653131b876abc979c2060bbdce3e5a50127af2c2
19
Commits
| Author | SHA1 | Message | Date | |
|---|---|---|---|---|
  TalonandClaude Sonnet 4.6
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2c8178fa02 |
chore: remove skeleton build mode and stub #ifdef scaffolding
Drop the M0 no-deps skeleton preset and all VOICECAT_HAS_NET/AUDIO/OPUS/NS guards that it required. Every subsystem is fully implemented; the stub #else paths were dead code that added noise to every header and source file. - CMakePresets.json: remove skeleton configure/build/test entries - CMakeLists.txt (root/core/tests): remove VOICECAT_USE_VCPKG_DEPS option and guards; all targets now build unconditionally - 17 C++ source files: unwrap HAS_* guards, delete stub #else blocks - apm_processor.cpp: delete ApmPassthrough no-op class; create() always returns RnnoiseProcessor - 18 test files: remove HAS_* guards and stub int main() skip bodies - docs/building.md: remove skeleton from preset table and prose VOICECAT_HAS_LOOPBACK (Windows WASAPI loopback platform gate) unchanged. 29/29 ctest green. Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com> |
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TalonandClaude Opus 4.8
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b44a200b95 |
fix(audio): apply receive-side NR to stereo mic streams
The per-listener noise-reduction toggle (vc_set_remote_stream) did nothing on Windows/macOS/iOS. The decode loop gated the RNNoise pass on dec_channels == 1 as a proxy for "this stream is voice" (assuming stereo => screen-share). The stereo-mic capture commit broke that: a stereo mic with send-side NR off transmits stereo Opus, so the receiver decoded two channels and skipped NR entirely. gain/mute have no channel guard, which is why only NR appeared broken. Thread the stream kind through init_recv_stream into RemoteStream::is_voice (set from si.kind() == STREAM_MIC), gate receive NR on is_voice instead of channel count, and fold a stereo voice frame to mono -> denoise -> duplicate back across both channels in place (symmetric with the send-side downmix; RNNoise is mono-only). Screen-audio shares are never denoised. New test test_recv_noise_reduction drives AudioEngine and asserts a stereo voice stream's noise floor collapses with NR on (RMS 1046 -> 0.1) while a screen-audio share stays unchanged. ctest --preset dev green 29/29. Docs: voice.md section 10. Shared-core fix; clients need only a rebuild. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com> |
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TalonandClaude Opus 4.8
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bad9c7533a |
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> |
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TalonandClaude Opus 4.8
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ce2035f271 |
fix(audio): bound playout depth to stop voice latency ratcheting up
Latency between speakers grew to multiple seconds and only reset on rejoining voice. Root cause was the receiver playout logic, not the codec settings: the playout clock free-ran in real time while the sender omitted silence from its timestamps (and set no header flags), and the only correction snapped the clock to the *oldest* buffered frame — which could only ever add standing latency. target_depth_ms_ was computed but never enforced, so latency could only grow or reset. Fix: bound playout against the stream's leading edge (newest frame). (Re)seed to the leading edge on start/marker/starve (no prebuffer, so latency stays low), and frame-skip catch-up trims any backlog beyond target+hysteresis — the missing downward force. Hardening: sender now stamps kFlagMarker (talkspurt start) and kFlagDtx, consumed on recv for clean resync; adaptive late-drop window; EWMA outlier rejection so silence gaps/stragglers don't poison the estimate; duplicate counting and ring-underrun diagnostics. New test_jitter_depth asserts depth stays bounded (<200ms) while arrivals outrun playback. ctest --preset dev green (27/27). Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com> |
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TalonandClaude Opus 4.8
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e155e342f4 |
feat(audio): channel sample_rate caps Opus bandwidth (narrowband/wideband)
The per-channel sample_rate field was inert after pinning the codec to 48 kHz. Make it meaningful without changing the 48 kHz clock: carry it as OpusParams::max_bandwidth_hz and apply OPUS_SET_MAX_BANDWIDTH in OpusEncoder::init (8000->narrowband, 16000->wideband, 24000->super-wideband, 48000->full). A low-bitrate room can now shed out-of-band content while every endpoint keeps a single 48 kHz clock. Make sample_rate channel-authoritative on the server: conn_session no longer overrides effective sample_rate with the client's always-48000 request (it now behaves like frame_ms/mode). vc_get_stream_audio_config reports the channel's configured rate for own streams too. New ctest channel_samplerate: a 7 kHz tone is attenuated ~1000x on an 8 kHz (narrowband) channel vs a 48 kHz (full-band) channel, proving the cap is in effect. ctest --preset dev 26/26. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com> |
