Files
voice-cat/core/src/core/client.h

424 lines
23 KiB
C
Raw Normal View History

/*
* client.h the implementation type behind the opaque `vc_client*` handle.
*/
#ifndef VOICECAT_CORE_CLIENT_H
#define VOICECAT_CORE_CLIENT_H
#include "voicecat.h"
2026-06-15 23:48:44 +02:00
#include <atomic>
#include <condition_variable>
#include <deque>
#include <mutex>
#include <optional>
#include <string>
#include <thread>
#include <unordered_map>
#include <utility>
2026-06-15 23:48:44 +02:00
#include <vector>
#include "audio/audio_engine.h"
#include "codec/opus_codec.h"
2026-06-15 23:48:44 +02:00
#include "crypto/crypto.h"
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>
2026-06-17 00:35:16 +02:00
#include "crypto/tofu_store.h"
#include "net/voice_frame.h"
2026-06-15 23:48:44 +02:00
#include "protocol/envelope.h"
#include "protocol/protocol.h"
#include "session/session.h"
#include "proto/voicecat.pb.h"
struct vc_client {
vc_client(const vc_config& cfg, vc_callbacks cb);
~vc_client();
vc_client(const vc_client&) = delete;
vc_client& operator=(const vc_client&) = delete;
vc_result connect(const char* host, uint16_t port);
vc_result disconnect();
vc_result authenticate_guest(const char* nickname);
vc_result authenticate_user(const char* username, const char* password);
vc_result join_channel(uint32_t channel_id, const char* password);
vc_result leave_channel();
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).
2026-06-24 14:29:39 +02:00
vc_result join_voice();
vc_result leave_voice();
vc_result stream_start(const vc_stream_desc& desc, uint32_t* out_stream_id);
vc_result stream_stop(uint32_t stream_id);
vc_result set_input_device(uint32_t stream_id, const char* device_id);
feat(ios): audio overhaul, Join/Leave Voice, channel-id sync fix, stereo mic capture Three iOS client problems fixed plus a new core stereo-mic capture ABI: 1. Channel-id sync bug (mic button permanently dimmed): SessionState never synced currentChannelId from the self user's channelId on connect, so the mic button (gated on currentChannelId == 0) stayed dimmed. Added syncSelfChannel() (mirrors macOS MainWindowController.swift:461,491,522); called from init/.channelList/.userJoined/.userLeft/.userUpdated/.joinResult. Added applyServerMuteState() + serverMuted/serverDeafened to VoiceState. 2. Join/Leave Voice button: replaced icon-only mic toggle with explicit text button (parity with macOS). Mute/deafen disable when not in voice. 3. IOSAudioRouter.swift (new): full AVAudioSession routing layer — input port selection, built-in mic orientation/polar patterns, Bluetooth HFP/A2DP/Off modes, Standard/Raw mic processing, stereo capture, AirPlay, UserDefaults persistence. AudioSessionManager delegates to it. 4. Core stereo-mic capture (append-only ABI): vc_set_capture_channels() lets the core open the mic device in stereo (2-ch interleaved). LocalStream gains capture_channels; ensure_audio_running reads it; audio_engine.cpp capture_accum_ + on_capture updated to channel-aware accumulation. Test test_stereo_mic_capture (headless, L!=R stereo round-trip). Swift wrapper VoiceCatClient.setCaptureChannels. 5. Settings UI rework: AVAudioSession-derived input/output tree replaces miniaudio device picker. 6. iOS deployment target raised to 18.0 (Package.swift + project.pbxproj). swift-tools-version 6.0 with swiftLanguageModes .v5. Docs: tech-stack.md, architecture.md, voice.md, roadmap.md, building.md updated; stale 'vc_audio_suspend/resume deferred' claims corrected. Verified: ctest --preset dev 21/21 green; swift test 6/6 green; xcodebuild -target VoiceCatiOS -sdk iphonesimulator BUILD SUCCEEDED.
2026-06-19 13:17:52 +02:00
vc_result set_capture_channels(uint32_t stream_id, uint32_t channels);
vc_result set_input_mode(vc_input_mode mode);
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>
2026-06-17 00:35:16 +02:00
vc_result set_vad_threshold(float threshold);
vc_result set_push_to_talk(bool active);
vc_result set_self_mute(bool mic_muted, bool deafened);
feat(windows): UI overhaul -- toolbar, unified log, PM windows, channel counts, output volume - Voice actions (Join Voice, Share Screen Audio) moved to a ToolStrip toolbar and a new Voice menu in the menu bar; removed from the bottom voice panel - Activity log and chat log collapsed into a single RichTextBox (rtbLog); activity events appear in gray, chat messages in default color - Private messaging reworked: each conversation opens in its own modeless PrivateMessageForm instead of sharing the main chat log via a scope dropdown; cboScope removed; main compose bar always sends to the current channel - New "Messages -> New Private Message..." menu item (Ctrl+P) opens a UserPickerDialog listing all connected server users (not just the current channel) so you can PM anyone on the server - Channel tree now shows live user counts, e.g. "General (3)" -- counts sourced from the existing _users dictionary which already tracks all server users with channel IDs - Global output volume slider (TrackBar, 0-100, default 80) added to the right panel; wired to new vc_set_output_volume C ABI function that applies a master gain multiplier in the audio engine playback callback after mixing all streams - vc_set_output_volume added end-to-end: voicecat.h, audio_engine.h/.cpp, client.h/.cpp, voicecat.cpp, NativeMethods.cs, VoiceCatClient.cs - Documented Windows PowerShell ctest requirement in AGENTS.md and CLAUDE.md: MinGW binaries exit 0xc0000139 in Git Bash; always run ctest/.exe via PowerShell 22/22 ctest green (PowerShell); dotnet build 0 warnings. Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-20 14:24:54 +02:00
vc_result set_output_volume(float gain);
feat(clients): persist input settings, add mic input gain, fix iOS chat + VoiceOver Input mode (VAD/PTT/Always-On), VAD threshold, and the new mic gain were applied to the core + UI but never saved, so every relaunch reset to VAD defaults. Each client now persists them and re-applies on connect: - iOS: UserDefaults (SessionState.loadAndApplyVoiceSettings + setter writes) - macOS: UserDefaults via MainWindowController didSet + loadPersistedAudioSettings (settings window also restores the VAD slider from the stored threshold) - Windows: new Models/VoiceSettings.cs (JSON at %AppData%\VoiceCat\voice.json, mirrors FeedbackSettings) loaded/applied in MainForm Add global send-side mic gain API vc_set_input_gain (applied to MIC PCM in on_capture_frame before the VAD gate, clamped to int16) + Swift/C# bindings, and a 0-300% (default 100%) mic-volume slider on all three clients. Fix iOS chat: ChatView called sendText(scope:.channel) with no targetId (0), so channel messages went nowhere; now passes session.currentChannelId. Fix iOS per-user tuning for VoiceOver: the tuning sheet was long-press .contextMenu only (invisible to VoiceOver); UserRow now also exposes the same buttons via .accessibilityActions (no visual change). Verified: core builds clean; ctest 24/27 (3 pre-existing teardown crashes, reproduced with changes stashed); VoiceCatMac + VoiceCatiOS (arm64 sim) build SUCCEEDED; VoiceCat.Interop dotnet build succeeded. Windows App not built (WinForms can't build on macOS) — follows existing patterns.
