voice-cat/docs/voice.md

# Voice & Media

Real-time audio runs over **UDP**, secured per [security.md](security.md). The control
channel (TCP/TLS) handles *signaling* — announcing streams, channel membership, talk state
— while UDP carries only the encoded audio frames. This split keeps media latency low and
independent of TCP head-of-line blocking.

## 1. The multi-stream model

A **user** publishes one or more **streams**. Each stream is an independent audio source
with its own encoder, its own `stream_id` (unique per user) and `ssrc` (media-plane id
assigned by the server), and is independently mutable/mutable at the receiver.

```
        User "Alex"                              Receiver "Sam"
   ┌────────────────────┐                   ┌──────────────────────────┐
   │ mic       → enc ───┼──ssrc 1001──▶     │ jitter(1001)→dec→┐        │
   │ desktop   → enc ───┼──ssrc 1002──▶     │ jitter(1002)→dec→┤        │
   │ 2nd mic   → enc ───┼──ssrc 1003──▶     │ jitter(1003)→dec→┴─mix──▶ out
   └────────────────────┘                   └──────────────────────────┘
```

Stream **kinds** (v1): `MIC`, `SCREEN_AUDIO` (system/desktop audio for listening together),
`AUX_DEVICE` (a second capture device). Receivers can set, per incoming stream: **gain**,
**mute**, and **noise reduction** (see §10) — so Sam can turn down Alex's desktop audio
while keeping the mic, *and* independently apply noise suppression to a third user who has a
loud fan. The mixer sums all active streams from all users in the channel into the local
playback device. All of these receiver-side controls are **local to the listener** and carry
no protocol traffic.

`SCREEN_AUDIO` capture is platform-specific and covered in §9 — it is supported on Windows,
macOS, **and iOS** (via a ReplayKit broadcast extension).

## 2. Voice frame format (UDP payload, inside the media AEAD)

A fixed binary header — no protobuf on the RT path. Multi-byte fields are big-endian.

```
 0      1      2      3      4      5      6      7      8 ...
┌──────┬──────┬──────┬──────┬──────┬──────┬──────┬──────┬───────────────┐
│ type │flags │  codec      │     ssrc (u32)                            │
├──────┴──────┴──────┴──────┼──────┬──────┬──────┬──────┬──────────────┤
│  seq (u16)  │       timestamp (u32, in samples @48k)   │  payload ... │
└─────────────┴──────────────────────────────────────────┴──────────────┘

type    u8   1 = VOICE, 2 = KEEPALIVE, 3 = UDP_BINDING (handshake)
flags   u8   bit0 marker (start of talkspurt) · bit1 FEC-present
             bit2 DTX/comfort-noise · bit3 last-frame-before-stop
codec   u16  0 = OPUS  (room for future codecs)
ssrc    u32  media-plane stream id. Client sends its own ssrc; the server
             validates it against the bound session and relays unchanged.
seq     u16  per-ssrc sequence number, wraps; drives loss detection + reorder
timestamp u32 RTP-style sample clock @48 kHz; drives the jitter buffer
payload      one Opus packet (the encoder's output for one frame)
```

This is intentionally RTP-shaped (familiar semantics: ssrc/seq/timestamp) without RTP's
full machinery. The **server relays the payload unmodified** — it only reads the header to
route by ssrc→channel and may restamp nothing (the client's ssrc is globally unique once
assigned at `StreamAnnounce`). No server-side decode.

### Why client-sends-ssrc is safe

The UDP 5-tuple is bound to an authenticated session (protocol.md §4). The server checks
that the ssrc in each frame belongs to a stream that session announced; spoofed ssrcs are
dropped. So identity is anchored by the session binding + transport encryption, not by
trusting the header.

