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# Architecture
## 1. The shared-core model
All non-UI logic lives in one C++ library, **`libvoicecat`**. The same library is linked
into every client and into the server. Platform UIs are thin and call the core through a
stable **C ABI** (`voicecat.h`).
```
┌───────────────────────────────────────────┐
macOS / iOS (Swift) │ │ Windows (C#)
┌──────────────────┐ │ libvoicecat (C++) │ ┌──────────────────┐
2026-06-17 00:52:02 +02:00
│ SwiftUI views │ │ ┌─────────────────────────────────────┐ │ │ WinForms (.NET 10│
│ AVAudioSession │──┼─▶│ C ABI (voicecat.h) │◀─┼──│ LibraryImport │
│ Swift↔C++ interop│ │ ├─────────────────────────────────────┤ │ │ P/Invoke │
└──────────────────┘ │ │ Session / Protocol state machine │ │ └──────────────────┘
│ │ Text + voice signaling │ │
Linux/macOS/Windows │ │ Audio engine: capture→encode→send, │ │
server │ │ recv→jitter→decode→mix→playback │ │
┌──────────────────┐ │ │ Codec layer (Opus 1.6) │ │
│ voicecat-server │──┼─▶│ Crypto + transport (TLS 1.3 + AEAD) │ │
│ (reuses core) │ │ │ Net I/O (Asio: TCP + UDP + timers) │ │
└──────────────────┘ │ └─────────────────────────────────────┘ │
└───────────────────────────────────────────┘
```
Why this shape:
- **Swift** (5.9+) can import C++ directly, but we still ship a C ABI because it is the
lowest-friction, most stable boundary and it is what **C#** needs (`LibraryImport`/
P/Invoke). One ABI serves both.
- The **server** is not a separate codebase. It links the same protocol, crypto, and Opus
code as the client, so framing/encryption can never drift between the two ends.
## 2. Layered design inside the core
From the OS up:
| Layer | Responsibility | Key deps |
|-------|----------------|----------|
| **Platform I/O** | Sockets, timers; audio device capture/playback | Asio, miniaudio |
| **Transport** | TLS 1.3 (TCP), exported-key ChaCha20-Poly1305 AEAD (UDP), framing, anti-replay | mbedTLS, libsodium |
| **Codec & DSP** | Opus encode/decode; APM (AEC/NS/AGC/VAD) send-side + per-user NR receive-side; resample; jitter buffer; mixer | libopus, webrtc-audio-processing, speexdsp |
| **Protocol** | Message (de)serialization, request/response correlation, state machine | protobuf |
| **Session/domain** | Channels, users, streams, permissions, text routing | — |
| **C ABI façade** | Handle-based API + event callbacks exposed to UIs | — |
A UI never sees a socket, an Opus packet, or a protobuf message. It sees: "connect",
"join channel", "start a stream from this device", "send this text", and a stream of
events ("user joined", "user is talking", "message received", "level meter = 0.4").
## 3. Threading model
Three classes of thread, with strict rules.
```
┌──────────────┐ lock-free ┌──────────────┐ lock-free ┌──────────────┐
│ Audio capture│ ──ring buffer─▶│ Net thread │ ──ring buffer─▶│Audio playback│
│ (RT, miniaudio│ │ (Asio loop) │ │ (RT, miniaudio│
│ callback) │◀───ring buffer─│ │◀───ring buffer─│ callback) │
│ capture→Opus │ │ TLS + AEAD, │ │ jitter→Opus │
│ encode │ │ route, relay │ │ decode→mix │
└──────────────┘ └──────────────┘ └──────────────┘
┌─────▼──────┐
│ Worker pool│ DB, Argon2id, file I/O,
│ (blocking) │ TLS handshakes, codec setup
└────────────┘
```
Rules:
- **Audio (real-time) threads** are driven by the OS audio callback. They must not
allocate, lock, log, or do syscalls beyond the ring-buffer hand-off. Opus encode/decode
runs here (it is allocation-free after init).