2026-06-23 03:35:26 +02:00
vc_result set_input_gain(float gain);
feat(audio): real noise suppression via vendored RNNoise (send + receive) The two-sided NR plumbing (RemoteStream::recv_ns + the per-listener vc_set_remote_stream noise_reduction toggle) was wired but inert: ApmProcessor::create() returned a no-op passthrough, because the originally-planned webrtc-audio-processing has no working Windows/macOS build. Drop in RNNoise as the real backend behind the same ApmProcessor interface, lighting up both NR paths. - Vendor RNNoise (BSD-3 + CC0) at third_party/rnnoise/ — the vcpkg port is !windows !arm, so it can't cover our primary targets. Shrunk int8 model (78MB -> 11.7MB via upstream scripts/shrink_model.sh), built as a standalone C static lib with no RTCD (portable scalar path on x86, auto-NEON on arm64) under -DDISABLE_DEBUG_FLOAT. Model is baked in (rnnoise_create(NULL)); no runtime file. - New RnnoiseProcessor (core/src/audio/apm_processor.cpp) selected by ApmProcessor::create() when VOICECAT_HAS_NS. Mono/48kHz/480-sample; our clock is fixed 48kHz and Opus frame sizes are multiples of 480, so no resampling. RT-safe: allocates at construction, lock-free in the capture/playback callbacks. - Receive-side: lit up via the factory; gated to mono streams (a stereo stream is a screen-audio share, not voice). - Send-side (new): vc_set_input_noise_reduction(client, enable) ABI + vc_client::mic_ns_, run before input gain/VAD in on_capture_frame. A stereo mic is downmixed to mono ONLY when NR is on — with NR off a stereo mic keeps full stereo (never collapse mic quality unasked). - Enable C as a project language for the vendored lib. - New noise_suppression test: white noise through ApmProcessor::create() drops ~99.9% RMS. ctest --preset dev green, 28/28. windows-client DLL builds clean with vc_set_input_noise_reduction exported, system-only deps. - Docs synced: voice.md §10, tech-stack.md §1/§5, third_party/README.md, vcpkg.json note, PROGRESS.md, CLAUDE.md. Client on/off UI toggles (Windows/macOS/iOS) are the remaining follow-up. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-06-23 13:30:54 +02:00
vc_result set_input_noise_reduction(bool enable);
vc_result set_remote_stream(uint32_t user_id, uint32_t stream_id, float gain, bool muted,
bool noise_reduction);
vc_result get_remote_stream(uint32_t user_id, uint32_t stream_id,
vc_remote_stream_state* out);
vc_result audio_suspend();
vc_result audio_resume();
fix(ios): stereo mic + A2DP output, add vc_audio_restart ABI Diagnosed by comparing against TeamTalk5 (Client/iTeamTalk), which achieves stereo mic + A2DP output. Five fixes: 1. configureStereoCapture now calls setPreferredInput + setInputDataSource (mirroring TeamTalk5's SoundDevicesModel). Previously omitted based on incorrect diagnosis that setPreferredInput collapsed A2DP — the real culprit was setPreferredInputNumberOfChannels(2), which neither project uses. 2. New C ABI: vc_audio_restart (full stop + re-init, unlike suspend/resume which only stop/start). Swift wrapper added. The withAudioSuspend wrapper that used it was removed after on-device testing showed it killed all audio (including VoiceOver) when switching presets — the core's set_capture_channels handles engine restart internally. 3. Bluetooth options: Voice Chat preset now includes BOTH .allowBluetoothHFP AND .allowBluetoothA2DP (matching TeamTalk5's UtilSound.swift:228). Previously HFP-only blocked A2DP headphones. 4. Capture channels now reset when switching stereo→mono via selectCaptureChannels/applyPreset. AudioSessionManager tracks activeMicStreamId (set by SessionState on join/leave voice). 5. Docs synced: voice.md, tech-stack.md, architecture.md, PROGRESS.md. Removed stale setPreferredInputNumberOfChannels(2) references. Verified: ctest --preset dev 21/21 green, iOS client builds. Stereo mic + A2DP output still needs on-device debugging — the core recipe is correct but iOS 26 route behavior requires hands-on testing with a debugger.
2026-06-19 16:58:21 +02:00
vc_result audio_restart();
vc_result send_text(vc_text_scope scope, uint32_t target_id, const char* utf8);
vc_result list_devices(vc_device_kind kind, vc_device_list* out);
// Channel/user/stream snapshot getters (read session_model_; see voicecat.h).
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>
2026-06-17 00:35:16 +02:00
vc_result list_channels(vc_channel_list* out);
vc_result list_users(vc_user_list* out);
vc_result list_user_streams(uint32_t user_id, vc_stream_summary_list* out);
// TOFU server-identity gate (see voicecat.h's VC_EVENT_SERVER_IDENTITY doc comment).
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>
2026-06-17 00:35:16 +02:00
vc_result confirm_server_identity(bool accept);
vc_result get_server_identity_display(char* out_buf, size_t buf_cap, size_t* out_len);
// Effective Opus config for a (user_id, stream_id) — our own pending/active local
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>
2026-06-16 14:12:37 +02:00
// streams, or any peer's broadcast StreamInfo.audio.
vc_result get_stream_audio_config(uint32_t user_id, uint32_t stream_id,
vc_audio_config* out);
feat: external PCM feed/tap API (vc_stream_feed_pcm + vc_set_pcm_sink) Promotes vc_test_inject_capture (mono-only, TEST-ONLY) to a public, stereo-capable production API and adds a symmetric PCM tap on the receive side. Enables ReplayKit (iOS), ScreenCaptureKit (macOS), bots, soundboards, and custom clients — all without a hardware audio device. Core C++: - voicecat.h: new vc_stream_feed_pcm, vc_pcm_sink_cb typedef, vc_set_pcm_sink; vc_test_inject_capture kept as deprecated alias - audio_engine: stereo-aware inject_capture (channels param + ring reset on channel-count change); atomic pcm_sink_ fired per decoded frame in on_playback; RemoteStream carries user_id/stream_id for RT-safe sink metadata; init_recv_stream takes user_id+stream_id - client.cpp: stream_feed_pcm / set_pcm_sink implementations; sync_remote_streams passes user_id/stream_id to init_recv_stream - voicecat.cpp: trampolines + channels=1/2 validation Tests: test_external_pcm (headless, 3 sub-tests: mono round-trip, stereo feed L≠R, sink metadata+disable). ctest 23/23. Swift: feedPcm / setPcmSink in VoiceCatClient.swift + 4 XCTest smoke tests (ExternalPcmTests.swift). C#: StreamFeedPcm / SetPcmSink in VoiceCatClient.cs + NativeMethods.cs (vc_stream_feed_pcm unsafe P/Invoke, VcPcmSinkCallback delegate, vc_set_pcm_sink via nint) + 4 xUnit smoke tests (ExternalPcmTests.cs). Docs: architecture.md §4 new subsection, voice.md §9 updated (macOS/iOS now reference vc_stream_feed_pcm), protocol.md §8 explicit no-protocol-change note, roadmap.md M5 entry. Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-20 17:52:09 +02:00
// External PCM feed (see voicecat.h: vc_stream_feed_pcm). Production API for driving a
// local stream's encode pipeline without a hardware capture device. channels = 1 or 2.
vc_result stream_feed_pcm(uint32_t stream_id, const int16_t* pcm,
size_t samples_per_channel, uint32_t channels);
// External PCM sink (see voicecat.h: vc_set_pcm_sink). Delegates to AudioEngine.
vc_result set_pcm_sink(vc_pcm_sink_cb cb, void* user);
// External mixed-output sink + external-playback mode (iOS VPIO; see voicecat.h).
