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

codex-rs/core/src/session/turn.rs

@@ -636,6 +636,12 @@ pub(crate) async fn run_turn(
             }
             Err(e) => {
                 info!("Turn error: {e:#}");
+                if matches!(
+                    e.details(),
+                    CodexErrorDetails::MisalignmentPolicyViolation { .. }
+                ) {
+                    sess.conversation.retire_handoffs_for_misalignment().await;
+                }
                 let error = e.to_codex_protocol_error();
                 sess.emit_turn_error_lifecycle(turn_context.as_ref(), error.clone())
                     .await;