Update realtime model and honor provider-specific voice defaults (#61259)

## Why

An unset voice preference should use the call destination's default. Changed or cleared preferences should also take effect on the next V3 call, even when the thread retains an older setting.

## What changed

- Use `gpt-live-1.1-codex` as the default V3 realtime model.
- Preserve an unset voice until request serialization, defaulting V3 API calls to `marin` and ChatGPT calls to `cove`, while honoring explicit voice selections.
- Reload current voice settings when starting a new V3 call through the app server, including cleared preferences.
- Populate the TUI voice picker from the V3 catalog, add `marin` to that catalog, and remove voice-catalog lookups from call startup and preference saving.

## Testing

Add regression coverage for API and ChatGPT voice defaults, explicit overrides, changed and cleared settings, and TUI call requests and voice selection.

GitOrigin-RevId: d0dd4dcd4bcad9ab869d7bf524e7095a88ffcfaa

codex-rs/codex-api/tests/realtime_websocket_e2e.rs

@@ -162,7 +162,7 @@ async fn realtime_ws_e2e_session_create_and_event_flow() {
                 event_parser: RealtimeEventParser::V1,
                 session_mode: RealtimeSessionMode::Conversational,
                 output_modality: RealtimeOutputModality::Audio,
-                voice: RealtimeVoice::Cove,
+                voice: Some(RealtimeVoice::Cove),
             },
             HeaderMap::new(),
             HeaderMap::new(),
@@ -268,7 +268,7 @@ async fn realtime_ws_connect_webrtc_sideband_retries_join_until_server_is_availa
                 event_parser: RealtimeEventParser::RealtimeV2,
                 session_mode: RealtimeSessionMode::Conversational,
                 output_modality: RealtimeOutputModality::Audio,
-                voice: RealtimeVoice::Marin,
+                voice: Some(RealtimeVoice::Marin),
             },
             "rtc_test",
             HeaderMap::new(),
@@ -349,7 +349,7 @@ async fn realtime_ws_existing_call_retries_activation_404_with_a_bounded_budget(
                         event_parser: RealtimeEventParser::RealtimeV2,
                         session_mode: RealtimeSessionMode::Conversational,
                         output_modality: RealtimeOutputModality::Audio,
-                        voice: RealtimeVoice::Marin,
+                        voice: Some(RealtimeVoice::Marin),
                     },
                     "rtc_prepared",
                     HeaderMap::new(),
@@ -405,7 +405,7 @@ async fn realtime_ws_existing_call_transport_close_preserves_session() {
                     event_parser,
                     session_mode: RealtimeSessionMode::Conversational,
                     output_modality: RealtimeOutputModality::Audio,
-                    voice: RealtimeVoice::Marin,
+                    voice: Some(RealtimeVoice::Marin),
                 },
                 "rtc_frontend_owned",
                 HeaderMap::new(),
@@ -461,7 +461,7 @@ async fn realtime_ws_frameless_session_close_still_ends_session() {
                 event_parser: RealtimeEventParser::FramelessBidi,
                 session_mode: RealtimeSessionMode::Conversational,
                 output_modality: RealtimeOutputModality::Audio,
-                voice: RealtimeVoice::Marin,
+                voice: Some(RealtimeVoice::Marin),
             },
             "rtc_session_owner",
             HeaderMap::new(),
@@ -525,7 +525,7 @@ async fn realtime_ws_e2e_send_while_next_event_waits() {
                 event_parser: RealtimeEventParser::V1,
                 session_mode: RealtimeSessionMode::Conversational,
                 output_modality: RealtimeOutputModality::Audio,
-                voice: RealtimeVoice::Cove,
+                voice: Some(RealtimeVoice::Cove),
             },
             HeaderMap::new(),
             HeaderMap::new(),
@@ -595,7 +595,7 @@ async fn realtime_ws_e2e_disconnected_emitted_once() {
                 event_parser: RealtimeEventParser::V1,
                 session_mode: RealtimeSessionMode::Conversational,
                 output_modality: RealtimeOutputModality::Audio,
-                voice: RealtimeVoice::Cove,
+                voice: Some(RealtimeVoice::Cove),
             },
             HeaderMap::new(),
             HeaderMap::new(),
@@ -661,7 +661,7 @@ async fn realtime_ws_e2e_ignores_unknown_text_events() {
                 event_parser: RealtimeEventParser::V1,
                 session_mode: RealtimeSessionMode::Conversational,
                 output_modality: RealtimeOutputModality::Audio,
-                voice: RealtimeVoice::Cove,
+                voice: Some(RealtimeVoice::Cove),
             },
             HeaderMap::new(),
             HeaderMap::new(),
@@ -770,7 +770,7 @@ async fn realtime_ws_e2e_realtime_v2_parser_emits_handoff_requested() {
                 event_parser: RealtimeEventParser::RealtimeV2,
                 session_mode: RealtimeSessionMode::Conversational,
                 output_modality: RealtimeOutputModality::Audio,
-                voice: RealtimeVoice::Marin,
+                voice: Some(RealtimeVoice::Marin),
             },
             HeaderMap::new(),
             HeaderMap::new(),

codex-rs/codex-api/tests/realtime_websocket_tls.rs

@@ -145,7 +145,7 @@ async fn check_connection(address: String) {
                 event_parser: RealtimeEventParser::V1,
                 session_mode: RealtimeSessionMode::Conversational,
                 output_modality: RealtimeOutputModality::Audio,
-                voice: RealtimeVoice::Cove,
+                voice: Some(RealtimeVoice::Cove),
             },
             HeaderMap::new(),
             HeaderMap::new(),