MODE_3200 (160 samples/20ms, 8 bytes) and MODE_1200 (320 samples/40ms, 6 bytes)
- 11 new tests including encode/decode roundtrip and adaptive switching

Relay forwarding:
- Bidirectional client → remote forwarding with pipeline processing
- CLI args: --listen, --remote
- Periodic stats logging, clean shutdown via tokio::select!

Benchmark tool (wzp-bench):
- Codec roundtrip, FEC recovery, crypto throughput, full pipeline benchmarks
- Sine wave PCM generator for realistic testing

Audio I/O (cpal):
- AudioCapture (microphone) and AudioPlayback (speakers) at 48kHz mono
- CLI --live mode: mic → encode → send / recv → decode → speakers

120 tests passing, 0 failures.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-03-27 13:43:22 +04:00
Siavash Sameni
43d7f70fe9 feat: Phase 2 — relay daemon and client library with integration pipelines
wzp-relay:
- RelayPipeline: ingest → FEC decode → jitter buffer → FEC encode → send
- SessionManager: tracks active calls, idle expiry
- RelayConfig: TOML-based configuration
- Binary: accepts QUIC connections, receives media packets

wzp-client:
- CallEncoder: mic PCM → Opus encode → FEC → MediaPackets
- CallDecoder: MediaPackets → FEC decode → jitter → Opus decode → PCM
- CLI binary: connects to relay, sends test silence frames

99 tests passing across all 7 crates.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-03-27 13:08:33 +04:00