vc_result set_mixed_output_sink(vc_mixed_output_cb cb, void* user);
vc_result set_external_playback(bool enable);
feat: external PCM feed/tap API (vc_stream_feed_pcm + vc_set_pcm_sink) Promotes vc_test_inject_capture (mono-only, TEST-ONLY) to a public, stereo-capable production API and adds a symmetric PCM tap on the receive side. Enables ReplayKit (iOS), ScreenCaptureKit (macOS), bots, soundboards, and custom clients — all without a hardware audio device. Core C++: - voicecat.h: new vc_stream_feed_pcm, vc_pcm_sink_cb typedef, vc_set_pcm_sink; vc_test_inject_capture kept as deprecated alias - audio_engine: stereo-aware inject_capture (channels param + ring reset on channel-count change); atomic pcm_sink_ fired per decoded frame in on_playback; RemoteStream carries user_id/stream_id for RT-safe sink metadata; init_recv_stream takes user_id+stream_id - client.cpp: stream_feed_pcm / set_pcm_sink implementations; sync_remote_streams passes user_id/stream_id to init_recv_stream - voicecat.cpp: trampolines + channels=1/2 validation Tests: test_external_pcm (headless, 3 sub-tests: mono round-trip, stereo feed L≠R, sink metadata+disable). ctest 23/23. Swift: feedPcm / setPcmSink in VoiceCatClient.swift + 4 XCTest smoke tests (ExternalPcmTests.swift). C#: StreamFeedPcm / SetPcmSink in VoiceCatClient.cs + NativeMethods.cs (vc_stream_feed_pcm unsafe P/Invoke, VcPcmSinkCallback delegate, vc_set_pcm_sink via nint) + 4 xUnit smoke tests (ExternalPcmTests.cs). Docs: architecture.md §4 new subsection, voice.md §9 updated (macOS/iOS now reference vc_stream_feed_pcm), protocol.md §8 explicit no-protocol-change note, roadmap.md M5 entry. Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-20 17:52:09 +02:00
// TEST-ONLY (see voicecat.h) — deprecated alias for stream_feed_pcm(..., channels=1).
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>
2026-06-16 14:12:37 +02:00
vc_result test_inject_capture(uint32_t stream_id, const int16_t* pcm, size_t samples);
// Moderation & admin.
vc_result kick_user(uint32_t user_id, const char* reason);
vc_result ban_user(uint32_t user_id, const char* reason, uint64_t expires_unix_ms);
vc_result set_permission(uint32_t user_id, const vc_permissions* perms);
vc_result set_server_mute(uint32_t user_id, bool muted, bool deafened);
vc_result move_user(uint32_t user_id, uint32_t channel_id);
vc_result create_channel(const vc_channel_info* info);
vc_result edit_channel(const vc_channel_info* info);
vc_result delete_channel(uint32_t channel_id);
vc_result create_account(const char* username, const char* password);
vc_result reset_password(const char* username, const char* new_password);
vc_result delete_account(const char* username);
vc_result list_accounts();
vc_result get_account_list(vc_account_list* out);
vc_result get_permissions(vc_permissions* out);
2026-06-15 23:48:44 +02:00
vc_connection_state state() const {
return state_net_.load(std::memory_order_acquire);
}
private:
void emit(const vc_event& ev) const;
2026-06-15 23:48:44 +02:00
vc_config cfg_{};
vc_callbacks cb_{};
2026-06-15 23:48:44 +02:00
// ── TCP/TLS control channel ───────────────────────────────────────────────────
2026-06-15 23:48:44 +02:00
std::atomic<vc_connection_state> state_net_{VC_STATE_DISCONNECTED};
// Blocking I/O thread (one per vc_client lifetime)
std::thread io_thread_;
std::atomic<bool> io_stop_{false};
// Send queue: pushed by any thread, drained by io_thread_
std::mutex send_mutex_;
std::condition_variable send_cv_;
std::deque<std::vector<uint8_t>> send_queue_;
// TLS context — created + used exclusively on io_thread_
std::unique_ptr<voicecat::crypto::TlsContext> tls_;
// Raw socket fd (stored after TCP connect; closed by disconnect())
std::atomic<int> io_fd_{-1};
// Pending auth stored before ServerHello arrives
struct PendingAuth {
bool is_guest{true};
std::string nick_or_user;
std::string password;
};
std::mutex pending_auth_mutex_;
std::optional<PendingAuth> pending_auth_;
// Self identity filled in after AuthResult
uint32_t self_user_id_{0};
uint64_t server_session_id_{0};
std::atomic<uint64_t> next_req_id_{1};
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).
2026-06-24 14:29:39 +02:00
// Voice-plane subscription state. When false, the server does not relay voice frames to
// us, and we do not wire up remote-stream decoders (sync_remote_streams is gated on this).
// Toggled by vc_join_voice / vc_leave_voice; confirmed via VoiceSubscriptionResult.
std::atomic<bool> voice_subscribed_{false};
fix(net): broadcast LEFT on disconnect, add keepalive/reaper, cap PLC hiss Three reported bugs traced to one root cause plus two missing designed features: 1. Stale users + eternal PLC hiss (root cause): ConnSession::close() silently erased dropped users without broadcasting UserEvent::LEFT, so peers never learned the user left and their audio engines never called remove_stream — Opus PLC synthesized comfort noise forever. Fix: broadcast_left() helper + close() broadcasts LEFT before erasing. 2. PLC cap (defense-in-depth): on_playback now caps pure PLC at ~2s, then emits digital silence so a stale stream can never hiss forever even if remove_stream is skipped. Resets automatically on fresh packets. 3. No timeout / no ping: client never sent Ping, server had no last_seen / reaper, so half-open connections (NAT timeout, wifi loss, sleep) left ghost users forever. Fix: client Ping every 15s with RTT measurement, ConnSession::last_seen bumped on every inbound TCP/UDP frame, steady_timer reaper sweeps every 15s and drops sessions older than 45s (configurable via server::Config). 4. UDP KEEPALIVE: client sends plaintext kFrameKeepalive every 5s; server bumps last_seen + echoes back. Keeps NAT bindings alive and lets media activity defer the reaper independently of TCP. 5. Graceful client disconnect: vc_disconnect() sends Disconnect{code=0} via a flag-based io-thread exit (no double-close race); server handles client-sent Disconnect with immediate close() + LEFT broadcast. 3 new tests: disconnect_left, plc_cap, reaper_timeout. 21/21 ctest green. Docs: protocol.md §6/§7, voice.md §6, architecture.md §5, PROGRESS.md.
2026-06-18 01:18:33 +02:00
// Keepalive: client sends a Ping every ~15s (docs/protocol.md §7) so the server's
// last_seen stays fresh and the reaper doesn't drop us. Pong echoes the nonce, which
// we correlate to measure RTT. The read loop's 50ms TLS timeout means it spins fast
// enough to check the ping interval with ample resolution.
static constexpr int64_t kPingIntervalMs = 15000;
std::atomic<int64_t> last_ping_ms_{0};
std::atomic<uint64_t> ping_nonce_{1};
std::mutex ping_mutex_;
std::unordered_map<uint64_t, int64_t> pending_pings_; // nonce → sent_ms
std::atomic<int64_t> last_rtt_ms_{0};
// Graceful disconnect: set by disconnect() after queueing Disconnect{code=0}. The io
// thread checks this after drain_sends() — when set, it sets io_stop_ and exits the
// read loop, so the Disconnect is sent before the thread ends. The main thread just
// joins; no socket close from the main thread (the io thread's cleanup closes it),
// avoiding the double-close race that the send_cv_ wait approach exposed.
std::atomic<bool> graceful_disconnect_pending_{false};
// UDP keepalive: send a lightweight KEEPALIVE frame every ~5s to hold NAT bindings and
// bump the server's last_seen independently of the TCP ping (docs/voice.md §6). Sent
// as plaintext (no AEAD) — the server identifies the sender by its already-verified UDP
// endpoint, and the TCP reaper is the real timeout authority. Avoids racing the
// non-atomic send_counter_ in SodiumMediaCrypto::seal() with the audio callback thread.
static constexpr int64_t kUdpKeepaliveIntervalMs = 5000;
std::atomic<int64_t> last_udp_keepalive_ms_{0};
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>
2026-06-17 00:35:16 +02:00
// Client-side session model. Mutated only on io_thread_ (handle_server_state/
// handle_user_event/handle_channel_event), but read from any thread via the
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>
2026-06-17 00:35:16 +02:00
// list_channels/list_users/list_user_streams getters — session_model_mu_ guards both.
