Files
codex/codex-rs/realtime-webrtc
Benjamin Carlsson b01c3986fd Add live WebRTC voice conversations to the TUI (#43581)
## What changed

- Add feature-gated `/voice`, `/voice mute`, and `/voice stop` commands with local WebRTC audio and app-server signaling.
- Show live transcripts, conversation status, and microphone and speaker levels. Preserve captions and undelivered answers across thread switches.
- Speak final answers from voice handoffs while keeping delegated reasoning and commentary hidden and typed answers unspoken.
- Retry eligible startup failures once and clean up voice sessions on thread switches and disconnects.
- Stop voice and block late handoffs after a misalignment policy violation. Remove realtime event payloads and spoken text from receipt and debug logs.

## Testing

Add coverage for voice command mapping, rejected starts, disconnect cleanup, delegated final-answer speech, late handoff rejection, and transcript and footer rendering.

GitOrigin-RevId: 11def1f86b0e023b21d0d92072d6636ccad89815
2026-09-07 20:25:30 +00:00
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