voice-cat/PROGRESS.md

# PROGRESS — VoiceCat

Living status. **Update this file in the same commit as your work** so the next agent picks
up instantly. Newest status at the top.

- **Date convention:** ISO (YYYY-MM-DD).
- Statuses: `[ ]` not started · `[~]` in progress · `[x]` done.

---

## ▶ Where we left off / next action

- **In progress:** **M5 — moderation & admin** (2026-06-17). Server-side and C ABI are
  implemented and tested: permissions, kick/ban/move/server-mute, channel CRUD, in-app account
  management. Four new tests pass: `test_m5_permissions`, `test_m5_kick_ban_move_mute`,
  `test_m5_admin_accounts`, `test_m5_channel_crud`. `vccli` now exposes all M5 operations via
  CLI flags (`--kick`, `--ban`, `--move`, `--server-mute`/`-unmute`/`-deafen`/`-undeafen`,
  `--set-permission`, `--create-channel`, `--edit-channel`, `--delete-channel`,
  `--create-account`, `--reset-password`, `--delete-account`, `--list-accounts`) plus
  `--username`/`--password` for account auth and `--self-mute`/`--self-deafen`. Docs updated:
  `docs/protocol.md` (envelope tags for `ServerMuteRequest`/`ListAccountsResult`, `User.server_deafened`,
  `GenericResult` usage), `docs/security.md` (BLAKE2b channel passwords, `bans` schema).
  `ctest --preset m1-dev` — **18/18 green**. Still to do: DRED/audio-quality polish and Windows
  admin/moderation UI.
- **Done:** **Fixed a *second* silent-playback bug — the playout clock free-ran and drifted off
  the stream** (2026-06-17, reported live: both `vccli` and the Windows client showed `talking=1/0`
  correctly on VAD/PTT, mic + screen-share were recognized by peers, but nothing was audible).
  Root cause: `RemoteStream::playout_ts` was only ever seeded to `0` and then advanced one Opus
  frame per playback callback **via the PLC path too** (`core/src/audio/audio_engine.cpp`
  `on_playback`), so it free-ran at ~1× wall-clock regardless of whether the sender was
  transmitting. The sender's frame timestamps only advance while it actually sends (the VAD/PTT
  gate in `core/src/core/client.cpp` returns before `ls.timestamp += samples`). Across a late join
  or any VAD/PTT silence gap the two clocks diverged without bound; once past the jitter buffer's
  500 ms late-drop window, every real frame was dropped-as-late (clock ahead) or never-due (clock
  behind) → permanent silence, while the talk indicator (driven by `push_recv_frame`, independent
  of the jitter buffer) stayed lit. The M3 E2E test missed it because clients there talked
  continuously right after joining, keeping the clocks aligned. **Fix:** `JitterBuffer` gained
  `peek_front_ts()` (try-lock, RT-safe); `on_playback` now seeds/re-syncs `playout_ts` to the
  earliest buffered frame on the first frame and whenever it has drifted past ±200/500 ms
  (`kResyncAheadSamples`/`kResyncBehindSamples`), which both seeds startup and recovers after every
  silence gap. New regression test `test_playout_resync` (`tests/test_vad_ptt_devices.cpp`):
  free-runs the clock ~2 s past the drop window, pushes a `ts=0` frame, asserts audible output —
  verified to fail (energy=0) with the fix disabled, pass (energy≈15M) with it. `ctest --test-dir
  build/m1-dev` — **14/14 green** (run via PowerShell; Git Bash exec gotcha for these binaries, see
  `docs/building.md`). **Not yet confirmed audible by ear** — pending the user re-running their
  live test.
- **Done:** **Fixed silent-playback bug in `AudioEngine::on_playback`** (2026-06-16, found via
  live manual test: two `vccli --voice` clients, control-plane events and VAD all correct, but
  zero audible output). Root cause: `opus_decode()`'s `max_samples` was being passed the
  *hardware playback callback's* frame count (miniaudio's own choice, frequently smaller than
  one Opus frame — e.g. ~480 samples on default low-latency WASAPI periods), instead of the
  decoder's fixed frame size (960 @ 20ms/48kHz). Since the real packet almost always decodes to
  more samples than that, `opus_decode` returned `OPUS_BUFFER_TOO_SMALL` on nearly every
  callback — frames were correctly received/decrypted/jitter-buffered, just never decoded into
  audible PCM. `mix_for_test()`'s white-box test masked this because it always called
  `on_playback` with `frames == frame_samples`, the one case where the bug is invisible.
  Fix: `RemoteStream` (`core/src/audio/audio_engine.h`) gained a small ring buffer
  (`init_ring`/`push_ring`/`pop_ring`) that decouples decode cadence from playback-callback
  cadence — `on_playback` (`core/src/audio/audio_engine.cpp`) now tops the ring up by decoding
  whole Opus frames (`decoder.frame_samples()`, never the hardware `frames`) and drains exactly
  `frames` samples-per-channel from it each callback, silence-padding (PLC) on underrun. Also
  fixes a latent `playout_ts` bug: it now advances by the actual decoded sample count per Opus
  frame, not by the hardware callback's (unrelated) frame count, which was the wrong unit for
  jitter-buffer timestamp comparisons. `ctest --test-dir build/m1-dev` — 12/12 green (run via
  PowerShell; Git Bash exec gotcha for these binaries, see `docs/building.md`). **Not yet
  confirmed audible by ear** — pending the user re-running their live two-`vccli` test.
- **Done:** **Post-M3 follow-up — device enumeration, VAD/PTT gate, stereo playback, WASAPI
  loopback** ✓ complete (2026-06-16). Closes all three items M3 explicitly carried forward as
  out of scope (see the dated section below for the full file-by-file change list).
  `ctest --test-dir build/m1-dev` — **12/12 tests** green (3 consecutive full-suite runs),
  including the new `test_vad_ptt_devices` (real `vc_client`s against a real server, plus a
  white-box `AudioEngine` stereo-mix check — same ABI-level-coverage lesson as M2/M3).
  Manually verified live: `vccli --list-devices` against real hardware, and
  `vccli --voice --input-mode vad` connecting/streaming without incident.
  **Still explicitly out of scope** (carried forward, not silently dropped):
  - Real `webrtc-audio-processing`/AEC — no working Windows/MSVC build upstream; v1 ships a
    lightweight energy/RMS VAD instead (see docs/roadmap.md §2, docs/voice.md §8/§11). There is
    **no AEC, NS, or AGC implementation at all**, not just a deferred VAD.
    `vc_set_remote_stream(..., noise_reduction)`'s per-stream NS toggle is unaffected by this
    pass and stays exactly as inert as it was after M3 (`ApmPassthrough`, no PCM modification).
  - macOS/iOS `SCREEN_AUDIO` capture (ScreenCaptureKit / ReplayKit) — this pass is
    Windows-only for real loopback capture; other platforms keep `vc_test_inject_capture` as
    the only way to feed `SCREEN_AUDIO`.
  - Process-specific WASAPI loopback — miniaudio's loopback mode captures the whole render
    endpoint (including this app's own incoming voice mix), not a single process.
  - The pre-existing RT-thread rule violation in `on_capture_frame`/`AudioEngine::on_capture`
    (mutex lock, heap allocation, blocking `sendto` on the miniaudio real-time callback
    thread) — predates this work, documented but not fixed; fixing it needs the lock-free
    ring-buffer hand-off `docs/architecture.md §3` specifies, a separate, larger refactor.
- **Done:** **M4 — Windows WinForms C# client** ✓ complete (2026-06-17). Full details in the
  M4 section below. `ctest --preset m1-dev` — **14/14 tests** green. `dotnet build` — 0
  warnings/errors across all three C# projects. Manually verified: saved servers, TOFU
  first-connect dialog, channel tree, join, voice (VAD/PTT/always-on + sensitivity slider +
  per-user gain/mute/NR), text chat (channel + private), device pickers, level meter.
- **Next:** macOS/iOS Swift client (M4 continued) and/or **M5** (admin UI, moderation, kick/
  ban). The C ABI is complete and stable through M4 — both directions are unblocked. See
  `docs/roadmap.md §M4–M5`.

