# Road to Video > Plan for adding video to WZP. Audio remains unchanged through Phase V1; video is additive. See `PROTOCOL-AUDIT.md` for the issues this plan addresses. ## Premise The transport, crypto, session, federation, and SFU layers are codec-agnostic. The work is concentrated in: 1. Wire format (CodecID width, MediaType, MiniHeader seq, simulcast hooks) 2. Framer / depacketizer (NAL fragmentation, access-unit reassembly) 3. Bandwidth estimator (Quinn cwnd + transport feedback) 4. Keyframe semantics (PLI, NACK, keyframe cache at SFU) 5. Capture / encode pipeline (VideoToolbox / MediaCodec / NVENC) ## Phase V1 — Wire format & negotiation (no new code paths yet) Bump protocol version. Land all wire changes together so compat breaks exactly once. ### Sizing decision (2026-05-11) Hypothetical benchmarks on 12 B packed vs 16 B byte-aligned showed the overhead delta is invisible across every realistic scenario: | Scenario | Δ overhead (12 B → 16 B) | Δ % of stream | |---|---|---| | Opus 24k audio (MiniHeader 49/50) | 4 B/s | 0.013 % | | Codec2 1200 audio | 2 B/s | 0.13 % | | H.264 SD 500 kbps video | 1.6 kbps | 0.32 % | | H.264 HD 2.5 Mbps video | 7.1 kbps | 0.28 % | | H.264 FHD 5 Mbps video | 14.1 kbps | 0.28 % | Trunking cap (10) binds before MTU for audio, so TrunkFrame layout is unaffected. ChaCha20-Poly1305 cost is dominated by AEAD setup, not byte count — 4 extra bytes per packet is < 0.1 % of AEAD CPU on Cortex-A55. **Decision: 16 B byte-aligned.** Bit-packing saves nothing material and costs recurring debug / fuzzer / evolution complexity. Reserves headroom for the next decade. ### `MediaHeader` v2 (16 B byte-aligned) ``` Byte 0: version (u8) currently 0x02 Byte 1: flags (u8) [T:1][Q:1][KeyFrame:1][FrameEnd:1][reserved:4] T = FEC repair Q = QualityReport trailer present KeyFrame = packet belongs to an I-frame (video) FrameEnd = last packet of an access unit (video) Byte 2: media_type (u8) 0=audio, 1=video, 2=data, 3=control Byte 3: codec_id (u8) widened from 4-bit (room for 256) Byte 4: stream_id (u8) simulcast layer; 0=base Byte 5: fec_ratio (u8) 0..200 → 0.0..2.0 Bytes 6-9: sequence (u32 BE) Bytes 10-13: timestamp_ms (u32 BE) Bytes 14-15: fec_block_id (u16 BE) audio: low 8 bits block_id, high 8 bits symbol_idx video: full u16 block_id (large blocks for I-frames) ``` - `version=2` is a hard switch — old clients receive a typed `Hangup::ProtocolVersionMismatch`. - `media_type` (W10) lets the SFU drop video first under load without a codec lookup. - `KeyFrame` lets a joining peer fast-forward to the next I-frame; SFU keyframe cache keys on it. - `FrameEnd` lets the depacketizer fire an access unit without counting packets. - `stream_id` is forward-compatible for simulcast (Phase V5). - `sequence` widened to u32 (W1) — also benefits audio. ### `MiniHeader` v2 (5 B) ``` [FRAME_TYPE_MINI = 0x01] Byte 0: seq_delta (u8) ← new (W4) Bytes 1-2: timestamp_delta_ms (u16 BE) Bytes 3-4: payload_len (u16 BE) ``` Audio-only in V1. Video pays the full 16 B header per packet (every frame is a new access unit; no clean periodic structure to compress). ### New codec IDs | ID | Codec | Notes | |---|---|---| | 9 | H.264 baseline | Universal HW encode coverage; ship first | | 10 | H.264 main | Slight quality win over baseline; same HW | | 11 | H.265 main | Apple A10+ universal, Snapdragon since ~2017, NVENC GTX 9xx+; ~30 % win vs H.264 | | 12 | AV1 | Apple M3/A17+, Snapdragon 8 Gen 3+, RTX 40+, Arc, RX 7000+; best efficiency, narrow HW | | 13 | VP9 | Reserved; may not implement | Negotiation: `CallOffer.supported_codecs: Vec`. Both sides pick the highest mutually supported codec from preference cascade `[AV1, H.265, H.264 main, H.264 baseline]`. ### `QualityProfile` extension Add: - `video_bitrate_kbps: Option` - `video_resolution: Option<(u16, u16)>` - `video_fps: Option` - `priority_mode: PriorityMode` (see Phase V5) `CallOffer` / `CallAnswer` already negotiate profiles — slot video into the same path. ### Acceptance - All 571 audio tests pass with `V=2` headers. - Old v1 clients refused gracefully (clear error in `CallAnswer`). ## Phase V2 — Transport additions **Decision (2026-05-11): all media on QUIC datagrams; no separate "reliable media" stream.