post-m3-vad-ptt-devices-stereo
6
Commits
| Author | SHA1 | Message | Date | |
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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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