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TalonandClaude Opus 4.8
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a460009a2f |
fix(audio): reframe send path to channel frame_ms; pin codec to 48 kHz
The AudioEngine capture clock is fixed at 48 kHz / 20 ms (960-sample frames), but a channel may set any Opus frame_ms (2.5..60 ms, voice.md §3) and the server enforces it unclamped. on_capture_frame handed the engine's 960-sample frame straight to an encoder configured for the channel's window: frame_ms > 20 was silently ignored, and frame_ms < 20 broke entirely (receiver sized its decode buffer too small -> OPUS_BUFFER_TOO_SMALL -> dead audio). Affected the hardware mic and vc_stream_feed_pcm alike. Reframe each captured/fed block to ls.frame_samples via a per-LocalStream accumulator (pre-sized at announce, no RT-thread alloc) before encode_and_send_frame; the 20 ms case stays a zero-copy fast path. Also pin the codec to 48 kHz in opus_params_from_audio_config — it was honoring a non-48k effective sample_rate against a 48 kHz PCM clock. New ctest frame_ms_reframe covers 40 ms (accumulate) and 10 ms (split) feed->encode->relay->decode->sink round trips. ctest --preset dev 25/25. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com> |
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Talon
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6c17881cc0 |
feat(ios): real echo cancellation/NR via native Voice-Processing engine
iOS "voice chat" had echo and no noise suppression: real iOS AEC/NS/AGC come only from Apple's Voice-Processing I/O unit (VPIO), but the core plays/captures via miniaudio's plain RemoteIO units, so .voiceChat mode alone never engaged AEC. Core (ABI PATCH 1->2): - vc_set_mixed_output_sink + vc_set_external_playback. In external mode the AudioEngine opens no hardware playback device; a mixer-timer thread drives on_playback (decode+mix) on a ~20ms cadence and ships the final mix to the sink. start() also skips the hardware capture device when the MIC stream is external_feed (AudioParams.external_capture). - New white-box test test_external_playback (drives the timer with no hw). iOS/Swift: - StreamDescriptor.externalFeed; VoiceCatClient.setMixedOutputSink / setExternalPlayback wrappers. - IOSVoiceProcessingEngine: AVAudioEngine + setVoiceProcessingEnabled; mic tap -> feedPcm, mixed-sink lock-free ring -> AVAudioSourceNode (both share the VPIO unit so AEC has its reference signal). - IOSAudioRouter.currentConfigUsesVoiceProcessing scopes VPIO to the AEC presets; SessionState join/leave + reconcileVoicePath() switch paths; Voice Chat defaults to speaker; Settings surfaces AEC/NS state. Known: pending on-device verification; a few bugs to fix afterward. |
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TalonandClaude Sonnet 4.6
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615d2a8e5f |
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> |
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TalonandClaude Sonnet 4.6
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fdcd8d1427 |
feat: DRED (Deep REDundancy) per-channel toggle
Adds Opus 1.6 DRED support end-to-end: encoder embeds 20 ms of ML redundancy in every packet when enabled; decoder recovers lost frames from the next buffered packet's DRED extension rather than falling back to PLC comfort noise. Protocol: bool dred = 11 added to AudioConfig (backward-compatible, defaults false). C ABI: int dred added to vc_audio_config. Encoder: OPUS_SET_DRED_DURATION(2) when dred=true. Decoder: OpusDREDDecoder + per-stream OpusDRED scratch pre-allocated off the RT thread; JitterBuffer::try_copy_front_payload peeks at the next packet without popping on every PLC step; opus_decoder_dred_decode reconstructs the lost frame if DRED data is present, otherwise falls back to PLC. New test: test_dred_toggle (22/22 ctest green). Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com> |
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Talon
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bcb7ae8ccb |
build(cmake): clean up presets, add release/apple presets, cross-platform triplets
Rationalize the preset set to match the project's actual state (past M5): - Rename dev->skeleton (no-deps stub smoke), m1-dev->dev (default dev preset) - Drop m2-dev (cache-identical to m1-dev) - Add release preset (optimized + tests on, symbols kept) - Strip server-release binaries (-s linker flag) - Add apple-dev/apple-ios/apple-ios-sim scaffolding presets for XCFramework Add cmake/voicecat-toolchain.cmake wrapper that auto-resolves the vcpkg triplet from the host platform (x64-mingw-static/x64-linux/arm64-osx) so the main presets work on Windows/Linux/macOS without per-OS variants. Update all docs (building.md, CLAUDE.md, README.md, AGENTS.md, deployment.md, tech-stack.md, client READMEs) and stale preset-name references in code comments. No C++ behavior changes — the core was already portable. |
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Talon
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487a561963 |
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.
