mirror of
https://github.com/affaan-m/everything-claude-code.git
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* Introduces /aside — a mid-task side conversation command inspired by
Claude Code's native /btw feature. Allows users to ask a question while
Claude is actively working without losing task context or touching any files.
Key behaviors:
- Freezes current task state before answering (read-only during aside)
- Delivers answers in a consistent ASIDE / Back to task format
- Auto-resumes the active task after answering
- Handles edge cases: no question given, answer reveals a blocker,
question implies a task redirect, chained asides, ambiguous questions,
and answers that suggest code changes without making them
* Two documentation inconsistencies fixed:
* Fixed 4 pre-existing lint errors in skills/videodb/ that were causing CI to fail across all PR checks: - api-reference.md: add blockquote continuation line to fix MD028 - capture-reference.md: wrap bare URL to fix MD034 - SKILL.md: wrap bare URL to fix MD034
408 lines
12 KiB
Markdown
408 lines
12 KiB
Markdown
# Capture Reference
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Code-level details for VideoDB capture sessions. For workflow guide, see [capture.md](capture.md).
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---
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## WebSocket Events
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Real-time events from capture sessions and AI pipelines. No webhooks or polling required.
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Use [scripts/ws_listener.py](../scripts/ws_listener.py) to connect and dump events to `${VIDEODB_EVENTS_DIR:-$HOME/.local/state/videodb}/videodb_events.jsonl`.
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### Event Channels
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| Channel | Source | Content |
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|---------|--------|---------|
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| `capture_session` | Session lifecycle | Status changes |
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| `transcript` | `start_transcript()` | Speech-to-text |
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| `visual_index` / `scene_index` | `index_visuals()` | Visual analysis |
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| `audio_index` | `index_audio()` | Audio analysis |
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| `alert` | `create_alert()` | Alert notifications |
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### Session Lifecycle Events
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| Event | Status | Key Data |
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|-------|--------|----------|
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| `capture_session.created` | `created` | — |
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| `capture_session.starting` | `starting` | — |
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| `capture_session.active` | `active` | `rtstreams[]` |
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| `capture_session.stopping` | `stopping` | — |
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| `capture_session.stopped` | `stopped` | — |
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| `capture_session.exported` | `exported` | `exported_video_id`, `stream_url`, `player_url` |
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| `capture_session.failed` | `failed` | `error` |
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### Event Structures
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**Transcript event:**
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```json
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{
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"channel": "transcript",
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"rtstream_id": "rts-xxx",
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"rtstream_name": "mic:default",
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"data": {
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"text": "Let's schedule the meeting for Thursday",
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"is_final": true,
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"start": 1710000001234,
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"end": 1710000002345
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}
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}
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```
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**Visual index event:**
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```json
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{
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"channel": "visual_index",
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"rtstream_id": "rts-xxx",
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"rtstream_name": "display:1",
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"data": {
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"text": "User is viewing a Slack conversation with 3 unread messages",
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"start": 1710000012340,
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"end": 1710000018900
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}
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}
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```
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**Audio index event:**
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```json
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{
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"channel": "audio_index",
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"rtstream_id": "rts-xxx",
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"rtstream_name": "mic:default",
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"data": {
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"text": "Discussion about scheduling a team meeting",
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"start": 1710000021500,
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"end": 1710000029200
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}
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}
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```
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**Session active event:**
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```json
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{
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"event": "capture_session.active",
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"capture_session_id": "cap-xxx",
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"status": "active",
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"data": {
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"rtstreams": [
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{ "rtstream_id": "rts-1", "name": "mic:default", "media_types": ["audio"] },
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{ "rtstream_id": "rts-2", "name": "system_audio:default", "media_types": ["audio"] },
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{ "rtstream_id": "rts-3", "name": "display:1", "media_types": ["video"] }
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]
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}
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}
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```
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**Session exported event:**
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```json
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{
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"event": "capture_session.exported",
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"capture_session_id": "cap-xxx",
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"status": "exported",
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"data": {
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"exported_video_id": "v_xyz789",
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"stream_url": "https://stream.videodb.io/...",
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"player_url": "https://console.videodb.io/player?url=..."
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}
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}
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```
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> For latest details, see [VideoDB Realtime Context docs](https://docs.videodb.io/pages/ingest/capture-sdks/realtime-context.md).
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---
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## Event Persistence
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Use `ws_listener.py` to dump all WebSocket events to a JSONL file for later analysis.