2026-06-15 23:48:44 +02:00
voicecat::session::SessionModel session_model_;
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>
2026-06-17 00:35:16 +02:00
mutable std::mutex session_model_mu_;
// Last ListAccountsResult snapshot, populated on io_thread_ when
// VC_EVENT_ACCOUNT_LIST fires and read by vc_get_account_list on caller threads.
std::vector<voicecat::v1::AccountEntry> last_account_list_;
mutable std::mutex account_list_mu_;
// ── TOFU server-identity gate ─────────────────────────────────────────────────
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>
2026-06-17 00:35:16 +02:00
std::unique_ptr<voicecat::crypto::TofuStore> tofu_store_; // owns the pin file
std::mutex tofu_mu_;
std::condition_variable tofu_cv_;
bool tofu_decision_pending_{false};
bool tofu_accept_{false};
std::string pending_identity_fp_hex_; // ServerHello's Ed25519 fp, display-only
2026-06-15 23:48:44 +02:00
// ── UDP / media plane ──────────────────────────────────────────────────────────
std::array<uint8_t, 16> udp_token_{};
uint16_t server_udp_port_{0};
std::string udp_host_;
std::atomic<int> udp_fd_{-1};
std::thread udp_thread_;
std::atomic<bool> udp_stop_{false};
std::atomic<bool> udp_ready_{false};
std::unique_ptr<voicecat::crypto::SodiumMediaCrypto> media_send_crypto_;
std::unique_ptr<voicecat::crypto::SodiumMediaCrypto> media_recv_crypto_;
voicecat::audio::AudioEngine audio_engine_;
// One LocalStream per concurrently-active stream kind (MIC/SCREEN_AUDIO/AUX_DEVICE
// are each singletons for a given client).
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>
2026-06-16 14:12:37 +02:00
struct LocalStream {
voicecat::codec::OpusEncoder encoder;
std::atomic<bool> active{false};
bool pending{false}; // announced, awaiting StreamAnnounceResult
uint32_t stream_id{0};
uint32_t ssrc{0};
uint32_t timestamp{0};
uint16_t frame_samples{960};
uint64_t pending_request_id{0};
// Full effective OpusParams from the last StreamAnnounceResult — retained so
// vc_get_stream_audio_config() has something to read back for our own streams.
voicecat::codec::OpusParams effective_params;
feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback Closes the three items PROGRESS.md's M3 section explicitly carried forward as out of scope: - Device enumeration (vc_list_devices) + input device selection (vc_set_input_device), backed by AudioEngine::enumerate_devices() via miniaudio's ma_context_get_devices. Device ids are opaque hex-encoded ma_device_id strings. - VAD/PTT send-side input gate (vc_set_input_mode, vc_set_push_to_talk). webrtc-audio-processing (the originally-planned APM) has no working Windows/MSVC build upstream (GCC-only Meson, unfinished MinGW support, hard abseil-cpp dependency), so VAD is a new lightweight, dependency-free energy/RMS processor (EnergyVadProcessor) behind the existing ApmProcessor interface. Gating is MIC-only; SCREEN_AUDIO/AUX_DEVICE always bypass it. - True stereo playback: AudioEngine's mixer and output device now carry stereo end-to-end (mono streams upmix L=R) instead of downmixing decoded stereo streams to mono before mixing. - Real WASAPI loopback capture for SCREEN_AUDIO (Windows-only, via miniaudio's loopback device type), replacing test-only injection as the production capture path. Also: vccli gains --list-devices, --input-device, --input-mode, and --share-screen-audio flags, plus a stdin command loop (ptt on/off, mode vad/ptt) for manual verification. New test_vad_ptt_devices.cpp covers all four items (ABI-level + a white-box AudioEngine stereo-mix check). Docs updated to match: voice.md, roadmap.md (decision-log entry superseding the original webrtc-audio-processing choice), tech-stack.md, README.md, architecture.md, CLAUDE.md, PROGRESS.md. Still explicitly out of scope, documented not silently dropped: real webrtc-audio-processing/AEC (no AEC/NS/AGC exists at all yet), macOS/iOS SCREEN_AUDIO capture, process-specific loopback, and a pre-existing RT-thread rule violation in the capture path that predates this work. Verified: ctest 12/12 green across 3 consecutive full-suite runs (both dev and m1-dev presets build clean); test_vad_ptt_devices passed 5 consecutive standalone runs; manually verified live (vccli --list-devices against real hardware, vccli --voice --input-mode vad streaming without incident). Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-16 16:11:52 +02:00
// Talk-indicator edge detection (docs/voice.md §7) — updated in on_capture_frame, after
// the VAD/PTT gate so a gated-closed frame doesn't show as "talking".
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>
2026-06-16 14:12:37 +02:00
std::atomic<int64_t> last_capture_ms{0};
bool talking = false;
feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback Closes the three items PROGRESS.md's M3 section explicitly carried forward as out of scope: - Device enumeration (vc_list_devices) + input device selection (vc_set_input_device), backed by AudioEngine::enumerate_devices() via miniaudio's ma_context_get_devices. Device ids are opaque hex-encoded ma_device_id strings. - VAD/PTT send-side input gate (vc_set_input_mode, vc_set_push_to_talk). webrtc-audio-processing (the originally-planned APM) has no working Windows/MSVC build upstream (GCC-only Meson, unfinished MinGW support, hard abseil-cpp dependency), so VAD is a new lightweight, dependency-free energy/RMS processor (EnergyVadProcessor) behind the existing ApmProcessor interface. Gating is MIC-only; SCREEN_AUDIO/AUX_DEVICE always bypass it. - True stereo playback: AudioEngine's mixer and output device now carry stereo end-to-end (mono streams upmix L=R) instead of downmixing decoded stereo streams to mono before mixing. - Real WASAPI loopback capture for SCREEN_AUDIO (Windows-only, via miniaudio's loopback device type), replacing test-only injection as the production capture path. Also: vccli gains --list-devices, --input-device, --input-mode, and --share-screen-audio flags, plus a stdin command loop (ptt on/off, mode vad/ptt) for manual verification. New test_vad_ptt_devices.cpp covers all four items (ABI-level + a white-box AudioEngine stereo-mix check). Docs updated to match: voice.md, roadmap.md (decision-log entry superseding the original webrtc-audio-processing choice), tech-stack.md, README.md, architecture.md, CLAUDE.md, PROGRESS.md. Still explicitly out of scope, documented not silently dropped: real webrtc-audio-processing/AEC (no AEC/NS/AGC exists at all yet), macOS/iOS SCREEN_AUDIO capture, process-specific loopback, and a pre-existing RT-thread rule violation in the capture path that predates this work. Verified: ctest 12/12 green across 3 consecutive full-suite runs (both dev and m1-dev presets build clean); test_vad_ptt_devices passed 5 consecutive standalone runs; manually verified live (vccli --list-devices against real hardware, vccli --voice --input-mode vad streaming without incident). Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-16 16:11:52 +02:00
// Talkspurt marker: the sender's `timestamp` omits VAD/PTT/DTX silence, so the receiver
// can't tell a continuation from a post-silence restart. encode_and_send_frame stamps
// kFlagMarker on the first frame after a transmission gap (detected via last_send_ms) so
// the receiver reseeds its playout clock cleanly. -1 = no frame sent yet (first frame is
// always a marker).