- **Net thread(s)** run the Asio event loop: TLS records, AEAD seal/open, protobuf parse, channel
routing, jitter-buffer feed. On the server, this is where the SFU relay copies packets
to subscribers.
- **Worker pool** absorbs anything that can block: SQLite, Argon2id verification, DNS,
TLS handshake CPU, codec (re)configuration.
- **Communication** between audio and net is single-producer/single-consumer lock-free
ring buffers (one per direction per stream). Control-plane events to the UI go through a
thread-safe queue drained on the UI's terms.
## 4. The C ABI (`voicecat.h`) — shape
Handle-based, opaque pointers, C-linkage. Illustrative (final names in implementation):
```c
typedef struct vc_client vc_client;
typedef struct {
void (*on_event)(void* user, const vc_event* ev); // state changes, messages
void (*on_level)(void* user, uint32_t stream_id, float rms); // meters (throttled)
void* user;
} vc_callbacks;
vc_client* vc_client_create(const vc_config* cfg, vc_callbacks cb);
void vc_client_destroy(vc_client*);
int vc_connect(vc_client*, const char* host, uint16_t port); // async; result via event
int vc_authenticate_guest(vc_client*, const char* nickname);
int vc_authenticate_user(vc_client*, const char* user, const char* password);
int vc_join_channel(vc_client*, uint32_t channel_id, const char* password /*nullable*/);
int vc_leave_channel(vc_client*);
// Streams (mic / screen audio / aux device)
int vc_stream_start(vc_client*, const vc_stream_desc* desc, uint32_t* out_stream_id);
int vc_stream_stop(vc_client*, uint32_t stream_id);
int vc_set_input_device(vc_client*, uint32_t stream_id, const char* device_id);
int vc_set_self_mute(vc_client*, bool mic_muted, bool deafened);
// Text
int vc_send_text(vc_client*, vc_text_scope scope, uint32_t target_id, const char* utf8);
// Enumeration helpers for UI device pickers
int vc_list_devices(vc_client*, vc_device_kind kind, vc_device_list* out);
```
Design notes:
- **Async, event-driven.** Calls return immediately; results and state changes arrive via
`on_event`. This maps cleanly onto SwiftUI/`async` and C# `event`/`Task` patterns.
- **The core owns audio.** Capture, encode, decode, mixing, and playback happen inside the
core via miniaudio. The UI only *selects devices*, *starts/stops streams*, and *renders
meters/state*. This keeps the real-time path identical on every OS. (iOS is the one
exception that needs UI-side cooperation — see below.)
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>
2026-06-16 16:11:52 +02:00
- **Device enumeration works pre-connect.** `vc_list_devices` needs no live session — device
pickers can populate before `vc_connect`. `vc_device.id` is an opaque, internally-encoded
handle (currently a hex-encoded `ma_device_id`) — always round-trip an id that came from
`vc_list_devices`/`vc_get_stream_audio_config`; never construct one by hand. Tolerate an
empty list (a machine can legitimately have zero input or output devices).
- **Strings are UTF-8 `const char*`; ownership is explicit.** Output buffers are
caller-allocated or returned with a paired `vc_free`.
### Per-platform binding notes
fix(ios): stereo mic + A2DP output, add vc_audio_restart ABI 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.
