Commit Graph

10 Commits

Author SHA1 Message Date
Siavash Sameni
54cbebd34e feat(codec): Phase 1 — enable DRED on all Opus profiles, disable inband FEC
Phase 1 of the DRED integration (docs/PRD-dred-integration.md). The Opus
encoder now emits DRED (Deep REDundancy) bytes in every packet, carrying
a neural-coded history of recent audio that the decoder can use to
reconstruct loss bursts up to the configured window. Opus inband FEC
(LBRR) is disabled because DRED does the same job better and running both
wastes bitrate on overlapping protection.

Tiered DRED duration policy per PRD:
  Studio  (Opus 32k/48k/64k): 10 frames = 100 ms
  Normal  (Opus 16k/24k):     20 frames = 200 ms
  Degraded (Opus 6k):         50 frames = 500 ms

Each profile switch (via adaptive quality) updates the DRED duration to
match the new tier. A 5% packet_loss floor is applied whenever DRED is
active, because libopus 1.5 gates DRED emission on non-zero packet_loss.
Real loss measurements from the quality adapter override upward.

Escape hatch: AUDIO_USE_LEGACY_FEC=1 reverts the encoder to Phase 0
behavior (inband FEC Mode1, DRED off, no loss floor). Read once at
OpusEncoder::new; call-scoped, not re-read mid-call. Trait-level
set_inband_fec becomes a no-op in DRED mode to preserve the invariant
even if external callers forget.

Observations from the bitrate probe test (dred_mode_roundtrip_voice_pattern):
  DRED mode:   3649 bytes/sec (~29.2 kbps) on Opus 24k + 300 Hz sine
  Legacy mode: 2383 bytes/sec (~19.1 kbps)
  Delta:       +10.1 kbps

The delta is considerably larger than the "+1 kbps flat" figure I carried
into the PRD from hazy memory of published DRED benchmarks. Likely because
the input (300 Hz sine) is very compressible so the base Opus rate in
legacy mode is well below the 24 kbps target, making the delta look
disproportionate. Signal-dependent — real speech would probably show a
different ratio. If production telemetry shows the overhead is excessive,
we can cut DRED duration on the normal tier from 200 ms to 100 ms as a
first tuning lever. Not blocking Phase 1 since the test still passes
within the reasonable 2000–8000 bytes/sec bounds.

Test changes (+8 tests, total wzp-codec: 61 passing):
- dred_duration_for_studio_tiers_is_100ms  (per-profile policy)
- dred_duration_for_normal_tiers_is_200ms
- dred_duration_for_degraded_tier_is_500ms
- dred_duration_for_codec2_is_zero
- default_mode_is_dred_not_legacy  (sanity check on fresh construction)
- dred_mode_roundtrip_voice_pattern  (observes DRED bitrate, asserts bounds)
- profile_switch_refreshes_dred_duration  (verifies set_profile updates DRED)
- set_inband_fec_noop_in_dred_mode  (trait-level inband FEC no-op)

Verification:
- cargo check --workspace: zero errors, no new warnings
- cargo test -p wzp-codec: 61/61 passing (53 pre-Phase-1 baseline + 8 new)
- Empirical DRED bitrate observed via `rtk proxy cargo test
  dred_mode_roundtrip_voice_pattern -- --nocapture`

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 20:02:35 +04:00
Siavash Sameni
86526a7ad4 feat(codec): Phase 0 — swap audiopus → opusic-c + opusic-sys (libopus 1.5.2)
Phase 0 of the DRED integration (docs/PRD-dred-integration.md). No behavior
change: inband FEC stays ON, no DRED, same bitrate, same quality. This
commit unblocks Phase 1+ by getting us onto libopus 1.5.2 where DRED lives.

Rationale for going straight to a custom DecoderHandle: opusic-c::Decoder's
inner *mut OpusDecoder pointer is pub(crate), so we cannot reach it for the
Phase 3 DRED reconstruction path. Running two parallel decoders (one for
audio, one for DRED) would drift because the DRED decoder wouldn't see
normal decode calls. Single unified DecoderHandle over raw opusic-sys is
the only correct architecture, so we build it in Phase 0 rather than
rewriting opus_dec.rs twice.