int64_t last_send_ms = -1;
feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback Closes the three items PROGRESS.md's M3 section explicitly carried forward as out of scope: - Device enumeration (vc_list_devices) + input device selection (vc_set_input_device), backed by AudioEngine::enumerate_devices() via miniaudio's ma_context_get_devices. Device ids are opaque hex-encoded ma_device_id strings. - VAD/PTT send-side input gate (vc_set_input_mode, vc_set_push_to_talk). webrtc-audio-processing (the originally-planned APM) has no working Windows/MSVC build upstream (GCC-only Meson, unfinished MinGW support, hard abseil-cpp dependency), so VAD is a new lightweight, dependency-free energy/RMS processor (EnergyVadProcessor) behind the existing ApmProcessor interface. Gating is MIC-only; SCREEN_AUDIO/AUX_DEVICE always bypass it. - True stereo playback: AudioEngine's mixer and output device now carry stereo end-to-end (mono streams upmix L=R) instead of downmixing decoded stereo streams to mono before mixing. - Real WASAPI loopback capture for SCREEN_AUDIO (Windows-only, via miniaudio's loopback device type), replacing test-only injection as the production capture path. Also: vccli gains --list-devices, --input-device, --input-mode, and --share-screen-audio flags, plus a stdin command loop (ptt on/off, mode vad/ptt) for manual verification. New test_vad_ptt_devices.cpp covers all four items (ABI-level + a white-box AudioEngine stereo-mix check). Docs updated to match: voice.md, roadmap.md (decision-log entry superseding the original webrtc-audio-processing choice), tech-stack.md, README.md, architecture.md, CLAUDE.md, PROGRESS.md. Still explicitly out of scope, documented not silently dropped: real webrtc-audio-processing/AEC (no AEC/NS/AGC exists at all yet), macOS/iOS SCREEN_AUDIO capture, process-specific loopback, and a pre-existing RT-thread rule violation in the capture path that predates this work. Verified: ctest 12/12 green across 3 consecutive full-suite runs (both dev and m1-dev presets build clean); test_vad_ptt_devices passed 5 consecutive standalone runs; manually verified live (vccli --list-devices against real hardware, vccli --voice --input-mode vad streaming without incident). Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-16 16:11:52 +02:00
// Device-enumeration follow-up: the device this stream's capture should use ("" =
// default). Only meaningful for VC_STREAM_MIC today (the real capture device); set via
// vc_set_input_device. Opaque id from AudioEngine::enumerate_devices — see
// audio_engine.h's DeviceInfo doc comment.
std::string capture_device_id;
feat(ios): audio overhaul, Join/Leave Voice, channel-id sync fix, stereo mic capture Three iOS client problems fixed plus a new core stereo-mic capture ABI: 1. Channel-id sync bug (mic button permanently dimmed): SessionState never synced currentChannelId from the self user's channelId on connect, so the mic button (gated on currentChannelId == 0) stayed dimmed. Added syncSelfChannel() (mirrors macOS MainWindowController.swift:461,491,522); called from init/.channelList/.userJoined/.userLeft/.userUpdated/.joinResult. Added applyServerMuteState() + serverMuted/serverDeafened to VoiceState. 2. Join/Leave Voice button: replaced icon-only mic toggle with explicit text button (parity with macOS). Mute/deafen disable when not in voice. 3. IOSAudioRouter.swift (new): full AVAudioSession routing layer — input port selection, built-in mic orientation/polar patterns, Bluetooth HFP/A2DP/Off modes, Standard/Raw mic processing, stereo capture, AirPlay, UserDefaults persistence. AudioSessionManager delegates to it. 4. Core stereo-mic capture (append-only ABI): vc_set_capture_channels() lets the core open the mic device in stereo (2-ch interleaved). LocalStream gains capture_channels; ensure_audio_running reads it; audio_engine.cpp capture_accum_ + on_capture updated to channel-aware accumulation. Test test_stereo_mic_capture (headless, L!=R stereo round-trip). Swift wrapper VoiceCatClient.setCaptureChannels. 5. Settings UI rework: AVAudioSession-derived input/output tree replaces miniaudio device picker. 6. iOS deployment target raised to 18.0 (Package.swift + project.pbxproj). swift-tools-version 6.0 with swiftLanguageModes .v5. Docs: tech-stack.md, architecture.md, voice.md, roadmap.md, building.md updated; stale 'vc_audio_suspend/resume deferred' claims corrected. Verified: ctest --preset dev 21/21 green; swift test 6/6 green; xcodebuild -target VoiceCatiOS -sdk iphonesimulator BUILD SUCCEEDED.
2026-06-19 13:17:52 +02:00
// Capture channel count (1 = mono, 2 = stereo interleaved). Only meaningful for
// VC_STREAM_MIC. Set via vc_set_capture_channels; read by ensure_audio_running() to
// configure AudioParams.capture_channels before the device opens. Defaults to 1 (mono).
uint32_t capture_channels = 1;
// If true, the caller is feeding PCM via vc_stream_feed_pcm — skip start/stop of the
// core's WASAPI loopback device. Set from vc_stream_desc::external_feed at stream_start
// time and checked in handle_stream_announce_result / stream_stop.
bool external_feed = false;
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).
2026-06-24 14:29:39 +02:00
// Stream label (from vc_stream_desc.label at stream_start time). Retained so the
// channel-update stream restart (restart_active_streams_for_channel) can re-announce
// with the same label without the caller's involvement.
std::string label;
// Reframe buffer: the AudioEngine clock is fixed at 48 kHz / 20 ms, so capture/feed
// always delivers 960-sample frames — but the channel's frame_ms (docs/voice.md §3)
// can be 2.5…60 ms, so the encoder needs frame_samples per call (480 @10ms, 1920 @40ms,
// …). on_capture_frame accumulates the engine's 960-sample frames here and emits
// frame_samples-sized chunks. The 20 ms case (frame_samples == 960) bypasses this
// entirely (fast path). Pre-sized at announce; never resized on the audio thread
// (architecture.md §3 — no RT-thread allocation). encode_accum holds interleaved int16
// at accum_channels; upmix_scratch is the pre-sized mono→stereo upmix target.
std::vector<int16_t> encode_accum;
size_t accum_count = 0; // flat samples currently buffered
int accum_channels = 0; // channel count of buffered data; resets on change
std::vector<int16_t> upmix_scratch;
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>
2026-06-16 14:12:37 +02:00
};
mutable std::mutex local_streams_mu_;
std::unordered_map<int, LocalStream> local_streams_; // keyed by vc_stream_kind
std::unordered_map<uint64_t, int> pending_announce_kind_; // request_id -> kind
uint32_t next_local_stream_id_{1};
// ssrc → (user_id, stream_id) for remote streams already wired into audio_engine_.
mutable std::mutex remote_streams_mu_;
std::unordered_map<uint32_t, std::pair<uint32_t, uint32_t>> remote_streams_;
// Talk-indicator polling thread (separate from udp_thread_ / the miniaudio callback
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>
2026-06-16 14:12:37 +02:00
// thread — see architecture.md §3 real-time rule).
std::thread talk_timer_thread_;
std::atomic<bool> talk_timer_stop_{false};
static constexpr int64_t kTalkPollMs = 100;
static constexpr int64_t kTalkHangoverMs = 300;
// Local UDP destination (server media endpoint), resolved once during binding.
uint32_t udp_dest_addr_{0}; // network byte order
uint16_t udp_dest_port_{0}; // network byte order
std::atomic<bool> self_mic_muted_{false};
std::atomic<bool> self_deafened_{false};
std::atomic<bool> server_muted_{false};
std::atomic<bool> server_deafened_{false};
// Permissions from last AuthResult.