2026-06-19 16:58:21 +02:00
- **Swift / Apple.** Import the C ABI via a **module map** (`module VoiceCatC { header "voicecat.h" }`) staged into the XCFramework headers by `clients/apple/scripts/build-xcframework.sh` — Swift gets a clean `import VoiceCatC` with all C enums/structs/functions available directly (no manual redeclaration, unlike the C# P/Invoke layer). A **Swift wrapper** (`VoiceCatCore` package at `clients/apple/`) provides Swift-idiomatic types (`VoiceCatResult`, `VoiceCatEvent`, `Channel`, `User`, etc.) on top, mirroring the C# `VoiceCat.Interop` layer. Callbacks use `@convention(c)` closures (plain C function pointers, not ARC-managed closures) + `Unmanaged.passUnretained(self)` as the `user` context (the Swift analog of C#'s `[UnmanagedCallersOnly]` + `GCHandle`). Events are delivered on `@MainActor` via a coalesced `DispatchQueue.main` drain (one async block scheduled at a time) — the Swift analog of C#'s `Channel<VoiceCatEvent>` + 30ms WinForms Timer pump. `deinit` calls `vc_client_destroy` (joins all threads) then frees native CString config storage (the core stores raw pointers, doesn't copy). **macOS UI: AppKit** (chosen over SwiftUI for the most mature VoiceOver accessibility story — same rationale as the Windows client's WinForms choice); **iOS UI: SwiftUI** (narrower control surface, sufficient VoiceOver support). On **iOS** the app owns `AVAudioSession` (category `.playAndRecord`), requests mic permission, and handles interruptions/route changes — the core exposes hooks (`vc_audio_suspend`/`vc_audio_resume`/`vc_audio_restart`, implemented) the Swift layer calls from `AVAudioSession` notifications and `IOSAudioRouter` setting changes. All iOS audio routing (input port selection, mic orientation/polar patterns, HFP vs A2DP, measurement/raw mode, stereo capture) is driven from the Swift `IOSAudioRouter` singleton via `AVAudioSession` *before* the core (miniaudio) opens its device — miniaudio does NOT touch `AVAudioSession` on iOS. The core is told the capture channel count via `vc_set_capture_channels` (append-only ABI). `vc_audio_restart` does a full stop + re-init (unlike `suspend`/`resume` which only stop/start) so devices reopen against a new route after `AVAudioSession` reconfiguration. iOS 18.0 deployment target. Background voice and VoIP push (CallKit/PushKit) are a later milestone. The XCFramework carries a **fat static library** (`libvoicecat-fat.a`) bundling `libvoicecat.a` + all vcpkg static deps so the Swift Package links a single self-contained `.a` per slice.
- **iOS screen / system-audio sharing** is supported via a **ReplayKit Broadcast Upload
Extension** (the same mechanism Discord uses; triggered from Control Center's screen-record
button via `RPSystemBroadcastPickerView`). The extension receives
`RPSampleBufferType.audioApp` (system/app audio) and `.audioMic`. We capture **`.audioApp`**
for the `SCREEN_AUDIO` "listen together" stream and ignore video. The extension runs in a
**separate process with a ~50 MB memory cap** — that cap is a problem only for video
frames, so audio-only stays well within budget. It links a *minimal* slice of the core
(Opus encode + media send), shares the session/credentials with the host app through an
**App Group**, and re-derives its own media keys. This is detailed in [voice.md](voice.md) §9.
2026-06-17 00:52:02 +02:00
- **C# / Windows.** `[LibraryImport]` (source-generated P/Invoke, .NET 7+) over the C ABI.
`[UnmanagedCallersOnly]` static methods for `on_event`/`on_level` to avoid delegate-lifetime
pitfalls. UI in **WinForms (.NET 10)** — chosen over WinUI 3/Avalonia for its mature,
predictable screen-reader (NVDA/JAWS/Narrator) UIA support (see roadmap.md §2).
Events are delivered via `System.Threading.Channels.Channel<VoiceCatEvent>`, drained by a
30ms `System.Windows.Forms.Timer` on the UI thread — simpler than a message-only HWND +
`PostMessage` with no meaningful latency cost. `VoiceCatClientHandle : SafeHandle` wraps
the `vc_client*` and guarantees `vc_client_destroy` runs on GC/Dispose.