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### Start Listener and Get WebSocket ID
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```bash
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# Start with --clear to clear old events (recommended for new sessions)
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python scripts/ws_listener.py --clear &
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# Append to existing events (for reconnects)
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python scripts/ws_listener.py &
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```
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Or specify a custom output directory:
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```bash
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python scripts/ws_listener.py --clear /path/to/output &
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# Or via environment variable:
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VIDEODB_EVENTS_DIR=/path/to/output python scripts/ws_listener.py --clear &
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```
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The script outputs `WS_ID=<connection_id>` on the first line, then listens indefinitely.
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**Get the ws_id:**
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```bash
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cat "${VIDEODB_EVENTS_DIR:-$HOME/.local/state/videodb}/videodb_ws_id"
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```
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**Stop the listener:**
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```bash
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kill "$(cat "${VIDEODB_EVENTS_DIR:-$HOME/.local/state/videodb}/videodb_ws_pid")"
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```
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**Functions that accept `ws_connection_id`:**
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| Function | Purpose |
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|----------|---------|
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| `conn.create_capture_session()` | Session lifecycle events |
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| RTStream methods | See [rtstream-reference.md](rtstream-reference.md) |
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**Output files** (in output directory, default `${XDG_STATE_HOME:-$HOME/.local/state}/videodb`):
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- `videodb_ws_id` - WebSocket connection ID
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- `videodb_events.jsonl` - All events
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- `videodb_ws_pid` - Process ID for easy termination
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**Features:**
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- `--clear` flag to clear events file on start (use for new sessions)
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- Auto-reconnect with exponential backoff on connection drops
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- Graceful shutdown on SIGINT/SIGTERM
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- Connection status logging
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### JSONL Format
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Each line is a JSON object with added timestamps:
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```json
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{"ts": "2026-03-02T10:15:30.123Z", "unix_ts": 1772446530.123, "channel": "visual_index", "data": {"text": "..."}}
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{"ts": "2026-03-02T10:15:31.456Z", "unix_ts": 1772446531.456, "event": "capture_session.active", "capture_session_id": "cap-xxx"}
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```
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### Reading Events
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```python
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import json
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import time
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from pathlib import Path
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events_path = Path.home() / ".local" / "state" / "videodb" / "videodb_events.jsonl"
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transcripts = []
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recent = []
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visual = []
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cutoff = time.time() - 600
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with events_path.open(encoding="utf-8") as handle:
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for line in handle:
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event = json.loads(line)
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if event.get("channel") == "transcript":
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transcripts.append(event)
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if event.get("unix_ts", 0) > cutoff:
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recent.append(event)
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if (
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event.get("channel") == "visual_index"
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and "code" in event.get("data", {}).get("text", "").lower()
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):
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visual.append(event)
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```
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---
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## WebSocket Connection
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Connect to receive real-time AI results from transcription and indexing pipelines.
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```python
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ws_wrapper = conn.connect_websocket()
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ws = await ws_wrapper.connect()
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ws_id = ws.connection_id
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```
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| Property / Method | Type | Description |
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|-------------------|------|-------------|
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| `ws.connection_id` | `str` | Unique connection ID (pass to AI pipeline methods) |
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| `ws.receive()` | `AsyncIterator[dict]` | Async iterator yielding real-time messages |
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---
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## CaptureSession
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### Connection Methods
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| Method | Returns | Description |
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|--------|---------|-------------|
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| `conn.create_capture_session(end_user_id, collection_id, ws_connection_id, metadata)` | `CaptureSession` | Create a new capture session |
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| `conn.get_capture_session(capture_session_id)` | `CaptureSession` | Retrieve an existing capture session |
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| `conn.generate_client_token()` | `str` | Generate a client-side authentication token |
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### Create a Capture Session
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```python
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from pathlib import Path
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ws_id = (Path.home() / ".local" / "state" / "videodb" / "videodb_ws_id").read_text().strip()
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session = conn.create_capture_session(
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end_user_id="user-123", # required
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collection_id="default",
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ws_connection_id=ws_id,
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metadata={"app": "my-app"},
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)
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print(f"Session ID: {session.id}")
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```
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> **Note:** `end_user_id` is required and identifies the user initiating the capture. For testing or demo purposes, any unique string identifier works (e.g., `"demo-user"`, `"test-123"`).