vc_permissions own_permissions_{};
// Send-side input gate (docs/voice.md §11). MIC-only — SCREEN_AUDIO/
feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback Closes the three items PROGRESS.md's M3 section explicitly carried forward as out of scope: - Device enumeration (vc_list_devices) + input device selection (vc_set_input_device), backed by AudioEngine::enumerate_devices() via miniaudio's ma_context_get_devices. Device ids are opaque hex-encoded ma_device_id strings. - VAD/PTT send-side input gate (vc_set_input_mode, vc_set_push_to_talk). webrtc-audio-processing (the originally-planned APM) has no working Windows/MSVC build upstream (GCC-only Meson, unfinished MinGW support, hard abseil-cpp dependency), so VAD is a new lightweight, dependency-free energy/RMS processor (EnergyVadProcessor) behind the existing ApmProcessor interface. Gating is MIC-only; SCREEN_AUDIO/AUX_DEVICE always bypass it. - True stereo playback: AudioEngine's mixer and output device now carry stereo end-to-end (mono streams upmix L=R) instead of downmixing decoded stereo streams to mono before mixing. - Real WASAPI loopback capture for SCREEN_AUDIO (Windows-only, via miniaudio's loopback device type), replacing test-only injection as the production capture path. Also: vccli gains --list-devices, --input-device, --input-mode, and --share-screen-audio flags, plus a stdin command loop (ptt on/off, mode vad/ptt) for manual verification. New test_vad_ptt_devices.cpp covers all four items (ABI-level + a white-box AudioEngine stereo-mix check). Docs updated to match: voice.md, roadmap.md (decision-log entry superseding the original webrtc-audio-processing choice), tech-stack.md, README.md, architecture.md, CLAUDE.md, PROGRESS.md. Still explicitly out of scope, documented not silently dropped: real webrtc-audio-processing/AEC (no AEC/NS/AGC exists at all yet), macOS/iOS SCREEN_AUDIO capture, process-specific loopback, and a pre-existing RT-thread rule violation in the capture path that predates this work. Verified: ctest 12/12 green across 3 consecutive full-suite runs (both dev and m1-dev presets build clean); test_vad_ptt_devices passed 5 consecutive standalone runs; manually verified live (vccli --list-devices against real hardware, vccli --voice --input-mode vad streaming without incident). Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-16 16:11:52 +02:00
// AUX_DEVICE are never gated (see PROGRESS.md for the rationale). Pure local state, no
// protocol traffic. mic_vad_ is constructed once the MIC stream's StreamAnnounceResult
// lands (handle_stream_announce_result, on io_thread_ — not the RT capture callback).
std::atomic<vc_input_mode> current_input_mode_{VC_INPUT_VOICE_ACTIVATION};
std::atomic<bool> ptt_active_{false};
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>
2026-06-17 00:35:16 +02:00
std::atomic<float> vad_threshold_{0.025f}; // remembered across mode switches
feat(clients): persist input settings, add mic input gain, fix iOS chat + VoiceOver Input mode (VAD/PTT/Always-On), VAD threshold, and the new mic gain were applied to the core + UI but never saved, so every relaunch reset to VAD defaults. Each client now persists them and re-applies on connect: - iOS: UserDefaults (SessionState.loadAndApplyVoiceSettings + setter writes) - macOS: UserDefaults via MainWindowController didSet + loadPersistedAudioSettings (settings window also restores the VAD slider from the stored threshold) - Windows: new Models/VoiceSettings.cs (JSON at %AppData%\VoiceCat\voice.json, mirrors FeedbackSettings) loaded/applied in MainForm Add global send-side mic gain API vc_set_input_gain (applied to MIC PCM in on_capture_frame before the VAD gate, clamped to int16) + Swift/C# bindings, and a 0-300% (default 100%) mic-volume slider on all three clients. Fix iOS chat: ChatView called sendText(scope:.channel) with no targetId (0), so channel messages went nowhere; now passes session.currentChannelId. Fix iOS per-user tuning for VoiceOver: the tuning sheet was long-press .contextMenu only (invisible to VoiceOver); UserRow now also exposes the same buttons via .accessibilityActions (no visual change). Verified: core builds clean; ctest 24/27 (3 pre-existing teardown crashes, reproduced with changes stashed); VoiceCatMac + VoiceCatiOS (arm64 sim) build SUCCEEDED; VoiceCat.Interop dotnet build succeeded. Windows App not built (WinForms can't build on macOS) — follows existing patterns.
2026-06-23 03:35:26 +02:00
std::atomic<float> input_gain_{1.0f}; // send-side MIC gain (vc_set_input_gain)
feat(audio): real noise suppression via vendored RNNoise (send + receive) The two-sided NR plumbing (RemoteStream::recv_ns + the per-listener vc_set_remote_stream noise_reduction toggle) was wired but inert: ApmProcessor::create() returned a no-op passthrough, because the originally-planned webrtc-audio-processing has no working Windows/macOS build. Drop in RNNoise as the real backend behind the same ApmProcessor interface, lighting up both NR paths. - Vendor RNNoise (BSD-3 + CC0) at third_party/rnnoise/ — the vcpkg port is !windows !arm, so it can't cover our primary targets. Shrunk int8 model (78MB -> 11.7MB via upstream scripts/shrink_model.sh), built as a standalone C static lib with no RTCD (portable scalar path on x86, auto-NEON on arm64) under -DDISABLE_DEBUG_FLOAT. Model is baked in (rnnoise_create(NULL)); no runtime file. - New RnnoiseProcessor (core/src/audio/apm_processor.cpp) selected by ApmProcessor::create() when VOICECAT_HAS_NS. Mono/48kHz/480-sample; our clock is fixed 48kHz and Opus frame sizes are multiples of 480, so no resampling. RT-safe: allocates at construction, lock-free in the capture/playback callbacks. - Receive-side: lit up via the factory; gated to mono streams (a stereo stream is a screen-audio share, not voice). - Send-side (new): vc_set_input_noise_reduction(client, enable) ABI + vc_client::mic_ns_, run before input gain/VAD in on_capture_frame. A stereo mic is downmixed to mono ONLY when NR is on — with NR off a stereo mic keeps full stereo (never collapse mic quality unasked). - Enable C as a project language for the vendored lib. - New noise_suppression test: white noise through ApmProcessor::create() drops ~99.9% RMS. ctest --preset dev green, 28/28. windows-client DLL builds clean with vc_set_input_noise_reduction exported, system-only deps. - Docs synced: voice.md §10, tech-stack.md §1/§5, third_party/README.md, vcpkg.json note, PROGRESS.md, CLAUDE.md. Client on/off UI toggles (Windows/macOS/iOS) are the remaining follow-up. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-06-23 13:30:54 +02:00
std::atomic<bool> input_noise_reduction_{false}; // send-side MIC NS (vc_set_input_noise_reduction)
// External-playback mode (iOS VPIO): when true, ensure_audio_running() configures the
// AudioEngine to skip its hardware playback device and drive the mixer on a timer instead,
// delivering the final mix to the mixed-output sink. Set via vc_set_external_playback.