---

## Milestones (see [docs/roadmap.md](docs/roadmap.md) for full detail)

- [x] **M0 — Scaffolding** ✓ complete
- [x] **M1 — Control plane** ✓ complete (2026-06-15)
- [x] **M2 — Voice, single stream** ✓ complete (2026-06-16)
- [x] **M3 — Multi-stream & per-channel tuning** ✓ complete (2026-06-16)
- [x] **M4 — Native clients** — Windows WinForms ✓ (2026-06-17); macOS/iOS Swift pending
- [~] **M5 — Moderation, polish, beyond** (perms, bans, DRED; then file transfer, E2EE, …)

---

## M0 — Scaffolding ✓ (completed)

- [x] Repo layout (`core/ server/ tools/ clients/ tests/`), CMake + presets, vcpkg manifest.
- [x] C ABI header `core/include/voicecat.h` (full surface, stubbed).
- [x] Protocol source-of-truth `core/proto/voicecat.proto` (matches docs/protocol.md).
- [x] Core stubs for all six subsystems (net/crypto/codec/protocol/session/audio) + `vc_client`.
- [x] `voicecat-server` (arg parsing, config, stub run) and `vccli` (drives the C ABI).
- [x] CTest **smoke test** asserting the C ABI contract (not just "it compiles").
- [x] `.gitattributes` (LF), `.gitignore`, `.clang-format`, onboarding docs.
- **Verified:** `cmake --preset dev && cmake --build --preset dev && ctest --preset dev` → green.

---

## M1 — Control plane ✓ (completed 2026-06-15)

**Exit criterion:** ✓ `test_m1_integration` — two clients authenticate over TLS 1.3 (guest
+ Argon2id password), exchange channel and private text messages. Passes in ~1 s.

