start_loopback_capture hardcoded channels=1, forcing miniaudio to downmix the system's stereo mix to mono before the encoder saw it -- on_capture_frame then upmixed L=R to produce fake stereo. Now the loopback device opens in the channel's mode (stereo when the channel is stereo), CaptureCallback carries an explicit channels param so the encoder gets real interleaved L/R, and a mono fallback covers unusual render endpoints. New test_loopback_stereo_capture asserts L!=R end-to-end; 18/18 ctest green.
A connected Windows client would randomly snap from its joined channel
back to Lobby. Root cause was a state-sync inconsistency, not a drop:
the server delivered self-initiated state changes (channel join/leave,
stream announce/stop) only as a private *Result to the actor and
broadcast the authoritative UserEvent::UPDATED to everyone else. The
core never applied the result to its SessionModel, so vc_list_users()
kept self in the old channel; the Windows HandleUserUpdated rebuilds
_currentChannelId from vc_list_users() on any user's UPDATED event, so
the next unrelated event surfaced the stale self-channel.
Fix, per the response-vs-broadcast contract now documented in
docs/protocol.md §6: the *Result is pure ack/correlation/actor-private
payload; the resulting state change is broadcast to every client
INCLUDING the actor, and clients apply it to their local model rather
than re-deriving own state from a *Result.
- server: join/leave/stream announce+stop broadcast with exclude=0
- server: text fan-out includes the sender (channel + private echo)
- core: response handlers no longer mutate session_model_
- windows: drop optimistic text echo; render own message via the relay
- docs/protocol.md §6: document the response-vs-broadcast contract
Registry-level admin broadcasts (move/mute/kick/channel CRUD) already
used exclude=0 and were correct. ctest build/m1-dev 18/18 green;
VoiceCat.App builds 0 warnings.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
- docs/building.md: explains what each CMake preset (dev, m1-dev, m2-dev,
server-release) is actually for, and how to build voicecat-server + vccli
for manual testing. Linked from CLAUDE.md's doc index.
- core/include/voicecat.h, core/src/voicecat.cpp, core/src/protocol/protocol.h,
server/src/main.cpp: doc-header comments still claimed M0-skeleton/stub
behavior (VC_ERR_NOT_IMPLEMENTED everywhere, "prints what it would do",
protobuf codegen "commented") that M1-M3 made real. Updated to describe
current behavior, with the dev-preset stub fallback noted explicitly where
it still applies.
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>
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>