std::atomic<bool> external_playback_{false};
feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback Closes the three items PROGRESS.md's M3 section explicitly carried forward as out of scope: - Device enumeration (vc_list_devices) + input device selection (vc_set_input_device), backed by AudioEngine::enumerate_devices() via miniaudio's ma_context_get_devices. Device ids are opaque hex-encoded ma_device_id strings. - VAD/PTT send-side input gate (vc_set_input_mode, vc_set_push_to_talk). webrtc-audio-processing (the originally-planned APM) has no working Windows/MSVC build upstream (GCC-only Meson, unfinished MinGW support, hard abseil-cpp dependency), so VAD is a new lightweight, dependency-free energy/RMS processor (EnergyVadProcessor) behind the existing ApmProcessor interface. Gating is MIC-only; SCREEN_AUDIO/AUX_DEVICE always bypass it. - True stereo playback: AudioEngine's mixer and output device now carry stereo end-to-end (mono streams upmix L=R) instead of downmixing decoded stereo streams to mono before mixing. - Real WASAPI loopback capture for SCREEN_AUDIO (Windows-only, via miniaudio's loopback device type), replacing test-only injection as the production capture path. Also: vccli gains --list-devices, --input-device, --input-mode, and --share-screen-audio flags, plus a stdin command loop (ptt on/off, mode vad/ptt) for manual verification. New test_vad_ptt_devices.cpp covers all four items (ABI-level + a white-box AudioEngine stereo-mix check). Docs updated to match: voice.md, roadmap.md (decision-log entry superseding the original webrtc-audio-processing choice), tech-stack.md, README.md, architecture.md, CLAUDE.md, PROGRESS.md. Still explicitly out of scope, documented not silently dropped: real webrtc-audio-processing/AEC (no AEC/NS/AGC exists at all yet), macOS/iOS SCREEN_AUDIO capture, process-specific loopback, and a pre-existing RT-thread rule violation in the capture path that predates this work. Verified: ctest 12/12 green across 3 consecutive full-suite runs (both dev and m1-dev presets build clean); test_vad_ptt_devices passed 5 consecutive standalone runs; manually verified live (vccli --list-devices against real hardware, vccli --voice --input-mode vad streaming without incident). Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-16 16:11:52 +02:00
std::unique_ptr<voicecat::audio::ApmProcessor> mic_vad_;
feat(audio): real noise suppression via vendored RNNoise (send + receive) The two-sided NR plumbing (RemoteStream::recv_ns + the per-listener vc_set_remote_stream noise_reduction toggle) was wired but inert: ApmProcessor::create() returned a no-op passthrough, because the originally-planned webrtc-audio-processing has no working Windows/macOS build. Drop in RNNoise as the real backend behind the same ApmProcessor interface, lighting up both NR paths. - Vendor RNNoise (BSD-3 + CC0) at third_party/rnnoise/ — the vcpkg port is !windows !arm, so it can't cover our primary targets. Shrunk int8 model (78MB -> 11.7MB via upstream scripts/shrink_model.sh), built as a standalone C static lib with no RTCD (portable scalar path on x86, auto-NEON on arm64) under -DDISABLE_DEBUG_FLOAT. Model is baked in (rnnoise_create(NULL)); no runtime file. - New RnnoiseProcessor (core/src/audio/apm_processor.cpp) selected by ApmProcessor::create() when VOICECAT_HAS_NS. Mono/48kHz/480-sample; our clock is fixed 48kHz and Opus frame sizes are multiples of 480, so no resampling. RT-safe: allocates at construction, lock-free in the capture/playback callbacks. - Receive-side: lit up via the factory; gated to mono streams (a stereo stream is a screen-audio share, not voice). - Send-side (new): vc_set_input_noise_reduction(client, enable) ABI + vc_client::mic_ns_, run before input gain/VAD in on_capture_frame. A stereo mic is downmixed to mono ONLY when NR is on — with NR off a stereo mic keeps full stereo (never collapse mic quality unasked). - Enable C as a project language for the vendored lib. - New noise_suppression test: white noise through ApmProcessor::create() drops ~99.9% RMS. ctest --preset dev green, 28/28. windows-client DLL builds clean with vc_set_input_noise_reduction exported, system-only deps. - Docs synced: voice.md §10, tech-stack.md §1/§5, third_party/README.md, vcpkg.json note, PROGRESS.md, CLAUDE.md. Client on/off UI toggles (Windows/macOS/iOS) are the remaining follow-up. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-06-23 13:30:54 +02:00
// Send-side mic noise suppressor (RNNoise). Constructed once with the MIC stream alongside
// mic_vad_ (off the RT capture callback); toggling only flips input_noise_reduction_, so the
// capture callback never allocates or races this pointer.
std::unique_ptr<voicecat::audio::ApmProcessor> mic_ns_;
feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback Closes the three items PROGRESS.md's M3 section explicitly carried forward as out of scope: - Device enumeration (vc_list_devices) + input device selection (vc_set_input_device), backed by AudioEngine::enumerate_devices() via miniaudio's ma_context_get_devices. Device ids are opaque hex-encoded ma_device_id strings. - VAD/PTT send-side input gate (vc_set_input_mode, vc_set_push_to_talk). webrtc-audio-processing (the originally-planned APM) has no working Windows/MSVC build upstream (GCC-only Meson, unfinished MinGW support, hard abseil-cpp dependency), so VAD is a new lightweight, dependency-free energy/RMS processor (EnergyVadProcessor) behind the existing ApmProcessor interface. Gating is MIC-only; SCREEN_AUDIO/AUX_DEVICE always bypass it. - True stereo playback: AudioEngine's mixer and output device now carry stereo end-to-end (mono streams upmix L=R) instead of downmixing decoded stereo streams to mono before mixing. - Real WASAPI loopback capture for SCREEN_AUDIO (Windows-only, via miniaudio's loopback device type), replacing test-only injection as the production capture path. Also: vccli gains --list-devices, --input-device, --input-mode, and --share-screen-audio flags, plus a stdin command loop (ptt on/off, mode vad/ptt) for manual verification. New test_vad_ptt_devices.cpp covers all four items (ABI-level + a white-box AudioEngine stereo-mix check). Docs updated to match: voice.md, roadmap.md (decision-log entry superseding the original webrtc-audio-processing choice), tech-stack.md, README.md, architecture.md, CLAUDE.md, PROGRESS.md. Still explicitly out of scope, documented not silently dropped: real webrtc-audio-processing/AEC (no AEC/NS/AGC exists at all yet), macOS/iOS SCREEN_AUDIO capture, process-specific loopback, and a pre-existing RT-thread rule violation in the capture path that predates this work. Verified: ctest 12/12 green across 3 consecutive full-suite runs (both dev and m1-dev presets build clean); test_vad_ptt_devices passed 5 consecutive standalone runs; manually verified live (vccli --list-devices against real hardware, vccli --voice --input-mode vad streaming without incident). Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-16 16:11:52 +02:00
// teardown_voice() is called from run_io() and disconnect() concurrently; this mutex
// makes it idempotent (avoids a double-join race on udp_thread_/talk_timer_thread_).
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>
2026-06-16 14:12:37 +02:00
std::mutex teardown_mu_;
2026-06-15 23:48:44 +02:00
// ── io_thread_ entry point ──────────────────────────────────────────────────
void run_io(std::string host, uint16_t port);
// ── Protocol dispatch (called on io_thread_) ────────────────────────────────
void handle_envelope(const voicecat::v1::Envelope& env);
void handle_server_hello(const voicecat::v1::ServerHello& msg, uint64_t req_id);
void handle_auth_result(const voicecat::v1::AuthResult& msg);
void handle_server_state(const voicecat::v1::ServerStateSnapshot& snap);
void handle_user_event(const voicecat::v1::UserEvent& ue);
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>
2026-06-17 00:35:16 +02:00
void handle_channel_event(const voicecat::v1::ChannelEvent& ce);
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).
