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).
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>
Input mode (VAD/PTT/Always-On), VAD threshold, and the new mic gain were
applied to the core + UI but never saved, so every relaunch reset to VAD
defaults. Each client now persists them and re-applies on connect:
- iOS: UserDefaults (SessionState.loadAndApplyVoiceSettings + setter writes)
- macOS: UserDefaults via MainWindowController didSet + loadPersistedAudioSettings
(settings window also restores the VAD slider from the stored threshold)
- Windows: new Models/VoiceSettings.cs (JSON at %AppData%\VoiceCat\voice.json,
mirrors FeedbackSettings) loaded/applied in MainForm
Add global send-side mic gain API vc_set_input_gain (applied to MIC PCM in
on_capture_frame before the VAD gate, clamped to int16) + Swift/C# bindings,
and a 0-300% (default 100%) mic-volume slider on all three clients.
Fix iOS chat: ChatView called sendText(scope:.channel) with no targetId (0),
so channel messages went nowhere; now passes session.currentChannelId.
Fix iOS per-user tuning for VoiceOver: the tuning sheet was long-press
.contextMenu only (invisible to VoiceOver); UserRow now also exposes the same
buttons via .accessibilityActions (no visual change).
Verified: core builds clean; ctest 24/27 (3 pre-existing teardown crashes,
reproduced with changes stashed); VoiceCatMac + VoiceCatiOS (arm64 sim) build
SUCCEEDED; VoiceCat.Interop dotnet build succeeded. Windows App not built
(WinForms can't build on macOS) — follows existing patterns.
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.
- Voice actions (Join Voice, Share Screen Audio) moved to a ToolStrip toolbar and
a new Voice menu in the menu bar; removed from the bottom voice panel
- Activity log and chat log collapsed into a single RichTextBox (rtbLog); activity
events appear in gray, chat messages in default color
- Private messaging reworked: each conversation opens in its own modeless
PrivateMessageForm instead of sharing the main chat log via a scope dropdown;
cboScope removed; main compose bar always sends to the current channel
- New "Messages -> New Private Message..." menu item (Ctrl+P) opens a UserPickerDialog
listing all connected server users (not just the current channel) so you can PM
anyone on the server
- Channel tree now shows live user counts, e.g. "General (3)" -- counts sourced from
the existing _users dictionary which already tracks all server users with channel IDs
- Global output volume slider (TrackBar, 0-100, default 80) added to the right panel;
wired to new vc_set_output_volume C ABI function that applies a master gain multiplier
in the audio engine playback callback after mixing all streams
- vc_set_output_volume added end-to-end: voicecat.h, audio_engine.h/.cpp,
client.h/.cpp, voicecat.cpp, NativeMethods.cs, VoiceCatClient.cs
- Documented Windows PowerShell ctest requirement in AGENTS.md and CLAUDE.md:
MinGW binaries exit 0xc0000139 in Git Bash; always run ctest/.exe via PowerShell
22/22 ctest green (PowerShell); dotnet build 0 warnings.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Diagnosed by comparing against TeamTalk5 (Client/iTeamTalk), which
achieves stereo mic + A2DP output. Five fixes:
1. configureStereoCapture now calls setPreferredInput +
setInputDataSource (mirroring TeamTalk5's SoundDevicesModel).
Previously omitted based on incorrect diagnosis that
setPreferredInput collapsed A2DP — the real culprit was
setPreferredInputNumberOfChannels(2), which neither project uses.
2. New C ABI: vc_audio_restart (full stop + re-init, unlike
suspend/resume which only stop/start). Swift wrapper added.
The withAudioSuspend wrapper that used it was removed after
on-device testing showed it killed all audio (including
VoiceOver) when switching presets — the core's
set_capture_channels handles engine restart internally.
3. Bluetooth options: Voice Chat preset now includes BOTH
.allowBluetoothHFP AND .allowBluetoothA2DP (matching TeamTalk5's
UtilSound.swift:228). Previously HFP-only blocked A2DP headphones.
4. Capture channels now reset when switching stereo→mono via
selectCaptureChannels/applyPreset. AudioSessionManager tracks
activeMicStreamId (set by SessionState on join/leave voice).
5. Docs synced: voice.md, tech-stack.md, architecture.md,
PROGRESS.md. Removed stale setPreferredInputNumberOfChannels(2)
references.
Verified: ctest --preset dev 21/21 green, iOS client builds.