feat: external PCM feed/tap API (vc_stream_feed_pcm + vc_set_pcm_sink) Promotes vc_test_inject_capture (mono-only, TEST-ONLY) to a public, stereo-capable production API and adds a symmetric PCM tap on the receive side. Enables ReplayKit (iOS), ScreenCaptureKit (macOS), bots, soundboards, and custom clients — all without a hardware audio device. Core C++: - voicecat.h: new vc_stream_feed_pcm, vc_pcm_sink_cb typedef, vc_set_pcm_sink; vc_test_inject_capture kept as deprecated alias - audio_engine: stereo-aware inject_capture (channels param + ring reset on channel-count change); atomic pcm_sink_ fired per decoded frame in on_playback; RemoteStream carries user_id/stream_id for RT-safe sink metadata; init_recv_stream takes user_id+stream_id - client.cpp: stream_feed_pcm / set_pcm_sink implementations; sync_remote_streams passes user_id/stream_id to init_recv_stream - voicecat.cpp: trampolines + channels=1/2 validation Tests: test_external_pcm (headless, 3 sub-tests: mono round-trip, stereo feed L≠R, sink metadata+disable). ctest 23/23. Swift: feedPcm / setPcmSink in VoiceCatClient.swift + 4 XCTest smoke tests (ExternalPcmTests.swift). C#: StreamFeedPcm / SetPcmSink in VoiceCatClient.cs + NativeMethods.cs (vc_stream_feed_pcm unsafe P/Invoke, VcPcmSinkCallback delegate, vc_set_pcm_sink via nint) + 4 xUnit smoke tests (ExternalPcmTests.cs). Docs: architecture.md §4 new subsection, voice.md §9 updated (macOS/iOS now reference vc_stream_feed_pcm), protocol.md §8 explicit no-protocol-change note, roadmap.md M5 entry. Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-20 17:52:09 +02:00
### External PCM feed/tap
Two API functions let callers bypass miniaudio entirely for a stream:
| Function | Direction | Contract |
|----------|-----------|----------|
| `vc_stream_feed_pcm(c, stream_id, pcm, samples_per_channel, channels)` | **Send** — caller → network | Caller supplies interleaved int16 at the stream's sample rate (`channels` = 1 mono, 2 stereo). The core frames, Opus-encodes, AEAD-seals, and sends over UDP — identical wire path to hardware capture. The stream must already be started with `vc_stream_start`. Thread-safe; may be called from any thread (audio callback, ReplayKit delegate, SCStream callback). |
| `vc_set_pcm_sink(c, cb, user)` | **Receive** — network → caller | `cb` is called on the audio (playback) thread once per decoded Opus frame per remote stream, with `(user_id, stream_id, pcm, samples_per_channel, channels, sample_rate)`. PCM is delivered to the sink **and** the hardware device — dual output; the hardware mix is unaffected. Pass `cb=NULL` to disable (default). **Must not block** — copy what you need and return. |
`vc_test_inject_capture` (the old TEST-ONLY mono-only predecessor) is a deprecated alias
for `vc_stream_feed_pcm(..., channels=1)` — kept for source compatibility.
**Use cases:** ReplayKit Broadcast Extension (iOS `SCREEN_AUDIO`), ScreenCaptureKit (macOS
`SCREEN_AUDIO`), music/TTS/relay bots, soundboards, transcription clients. The extension or
bot links Opus + the feed entry point — no `ma_device`, no hardware, headless.
**Threading:** the feed path is thread-safe (ring buffer, no lock on the RT path). The sink
callback runs on the miniaudio playback thread — observe the same rules as the capture
callback: no allocations, no blocking calls.
## 5. Server architecture
`voicecat-server` is a headless process linking the core.
```
TCP/TLS 1.3 UDP + media AEAD
│ │
┌────────▼─────────┐ ┌─────────▼──────────┐
│ Connection mgr │ │ UDP demux │
│ (accept, TLS, │ │ 5-tuple → session │
│ per-conn state) │ │ anti-replay window │
└────────┬─────────┘ └─────────┬──────────┘
│ │
┌────────▼───────────────────────────────────▼──────────┐
│ Session registry (session_id ↔ TCP conn ↔ UDP tuple) │
└────────┬───────────────────────────────┬───────────────┘
│ │
┌────────▼─────────┐ ┌──────────────┐ ┌▼─────────────────┐
│ Channel manager │ │ Text router │ │ Voice router/SFU │
│ tree, configs, │ │ channel + PM │ │ relay Opus to │
│ membership, perms│ │ │ │ channel members │
└────────┬─────────┘ └──────────────┘ └──────────────────┘
┌────────▼─────────┐
│ Persistence │ accounts (Argon2id), channels, bans, config
│ SQLite │
└──────────────────┘
```
- **Voice router is a relay, not a mixer.** For each incoming voice frame it looks up the
sender's channel and forwards the *unmodified Opus payload* (restamped with the sender's
user id) to every other subscribed member. No server-side decode/transcode → low CPU,
low latency, and end-to-content is just Opus. Per-channel Opus params are enforced so all
members are mutually decodable.
