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>
The CallDecoder now inspects each incoming packet's codec_id and
automatically switches the audio decoder if it differs from the
current profile. This enables cross-codec interop where one client
sends Opus and the other sends Codec2 — previously the receiver
would try to decode with the wrong codec, producing garbled audio.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- 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>
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>
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>
- 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>