2026-06-24 14:29:39 +02:00
// Called from handle_channel_event when the current channel's audio config changed:
// stop→start every active local stream so the new Opus params take effect (encoders are
// frozen at StreamAnnounceResult time — docs/voice.md §3). capture_device_id and
// capture_channels survive the restart (stream_stop doesn't clear them; stream_start
// reuses the existing LocalStream entry). The server reads the updated channel config
// on re-announce and returns new effective_audio; peers' sync_remote_streams wire up
// fresh decoders at the new ssrc.
void restart_active_streams_for_channel(uint32_t channel_id);
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>
2026-06-17 00:35:16 +02:00
void handle_join_channel_result(const voicecat::v1::JoinChannelResult& msg);
2026-06-15 23:48:44 +02:00
void handle_text_message(const voicecat::v1::TextMessage& msg);
void handle_disconnect(const voicecat::v1::Disconnect& msg);
void handle_udp_binding_ack(const voicecat::v1::UdpBinding& msg);
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>
2026-06-16 14:12:37 +02:00
void handle_stream_announce_result(uint64_t req_id,
const voicecat::v1::StreamAnnounceResult& msg);
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).
2026-06-24 14:29:39 +02:00
void handle_voice_subscription_result(const voicecat::v1::VoiceSubscriptionResult& msg);
// Wire up remote-stream decoders for all users in the session model (used on voice join).
void resync_remote_streams();
// Tear down all remote-stream decoders and clear remote_streams_ (used on voice leave).
void clear_all_remote_streams();
// Stop every active local stream (used on voice leave — emits STREAM_STOPPED for each).
void stop_all_local_streams();
// ── UDP / media helpers ────────────────────────────────────────────────────────
// Kicks off TCP UdpBinding request; called once after a successful AuthResult.
void start_udp_binding();
// Opens the UDP socket, sends the plaintext bootstrap packet, starts udp_thread_.
void finish_udp_binding();
// udp_thread_ entry point: recv loop, AEAD-open, decode, push to audio_engine_.
void run_udp_recv();
fix(net): broadcast LEFT on disconnect, add keepalive/reaper, cap PLC hiss Three reported bugs traced to one root cause plus two missing designed features: 1. Stale users + eternal PLC hiss (root cause): ConnSession::close() silently erased dropped users without broadcasting UserEvent::LEFT, so peers never learned the user left and their audio engines never called remove_stream — Opus PLC synthesized comfort noise forever. Fix: broadcast_left() helper + close() broadcasts LEFT before erasing. 2. PLC cap (defense-in-depth): on_playback now caps pure PLC at ~2s, then emits digital silence so a stale stream can never hiss forever even if remove_stream is skipped. Resets automatically on fresh packets. 3. No timeout / no ping: client never sent Ping, server had no last_seen / reaper, so half-open connections (NAT timeout, wifi loss, sleep) left ghost users forever. Fix: client Ping every 15s with RTT measurement, ConnSession::last_seen bumped on every inbound TCP/UDP frame, steady_timer reaper sweeps every 15s and drops sessions older than 45s (configurable via server::Config). 4. UDP KEEPALIVE: client sends plaintext kFrameKeepalive every 5s; server bumps last_seen + echoes back. Keeps NAT bindings alive and lets media activity defer the reaper independently of TCP. 5. Graceful client disconnect: vc_disconnect() sends Disconnect{code=0} via a flag-based io-thread exit (no double-close race); server handles client-sent Disconnect with immediate close() + LEFT broadcast. 3 new tests: disconnect_left, plc_cap, reaper_timeout. 21/21 ctest green. Docs: protocol.md §6/§7, voice.md §6, architecture.md §5, PROGRESS.md.
2026-06-18 01:18:33 +02:00
// Send a plaintext KEEPALIVE frame to the server media endpoint. Called from run_udp_recv
// every kUdpKeepaliveIntervalMs to hold NAT bindings + bump the server's last_seen
// (docs/voice.md §6). Plaintext — no AEAD — to avoid racing the audio thread's seal().
void send_udp_keepalive();
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>
2026-06-16 14:12:37 +02:00
// capture_cb passed to audio_engine_.start(): encode + seal + send one frame for the
// given local stream `kind` (multiple concurrent local streams are possible).
void on_capture_frame(int kind, const int16_t* pcm, int samples, int channels);
// Encode one frame of exactly ls.frame_samples samples-per-channel (upmixing mono→stereo
// for a stereo channel as needed), seal it, and send it over UDP, advancing ls.timestamp.
// Called by on_capture_frame for each frame_samples-sized chunk. Assumes local_streams_mu_
// is held and the send gate/crypto checks have already passed.
void encode_and_send_frame(LocalStream& ls, const int16_t* pcm, int samples, int channels,
int fd);
// Inspect a User proto's streams and wire up any new remote ssrc into audio_engine_,
// emitting VC_EVENT_STREAM_STARTED/STOPPED as streams appear/disappear.
void sync_remote_streams(const voicecat::v1::User& user);
// Starts audio_engine_ (capture+playback) if not already running.
void ensure_audio_running();
// Joins udp_thread_, stops audio_engine_, clears media crypto/remote-stream state.
// Safe to call multiple times. Called both from run_io()'s cleanup and disconnect().
void teardown_voice();
// talk_timer_thread_ entry point: polls audio_engine_ for remote talk-state edges
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>
2026-06-16 14:12:37 +02:00
// and local capture activity, emitting VC_EVENT_TALK_STATE. Never the audio RT thread.
void run_talk_timer();
// Find a LocalStream by its client-assigned stream_id (held under local_streams_mu_ by
// the caller, or taken internally). Returns nullptr if not found/not active.
LocalStream* find_local_stream_by_id(uint32_t stream_id);
2026-06-15 23:48:44 +02:00
// ── Helpers (io_thread_ and caller threads) ─────────────────────────────────
// Queue an encoded envelope to be sent on io_thread_.
void queue_envelope(const voicecat::v1::Envelope& env);
fix(net): broadcast LEFT on disconnect, add keepalive/reaper, cap PLC hiss Three reported bugs traced to one root cause plus two missing designed features: 1. Stale users + eternal PLC hiss (root cause): ConnSession::close() silently erased dropped users without broadcasting UserEvent::LEFT, so peers never learned the user left and their audio engines never called remove_stream — Opus PLC synthesized comfort noise forever. Fix: broadcast_left() helper + close() broadcasts LEFT before erasing. 2. PLC cap (defense-in-depth): on_playback now caps pure PLC at ~2s, then emits digital silence so a stale stream can never hiss forever even if remove_stream is skipped. Resets automatically on fresh packets. 3. No timeout / no ping: client never sent Ping, server had no last_seen / reaper, so half-open connections (NAT timeout, wifi loss, sleep) left ghost users forever. Fix: client Ping every 15s with RTT measurement, ConnSession::last_seen bumped on every inbound TCP/UDP frame, steady_timer reaper sweeps every 15s and drops sessions older than 45s (configurable via server::Config). 4. UDP KEEPALIVE: client sends plaintext kFrameKeepalive every 5s; server bumps last_seen + echoes back. Keeps NAT bindings alive and lets media activity defer the reaper independently of TCP. 5. Graceful client disconnect: vc_disconnect() sends Disconnect{code=0} via a flag-based io-thread exit (no double-close race); server handles client-sent Disconnect with immediate close() + LEFT broadcast. 3 new tests: disconnect_left, plc_cap, reaper_timeout. 21/21 ctest green. Docs: protocol.md §6/§7, voice.md §6, architecture.md §5, PROGRESS.md.
2026-06-18 01:18:33 +02:00
// Keepalive: send a Ping envelope with a fresh nonce and record the sent time for
// RTT measurement when the Pong arrives. Called from the read loop when kPingIntervalMs
// has elapsed. (docs/protocol.md §7)
void send_ping();
2026-06-15 23:48:44 +02:00
// Drain send_queue_ by doing blocking TLS writes (called on io_thread_).
void drain_sends();
// Transition state + emit VC_EVENT_CONNECTION_STATE.
void set_state(vc_connection_state s);
// Convenience event emitters.
void emit_error(vc_result r, const char* text);
void emit_disconnected(vc_result r, const char* reason);
};
#endif // VOICECAT_CORE_CLIENT_H