Changes:
- Cargo.toml (workspace + wzp-codec): remove audiopus 0.3.0-rc.0, add
  opusic-c 1.5.5 (bundled + dred features), opusic-sys 0.6.0 (bundled),
  bytemuck 1. Pinned exactly for reproducible libopus 1.5.2.
- opus_enc.rs: rewritten against opusic_c::Encoder. Argument order for
  Encoder::new swapped (Channels first). set_inband_fec(bool) now maps
  to InbandFec::Mode1 (the libopus 1.5 equivalent of 1.3's LBRR). encode
  uses bytemuck::cast_slice<i16,u16> at the &[u16] boundary.
- dred_ffi.rs (new): DecoderHandle wrapping *mut OpusDecoder directly via
  opusic-sys. Owns the allocation, frees on Drop. Exposes decode,
  decode_lost, and a pub(crate) as_raw_ptr() for the future Phase 3 DRED
  reconstruction. Send+Sync justified via &mut self access discipline.
- opus_dec.rs: rewritten as a thin AudioDecoder impl over DecoderHandle.
  Behavior identical to pre-swap.

Verification (Phase 0 acceptance gates):
- cargo check --workspace: clean (30 pre-existing warnings in jni_bridge.rs
  unrelated to this work; zero in changed files).
- cargo test -p wzp-codec: 53 tests pass (50 pre-swap + 6 new: 3 in
  dred_ffi.rs for DecoderHandle lifecycle, 3 in opus_enc.rs for version
  check and roundtrip).
- linked_libopus_is_1_5 test asserts opusic_c::version() contains "1.5" —
  hard signal that the swap landed correctly.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 20:02:15 +04:00
Siavash Sameni
a8c2011445 feat: add Opus 32k/48k/64k studio quality tiers
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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>
2026-04-07 18:31:05 +04:00
Siavash Sameni
d1c96cd71f feat: macOS VoiceProcessingIO for hardware AEC + delay-compensated NLMS
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- Add --os-aec flag: uses Apple VoiceProcessingIO audio unit for
  hardware echo cancellation (same engine as FaceTime)
- New vpio feature + audio_vpio.rs: combined capture+playback via VPIO
- Improved software AEC: delay-compensated leaky NLMS with Geigel DTD
  (60ms tail, 40ms delay, configurable via --aec-delay)
- Add --aec-delay flag for tuning software AEC delay compensation
- Add dev-fast Cargo profile (opt-level 2 with incremental compilation)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 11:10:10 +04:00
Siavash Sameni
1b00b5e2a4 feat: improved AEC, keyboard shortcuts, dedup participants, dev-fast profile
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AEC improvements:
- Reduce echo tail from 100ms to 30ms (3.3x faster, suited for laptops)
- Add double-talk detection: freeze adaptation when near-end speaks
- Add residual echo suppression
- Disable AEC by default in --android mode (macOS has built-in AEC)

CLI features:
- Keyboard shortcuts: m=mic mute, s=speaker mute, q=quit (raw terminal mode)
- Dedup participants in RoomUpdate display (same fingerprint+alias shown once)
- Add dev-fast profile (opt-level 2 with incremental compilation)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 10:15:23 +04:00
Claude
26e9c55f1f feat: Android VoIP client — Phase 1 (audio quality, network adaptation, crate skeleton)
- 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>
2026-04-04 18:07:55 +00:00
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
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
51e893590c feat: WarzonePhone lossy VoIP protocol — Phase 1 complete
Rust workspace with 7 crates implementing a custom VoIP protocol
designed for extremely lossy connections (5-70% loss, 100-500kbps,
300-800ms RTT). 89 tests passing across all crates.

Crates:
- wzp-proto: Wire format, traits, adaptive quality controller, jitter buffer, session FSM
- wzp-codec: Opus encoder/decoder (audiopus), Codec2 stubs, adaptive switching, resampling
- wzp-fec: RaptorQ fountain codes, interleaving, block management (proven 30-70% loss recovery)
- wzp-crypto: X25519+ChaCha20-Poly1305, Warzone identity compatible, anti-replay, rekeying
- wzp-transport: QUIC via quinn with DATAGRAM frames, path monitoring, signaling streams
- wzp-relay: Integration stub (Phase 2)
- wzp-client: Integration stub (Phase 2)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-03-27 12:45:07 +04:00