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Talon
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3990f63f0f |
M5: moderation, permissions, channel CRUD, in-app account management
- Server-side moderation & permissions (kick/ban/move/server-mute, channel CRUD). - Database schema v2: channels, bans; BLAKE2b channel passwords, Argon2id accounts. - C ABI additions and client-side handling (vc_kick_user, vc_ban_user, vc_set_permission, vc_set_server_mute, vc_move_user, vc_create/edit/delete_channel, vc_create/reset/delete/list_account). - vccli flags for all M5 operations plus --username/--password auth. - Four new tests covering permissions, kick/ban/move/mute, admin accounts, channel CRUD. - Docs: protocol.md envelope updates, security.md channel-password hashing, PROGRESS.md. |
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TalonandClaude Sonnet 4.6
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63b241cc2e |
feat(M4): Windows WinForms client, TOFU identity pinning, VAD threshold + always-on mode
Core ABI extensions (voicecat.h): - vc_list_channels / vc_list_users / vc_list_user_streams — pull-based snapshot getters for the channel-tree and user-list UI; session_model_mu_ guards cross-thread reads - VC_EVENT_JOIN_RESULT / vc_join_channel — channel join with optional password - VC_EVENT_SERVER_IDENTITY + vc_confirm_server_identity — TOFU gate that blocks io_thread_ until the UI approves or rejects; pins TLS leaf-cert SHA-256 (not declared Ed25519) - vc_get_server_identity_display — Ed25519 fingerprint for human-readable display only - VC_INPUT_ALWAYS_ON = 2 in vc_input_mode — transmit unconditionally, no VAD gate - vc_set_vad_threshold — live RMS threshold update (0.0–1.0); EnergyVadProcessor stores it atomically so the audio RT path reads without a lock C++ implementation: - SessionModel::apply_snapshot / apply_channel_event fixed to populate parent_id, password_protected, and max_users (were permanently zeroed) - TlsContext::peer_cert_fingerprint — SHA-256 of peer leaf cert DER via mbedTLS - TofuStore split into peek (read-only) + pin (write) so first-connect only persists after user approval; tofu_store_path in vc_config for per-user pin file location - TcpAcceptor uses dual-stack IPv6+IPv4 fallback (fixes localhost → ::1 on Windows) - windows-client CMake preset: Release shared DLL, static MinGW runtime, no tools/tests - New C++ tests: test_channel_user_list_abi, test_tofu_flow (14/14 green) Windows client (clients/windows/ — .NET 10 WinForms): - VoiceCat.Interop: LibraryImport P/Invoke surface, UnmanagedCallersOnly callbacks, Channel<VoiceCatEvent> event delivery drained by 30ms WinForms Timer - VoiceCat.App: ConnectDialog (saved servers, DPAPI password storage), ServerIdentity- Dialog (TOFU first-connect / mismatch warning), MainForm (channel TreeView, user ListBox, RichTextBox chat, voice controls, device pickers, VAD/PTT/always-on mode, per-user gain/mute/NR tuning, VAD sensitivity TrackBar, level meter ProgressBar) - PttKeyCaptureDialog — focus-scoped PTT key capture (documented limitation) - PerUserTuningDialog — real-time gain/mute/NR applied to all of a user's streams - Accessibility: explicit AccessibleName/Description on every control, & mnemonics, Activity log ListBox as durable screen-reader record, AutomationNotification for curated live announcements Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com> |
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TalonandClaude Sonnet 4.6
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5f6c223526 |
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> |
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TalonandClaude Sonnet 4.6
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867557eda1 |
feat(M3): multi-stream & per-channel tuning
