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.
225 lines
16 KiB
Markdown
225 lines
16 KiB
Markdown
# Architecture
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## 1. The shared-core model
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All non-UI logic lives in one C++ library, **`libvoicecat`**. The same library is linked
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into every client and into the server. Platform UIs are thin and call the core through a
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stable **C ABI** (`voicecat.h`).
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```
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┌───────────────────────────────────────────┐
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macOS / iOS (Swift) │ │ Windows (C#)
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┌──────────────────┐ │ libvoicecat (C++) │ ┌──────────────────┐
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│ SwiftUI views │ │ ┌─────────────────────────────────────┐ │ │ WinForms (.NET 10│
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│ AVAudioSession │──┼─▶│ C ABI (voicecat.h) │◀─┼──│ LibraryImport │
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│ Swift↔C++ interop│ │ ├─────────────────────────────────────┤ │ │ P/Invoke │
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└──────────────────┘ │ │ Session / Protocol state machine │ │ └──────────────────┘
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│ │ Text + voice signaling │ │
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Linux/macOS/Windows │ │ Audio engine: capture→encode→send, │ │
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server │ │ recv→jitter→decode→mix→playback │ │
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┌──────────────────┐ │ │ Codec layer (Opus 1.6) │ │
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│ voicecat-server │──┼─▶│ Crypto + transport (TLS 1.3 + AEAD) │ │
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│ (reuses core) │ │ │ Net I/O (Asio: TCP + UDP + timers) │ │
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└──────────────────┘ │ └─────────────────────────────────────┘ │
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└───────────────────────────────────────────┘
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```
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Why this shape:
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- **Swift** (5.9+) can import C++ directly, but we still ship a C ABI because it is the
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lowest-friction, most stable boundary and it is what **C#** needs (`LibraryImport`/
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P/Invoke). One ABI serves both.
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- The **server** is not a separate codebase. It links the same protocol, crypto, and Opus
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code as the client, so framing/encryption can never drift between the two ends.
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## 2. Layered design inside the core
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From the OS up:
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| Layer | Responsibility | Key deps |
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|-------|----------------|----------|
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| **Platform I/O** | Sockets, timers; audio device capture/playback | Asio, miniaudio |
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| **Transport** | TLS 1.3 (TCP), exported-key ChaCha20-Poly1305 AEAD (UDP), framing, anti-replay | mbedTLS, libsodium |
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| **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 |
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| **Protocol** | Message (de)serialization, request/response correlation, state machine | protobuf |
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| **Session/domain** | Channels, users, streams, permissions, text routing | — |
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| **C ABI façade** | Handle-based API + event callbacks exposed to UIs | — |
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A UI never sees a socket, an Opus packet, or a protobuf message. It sees: "connect",
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"join channel", "start a stream from this device", "send this text", and a stream of
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events ("user joined", "user is talking", "message received", "level meter = 0.4").
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## 3. Threading model
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Three classes of thread, with strict rules.
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```
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┌──────────────┐ lock-free ┌──────────────┐ lock-free ┌──────────────┐
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│ Audio capture│ ──ring buffer─▶│ Net thread │ ──ring buffer─▶│Audio playback│
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│ (RT, miniaudio│ │ (Asio loop) │ │ (RT, miniaudio│
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│ callback) │◀───ring buffer─│ │◀───ring buffer─│ callback) │
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│ capture→Opus │ │ TLS + AEAD, │ │ jitter→Opus │
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│ encode │ │ route, relay │ │ decode→mix │
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└──────────────┘ └──────────────┘ └──────────────┘
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│
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┌─────▼──────┐
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│ Worker pool│ DB, Argon2id, file I/O,
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│ (blocking) │ TLS handshakes, codec setup
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└────────────┘
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```
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Rules:
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- **Audio (real-time) threads** are driven by the OS audio callback. They must not
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allocate, lock, log, or do syscalls beyond the ring-buffer hand-off. Opus encode/decode
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runs here (it is allocation-free after init).