- [x] vcpkg baseline + `m1-dev` preset; `find_package` for protobuf/mbedTLS/libsodium/asio/sqlite3.
- [x] `FrameCodec` feed + emit; `encode_envelope` / `decode_envelope`.
- [x] Asio TCP acceptor + `TcpServerConn` (TLS path: blocking handshake thread + `tls_read_loop`).
- [x] `TlsContext` (mbedTLS 1.3, server cert/identity, ECDSA-P256 self-signed, TOFU on client).
- [x] `WorkerPool` (3 threads, used for Argon2id).
- [x] `Database` — SQLite, Argon2id via libsodium, `create_account` / `authenticate` / bootstrap admin.
- [x] `voicecat-admin` — account add/reset/del/list against live DB file.
- [x] `ServerIdentityManager` — generate/persist Ed25519 key + cert; fingerprint display.
- [x] `ConnSession` — WaitingHello → WaitingAuth → Authenticated state machine; full protocol relay.
- [x] `SessionRegistry` — channel tree, user map, broadcast, text routing.
- [x] `vc_client` (`client.cpp`) — full M1 C ABI: connect/TLS/ClientHello/AuthRequest/text/disconnect.
- [x] `Server::run()` — io_context, acceptor, worker pool, signal handling, `on_ready` callback.
- [x] `test_m1_integration` — M1 exit criterion. Verified green 2026-06-15.

**Key bug fixed:** double-framing in `ConnSession::send_envelope` — `encode_envelope` was
pre-framing the protobuf, then `TcpServerConn::send_frame` re-framed it. Fixed by serializing
raw protobuf bytes directly and letting `send_frame` add the single `[4-byte len]` prefix.

---

---

## M2 — Voice, single stream ✓ (completed 2026-06-16)

**Exit criterion:** ✓ `test_m2_voice` — two headless clients authenticate over TLS, bind UDP,
announce a MIC stream, send 50 encrypted Opus frames; server SFU relay re-encrypts + forwards
to the second client; B receives ≥ 25 frames and all decrypt correctly. Passes in ~4 s.

- [x] `m2-dev` preset (inherits `vcpkg-base`, binaryDir `build/m2-dev`); `m1-dev` also builds all M2 code.
- [x] `core/CMakeLists.txt` — `find_package(Opus)`, `find_path(MINIAUDIO_INCLUDE_DIR)`.
- [x] `core/src/net/voice_frame.h` — 14-byte UDP header (type/flags/codec/ssrc/seq/ts), serialize/parse, `make_udp_binding_packet`.
- [x] `SodiumMediaCrypto` — ChaCha20-Poly1305 AEAD; counter-nonce; 64-bit sliding-window anti-replay; `derive_send/recv` from TLS RFC 5705 exporter.
- [x] `OpusEncoder` / `OpusDecoder` — libopus 1.6, FEC, DTX, PLC (free; nullptr → decoder extrapolates).
- [x] `UdpMediaChannel` — async UDP socket (asio); thread-safe `send_to`; async recv loop.
- [x] `JitterBuffer` — per-ssrc, EWMA jitter estimation, adaptive depth 20–200 ms, late-drop at 500 ms.
- [x] `AudioEngine` — miniaudio capture+playback; `inject_capture()` bypass for headless tests; per-ssrc RemoteStream with OpusDecoder + JitterBuffer.
- [x] `ApmProcessor` — `ApmPassthrough` stub (VAD always open); WebRTC APM deferred until M3.
- [x] `on_tls_ready` callback in `TcpChannelCallbacks` — server derives and stores media AEAD keys immediately after TLS handshake.
- [x] `ConnSession` M2 — `udp_token` generated at construction; included in `AuthResult`; `handle_udp_binding` (verifies token, TCP ack); `handle_stream_announce` (assigns SSRC via registry); `udp_media_port` in `ServerHello`.
- [x] `SessionRegistry` M2 — `register_udp_token`, `find_by_udp_token`, `register_udp_endpoint`, `find_by_udp_endpoint`, `assign_ssrc`, `find_channel_sessions`, `user_channel`.
- [x] `MediaRelay` — SFU UDP relay; `kFrameUdpBinding` → endpoint binding; `kFrameVoice` → decrypt/re-encrypt/forward to channel members.
- [x] `Server::run()` — creates and binds `MediaRelay`; passes media port to `ConnSession`; wires `on_tls_ready` to derive per-connection media AEAD keys.
- [x] `test_voice_frame` — header round-trip, big-endian layout, binding packet format.
- [x] `test_media_aead` — seal/open round-trip, anti-replay, tamper detection, multi-packet sequence.
- [x] `test_opus_codec` — encode/decode round-trip energy check (within 3 dB), PLC, frame-samples helper.
- [x] `test_m2_voice` — M2 exit criterion (raw-socket harness). Verified green 2026-06-16.

**Follow-up (same day):** the above made `test_m2_voice` pass, but `vc_client`'s public voice
methods were still stubs — the *actual* M2 exit criterion ("two vccli/early-GUI clients talk")
wasn't met. Closed the gap:

- [x] `core/src/core/client.cpp` — real `stream_start`/`stream_stop`/`set_self_mute`/
  `set_remote_stream`; UDP-binding handshake (`start_udp_binding`/`handle_udp_binding_ack`/
  `finish_udp_binding`); media key derivation from `tls_` (RFC 5705 exporter); `run_udp_recv`
  (AEAD-open → `JitterBuffer::Frame` → `audio_engine_.push_recv_frame`); `on_capture_frame`
  (encode → seal → `sendto`); `sync_remote_streams` (diffs a `User` proto's `streams` against
  `remote_streams_`, wiring up `OpusDecoder`s and emitting `STREAM_STARTED`/`STOPPED`).
  `set_input_device`/`set_input_mode`/`set_push_to_talk`/`list_devices` remain
  `VC_ERR_NOT_IMPLEMENTED` — no device-enumeration backend yet; scoped to M3 (VAD/PTT).
- [x] `core/src/session/session.cpp/h` — `SessionModel::find_user`, `find_user_by_ssrc`,
  `Stream{stream_id, ssrc, kind, label, sample_rate, frame_ms}`.
- [x] `server/src/conn_session.cpp/h` — `handle_stream_announce`/`handle_stream_stop` now
  broadcast via `SessionRegistry::set_user_stream`/`clear_user_stream` → `UserEvent::UPDATED`.
- [x] `server/src/session_registry.cpp/h` — `set_user_stream`/`clear_user_stream` (mutate a
  user's `StreamInfo` list, return the updated `User` proto for broadcast).
- [x] `tests/test_voice_client_abi.cpp` — drives two real `vc_client` instances through
  `vc_connect`/`vc_authenticate_guest`/`vc_stream_start`/`vc_stream_stop`; asserts client B
  observes client A's `STREAM_STARTED`/`STOPPED` events. Verified green 2026-06-16.
- [x] `tools/vccli/src/main.cpp` — argv parsing (`--host/--port/--nick/--channel/--voice/
  --mute/--text`); `--voice` starts a MIC stream and blocks on SIGINT, printing `on_event`
  callbacks live (unbuffered stdout — MinGW/MSVCRT treat `_IOLBF` as full buffering for
  non-console streams). Dropped the originally-planned `--voice-loopback` and the
  `tx=N rx=M lost=K jitter=J` stats line: `voicecat.h` exposes no PCM-injection hook or
  jitter/loss stats getter publicly, only `on_event` + `on_level` (RMS). Manually verified:
  two `vccli --voice` instances see each other's stream start in real time.

---

## M3 — Multi-stream & per-channel tuning ✓ (completed 2026-06-16)

**Exit criterion:** ✓ `test_m3_multistream` — a real `vc_client` (A) runs two concurrent local
streams (MIC + SCREEN_AUDIO) with distinct stream ids; a second client (B) sees both as
separate `STREAM_STARTED` events and a `VC_EVENT_TALK_STATE` talking edge for A's MIC stream;
B independently sets gain/mute/noise-reduction on each of A's streams without one call
affecting the other; A then joins "Music Room" (channel 2: stereo/128kbps/`OPUS_AUDIO`/no
DTX) and announces a fresh MIC stream there, while B stays in "Lobby" (channel 1: mono/24kbps/
`OPUS_VOIP`/DTX on) — `vc_get_stream_audio_config` shows their effective Opus config differs
exactly as the server enforces per channel. Passes in ~2.4s; verified across 8 consecutive
standalone runs + 3 consecutive full-suite runs with no flakes.

Exploration before implementing turned up several bugs/gaps where the wire format already
supported this milestone but the client/server logic didn't — these were fixed as part of M3,
not treated as pre-existing-and-out-of-scope:

- [x] **Server `stream_id` bug** — `handle_stream_announce` always wrote `stream_id=1`, so a
  second stream from the same user silently overwrote the first in
  `SessionRegistry::set_user_stream`'s replace-by-id logic. Fixed with a per-session counter
  (`ConnSession::next_stream_id_`) + `announced_stream_ids_` (also now validated in
  `handle_stream_stop`, rejecting stops for ids the session never announced).
- [x] **Per-channel `AudioConfig` was modeled but never populated/enforced.**
  `SessionRegistry::init_default_channels()` now seeds Lobby (id=1: mono, 24kbps, `OPUS_VOIP`,
  FEC+DTX on) and a new "Music Room" (id=2: stereo, 128kbps, `OPUS_AUDIO`, FEC+DTX off) with
  real `AudioConfig`s; new `SessionRegistry::channel_audio_config(channel_id)` accessor (there
  was no per-id channel getter before, only `channel_snapshot()`). `handle_stream_announce`
  now treats the channel's config as authoritative (mode/frame_ms/application/fec/dtx/
  complexity), clamping (not overriding) `bitrate_bps` to the channel's ceiling.
- [x] **Client silently dropped `mode`/`dtx`/`complexity`/`application` from `effective_audio`**
  even for the single M2 stream — `handle_stream_announce_result` and `sync_remote_streams`
  only copied `sample_rate`/`bitrate_bps`/`frame_ms`/`fec` into `OpusParams`. New shared
  `opus_params_from_audio_config()` helper (`client.cpp`) fixes both the send and receive
  paths.
- [x] `core/src/codec/opus_codec.h/.cpp` — new `OpusApplication` enum + `OpusParams::application`
  field; `OpusEncoder::init` now honors it instead of hardcoding `OPUS_APPLICATION_VOIP`.
- [x] `core/src/session/session.h/.cpp` — `Stream` struct extended with the full `AudioConfig`
  (mode/bitrate_bps/application/fec/expected_packet_loss/dtx/complexity), not just
  sample_rate/frame_ms; `copy_streams()` now copies all of it.
- [x] `core/src/core/client.h/.cpp` — local-stream state is now a `std::unordered_map<int,
  LocalStream>` keyed by `vc_stream_kind` (one active stream per kind — MIC/SCREEN_AUDIO/
  AUX_DEVICE are each singletons for a client), replacing the M2 single-stream fields.
  `StreamAnnounce`/`StreamAnnounceResult` round-trips are now correlated by `request_id`
  (already round-tripped on the wire; just wasn't read) via `pending_announce_kind_`, so
  multiple concurrent announces from one client resolve to the right `LocalStream`.
  `on_capture_frame` takes a `kind` parameter and upmixes mono capture to stereo (duplicate
  L=R) when a stream's channel config calls for it. `vc_set_self_mute`'s `mic_muted` only
  gates the `MIC` kind — a concurrent `SCREEN_AUDIO` share keeps playing while muted.
  `set_remote_stream` now actually wires `noise_reduction` through (previously parsed and
  discarded). New `run_talk_timer()` (a small dedicated thread, started alongside the UDP
  media path, never the miniaudio callback thread) polls both remote talk-state edges
  (`AudioEngine::poll_talk_transitions()`) and local capture-activity edges, emitting
  `VC_EVENT_TALK_STATE`.
- [x] **Fixed a thread-join race in `teardown_voice()`** — it's called both from `run_io()`'s
  own cleanup and from `disconnect()`, on different threads; without serialization both could
  see `udp_thread_`/`talk_timer_thread_` as `joinable()` simultaneously and race to `join()`
  the same `std::thread` (UB; surfaced as an intermittent `std::system_error: No such process`
  under `ctest`). Added a `teardown_mu_` guard around the whole function. This pre-existed for
  `udp_thread_` alone (likely the same root cause as the `test_m1_integration`/`test_m2_voice`
  cleanup-path flake noted in the M2 section above) — adding `talk_timer_thread_`'s join just
  made it surface more often, so it was fixed properly here rather than carried forward again.
- [x] `core/src/audio/audio_engine.h/.cpp` — `CaptureCallback` gained a `kind` parameter
  (the real miniaudio capture device is always tagged `kind=0`/MIC; a second concurrent local
  stream is fed via its own `inject_capture(kind, ...)` ring buffer — `inject_taps_`, keyed by
  kind — since there is only one real hardware capture device in M3). Fixed a buffer-sizing
  bug in `on_playback`'s per-stream decode (`opus_decode`'s `frame_size` parameter is
  samples-*per-channel*, not total samples — the old code passed `frames * params_.channels`,
  which would have overflowed the decode buffer for any stereo stream). Stereo decoder output
  is downmixed (avg L/R) into the engine's mono mix accumulator immediately after decode.
  `RemoteStream` gained `recv_ns`/`noise_reduction_enabled` (lazy `ApmProcessor` instantiation
  — freed on disable, so no separate instance cap is needed per the roadmap's guidance) and
  `last_voice_ms`/`talking` (talk-indicator edge state, updated in `push_recv_frame`); new
  `set_stream_noise_reduction()` and `poll_talk_transitions()`. Note: until `VOICECAT_HAS_APM`
  is wired to a real WebRTC APM build, the NS toggle is plumbed end-to-end but behaviorally a
  passthrough no-op (`ApmPassthrough` doesn't touch PCM) — same situation send-side APM has
  been in since M2; M3's job was the plumbing, not the DSP backend.
- [x] **New C ABI surface** (`core/include/voicecat.h`, additive only):
  `vc_audio_config` struct + `vc_get_stream_audio_config(c, user_id, stream_id, out)` — the
  effective Opus config for a stream you own or a peer's, reading from the (now richer)
  `LocalStream`/`session::Stream`. `vc_test_inject_capture(c, stream_id, pcm, samples)` —
  clearly-marked **test-only**, forwards to `AudioEngine::inject_capture`, so
  `test_m3_multistream` can drive two concurrent synthetic-audio streams through the real ABI
  without a microphone.
- [x] `tests/test_m3_multistream.cpp` — the M3 exit criterion (ABI-level, mirrors
  `test_voice_client_abi.cpp`'s approach per the M2 lesson). Registered in `tests/CMakeLists.txt`.

**Explicitly out of scope for this pass** (confirmed with the user before implementing):
- `vc_set_input_device`/`vc_set_input_mode`/`vc_set_push_to_talk`/`vc_list_devices` (device
  enumeration + VAD/PTT input gate) — still `VC_ERR_NOT_IMPLEMENTED`. These were mentioned as
  "scoped to M3" in the M2 follow-up notes above, but docs/roadmap.md's M3 bullets never
  actually listed them — deferred again, now tracked explicitly rather than implicitly.
- Real WASAPI desktop-audio loopback capture for `SCREEN_AUDIO` — the engine now supports
  feeding a second concurrent local stream via `inject_capture`, but only synthetic PCM is
  wired up; a real loopback capture device is a follow-up.
- True stereo *playback output* — `AudioEngine`'s mixer/output device stays mono; stereo
  streams are downmixed after decode (see above). The Opus wire format itself is fully
  stereo-correct.

---

## Post-M3 follow-up — device enumeration, VAD/PTT gate, stereo playback, WASAPI loopback ✓ (completed 2026-06-16)

Closes all three items the M3 section above explicitly carried forward as out of scope.

