| src | |
| .gitignore | |
| CLAUDE.md | |
| package-lock.json | |
| package.json | |
| README.md | |
| tsconfig.json |
commonroom-recorder#
Record a commonroom call from the command line, for transcribing the meeting afterwards.
The recorder joins a room as an ordinary, visible participant (default name "Recorder") that stays muted the whole time. It receives every other participant's audio and writes one WAV file per participant β so a transcript with speaker attribution needs no diarization β plus the room chat and a machine-readable event log. By default it also posts a one-line chat notice to each participant so everyone knows the meeting is being recorded.
Usage#
npm install
npm run build
node dist/cli.js <room> [options]
Options:
--name <name> Display name in the room (default: Recorder)
--out <dir> Output directory (default: ./recordings/<room>-<timestamp>)
--duration <sec> Stop automatically after this many seconds
--notice <text> Chat line sent to each participant on connect
(default: "π΄ This meeting is being recorded.")
--no-notice Don't send any recording notice
Stop with Ctrl-C. Requires Node >= 22 (built-in WebSocket). The WebRTC stack
is @roamhq/wrtc, which
ships prebuilt binaries for Linux and macOS.
Output#
<out>/
audio/<name>-<peer8>-segN.wav one file per participant per connection
(48 kHz mono s16 PCM, typically)
chat.txt human-readable chat + join/left log
events.jsonl every event with ISO timestamps: join, left,
chat, mute/unmute, segment start/end
manifest.json session summary: room, participants,
segments with start/end times and durations
Everything is written incrementally (tail -f chat.txt works live; the
manifest is rewritten at every segment boundary and every 30 s), so a crash
loses at most about a second of audio. A file only starts when a participant's
first real audio arrives β someone who never unmutes produces no file. If a
participant disconnects and returns, they get a new numbered segment; the
manifest's per-segment start times let a transcript interleave speakers on one
timeline. During a segment, silence is padded by wall clock, so a sample's
position in the file always tracks elapsed time.
To transcribe: run each audio/*.wav through your transcriber of choice
(e.g. whisper), offset each result by its segment's startedAt from
manifest.json, and merge.
How it works#
The p2p layer is commonroom's, ported to Node: the same nostr presence/signaling topics (knowing the room name IS the key), the same schnorr-signed events (with a fresh ephemeral keypair per run), the same deterministic-initiator WebRTC mesh and control data channel (hello, mute notices, chat, bye). To the browsers in the room the recorder is indistinguishable from a participant whose mic and camera are muted β it counts toward the room cap of 8 and appears in the participant list.
Two deliberate deviations from the browser client:
- Receive-only media. The video m-line is negotiated
sendonlyfrom the recorder's side (a placeholder track that never produces a frame), so no video is ever sent to the recorder β with up to 7 participants that saves several Mbit/s and all the decode CPU. Audio is symmetric (a silent placeholder goes out, like any muted mic). - Files instead of tiles. Each remote audio track feeds an
RTCAudioSinkwhose PCM goes straight to an incrementally-written WAV.
Testing#
npm run test:loopback runs an end-to-end test with no browser: it starts the
recorder and a synthetic participant that "speaks" a 440 Hz sine and sends a
chat message, then verifies the WAV really contains the tone and the chat made
it to disk. It uses the real public nostr relays, so it needs network access.
For a real-world test, run the recorder and join the same room at
https://concept-collection.github.io/commonroom/ from a browser.