## 3. Per-channel audio configuration

Opus is configured **per channel** and pushed to clients in `JoinChannelResult.audio` /
`StreamAnnounceResult.effective_audio`. All members of a channel encode with mutually
decodable parameters.

```proto
message AudioConfig {
  uint32 codec = 1;               // 0 = OPUS
  ChannelMode mode = 2;           // MONO / STEREO
  uint32 sample_rate = 3;         // 8000/12000/16000/24000/48000 (48000 recommended)
  uint32 bitrate_bps = 4;         // e.g. 24000 (speech) … 128000 (music/stereo)
  uint32 frame_ms = 5;            // 2.5/5/10/20/40/60 (20 default)
  OpusApplication application = 6;// VOIP / AUDIO / LOWDELAY
  bool   fec = 7;                 // in-band forward error correction
  uint32 expected_packet_loss = 8;// %, tunes FEC aggressiveness
  bool   dtx = 9;                 // discontinuous transmission (silence suppression)
  uint32 complexity = 10;         // 0..10 encoder complexity
}
```

Guidance baked into defaults / docs:

- **Sample rate: always run Opus at 48 kHz internally.** Opus resamples internally anyway;
  48 kHz avoids surprises. The `sample_rate` field mainly constrains capture/narrowband
  modes for very low bitrate channels. Default **48000**.
- **Frame size: 20 ms default.** Smaller (10 ms) lowers latency at the cost of more
  per-packet overhead and CPU; larger (40/60 ms) improves efficiency and loss resilience at
  the cost of latency. Expose it per channel for "low-latency talk" vs "stable music" rooms.
- **Mode/bitrate:** speech channels → `MONO`, `VOIP`, 24–32 kbps, DTX on, FEC on.
  Music/screen-audio channels → `STEREO`, `AUDIO`, 96–128 kbps, DTX off, FEC optional.
- **`application`:** `VOIP` for talk, `AUDIO` for music/screen-share, `LOWDELAY` for
  monitoring use cases.

## 4. Packet-loss resilience (Opus 1.6)

Layered, all configurable per channel:

1. **In-band FEC** — Opus embeds a low-bitrate copy of the previous frame; the decoder
   recovers a lost packet from the *next* one (costs one frame of latency on recovery).
   Tuned by `expected_packet_loss`.
2. **PLC (packet loss concealment)** — decoder synthesizes a plausible frame for an
   unrecovered loss; always on, free.
3. **DTX** — sender stops transmitting during silence and sends sparse comfort-noise
   updates; cuts bandwidth and is bandwidth-friendly on busy channels.
4. **DRED (Deep REDundancy, optional/feature-gated)** — Opus 1.6's ML redundancy carries
   acoustic features so the decoder can reconstruct longer loss bursts. Heavier CPU; gate
   behind a `features` flag and per-channel toggle, off by default.

## 5. Jitter buffer

Each receiver keeps an **adaptive jitter buffer per ssrc**.

- Frames are inserted by `timestamp`; playback reads in order at the device callback rate.
- Target depth adapts to observed network jitter between a configurable **min/max latency**
  (channel-level "stability vs latency" knob). A "low-latency" channel runs a shallow
  buffer; a "stable" channel runs deeper.
- Late frames past the playout point are dropped; gaps are filled by FEC (if the next frame
  arrived) or PLC.
- The `marker` flag (start of talkspurt) lets the buffer resynchronize cleanly after
  silence/DTX without accumulating drift.

```
 incoming (out of order) ──▶ [ reorder by ts | adaptive depth ] ──▶ Opus decode ──▶ mixer
                                     ▲
                             jitter estimate feeds depth
```

## 6. UDP keepalive & NAT

- A `KEEPALIVE` (type 2) frame flows both directions on the media channel every ~5 s to
  hold NAT bindings and measure media-path RTT/loss independent of TCP. The frame is
  plaintext (14-byte header, no payload, no AEAD) — the server identifies the sender by
  its already-verified UDP endpoint (established during the `UdpBinding` handshake). On
  receipt the server bumps the sender's `last_seen` (so media activity defers the TCP
  reaper independently of control-channel traffic) and echoes the frame back so the
  client can measure media-path RTT.
- If the media path dies but TCP is alive, the client surfaces a "voice disconnected"
  state and attempts UDP re-binding (re-derive media keys + fresh `UdpBinding`) without dropping
  the control session.
- No ICE/STUN/TURN. The expectation matches TeamSpeak/Mumble: the **server** is reachable
  (public IP or port-forward); **clients** sit behind NAT and initiate, so their bindings
  are created by their outbound first packet.