**Exit verification:** `ctest --test-dir build/m1-dev` — **12/12 tests** green (3 consecutive
full-suite runs), including the new `test_vad_ptt_devices` (device enumeration + VAD/PTT gate
through real `vc_client`s against a real server, plus a white-box `AudioEngine` stereo-mix
check — no audio hardware needed for that last part). Also verified 5 consecutive standalone
runs of the new test alone, no flakes. Manually verified live on Windows: `vccli
--list-devices` against real hardware (3 input / 4 output devices, correct `is_default`
flags), and `vccli --voice --input-mode vad` connecting + streaming without incident.

- [x] **Device enumeration** (`vc_list_devices`) — `AudioEngine::enumerate_devices(bool
  capture)` (static, works without a running engine — inits a throwaway `ma_context` via
  `ma_context_get_devices`). `device_id`/`vc_device.id` is an opaque hex-encoded raw
  `ma_device_id` (not the device name — names aren't guaranteed unique); documented as an
  internal contract callers must round-trip, never construct by hand. `vc_client::list_devices`
  works in any connection state (no `VC_STATE_CONNECTED` gate) since device pickers need to
  populate pre-connect. `vc_free_device_list` is now a real free (was a no-op stub).
- [x] **Input device selection** (`vc_set_input_device`) — stores the device id on the
  targeted `LocalStream` (new field); for the MIC stream, if the engine is already running,
  restarts it (`stop()` + `ensure_audio_running()`) to pick up the new device. Simplified: it
  restarts unconditionally rather than trying to detect whether the device id actually
  changed (`AudioEngine` has no getter for "current device").
- [x] **VAD/PTT input gate** (`vc_set_input_mode`, `vc_set_push_to_talk`) — new
  `EnergyVadProcessor` (`core/src/audio/apm_processor.cpp`) implementing the existing
  `ApmProcessor` interface: energy/RMS threshold (default ~0.025 normalized) + hang-time
  (default 300 ms, matching `kTalkHangoverMs`). New factory `ApmProcessor::create_vad()`,
  kept separate from `create()` (which recv-side per-stream NS still uses, unaffected by this
  pass). `vc_client` gained `current_input_mode_`/`ptt_active_`/`mic_vad_`; the gate is
  inserted in `on_capture_frame`, **MIC-only** — `SCREEN_AUDIO`/`AUX_DEVICE` always bypass it
  (gating a desktop-audio share on the user's own voice activity would silently drop shared
  music/video audio). `last_capture_ms` (drives the talk indicator) is now updated *after* the
  gate check, not before, so a VAD/PTT-closed frame never shows as "talking". `mic_vad_` is
  constructed once the MIC stream's `StreamAnnounceResult` lands (on `io_thread_`), not lazily
  inside the capture path.
- [x] **True stereo playback** — `AudioParams::channels` split into `capture_channels` (stays
  1) and `playback_channels` (now 2, unconditionally). `AudioEngine::on_playback` no longer
  downmixes decoded stereo streams to mono before mixing — stereo decode output is mixed
  directly (L→L, R→R); mono decode output is upmixed (duplicated into both channels). Falls
  back to a 1-channel playback device once if the 2-channel `ma_device_init` fails (unusual
  hardware). New test-only `AudioEngine::mix_for_test()` exposes the mixer for white-box
  testing without a real `ma_device`.
- [x] **WASAPI loopback capture for `SCREEN_AUDIO`** — new `VOICECAT_HAS_LOOPBACK` macro
  (`core/CMakeLists.txt`, Windows-only). `AudioEngine` gained a separate `loopback_device_`
  (own lifecycle, decoupled from the mic capture/playback devices) with
  `start_loopback_capture()`/`stop_loopback_capture()`, using miniaudio's
  `ma_device_type_loopback` against the default render endpoint. Its callback feeds
  `capture_cb_` directly (same pattern as the real mic capture device), **not** through
  `inject_capture()`'s test-only ring. Wired into `vc_client::handle_stream_announce_result`
  (start, alongside `ensure_audio_running()`) and `stream_stop` (stop) for
  `VC_STREAM_SCREEN_AUDIO`. Non-Windows builds keep `vc_test_inject_capture` as the only way to
  feed `SCREEN_AUDIO`.
- [x] `tools/vccli/src/main.cpp` — new flags `--list-devices`, `--input-device`,
  `--input-mode vad|ptt`, `--share-screen-audio`; while `--voice` is running, a background
  stdin-reader thread accepts `ptt on`/`ptt off`/`mode vad`/`mode ptt` (the most portable way
  to drive PTT interactively from a headless CLI — no SIGUSR1 equivalent on Windows). Also
  prints `VC_EVENT_TALK_STATE`. Known minor caveat: on Windows the stdin-reader thread is
  detached (not joined) on exit, since `std::getline` can't be interrupted from another thread
  — a `vc_client*` use-after-free is theoretically possible if a command line arrives in the
  brief window between teardown and process exit; acceptable for a headless test/dev tool.
- [x] `tests/test_vad_ptt_devices.cpp` — new test covering all four items above; registered in
  `tests/CMakeLists.txt`. `tests/test_smoke.cpp`'s device-list assertion is now conditional on
  `VOICECAT_HAS_AUDIO` (was a hard `VC_ERR_NOT_IMPLEMENTED` assertion) — `VC_OK` only, never
  `count > 0` (a headless CI build agent may legitimately report zero audio devices).

