Removes leftover debug scaffolding (stray Console.WriteLine/NSLog traces,
dead nick_buf_ptr, a no-op --print-config flag now implemented for real),
fixes stale/misleading comments (channel passwords are no longer a "future
M5+" feature, a wrong cross-reference, a stale TlsContext::close() mention,
an incomplete BanRecord::subject_type doc, and a smoke test pointing at a
build/m1-dev preset that no longer exists), strips internal M1-M5 milestone
jargon from comments now that the roadmap is done, trims comments that just
restated the following line, and consolidates a few "why" explanations that
were duplicated 2-3 times in the same file.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
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>
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).
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>
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>
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>
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>
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>