## 7. Talk-state signaling

"Who is talking" can be derived two ways; we use both:

- **Implicit:** presence of recent voice frames for an ssrc → that stream is "active". The
  receiver drives talk indicators from the jitter buffer, so they're accurate and need no
  extra messages.
- **Explicit (optional):** `StreamStateUpdate` on TCP for coarse UI state (muted, hold) and
  for users not currently subscribed to the media. Server-side mute/deafen is authoritative
  and always signaled on TCP.

## 8. Capture/playback pipeline (inside the core)

```
 mic device ─(miniaudio capture, 48k, mono)→ resample? → send-side VAD/PTT gate
        → Opus encode → frame header → AEAD → UDP send

 screen audio ─(WASAPI loopback, 48k, mono or stereo per channel mode)→ Opus encode
        → frame header → AEAD → UDP send

 UDP recv → AEAD open → parse header → jitter(ssrc) → Opus decode
         → per-stream recv-side NS (optional, per user) → per-stream gain/mute
         → mixer (sum all ssrc, stereo; mono streams upmixed L=R) → (miniaudio playback,
           48k, stereo) → device
```

- Capture and playback run on miniaudio's real-time callbacks (WASAPI / CoreAudio / ALSA).
  Playback is genuinely stereo end-to-end. **Mic capture stays mono** (no stereo mic in v1);
  a mono mic frame on a stereo channel is upmixed L=R before encoding so the Opus bitstream
  is still spec-correct stereo. **Screen-audio (`SCREEN_AUDIO`) loopback** captures in the
  channel's mode — stereo when the channel is stereo (real interleaved L/R, no downmix), mono
  when the channel is mono — so a stereo music/screen-share channel gets genuine stereo
  end-to-end. See §9 for the platform-specific loopback mechanism.
- **DSP engine: see §11.** The original plan was `webrtc-audio-processing` (AEC + NS + AGC +
  VAD in one tuned module, BSD-licensed) — but it has no working Windows/MSVC build upstream
  (confirmed via its own issue tracker: GCC-only Meson build, MinGW support unfinished, hard
  `abseil-cpp` dependency, Linux-tested only —
  [gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing#1](https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/issues/1)).
  v1 ships a lightweight, dependency-free energy/RMS VAD instead (§11); there is **no AEC, NS,
  or AGC implementation at all yet** — not just a deferred VAD, the whole APM is unbuilt. Real
  `webrtc-audio-processing` stays a tracked future swap, behind the same `ApmProcessor`
  interface (`core/src/audio/apm_processor.h`), revisit if/when a Linux build target exists or
  upstream Windows support matures.
- The mixer sums decoded streams; clipping is handled by soft limiting on the master bus.

## 10. Noise reduction — two-sided

Noise reduction can be applied **at the sender, at the listener, or both** — they are
independent.

- **Sender-side** (the talker's choice): the publishing client runs APM noise suppression on
  its mic before encoding, controlled by that user's own settings. This cleans the signal for
  *everyone* and saves bitrate.
- **Listener-side, per user** (the listener's choice): on the receive path, *after* decoding
  each stream and *before* mixing, the listener can enable an **additional** NS pass on a
  **specific** sender's stream. So even if Alex chose not to denoise his mic, Sam can locally
  suppress Alex's background noise without affecting how anyone else hears Alex.

Implementation: a per-`ssrc` APM NS instance on the receive path, instantiated lazily only
for streams the listener has flagged. State lives entirely on the listener's machine; toggling
it is a local UI action with **no protocol message** and no effect on other listeners. Because
each receive stream is decoded independently before the mixer (voice.md §1), per-user receive
NS is a clean drop-in on that per-stream stage.