Stereo mic + A2DP output still needs on-device debugging — the
core recipe is correct but iOS 26 route behavior requires
hands-on testing with a debugger.
Three iOS client problems fixed plus a new core stereo-mic capture ABI:
1. Channel-id sync bug (mic button permanently dimmed): SessionState never
synced currentChannelId from the self user's channelId on connect, so the
mic button (gated on currentChannelId == 0) stayed dimmed. Added
syncSelfChannel() (mirrors macOS MainWindowController.swift:461,491,522);
called from init/.channelList/.userJoined/.userLeft/.userUpdated/.joinResult.
Added applyServerMuteState() + serverMuted/serverDeafened to VoiceState.
2. Join/Leave Voice button: replaced icon-only mic toggle with explicit
text button (parity with macOS). Mute/deafen disable when not in voice.
3. IOSAudioRouter.swift (new): full AVAudioSession routing layer — input
port selection, built-in mic orientation/polar patterns, Bluetooth
HFP/A2DP/Off modes, Standard/Raw mic processing, stereo capture, AirPlay,
UserDefaults persistence. AudioSessionManager delegates to it.
4. Core stereo-mic capture (append-only ABI): vc_set_capture_channels()
lets the core open the mic device in stereo (2-ch interleaved). LocalStream
gains capture_channels; ensure_audio_running reads it; audio_engine.cpp
capture_accum_ + on_capture updated to channel-aware accumulation. Test
test_stereo_mic_capture (headless, L!=R stereo round-trip). Swift wrapper
VoiceCatClient.setCaptureChannels.
5. Settings UI rework: AVAudioSession-derived input/output tree replaces
miniaudio device picker.
6. iOS deployment target raised to 18.0 (Package.swift + project.pbxproj).
swift-tools-version 6.0 with swiftLanguageModes .v5.
Docs: tech-stack.md, architecture.md, voice.md, roadmap.md, building.md
updated; stale 'vc_audio_suspend/resume deferred' claims corrected.
Verified: ctest --preset dev 21/21 green; swift test 6/6 green;
xcodebuild -target VoiceCatiOS -sdk iphonesimulator BUILD SUCCEEDED.
Full SwiftUI app at clients/apple/iOS/VoiceCatiOS.xcodeproj:
- 24 Swift source files: AppState + SessionState (@Observable @MainActor),
AudioSessionManager (AVAudioSession owner + interruption/route handling),
ServerListStore/SavedServer (App Group container + Keychain sharing),
and 14 SwiftUI views covering the full feature set
- NavigationSplitView on iPad, TabView on iPhone (horizontalSizeClass)
- Channel tree via OutlineGroup, user list with context menu admin actions
- PTT via DragGesture(minimumDistance: 0) + @GestureState
- onEvent closures hop to MainActor via Task { @MainActor in ... }
- App Group: group.cat.voice.VoiceCat (shared with future ReplayKit extension)
C ABI: add vc_audio_suspend / vc_audio_resume (AudioEngine::suspend/resume)
called by AudioSessionManager on AVAudioSession interruption events.
XCFramework: add ios-arm64 and ios-arm64-simulator slices to build-xcframework.sh;
Package.swift gains .iOS(.v17) platform; CMakePresets.json adds apple-ios /
apple-ios-sim presets with arm64-ios / arm64-ios-simulator vcpkg triplets.
Verified: xcodebuild -target VoiceCatiOS -sdk iphonesimulator26.5 BUILD SUCCEEDED.
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.
PerUserTuningDialog previously broadcast one gain/mute/NR set to *all* of a
user's streams, even though the core mixer (AudioEngine::RemoteStream) and
the C ABI (vc_set_remote_stream) were already per-stream. The UI had no
per-mix controls anywhere.
Reworks the dialog to enumerate ListUserStreams on open and render one row
per stream (kind + label + Gain + Mute + NR), each wiring only to its own
stream_id. Adds a read-back ABI counterpart, vc_get_remote_stream, so the
dialog opens at the listener's actual current per-stream settings (defaults
1.0/unmuted/NR-off) rather than always 100%. Additive ABI change only; no
existing symbols touched.
Tests: test_m3_multistream extended with getter round-trip assertions; new
C# smoke test exercises the full P/Invoke marshaling path with two clients.
Docs: voice.md §10 notes the getter. NR checkbox keeps its honest
'passthrough' label (NS DSP still unbuilt per §8).
- 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>
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>
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>
