Add realtime output modality and transcript events (#17701)

- Add outputModality to thread/realtime/start and wire text/audio output
selection through app-server, core, API, and TUI.\n- Rename the realtime
transcript delta notification and add a separate transcript done
notification that forwards final text from item done without correlating
it with deltas.

codex-rs/protocol/src/protocol.rs

@@ -134,6 +134,8 @@ pub struct McpServerRefreshConfig {
 
 #[derive(Debug, Clone, Deserialize, Serialize, PartialEq, JsonSchema, TS)]
 pub struct ConversationStartParams {
+    /// Selects whether the realtime session should produce text or audio output.
+    pub output_modality: RealtimeOutputModality,
     #[serde(
         default,
         deserialize_with = "conversation_start_prompt_serde::deserialize",
@@ -157,6 +159,13 @@ pub enum ConversationStartTransport {
     Webrtc { sdp: String },
 }
 
+#[derive(Debug, Clone, Copy, Deserialize, Serialize, PartialEq, Eq, JsonSchema, TS)]
+#[serde(rename_all = "snake_case")]
+pub enum RealtimeOutputModality {
+    Text,
+    Audio,
+}
+
 mod conversation_start_prompt_serde {
     use serde::Deserializer;
     use serde::Serializer;
@@ -290,6 +299,11 @@ pub struct RealtimeTranscriptDelta {
     pub delta: String,
 }
 
+#[derive(Debug, Clone, Deserialize, Serialize, PartialEq, Eq, JsonSchema, TS)]
+pub struct RealtimeTranscriptDone {
+    pub text: String,
+}
+
 #[derive(Debug, Clone, Deserialize, Serialize, PartialEq, Eq, JsonSchema, TS)]
 pub struct RealtimeTranscriptEntry {
     pub role: String,
@@ -332,7 +346,9 @@ pub enum RealtimeEvent {
     },
     InputAudioSpeechStarted(RealtimeInputAudioSpeechStarted),
     InputTranscriptDelta(RealtimeTranscriptDelta),
+    InputTranscriptDone(RealtimeTranscriptDone),
     OutputTranscriptDelta(RealtimeTranscriptDelta),
+    OutputTranscriptDone(RealtimeTranscriptDone),
     AudioOut(RealtimeAudioFrame),
     ResponseCreated(RealtimeResponseCreated),
     ResponseCancelled(RealtimeResponseCancelled),
@@ -4594,12 +4610,14 @@ mod tests {
             },
         });
         let start = Op::RealtimeConversationStart(ConversationStartParams {
+            output_modality: RealtimeOutputModality::Audio,
             prompt: Some(Some("be helpful".to_string())),
             session_id: Some("conv_1".to_string()),
             transport: None,
             voice: None,
         });
         let webrtc_start = Op::RealtimeConversationStart(ConversationStartParams {
+            output_modality: RealtimeOutputModality::Audio,
             prompt: Some(Some("be helpful".to_string())),
             session_id: Some("conv_1".to_string()),
             transport: Some(ConversationStartTransport::Webrtc {
@@ -4612,12 +4630,14 @@ mod tests {
         });
         let close = Op::RealtimeConversationClose;
         let default_prompt_start = Op::RealtimeConversationStart(ConversationStartParams {
+            output_modality: RealtimeOutputModality::Audio,
             prompt: None,
             session_id: None,
             transport: None,
             voice: None,
         });
         let null_prompt_start = Op::RealtimeConversationStart(ConversationStartParams {
+            output_modality: RealtimeOutputModality::Audio,
             prompt: Some(None),
             session_id: None,
             transport: None,
@@ -4629,6 +4649,7 @@ mod tests {
             serde_json::to_value(&start).unwrap(),
             json!({
                 "type": "realtime_conversation_start",
+                "output_modality": "audio",
                 "prompt": "be helpful",
                 "session_id": "conv_1"
             })
@@ -4636,19 +4657,22 @@ mod tests {
         assert_eq!(
             serde_json::to_value(&default_prompt_start).unwrap(),
             json!({
-                "type": "realtime_conversation_start"
+                "type": "realtime_conversation_start",
+                "output_modality": "audio"
             })
         );
         assert_eq!(
             serde_json::to_value(&null_prompt_start).unwrap(),
             json!({
                 "type": "realtime_conversation_start",
+                "output_modality": "audio",
                 "prompt": null
             })
         );
         assert_eq!(
             serde_json::from_value::<Op>(json!({
-                "type": "realtime_conversation_start"
+                "type": "realtime_conversation_start",
+                "output_modality": "audio"
             }))
             .unwrap(),
             default_prompt_start
@@ -4656,6 +4680,7 @@ mod tests {
         assert_eq!(
             serde_json::from_value::<Op>(json!({
                 "type": "realtime_conversation_start",
+                "output_modality": "audio",
                 "prompt": null
             }))
             .unwrap(),
@@ -4701,6 +4726,7 @@ mod tests {
             serde_json::to_value(&webrtc_start).unwrap(),
             json!({
                 "type": "realtime_conversation_start",
+                "output_modality": "audio",
                 "prompt": "be helpful",
                 "session_id": "conv_1",
                 "transport": {