## 11. Input activation — VAD and PTT (client-configurable)

Whether the mic transmits is decided locally by the **input gate**, and the client supports
**both** modes, switchable per client (`vc_set_input_mode`):

- **Voice activation (VAD):** v1 implements this as a lightweight, dependency-free
  energy/RMS-threshold VAD (`EnergyVadProcessor`, `core/src/audio/apm_processor.cpp`) — no
  external DSP dependency, since real `webrtc-audio-processing` has no working Windows/MSVC
  build (see §8). It opens the gate when a frame's RMS exceeds a configurable threshold
  (default ~0.025, normalized to int16 range), with a configurable hang-time (default 300 ms,
  matching the talk-indicator hangover so "talking" and "gate open" agree) to avoid clipping
  word tails. DTX naturally complements this — when the gate is closed nothing (or only
  comfort noise) is sent. This implementation has **no AEC** — a real limitation versus the
  originally-planned APM, not just a deferred VAD.
- **Push-to-talk (PTT):** `vc_set_push_to_talk(active)` opens/closes the gate directly. The UI
  exposes a configurable keybind; the core just receives gate open/close.

Gating applies to the **MIC stream only** — `SCREEN_AUDIO`/`AUX_DEVICE` always bypass it
(gating a desktop-audio share on the user's own voice activity would silently drop shared
music/video audio whenever the user isn't talking, which defeats the feature).

This is purely a send-side, client-local concern — it gates what gets encoded and sent. It
needs **no protocol support**; remote talk indicators are still derived from the presence of
received frames (§7), so they work identically under VAD or PTT.

## 9. System / screen audio capture (`SCREEN_AUDIO`)

"Listen together" needs to capture the audio another app is playing. The capture mechanism
differs per OS, but it always feeds the **same** Opus-encode → media-AEAD → UDP path as a
normal stream; only the *source* is platform-specific.

| Platform | Mechanism | Notes |
|----------|-----------|-------|
| **Windows** | **WASAPI loopback** capture of the default render endpoint (via miniaudio's loopback mode) | **Implemented.** Captures in the channel's mode — stereo (interleaved L/R) when the channel is stereo, mono when the channel is mono — so a stereo music/screen-share channel gets genuine stereo end-to-end (no downmix). Whole-device capture, not process-specific — it inherently captures this app's own incoming voice mix along with everything else playing (an accepted self-echo-loop characteristic of desktop-audio capture, not a bug). Windows 10 2004+'s process-specific loopback (`AUDIOCLIENT_ACTIVATION_PARAMS`) would avoid this but miniaudio doesn't expose it — a future enhancement. |
| **macOS** | **ScreenCaptureKit** system-audio capture (macOS 13+), or a virtual audio device fallback on older OSes | OS requires screen-recording permission; capture happens in the main app. |
| **iOS** | **ReplayKit Broadcast Upload Extension** (the Discord mechanism) | See below — separate process, App Group, ~50 MB cap (fine for audio-only). |

### iOS detail

- The user starts a broadcast from Control Center's screen-record button; we surface it via
  `RPSystemBroadcastPickerView` from inside the app for one-tap start.
- The **Broadcast Upload Extension** receives `RPSampleBufferType.audioApp` (system/app
  audio) and `.audioMic`. We consume **`.audioApp`** for `SCREEN_AUDIO` and drop the video
  buffers entirely — video is what blows the **~50 MB** extension memory budget, so an
  audio-only consumer stays comfortably inside it.
- The extension is a *separate process*. It links a **minimal slice of the core** (Opus
  encode + media send only — not the full client), reads the active session token and
  server endpoint from a shared **App Group** container that the host app wrote at join
  time, derives its own media keys, and publishes the `SCREEN_AUDIO` stream directly. The
  host app announces the stream over its control channel (`StreamAnnounce`) so the server and
  peers learn about it.
- Mic + voice continue to run in the **host app**; only the system-audio share lives in the
  extension. When the broadcast stops (`broadcastFinished`), the extension sends a final
  frame with the `last-frame-before-stop` flag and the host app emits `StreamStop`.