Implements docs/roadmap.md M3: multiple concurrent streams per user (MIC + SCREEN_AUDIO + AUX_DEVICE), independent per-stream receiver gain/mute/noise- reduction, talk indicators, and enforced per-channel Opus configurability (mono/stereo, bitrate, frame size, FEC/DTX, application). Bugs fixed along the way (found while implementing, not pre-existing scope): - Server hard-coded stream_id=1 for every announce, so a second stream from the same user silently overwrote the first in SessionRegistry::set_user_stream. Now a per-session counter (ConnSession::next_stream_id_); handle_stream_stop validates against announced_stream_ids_ before clearing. - Client dropped mode/dtx/complexity/application from effective_audio even for the single M2 stream -- only sample_rate/bitrate_bps/frame_ms/fec were ever applied to OpusParams. Fixed on both the send (handle_stream_announce_result) and receive (sync_remote_streams) paths via a shared opus_params_from_audio_config() helper. - OpusEncoder always used OPUS_APPLICATION_VOIP; added OpusParams::application and wired it through. - on_playback's per-stream decode passed the wrong frame_size to opus_decode (total samples instead of samples-per-channel), which would have overflowed the decode buffer for any stereo stream. - teardown_voice() raced when called concurrently from run_io()'s own cleanup and from disconnect() on a different thread -- both could see udp_thread_/talk_timer_thread_ as joinable() at once and race to join() the same std::thread (intermittent std::system_error under ctest). Fixed with a teardown_mu_ guard instead of carrying the flake forward. New: - Per-channel AudioConfig: SessionRegistry now seeds Lobby (mono/24kbps/VOIP/ FEC+DTX) and a new "Music Room" channel (stereo/128kbps/AUDIO/no DTX); handle_stream_announce enforces the channel's config, clamping (not overriding) bitrate_bps to its ceiling. - core/src/core/client.h/.cpp: local-stream state is now a std::unordered_map<int, LocalStream> keyed by vc_stream_kind, with request_id-correlated announce/result handling (request_id already round-tripped on the wire; just wasn't read before). on_capture_frame is kind-aware and upmixes mono capture to stereo when a stream's config calls for it. set_self_mute's mic_muted now only gates the MIC kind. NS is wired through set_remote_stream. New run_talk_timer() thread emits VC_EVENT_TALK_STATE from both remote and local edge detection. - core/src/audio/audio_engine.h/.cpp: kind-keyed injection taps (inject_capture), stereo-to-mono downmix at the decode/mix boundary, RemoteStream gains recv_ns (lazy ApmProcessor) + noise_reduction_enabled and last_voice_ms/talking; new set_stream_noise_reduction() and poll_talk_transitions(). - core/src/session/session.h/.cpp: Stream now carries the full AudioConfig, not just sample_rate/frame_ms. - New additive C ABI (core/include/voicecat.h): vc_audio_config + vc_get_stream_audio_config (effective Opus config for any stream you own or a peer's); vc_test_inject_capture (test-only synthetic PCM injection, clearly marked, mirrors AudioEngine::inject_capture). - tests/test_m3_multistream.cpp: the M3 exit criterion through the real ABI (mirrors test_voice_client_abi.cpp's approach, not raw sockets) -- two concurrent local streams, independent gain/mute/NS control, per-channel config divergence via vc_get_stream_audio_config, talk indicators. Explicitly out of scope for this pass (tracked in PROGRESS.md, not silently dropped): VAD/PTT input gate + device enumeration; real WASAPI loopback capture for SCREEN_AUDIO (synthetic injection only); true stereo playback output (AudioEngine's mixer/output device stays mono -- Opus itself is fully stereo-correct on the wire). ctest --test-dir build/m1-dev: 11/11 green, verified across 3 consecutive full-suite runs plus 8 standalone runs of the new test. Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com> |
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TalonandClaude Sonnet 4.6
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c693cab35c |
fix(M2): wire vc_client's real voice plane through the C ABI, not just raw sockets
test_m2_voice passed against raw BSD sockets, but vc_client::stream_start/stop, UDP binding, and capture/recv were still VC_ERR_NOT_IMPLEMENTED stubs -- meaning vccli and any GUI client still couldn't actually talk. Implements the real client-side UDP-binding handshake, media key derivation, capture->encode->seal-> send and recv->open->decode->playback paths, plus server-side StreamInfo broadcast so peers learn about each other's streams via sync_remote_streams(). Adds test_voice_client_abi (two real vc_client instances, not raw sockets) and vccli --voice/--mute/--text flags, manually verified live between two instances. Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com> |