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- **Net thread(s)** run the Asio event loop: TLS records, AEAD seal/open, protobuf parse, channel
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routing, jitter-buffer feed. On the server, this is where the SFU relay copies packets
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to subscribers.
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- **Worker pool** absorbs anything that can block: SQLite, Argon2id verification, DNS,
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TLS handshake CPU, codec (re)configuration.
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- **Communication** between audio and net is single-producer/single-consumer lock-free
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ring buffers (one per direction per stream). Control-plane events to the UI go through a
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thread-safe queue drained on the UI's terms.
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## 4. The C ABI (`voicecat.h`) — shape
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Handle-based, opaque pointers, C-linkage. Illustrative (final names in implementation):
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```c
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typedef struct vc_client vc_client;
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typedef struct {
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void (*on_event)(void* user, const vc_event* ev); // state changes, messages
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void (*on_level)(void* user, uint32_t stream_id, float rms); // meters (throttled)
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void* user;
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} vc_callbacks;
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vc_client* vc_client_create(const vc_config* cfg, vc_callbacks cb);
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void vc_client_destroy(vc_client*);
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int vc_connect(vc_client*, const char* host, uint16_t port); // async; result via event
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int vc_authenticate_guest(vc_client*, const char* nickname);
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int vc_authenticate_user(vc_client*, const char* user, const char* password);
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int vc_join_channel(vc_client*, uint32_t channel_id, const char* password /*nullable*/);
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int vc_leave_channel(vc_client*);
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// Streams (mic / screen audio / aux device)
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int vc_stream_start(vc_client*, const vc_stream_desc* desc, uint32_t* out_stream_id);
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int vc_stream_stop(vc_client*, uint32_t stream_id);
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int vc_set_input_device(vc_client*, uint32_t stream_id, const char* device_id);
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int vc_set_self_mute(vc_client*, bool mic_muted, bool deafened);
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// Text
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int vc_send_text(vc_client*, vc_text_scope scope, uint32_t target_id, const char* utf8);
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// Enumeration helpers for UI device pickers
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int vc_list_devices(vc_client*, vc_device_kind kind, vc_device_list* out);
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```
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Design notes:
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- **Async, event-driven.** Calls return immediately; results and state changes arrive via
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`on_event`. This maps cleanly onto SwiftUI/`async` and C# `event`/`Task` patterns.
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- **The core owns audio.** Capture, encode, decode, mixing, and playback happen inside the
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core via miniaudio. The UI only *selects devices*, *starts/stops streams*, and *renders
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meters/state*. This keeps the real-time path identical on every OS. (iOS is the one
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exception that needs UI-side cooperation — see below.)
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- **Device enumeration works pre-connect.** `vc_list_devices` needs no live session — device
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pickers can populate before `vc_connect`. `vc_device.id` is an opaque, internally-encoded
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handle (currently a hex-encoded `ma_device_id`) — always round-trip an id that came from
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`vc_list_devices`/`vc_get_stream_audio_config`; never construct one by hand. Tolerate an
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empty list (a machine can legitimately have zero input or output devices).
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- **Strings are UTF-8 `const char*`; ownership is explicit.** Output buffers are
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caller-allocated or returned with a paired `vc_free`.
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### Per-platform binding notes
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- **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.
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- **iOS screen / system-audio sharing** is supported via a **ReplayKit Broadcast Upload
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Extension** (the same mechanism Discord uses; triggered from Control Center's screen-record
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button via `RPSystemBroadcastPickerView`). The extension receives
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`RPSampleBufferType.audioApp` (system/app audio) and `.audioMic`. We capture **`.audioApp`**
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for the `SCREEN_AUDIO` "listen together" stream and ignore video. The extension runs in a
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**separate process with a ~50 MB memory cap** — that cap is a problem only for video
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frames, so audio-only stays well within budget. It links a *minimal* slice of the core
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(Opus encode + media send), shares the session/credentials with the host app through an
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**App Group**, and re-derives its own media keys. This is detailed in [voice.md](voice.md) §9.
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- **C# / Windows.** `[LibraryImport]` (source-generated P/Invoke, .NET 7+) over the C ABI.
