Russell Ballestrini
0cbbc37393
Fox: 'I want the sound to be epic and to fucking choppy ever.'
Two changes that compose:
1. Default-on DJ mode for listeners. On role landing (or transition
INTO listener), auto-flip every speaker's playback path off the
WebRTC subscribe and onto the existing HTTP Ogg/Opus tap from
the SFU. The browser's <audio> element keeps a deep media buffer
(~30s in Chrome) that absorbs glitches WebRTC can't. Speakers
stay on WebRTC for low-latency conversational audio. Listener
can still opt-out per-speaker via the stream button — gating
expanded from {self|host} to {self|host|listener}.
Auto-enrol triggers at:
- onRoleEntered (initial join as listener)
- peer-joined (new speaker arrives while we're listening)
- role-change of OTHER peer (they became speaker)
- onRoleChanged self speaker → listener
Auto-disable: onRoleChanged self listener → speaker
(cannot afford 2s+ buffer when you have to talk back).
2. RECV_PLAYOUT_DELAY_SEC 2.0 → 4.0 on both mesh and SFU receive
paths. Conversational latency is now ~4s for speakers; this is
the cheap-but-real win against chop on the WebRTC path for
everyone who still uses it. Music/DJ rooms generally don't care
about 4s — the room's already committed to a stream-mode 2s+.
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