44a336cc894d7650e7e4d0bf0a40fa34dbbb585f
4 Commits
| Author | SHA1 | Message | Date | |
|---|---|---|---|---|
| 6fe7bf0158 |
feat: fix voice join/leave, channel edit defaults, channel-update stream restart
Three bugs fixed across the full stack (proto/server/core/ABI/Win/macOS/iOS): 1. Join/Leave Voice now truly subscribes/unsubscribes from the voice plane. Previously the button only toggled the local mic — receiving was always on (gated by channel membership alone). Added a protocol-level voice subscription concept: new SubscribeVoiceRequest/UnsubscribeVoiceRequest/VoiceSubscriptionResult proto messages, User.voice_subscribed field, vc_join_voice/vc_leave_voice C ABI functions, VC_EVENT_VOICE_STATE event, server-side voice_subscribed flag checked by the SFU relay recipient filter, and core-client gating of remote-stream decoder setup. All three clients rewired to subscribe+mic on Join / unsubscribe on Leave. Text chat works regardless of voice subscription. 2. Channel edit dialog now shows the channel's actual current settings. The read struct vc_channel was missing sort_order and audio fields — only the write struct vc_channel_info had them. Extended vc_channel with both (additive, no ABI break), updated the session model and list_channels marshaling to populate them, and updated all three clients' edit callers to use actual channel info instead of hardcoded defaults. 3. Channel parameter updates now automatically restart everyone's streams. Previously editing a channel's audio config persisted and broadcast a ChannelEvent::UPDATED, but no layer restarted streams — encoders/decoders are frozen at announce time. handle_channel_event now detects audio-config changes on the user's current channel and stop->starts each active local stream. The server reads the updated config on re-announce; peers wire up fresh decoders at the new ssrc. All 29 CTest tests pass; Windows DLL + C# client build clean. Apple clients not yet compile-verified (Windows environment). |
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| d397731db9 |
feat(audio): per-stream mix controls in Windows client + vc_get_remote_stream getter
PerUserTuningDialog previously broadcast one gain/mute/NR set to *all* of a user's streams, even though the core mixer (AudioEngine::RemoteStream) and the C ABI (vc_set_remote_stream) were already per-stream. The UI had no per-mix controls anywhere. Reworks the dialog to enumerate ListUserStreams on open and render one row per stream (kind + label + Gain + Mute + NR), each wiring only to its own stream_id. Adds a read-back ABI counterpart, vc_get_remote_stream, so the dialog opens at the listener's actual current per-stream settings (defaults 1.0/unmuted/NR-off) rather than always 100%. Additive ABI change only; no existing symbols touched. Tests: test_m3_multistream extended with getter round-trip assertions; new C# smoke test exercises the full P/Invoke marshaling path with two clients. Docs: voice.md §10 notes the getter. NR checkbox keeps its honest 'passthrough' label (NS DSP still unbuilt per §8). |
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| 63b241cc2e |
feat(M4): Windows WinForms client, TOFU identity pinning, VAD threshold + always-on mode
Core ABI extensions (voicecat.h): - vc_list_channels / vc_list_users / vc_list_user_streams — pull-based snapshot getters for the channel-tree and user-list UI; session_model_mu_ guards cross-thread reads - VC_EVENT_JOIN_RESULT / vc_join_channel — channel join with optional password - VC_EVENT_SERVER_IDENTITY + vc_confirm_server_identity — TOFU gate that blocks io_thread_ until the UI approves or rejects; pins TLS leaf-cert SHA-256 (not declared Ed25519) - vc_get_server_identity_display — Ed25519 fingerprint for human-readable display only - VC_INPUT_ALWAYS_ON = 2 in vc_input_mode — transmit unconditionally, no VAD gate - vc_set_vad_threshold — live RMS threshold update (0.0–1.0); EnergyVadProcessor stores it atomically so the audio RT path reads without a lock C++ implementation: - SessionModel::apply_snapshot / apply_channel_event fixed to populate parent_id, password_protected, and max_users (were permanently zeroed) - TlsContext::peer_cert_fingerprint — SHA-256 of peer leaf cert DER via mbedTLS - TofuStore split into peek (read-only) + pin (write) so first-connect only persists after user approval; tofu_store_path in vc_config for per-user pin file location - TcpAcceptor uses dual-stack IPv6+IPv4 fallback (fixes localhost → ::1 on Windows) - windows-client CMake preset: Release shared DLL, static MinGW runtime, no tools/tests - New C++ tests: test_channel_user_list_abi, test_tofu_flow (14/14 green) Windows client (clients/windows/ — .NET 10 WinForms): - VoiceCat.Interop: LibraryImport P/Invoke surface, UnmanagedCallersOnly callbacks, Channel<VoiceCatEvent> event