** A QUIC stream for I-frames was considered and rejected. A 200 KB I-frame on a 1 Mbps mobile link takes ~1.6 s to transit a stream, and the next I-frame queues behind it (HoL blocking by design). Datagrams + NACK + dynamic per-keyframe FEC degrade more gracefully on the lossy links we care about. 1. **All media on datagrams.** Uniform wire format; no HoL. 2. **NACK loop for video P-frames.** When `RTT < 2 × frame_interval`, receiver NACKs missing P-frame packets via `SignalMessage::Nack { stream_id, seqs }`. Otherwise (high RTT) skip NACK and request a keyframe via `PictureLossIndication`. 3. **Dynamic FEC boost on I-frames.** Encoder bumps `fec_ratio` to ~0.5 for keyframe packets (k=20 source → r=10 repair). Recovers most I-frame loss without a round trip. 4. **SPS/PPS / parameter sets on the existing signal stream.** Reliable, ordered, one-time at session start. Re-sent on codec switch. No new stream needed. 5. **`SignalMessage::TransportFeedback`** — `{ acked_seqs: Vec, nacked_seqs: Vec, remb_bps: u32, recv_time_us: u64 }`. Sent every 50 ms or every N packets, whichever first. Feeds BWE. 6. **`BandwidthEstimator` in `wzp-proto`** — consumes Quinn `cwnd`, `bytes_in_flight`, plus `TransportFeedback`. Output: `target_send_bps = min(cwnd_bps * 0.9, remb_bps)`. ### Acceptance - Audio adapts to bandwidth (not just loss/RTT); fewer oscillations between 24 k and 32 k Opus on stable links. - BWE output is on Prometheus. - NACK round-trip recovery verified under 1–5 % packet loss at RTT ≤ 100 ms. ## Phase V3 — `wzp-video` crate New crate parallel to `wzp-codec`: ``` wzp-video/ src/ encoder.rs # trait VideoEncoder; VideoToolboxEncoder, MediaCodecEncoder, # OpenH264Encoder fallback decoder.rs # trait VideoDecoder framer.rs # NAL unit fragmentation to MTU-sized chunks # (simpler than RFC 6184 FU-A — we own both ends) depacketizer.rs # Reassemble NALs, emit access units keyframe.rs # Keyframe request handling ``` Framing rules: - One access unit → N packets, each ≤ MTU − 12 (MediaHeader) − 16 (AEAD tag). - `sequence` global per stream; `timestamp_ms` is presentation time. - `KeyFrame` bit set on every packet of an I-frame. - Last packet of frame: "frame end" bit (steal from `StreamId` or repurpose `reserved`). Platform encoders: - macOS / iOS: VideoToolbox - Android: MediaCodec (surface texture path, no CPU copy) - Windows: MediaFoundation → NVENC / QSV / AMF - Linux: VAAPI / NVENC; OpenH264 software fallback ### Acceptance - Unidirectional H.264 call working between two desktop clients. - CPU usage on M1 < 5 % at 720p30; on Android mid-tier < 15 %. ## Phase V4 — Keyframe & loss policy - On packet loss inside a P-frame: NACK if RTT < 2× frame interval, otherwise request keyframe via `SignalMessage::PictureLossIndication { stream_id }`. - Joining peer: relay sends most recent keyframe from its cache. - Tier downgrade: drop to lower simulcast layer, request keyframe for the new layer. ### Acceptance - Black-screen-on-join < 200 ms when keyframe cache is warm. - < 1 keyframe / 2 s on stable links; bursty on lossy links. ## Phase V5 — Video adaptive controller + PriorityMode ### `PriorityMode` on `QualityProfile` ```rust pub enum PriorityMode { AudioFirst, // default for calls: audio absolute priority, video elastic VideoFirst, // user override: video priority, audio degrades second ScreenShare, // video + slide-fallback; audio = intelligible speech only Balanced, // proportional split, no absolute priority } ``` Selected at call setup. Mutable mid-call via `SignalMessage::SetPriorityMode { mode }`. Defaults to `AudioFirst` for voice/video calls; presentation apps set `ScreenShare`; users can override to `VideoFirst` from settings. ### `VideoQualityController` ``` inputs: bwe_bps, loss_pct, rtt_ms, encoder_queue_ms, priority_mode outputs: target_bitrate, target_fps, target_resolution, simulcast_layer allocation gate (per PriorityMode): AudioFirst: audio_budget = max(24 