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TalonandClaude Sonnet 4.6
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694494a5be |
feat(M2): UDP voice/media plane -- SFU relay, Opus, AEAD, jitter buffer
Adds the full voice pipeline: 14-byte binary frame header, ChaCha20-Poly1305 AEAD keyed from the TLS exporter, libopus encode/decode with FEC/PLC/DTX, an adaptive per-ssrc jitter buffer, a miniaudio capture/playback engine, an APM passthrough stub, and the UdpBinding/StreamAnnounce signaling chain wired through ConnSession/SessionRegistry into a new server-side SFU (MediaRelay) that decrypts and re-encrypts frames per channel member. Exit criterion verified: test_m2_voice — two headless clients relay 50 encrypted Opus frames through the server; ctest --preset m1-dev is 9/9 green. Also corrects protocol.md's UdpBinding diagram, which described the UDP-side binding packet as AEAD-sealed when it is in fact a plaintext bootstrap frame (separate from the TCP/TLS UdpBinding ack). Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com> |
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TalonandClaude Sonnet 4.6
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63f457fc54 |
feat(M1): TCP/TLS control plane -- auth, channels, ephemeral text
Implements the full M1 milestone. Two clients authenticate over TLS 1.3 (guest + Argon2id password) and exchange channel + private text messages through a real server. All five ctest --preset m1-dev tests pass in ~1 s. Key components added: - vcpkg baseline + m1-dev preset (protobuf/mbedTLS/libsodium/asio/sqlite3) - FrameCodec feed+emit, encode/decode_envelope, protobuf codegen - TcpServerConn with blocking TLS handshake thread + tls_read_loop - TlsContext (mbedTLS 1.3, ECDSA-P256 self-signed cert, TOFU on client) - WorkerPool (3 threads, used for Argon2id) - Database: SQLite + libsodium Argon2id, account lifecycle, bootstrap admin - ServerIdentityManager: Ed25519 key + cert generate/persist/fingerprint - ConnSession state machine: WaitingHello -> WaitingAuth -> Authenticated - SessionRegistry: channel tree, user map, text routing, broadcast - vc_client full M1 C ABI: connect/TLS/handshake/auth/text/disconnect - voicecat-admin CLI: account add/reset/del/list - test_m1_integration: M1 exit criterion, verified green Bug fixed: double-framing in ConnSession::send_envelope -- encode_envelope was adding the [4-byte len] prefix, then TcpServerConn::send_frame added a second one, causing the client to parse [len][proto] as protobuf (silent failure). Fixed by serializing raw protobuf bytes in send_envelope and letting send_frame apply the single length prefix. Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com> |
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TalonandClaude Opus 4.8
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b332b0972b |
scaffold: M0 skeleton + agent onboarding (build, architecture, progress)
Turn the design into a buildable, dependency-free M0 skeleton plus the
onboarding layer so a new agent can pick up instantly.
Build system:
- CMake + CMakePresets (dev = no deps; server-release = vcpkg) + vcpkg.json
- Skeleton builds with just a C++20 compiler; deps stay off until needed
- .gitattributes (LF), .gitignore, .clang-format
Core (libvoicecat):
- core/include/voicecat.h: full C ABI (the client/server contract), stubbed
- core/proto/voicecat.proto: control-plane wire format, matches docs/protocol.md
- src/{net,crypto,codec,protocol,session,audio,core}: subsystem stubs that
return VC_ERR_NOT_IMPLEMENTED, each pointing to its design doc
- server/ (voicecat-server) and tools/vccli/ link the core
- tests/: CTest smoke test asserting the C ABI contract (behavior, not just build)
- clients/{apple,windows}: M4 placeholders
Onboarding for agents:
- CLAUDE.md: hub — build/test commands, architecture at a glance, doc map, rules
- AGENTS.md: working method (behavior-driven; clean compile is the floor not the goal)
- PROGRESS.md: living tracker — M0 done, M1 task checklist, "where we left off"
Verified: cmake --preset dev && cmake --build --preset dev && ctest --preset dev → green.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
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