Talon
867557eda1
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>
VoiceCat
Self-hosted, native voice & text chat in the spirit of classic TeamSpeak / Mumble —
channel-based voice, channel + private text, one server you run yourself. Plain TCP
(control) and UDP (media), no WebRTC. Encrypted by default. A shared C++ core
(libvoicecat) drives native clients (Swift on macOS/iOS, C# on Windows) and the server.
Status: pre-implementation. The design is complete in
docs/. The code is an M0 skeleton — it compiles and links, but every subsystem is a stub. SeeAGENTS.mdto start building, anddocs/roadmap.mdfor the milestones.
Read the design first
The docs/ folder is the source of truth. Start at docs/README.md,
then architecture → protocol → voice → security → tech-stack → deployment →
roadmap.
Build the skeleton (no dependencies needed yet)
The M0 skeleton builds with just a C++20 compiler + CMake + Ninja — no vcpkg, no third-party libraries, because every subsystem is currently a stub.
cmake --preset dev
cmake --build --preset dev
ctest --preset dev # runs the smoke test (links the core, calls the C ABI)
Artifacts land in build/dev/bin/ (voicecat-server, vccli).
When you start implementing a subsystem that needs real libraries, build with vcpkg deps:
# one-time: git clone https://github.com/microsoft/vcpkg && ./vcpkg/bootstrap-vcpkg.sh
export VCPKG_ROOT=/path/to/vcpkg # set VCPKG_ROOT (works on Linux/macOS/Windows)
cmake --preset server-release # auto-installs deps from vcpkg.json
cmake --build --preset server-release
Layout
docs/ design spec (read this)
core/ libvoicecat — the shared C++ core
include/ voicecat.h (the C ABI all clients call)
proto/ voicecat.proto (control-plane wire format, source of truth)
src/ net/ crypto/ codec/ protocol/ session/ audio/ (stubs today)
server/ voicecat-server (headless; links the core)
tools/vccli/ headless test client — drives the protocol from M1 on
clients/ apple/ (Swift, M4) windows/ (C#, M4) — placeholders for now
tests/ CTest targets
License
Permissive-only dependencies (no GPL/LGPL) so the project can be redistributed freely,
including closed-source. Project license: TBD (see docs/tech-stack.md §5).