kbps, audio_tier_min) video_budget = bwe_bps - audio_budget Under congestion: video → 0 before audio degrades. VideoFirst: video_budget = max(video_floor, target_video_kbps) audio_budget = bwe_bps - video_budget Audio degrades first to Opus 16 k; video held at floor. ScreenShare: video_budget = bwe_bps - 16 kbps // audio gets just Opus 16 k floor If video_budget < SD floor: switch encoder to slide mode (single high-quality I-frame every 2-5s instead of continuous video). Audio floor in this mode is Opus 16 k (speech only, no music). Balanced: audio_budget = bwe_bps * 0.15 video_budget = bwe_bps * 0.85 Both degrade proportionally. ``` Slide mode in `ScreenShare` is an encoder policy on the existing `wzp-video` framer (lower fps, higher per-frame quality, prefer HEVC/AV1 for text). No wire format change. ### Acceptance - On a 100 kbps link in `AudioFirst`, audio stays at Opus 24 k and video drops to 0. - On a 100 kbps link in `ScreenShare`, slide mode emits one I-frame every 3 s and audio holds Opus 16 k. - On a 5 Mbps link, video ramps to top simulcast layer within 10 s. - `SetPriorityMode` mid-call is honored within 1 s. ## Phase V6 — SFU changes - **Per-room keyframe cache.** Latest I-frame per `(sender, stream_id)`. Sent to new joiners immediately. Eliminates "black screen for 2 seconds" on join. - **Per-receiver layer selection.** Sender uploads ~3 simulcast layers; relay decides which to forward to each receiver based on their last `QualityReport`. Critical for N > 3 rooms. - **PLI suppression.** If 10 receivers PLI within 200 ms, send one `KeyframeRequest` upstream, not 10. ### Acceptance - 8-peer room with mixed link quality; high-quality peers see HD, low-quality peers see SD, no peer holds the room back. - PLI traffic at SFU upstream < 1 / s under simulated mass packet loss. ## Phase V7 — Capture pipeline (platform-specific) - macOS: `AVCaptureSession` → VideoToolbox → `wzp-video`. Wire into Tauri backend. - Android: Camera2 → MediaCodec → JNI bridge into `wzp-native` or sibling cdylib. Surface texture path. - Desktop Tauri (Windows): MediaFoundation → NVENC. ### Acceptance - Camera permission flows on all platforms. - < 50 ms end-to-end capture-to-encode latency on M1. ## Deferred - **SVC** (per-layer temporal scalability in one bitstream). Simulcast (separate streams per layer) is enough for v1; wire format already supports it via `StreamId`. - **Screen sharing.** Same codec path with a different capture source. - **Group video keys.** Existing X25519 session key works; no protocol change needed. ## Suggested order of work | Step | Effort | Output | |---|---|---| | 1. Wire format v2: 16 B MediaHeader, 5 B MiniHeader, MediaType, KeyFrame, FrameEnd, u32 seq, 8-bit CodecID | ~1 day | Audio still works under new header layout | | 2. TransportFeedback + BandwidthEstimator (Quinn cwnd + remb) | 3–4 days | Audio adaptation improves; BWE on Prom | | 3. `wzp-video` crate, H.264 baseline single-layer | 1–2 weeks | Unidirectional video call works | | 4. NACK path + dynamic FEC boost on I-frames | 4–5 days | Loss recovery for video | | 5. Keyframe cache at SFU + PLI suppression | 1 week | Fast join, low PLI traffic | | 6. H.265 codec support (reuse framer) | 3 days | ~30 % quality win on Apple HW | | 7. Simulcast + per-receiver layer selection | 1 week | Mixed-quality rooms work | | 8. `VideoQualityController` + PriorityMode (incl. ScreenShare slide mode) | 1 week | Graceful degradation under congestion, user choice | | 9. AV1 codec (gated on HW telemetry) | 4–5 days | Top-tier efficiency on capable devices | | 10. Native capture pipelines (VideoToolbox / MediaCodec / NVENC) | 2 weeks | Production camera support per OS | Step 1 is the lowest-regret, highest-leverage change and unlocks everything else. Steps 3 + 6 + 9 form the codec rollout: ship H.264 first (works everywhere → unblocks integration testing on every device), add H.265 once framer is stable (low-effort, big Apple win), gate AV1 on real device telemetry. By 2028 we should be in a position to deprecate H.264 if telemetry says < 5 % of sessions still need it.