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### CaptureSession Properties
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| Property | Type | Description |
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|----------|------|-------------|
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| `session.id` | `str` | Unique capture session ID |
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### CaptureSession Methods
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| Method | Returns | Description |
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|--------|---------|-------------|
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| `session.get_rtstream(type)` | `list[RTStream]` | Get RTStreams by type: `"mic"`, `"screen"`, or `"system_audio"` |
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### Generate a Client Token
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```python
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token = conn.generate_client_token()
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```
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---
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## CaptureClient
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The client runs on the user's machine and handles permissions, channel discovery, and streaming.
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```python
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from videodb.capture import CaptureClient
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client = CaptureClient(client_token=token)
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```
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### CaptureClient Methods
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| Method | Returns | Description |
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|--------|---------|-------------|
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| `await client.request_permission(type)` | `None` | Request device permission (`"microphone"`, `"screen_capture"`) |
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| `await client.list_channels()` | `Channels` | Discover available audio/video channels |
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| `await client.start_capture_session(capture_session_id, channels, primary_video_channel_id)` | `None` | Start streaming selected channels |
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| `await client.stop_capture()` | `None` | Gracefully stop the capture session |
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| `await client.shutdown()` | `None` | Clean up client resources |
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### Request Permissions
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```python
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await client.request_permission("microphone")
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await client.request_permission("screen_capture")
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```
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### Start a Session
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```python
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selected_channels = [c for c in [mic, display, system_audio] if c]
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await client.start_capture_session(
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capture_session_id=session.id,
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channels=selected_channels,
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primary_video_channel_id=display.id if display else None,
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)
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```
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### Stop a Session
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```python
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await client.stop_capture()
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await client.shutdown()
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```
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---
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## Channels
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Returned by `client.list_channels()`. Groups available devices by type.
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```python
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channels = await client.list_channels()
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for ch in channels.all():
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print(f" {ch.id} ({ch.type}): {ch.name}")
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mic = channels.mics.default
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display = channels.displays.default
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system_audio = channels.system_audio.default
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```
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### Channel Groups
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| Property | Type | Description |
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|----------|------|-------------|
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| `channels.mics` | `ChannelGroup` | Available microphones |
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| `channels.displays` | `ChannelGroup` | Available screen displays |
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| `channels.system_audio` | `ChannelGroup` | Available system audio sources |
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### ChannelGroup Methods & Properties
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| Member | Type | Description |
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|--------|------|-------------|
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| `group.default` | `Channel` | Default channel in the group (or `None`) |
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| `group.all()` | `list[Channel]` | All channels in the group |
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### Channel Properties
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| Property | Type | Description |
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|----------|------|-------------|
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| `ch.id` | `str` | Unique channel ID |
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| `ch.type` | `str` | Channel type (`"mic"`, `"display"`, `"system_audio"`) |
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| `ch.name` | `str` | Human-readable channel name |
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| `ch.store` | `bool` | Whether to persist the recording (set to `True` to save) |
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Without `store = True`, streams are processed in real-time but not saved.
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---
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## RTStreams and AI Pipelines
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After session is active, retrieve RTStream objects with `session.get_rtstream()`.
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For RTStream methods (indexing, transcription, alerts, batch config), see [rtstream-reference.md](rtstream-reference.md).
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---
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## Session Lifecycle
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```
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create_capture_session()
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│
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v
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┌───────────────┐
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│ created │
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└───────┬───────┘
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│ client.start_capture_session()
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v
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┌───────────────┐ WebSocket: capture_session.starting
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│ starting │ ──> Capture channels connect
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└───────┬───────┘
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│
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v
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┌───────────────┐ WebSocket: capture_session.active
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│ active │ ──> Start AI pipelines
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└───────┬──────────────┐
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│ │
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│ v
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│ ┌───────────────┐ WebSocket: capture_session.failed
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│ │ failed │ ──> Inspect error payload and retry setup
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│ └───────────────┘
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│ unrecoverable capture error
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│
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│ client.stop_capture()
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v
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┌───────────────┐ WebSocket: capture_session.stopping
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│ stopping │ ──> Finalize streams
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└───────┬───────┘
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│
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v
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┌───────────────┐ WebSocket: capture_session.stopped
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│ stopped │ ──> All streams finalized
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└───────┬───────┘
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│ (if store=True)
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v
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┌───────────────┐ WebSocket: capture_session.exported
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│ exported │ ──> Access video_id, stream_url, player_url
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└───────────────┘
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```
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