- **Subscriptions.** Clients implicitly subscribe to their current channel's voice; text
and presence can be subscribed more broadly. This keeps fan-out bounded on big servers.
- **Stateless-ish media.** UDP carries no auth per packet beyond the media-AEAD session;
the 5-tuple→session binding is established once via a token (see protocol.md §4).
fix(net): broadcast LEFT on disconnect, add keepalive/reaper, cap PLC hiss Three reported bugs traced to one root cause plus two missing designed features: 1. Stale users + eternal PLC hiss (root cause): ConnSession::close() silently erased dropped users without broadcasting UserEvent::LEFT, so peers never learned the user left and their audio engines never called remove_stream — Opus PLC synthesized comfort noise forever. Fix: broadcast_left() helper + close() broadcasts LEFT before erasing. 2. PLC cap (defense-in-depth): on_playback now caps pure PLC at ~2s, then emits digital silence so a stale stream can never hiss forever even if remove_stream is skipped. Resets automatically on fresh packets. 3. No timeout / no ping: client never sent Ping, server had no last_seen / reaper, so half-open connections (NAT timeout, wifi loss, sleep) left ghost users forever. Fix: client Ping every 15s with RTT measurement, ConnSession::last_seen bumped on every inbound TCP/UDP frame, steady_timer reaper sweeps every 15s and drops sessions older than 45s (configurable via server::Config). 4. UDP KEEPALIVE: client sends plaintext kFrameKeepalive every 5s; server bumps last_seen + echoes back. Keeps NAT bindings alive and lets media activity defer the reaper independently of TCP. 5. Graceful client disconnect: vc_disconnect() sends Disconnect{code=0} via a flag-based io-thread exit (no double-close race); server handles client-sent Disconnect with immediate close() + LEFT broadcast. 3 new tests: disconnect_left, plc_cap, reaper_timeout. 21/21 ctest green. Docs: protocol.md §6/§7, voice.md §6, architecture.md §5, PROGRESS.md.
2026-06-18 01:18:33 +02:00
- **Keepalive reaper.** An `asio::steady_timer` sweeps every 15 s and drops any session
whose `last_seen` (bumped on every inbound TCP or UDP frame) is older than 45 s. Each
drop broadcasts `UserEvent::LEFT` so peers clean up immediately. This catches half-open
connections that never produce a TCP EOF. Configurable via `server::Config`.
- **Single process, scalable later.** v1 is one process, one machine. The session registry
and router are written behind interfaces so a future build can sit them behind a shared
bus for multi-node, but that is explicitly out of scope for now.
## 6. Repository layout (proposed)
```
voice-cat/
├── docs/ # this folder
├── core/ # libvoicecat (C++)
│ ├── include/voicecat.h # the C ABI
│ ├── src/{net,crypto,codec,protocol,session,audio}/
│ └── proto/ # .proto definitions (shared source of truth)
├── server/ # voicecat-server (C++, links core)
├── clients/
│ ├── apple/ # Swift package + Xcode project (macOS + iOS)
│ └── windows/ # .NET solution (C#)
├── tools/
│ └── vccli/ # headless test client (C++), for protocol bring-up
├── third_party/ # vendored / vcpkg manifest
└── CMakeLists.txt
```
Build is **CMake** with **vcpkg** (manifest mode) for C/C++ deps; the Apple and Windows UI
projects consume the built core as a binary + headers. See [tech-stack.md](tech-stack.md).