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.

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▶ Where we left off / next action

• 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_clients 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.
• Next: M4 — native clients (Windows C#, macOS/iOS Swift). See docs/roadmap.md §M4.

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

• M0 — Scaffolding ✓ complete
• M1 — Control plane ✓ complete (2026-06-15)
• M2 — Voice, single stream ✓ complete (2026-06-16)
• M3 — Multi-stream & per-channel tuning ✓ complete (2026-06-16)
• M4 — Native clients (Windows C#, macOS/iOS Swift) ← next
• M5 — Moderation, polish, beyond (perms, bans, DRED; then file transfer, E2EE, …)

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M0 — Scaffolding ✓ (completed)

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

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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.

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

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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.

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

• 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 OpusDecoders 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).
• core/src/session/session.cpp/h — SessionModel::find_user, find_user_by_ssrc, Stream{stream_id, ssrc, kind, label, sample_rate, frame_ms}.
• server/src/conn_session.cpp/h — handle_stream_announce/handle_stream_stop now broadcast via SessionRegistry::set_user_stream/clear_user_stream → UserEvent::UPDATED.
• 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).
• 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.
• 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.

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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:

• 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).
• 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 AudioConfigs; 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.
• 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.
• core/src/codec/opus_codec.h/.cpp — new OpusApplication enum + OpusParams::application field; OpusEncoder::init now honors it instead of hardcoding OPUS_APPLICATION_VOIP.
• 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.
• 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.
• 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.
• 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.
• 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.
• 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.

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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_clients 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.

• 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).
• 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").
• 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.
• 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.
• 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.
• 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.
• 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.

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Decisions log

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

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How to update this file

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