**Still explicitly out of scope** (carried forward, not silently dropped):
- Real `webrtc-audio-processing`/AEC — no working Windows/MSVC build upstream (see
  docs/roadmap.md §2's superseding decision-log entry). There is **no AEC, NS, or AGC
  implementation at all**, not just a deferred VAD. The per-stream NS toggle
  (`vc_set_remote_stream(..., noise_reduction)`) is unaffected by this pass and stays exactly
  as inert as it was after M3 (`ApmPassthrough`, no PCM modification) — don't mistake this
  pass for having fixed it.
- macOS/iOS `SCREEN_AUDIO` capture (ScreenCaptureKit / ReplayKit) — Windows-only loopback in
  this pass.
- Process-specific WASAPI loopback — whole-device capture only; inherently captures this
  app's own incoming voice mix along with everything else playing.
- The pre-existing RT-thread rule violation in `on_capture_frame`/`AudioEngine::on_capture`
  (mutex lock, heap allocation for the stereo-upmix path, blocking `sendto`, all on the
  miniaudio real-time callback thread) — predates this work (was already present in M2/M3);
  documented here explicitly rather than silently carried forward again. Fixing it properly
  needs the lock-free ring-buffer hand-off `docs/architecture.md §3` specifies — a separate,
  larger refactor, out of scope for this pass.

---

## M4 — Windows WinForms C# client ✓ (completed 2026-06-17)

**Exit criterion:** ✓ `ctest --preset m1-dev` — **14/14 tests** green (existing 12 + 2 new
C++ tests: `test_channel_user_list_abi`, `test_tofu_flow`). `dotnet build` — 0 warnings/errors.
Manually verified: connect, TOFU first-connect dialog, channel tree, join, voice, text, device
pickers, level meter. Accessibility: explicit `AccessibleName`/`AccessibleDescription` on every
control, `&` mnemonics on every button, activity-log `ListBox` as screen-reader record.

**New C++ ABI surface** (all additive, backward-compatible):
- `vc_list_channels` / `vc_list_users` / `vc_list_user_streams` — pull-based snapshot getters
  for the channel tree + user list; `session_model_mu_` added for cross-thread safety.
  `SessionModel::apply_snapshot`/`apply_channel_event` fixed to populate `parent_id`,
  `password_protected`, `max_users` (were permanently zeroed despite the struct declaring them).
- `vc_join_channel(channel_id, password)` — join with optional password; server replies via
  new `VC_EVENT_JOIN_RESULT`.
- `VC_EVENT_SERVER_IDENTITY` + `vc_confirm_server_identity(accept)` — TOFU gate that blocks
  `io_thread_` until the UI approves or rejects. Pins the TLS leaf-cert SHA-256 fingerprint
  (verifiable directly at handshake), **not** the declared Ed25519 value (see docs/security.md
  §1.1 for why — the TLS cert and Ed25519 key are generated independently, no binding).
  `vc_get_server_identity_display` exposes the Ed25519 fingerprint for human-readable display.
- `vc_config::tofu_store_path` — optional per-user pin file path; defaults to a relative
  `"./voicecat_tofu_pins.txt"` so existing tests need no change.
- `TcpAcceptor` now dual-stacks (IPv6 + IPv4 fallback) so `localhost` → `::1` on Windows
  connects correctly without forcing users to type `127.0.0.1`.
- `VC_INPUT_ALWAYS_ON = 2` in `vc_input_mode` — transmit unconditionally (no VAD gate).
- `vc_set_vad_threshold(float)` — live VAD threshold update; `EnergyVadProcessor` stores it
  atomically so the RT capture path reads without a lock or allocation.

**New C++ tests:**
- `test_channel_user_list_abi` — snapshot getters, `parent_id`/`password_protected`/`max_users`
  regression, per-user stream list, invalid-user-id error, double-free idempotency.
- `test_tofu_flow` — first-connect blocks until confirmed; reject doesn't persist; reconnect to
  same identity reports `MATCHED`; rotated identity reports `MISMATCH`; `vc_confirm_*` with
  nothing pending returns an error.

**`windows-client` CMake preset** — Release, `VOICECAT_BUILD_SHARED=ON`, static MinGW runtime
(`-static-libgcc -static-libstdc++ -static -lwinpthread`), no tools/tests. Outputs
`build/windows-client/bin/voicecat.dll` with zero MinGW DLL dependencies (only Windows system
DLLs remain — verified via `objdump -p`).

**C# solution** (`clients/windows/`, .NET 10 LTS `net10.0-windows`):
- `VoiceCat.Interop` — `[LibraryImport]` P/Invoke surface, `[UnmanagedCallersOnly]` callbacks,
  `System.Threading.Channels.Channel<VoiceCatEvent>` event delivery drained by 30ms WinForms
  Timer; `VoiceCatClientHandle : SafeHandle` guarantees `vc_client_destroy`.
- `VoiceCat.App` — WinForms UI:
  - `ConnectDialog` — saved-server `ListBox`, Add/Remove/Edit; servers persisted to
    `%AppData%\VoiceCat\servers.json`; passwords DPAPI-encrypted (`ProtectedData`, opt-in).
  - `ServerIdentityDialog` — shown only on `FIRST_CONNECT`/`MISMATCH` (never `MATCHED`);
    mismatch text and button ordering are starkly different ("WARNING" framing, Cancel default).
  - `MainForm` — `TreeView` channel tree, `ListBox` user list, `RichTextBox` chat, scope
    `ComboBox` (Channel/Private), activity-log `ListBox`, voice panel with mic toggle,
    mute/deafen checkboxes, VAD/PTT/Always-On radio group, VAD sensitivity `TrackBar`
    (1–100, hidden for non-VAD modes), device `ComboBox` + refresh, level `ProgressBar`.
  - `PerUserTuningDialog` — real-time gain `TrackBar` + mute/NR checkboxes; applied to all
    of a user's streams immediately (no OK/Cancel round-trip).
  - `PttKeyCaptureDialog` — focus-scoped PTT key capture.
- `VoiceCat.Interop.Tests` — xunit smoke test: connect → TOFU → guest auth → list channels
  purely via P/Invoke against a live `voicecat-server.exe`.