delivery drained by 30ms WinForms Timer - VoiceCat.App: ConnectDialog (saved servers, DPAPI password storage), ServerIdentity- Dialog (TOFU first-connect / mismatch warning), MainForm (channel TreeView, user ListBox, RichTextBox chat, voice controls, device pickers, VAD/PTT/always-on mode, per-user gain/mute/NR tuning, VAD sensitivity TrackBar, level meter ProgressBar) - PttKeyCaptureDialog — focus-scoped PTT key capture (documented limitation) - PerUserTuningDialog — real-time gain/mute/NR applied to all of a user's streams - Accessibility: explicit AccessibleName/Description on every control, & mnemonics, Activity log ListBox as durable screen-reader record, AutomationNotification for curated live announcements Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com> |
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| 867557eda1 |
feat(M3): multi-stream & per-channel tuning
Implements docs/roadmap.md M3: multiple concurrent streams per user (MIC + SCREEN_AUDIO + AUX_DEVICE), independent per-stream receiver gain/mute/noise- reduction, talk indicators, and enforced per-channel Opus configurability (mono/stereo, bitrate, frame size, FEC/DTX, application). Bugs fixed along the way (found while implementing, not pre-existing scope): - Server hard-coded stream_id=1 for every announce, so a second stream from the same user silently overwrote the first in SessionRegistry::set_user_stream. Now a per-session counter (ConnSession::next_stream_id_); handle_stream_stop validates against announced_stream_ids_ before clearing. - Client dropped mode/dtx/complexity/application from effective_audio even for the single M2 stream -- only sample_rate/bitrate_bps/frame_ms/fec were ever applied to OpusParams. Fixed on both the send (handle_stream_announce_result) and receive (sync_remote_streams) paths via a shared opus_params_from_audio_config() helper. - OpusEncoder always used OPUS_APPLICATION_VOIP; added OpusParams::application and wired it through. - on_playback's per-stream decode passed the wrong frame_size to opus_decode (total samples instead of samples-per-channel), which would have overflowed the decode buffer for any stereo stream. - teardown_voice() raced when called concurrently from run_io()'s own cleanup and from disconnect() on a different thread -- both could see udp_thread_/talk_timer_thread_ as joinable() at once and race to join() the same std::thread (intermittent std::system_error under ctest). Fixed with a teardown_mu_ guard instead of carrying the flake forward. New: - Per-channel AudioConfig: SessionRegistry now seeds Lobby (mono/24kbps/VOIP/ FEC+DTX) and a new "Music Room" channel (stereo/128kbps/AUDIO/no DTX); handle_stream_announce enforces the channel's config, clamping (not overriding) bitrate_bps to its ceiling. - core/src/core/client.h/.cpp: local-stream state is now a std::unordered_map<int, LocalStream> keyed by vc_stream_kind, with request_id-correlated announce/result handling (request_id already round-tripped on the wire; just wasn't read before). on_capture_frame is kind-aware and upmixes mono capture to stereo when a stream's config calls for it. set_self_mute's mic_muted now only gates the MIC kind. NS is wired through set_remote_stream. New run_talk_timer() thread emits VC_EVENT_TALK_STATE from both remote and local edge detection. - core/src/audio/audio_engine.h/.cpp: kind-keyed injection taps (inject_capture), stereo-to-mono downmix at the decode/mix boundary, RemoteStream gains recv_ns (lazy ApmProcessor) + noise_reduction_enabled and last_voice_ms/talking; new set_stream_noise_reduction() and poll_talk_transitions(). - core/src/session/session.h/.cpp: Stream now carries the full AudioConfig, not just sample_rate/frame_ms. - New additive C ABI (core/include/voicecat.h): vc_audio_config + vc_get_stream_audio_config (effective Opus config for any stream you own or a peer's); vc_test_inject_capture (test-only synthetic PCM injection, clearly marked, mirrors AudioEngine::inject_capture). - tests/test_m3_multistream.cpp: the M3 exit criterion through the real ABI (mirrors test_voice_client_abi.cpp's approach, not raw sockets) -- two concurrent local streams, independent gain/mute/NS control, per-channel config divergence via vc_get_stream_audio_config, talk indicators. Explicitly out of scope for this pass (tracked in PROGRESS.md, not silently dropped): VAD/PTT input gate + device enumeration; real WASAPI loopback capture for SCREEN_AUDIO (synthetic injection only); true stereo playback output (AudioEngine's mixer/output device stays mono -- Opus itself is fully stereo-correct on the wire). ctest --test-dir build/m1-dev: 11/11 green, verified across 3 consecutive full-suite runs plus 8 standalone runs of the new test. Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com> |