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`[UnmanagedCallersOnly]` static methods for `on_event`/`on_level` to avoid delegate-lifetime
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pitfalls. UI in **WinForms (.NET 10)** — chosen over WinUI 3/Avalonia for its mature,
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predictable screen-reader (NVDA/JAWS/Narrator) UIA support (see roadmap.md §2).
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Events are delivered via `System.Threading.Channels.Channel<VoiceCatEvent>`, drained by a
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30ms `System.Windows.Forms.Timer` on the UI thread — simpler than a message-only HWND +
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`PostMessage` with no meaningful latency cost. `VoiceCatClientHandle : SafeHandle` wraps
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the `vc_client*` and guarantees `vc_client_destroy` runs on GC/Dispose.
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## 5. Server architecture
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`voicecat-server` is a headless process linking the core.
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```
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TCP/TLS 1.3 UDP + media AEAD
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│ │
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┌────────▼─────────┐ ┌─────────▼──────────┐
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│ Connection mgr │ │ UDP demux │
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│ (accept, TLS, │ │ 5-tuple → session │
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│ per-conn state) │ │ anti-replay window │
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└────────┬─────────┘ └─────────┬──────────┘
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│ │
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┌────────▼───────────────────────────────────▼──────────┐
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│ Session registry (session_id ↔ TCP conn ↔ UDP tuple) │
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└────────┬───────────────────────────────┬───────────────┘
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│ │
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┌────────▼─────────┐ ┌──────────────┐ ┌▼─────────────────┐
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│ Channel manager │ │ Text router │ │ Voice router/SFU │
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│ tree, configs, │ │ channel + PM │ │ relay Opus to │
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│ membership, perms│ │ │ │ channel members │
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└────────┬─────────┘ └──────────────┘ └──────────────────┘
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│
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┌────────▼─────────┐
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│ Persistence │ accounts (Argon2id), channels, bans, config
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│ SQLite │
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└──────────────────┘
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```
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- **Voice router is a relay, not a mixer.** For each incoming voice frame it looks up the
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sender's channel and forwards the *unmodified Opus payload* (restamped with the sender's
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user id) to every other subscribed member. No server-side decode/transcode → low CPU,
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low latency, and end-to-content is just Opus. Per-channel Opus params are enforced so all
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members are mutually decodable.
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- **Subscriptions.** Clients implicitly subscribe to their current channel's voice; text
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and presence can be subscribed more broadly. This keeps fan-out bounded on big servers.
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- **Stateless-ish media.** UDP carries no auth per packet beyond the media-AEAD session;
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the 5-tuple→session binding is established once via a token (see protocol.md §4).
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- **Keepalive reaper.** An `asio::steady_timer` sweeps every 15 s and drops any session
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whose `last_seen` (bumped on every inbound TCP or UDP frame) is older than 45 s. Each
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drop broadcasts `UserEvent::LEFT` so peers clean up immediately. This catches half-open
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connections that never produce a TCP EOF. Configurable via `server::Config`.
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- **Single process, scalable later.** v1 is one process, one machine. The session registry
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and router are written behind interfaces so a future build can sit them behind a shared
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bus for multi-node, but that is explicitly out of scope for now.
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## 6. Repository layout (proposed)
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```
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voice-cat/
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├── docs/ # this folder
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├── core/ # libvoicecat (C++)
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│ ├── include/voicecat.h # the C ABI
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│ ├── src/{net,crypto,codec,protocol,session,audio}/
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│ └── proto/ # .proto definitions (shared source of truth)
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├── server/ # voicecat-server (C++, links core)
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├── clients/
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│ ├── apple/ # Swift package + Xcode project (macOS + iOS)
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│ └── windows/ # .NET solution (C#)
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├── tools/
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│ └── vccli/ # headless test client (C++), for protocol bring-up
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├── third_party/ # vendored / vcpkg manifest
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└── CMakeLists.txt
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```
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Build is **CMake** with **vcpkg** (manifest mode) for C/C++ deps; the Apple and Windows UI
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projects consume the built core as a binary + headers. See [tech-stack.md](tech-stack.md).
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