**Explicitly out of scope for this pass:**
- macOS/iOS Swift client — pending.
- Admin/moderation UI (kick/ban/permissions/account provisioning) — server-side dispatch for
  these messages is M5's job; building the UI now would require building the server side too.
- PTT hotkey is **focus-scoped only** (works while VoiceCat window has focus). A system-wide
  `WH_KEYBOARD_LL` hook would require escalated permissions and risk AV flagging — documented
  limitation, not silently omitted.
- Receive-side noise reduction (`vc_set_remote_stream(..., noise_reduction)`) is end-to-end
  plumbed but behaviorally a passthrough no-op (`ApmPassthrough`, no PCM modification) —
  same as before M4. The per-user NR checkbox in `PerUserTuningDialog` is labeled accordingly.

---

## M5 — Moderation, polish, and beyond [~] (in progress 2026-06-17)

**Exit criterion:** four ABI-level tests green (`test_m5_permissions`,
`test_m5_kick_ban_move_mute`, `test_m5_admin_accounts`, `test_m5_channel_crud`);
`vccli` can drive all moderation/admin/channel operations against a live server.

- [x] **Server-side moderation & permissions:**
  - `server/src/session_registry.h/.cpp` — per-session `Permissions`, permission helpers
    (`can_kick`, `can_ban`, etc.), kick/ban/move/server-mute, channel CRUD, DB-backed channel
    tree load/save, in-memory channel state.
  - `server/src/conn_session.cpp` — M5 dispatch handlers, permission checks, channel-password
    + `max_users` enforcement, `UserEvent::UPDATED` broadcast on join/leave.
  - `core/proto/voicecat.proto` — `ServerMuteRequest`, `UserEvent.reason`, `User.server_deafened`,
    `ListAccountsResult`, `AccountEntry`.
- [x] **C ABI / client-side:**
  - `core/include/voicecat.h` — `vc_permissions`, `vc_channel_info`, `vc_kick_user`,
    `vc_ban_user`, `vc_set_permission`, `vc_set_server_mute`, `vc_move_user`,
    `vc_create_channel`, `vc_edit_channel`, `vc_delete_channel`, `vc_create_account`,
    `vc_reset_password`, `vc_delete_account`, `vc_list_accounts`, `vc_get_permissions`;
    new events `VC_EVENT_GENERIC_RESULT` and `VC_EVENT_ACCOUNT_LIST`.
  - `core/src/voicecat.cpp`, `core/src/core/client.h/.cpp` — implementations + server-mute/deafen
    gating on the client.
- [x] **Database:** `server/src/db.h/.cpp` schema v2 (`channels`, `bans`), Argon2id accounts,
  BLAKE2b channel passwords, migrations.
- [x] **Tests:** four new M5 tests registered in `tests/CMakeLists.txt`:
  - `test_m5_permissions` — grant/revoke permissions, verify enforcement.
  - `test_m5_kick_ban_move_mute` — kick, ban, move, server-mute/deafen.
  - `test_m5_admin_accounts` — create/reset/delete/list accounts.
  - `test_m5_channel_crud` — create/edit/delete channels, password + max_users enforcement.
- [x] **vccli** (`tools/vccli/src/main.cpp`) — all M5 operations exposed via flags; account auth
  via `--username`/`--password`; async `VC_EVENT_GENERIC_RESULT`/`VC_EVENT_ACCOUNT_LIST` handling.
- [x] **Docs** kept in sync: `docs/protocol.md`, `docs/security.md`, `PROGRESS.md`.

**Key bug fixed:** `test_m5_channel_crud` failed because `SessionRegistry::create_channel`
broadcast `ChannelEvent::CREATED` from a moved-from `entry.proto` after
`channels_[id] = std::move(entry)`. Fixed by building the event before moving into the map.

**Still to do:**
- DRED/audio-quality polish.
- Windows admin/moderation UI in the WinForms client.
- macOS/iOS Swift client (carried from M4).

---

## Decisions log

All architecture/scope decisions are settled and recorded in
[docs/roadmap.md §2 "Resolved decisions"](docs/roadmap.md) and reflected across `docs/`.
If you make a *new* decision, record it there and link it here.

---

## How to update this file

1. Check off tasks as you complete them; flip a milestone to `[x]` only when its **exit
   criterion test** passes.
2. Keep the **"Where we left off / next action"** block at the top accurate — it's the first
   thing the next agent reads.
3. When you start a milestone, copy its task list from `docs/roadmap.md` into a section here.