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[stack 2/4] Align main realtime v2 wire and runtime flow (#14830)

Browse source 
## Stack Position
2/4. Built on top of #14828.

## Base
- #14828

## Unblocks
- #14829
- #14827

## Scope
- Port the realtime v2 wire parsing, session, app-server, and
conversation runtime behavior onto the split websocket-method base.
- Branch runtime behavior directly on the current realtime session kind
instead of parser-derived flow flags.
- Keep regression coverage in the existing e2e suites.

---------

Co-authored-by: Codex <noreply@openai.com>
This commit is contained in:
Ahmed Ibrahim 2026-03-16 21:38:07 -07:00 • committed by GitHub
parent 1d85fe79ed
commit fbd7f9b986
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GPG key ID: B5690EEEBB952194
28 changed files with 807 additions and 140 deletions
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6
codex-rs/app-server-protocol/schema/json/ClientRequest.json
View file

@ -2779,6 +2779,12 @@
"data": {
"type": "string"
},
"itemId": {
"type": [
"string",
"null"
]
},
"numChannels": {
"format": "uint16",
"minimum": 0.0,

6
codex-rs/app-server-protocol/schema/json/ServerNotification.json
View file

@ -2750,6 +2750,12 @@
"data": {
"type": "string"
},
"itemId": {
"type": [
"string",
"null"
]
},
"numChannels": {
"format": "uint16",
"minimum": 0.0,

6
codex-rs/app-server-protocol/schema/json/codex_app_server_protocol.schemas.json
View file

@ -12817,6 +12817,12 @@
"data": {
"type": "string"
},
"itemId": {
"type": [
"string",
"null"
]
},
"numChannels": {
"format": "uint16",
"minimum": 0.0,

6
codex-rs/app-server-protocol/schema/json/codex_app_server_protocol.v2.schemas.json
View file

@ -10577,6 +10577,12 @@
"data": {
"type": "string"
},
"itemId": {
"type": [
"string",
"null"
]
},
"numChannels": {
"format": "uint16",
"minimum": 0.0,

6
codex-rs/app-server-protocol/schema/json/v2/ThreadRealtimeOutputAudioDeltaNotification.json
View file

@ -7,6 +7,12 @@
"data": {
"type": "string"
},
"itemId": {
"type": [
"string",
"null"
]
},
"numChannels": {
"format": "uint16",
"minimum": 0.0,

2
codex-rs/app-server-protocol/schema/typescript/v2/ThreadRealtimeAudioChunk.ts
View file

@ -5,4 +5,4 @@
/**
* EXPERIMENTAL - thread realtime audio chunk.
*/
export type ThreadRealtimeAudioChunk = { data: string, sampleRate: number, numChannels: number, samplesPerChannel: number | null, };
export type ThreadRealtimeAudioChunk = { data: string, sampleRate: number, numChannels: number, samplesPerChannel: number | null, itemId: string | null, };

5
codex-rs/app-server-protocol/src/protocol/common.rs
View file

@ -1577,6 +1577,7 @@ mod tests {
sample_rate: 24_000,
num_channels: 1,
samples_per_channel: Some(512),
item_id: None,
},
},
);
@ -1589,7 +1590,8 @@ mod tests {
"data": "AQID",
"sampleRate": 24000,
"numChannels": 1,
"samplesPerChannel": 512
"samplesPerChannel": 512,
"itemId": null
}
}
}),
@ -1641,6 +1643,7 @@ mod tests {
sample_rate: 24_000,
num_channels: 1,
samples_per_channel: Some(512),
item_id: None,
},
},
);

5
codex-rs/app-server-protocol/src/protocol/v2.rs
View file

@ -3659,6 +3659,7 @@ pub struct ThreadRealtimeAudioChunk {
pub sample_rate: u32,
pub num_channels: u16,
pub samples_per_channel: Option<u32>,
pub item_id: Option<String>,
}
impl From<CoreRealtimeAudioFrame> for ThreadRealtimeAudioChunk {
@ -3668,12 +3669,14 @@ impl From<CoreRealtimeAudioFrame> for ThreadRealtimeAudioChunk {
sample_rate,
num_channels,
samples_per_channel,
item_id,
} = value;
Self {
data,
sample_rate,
num_channels,
samples_per_channel,
item_id,
}
}
}
@ -3685,12 +3688,14 @@ impl From<ThreadRealtimeAudioChunk> for CoreRealtimeAudioFrame {
sample_rate,
num_channels,
samples_per_channel,
item_id,
} = value;
Self {
data,
sample_rate,
num_channels,
samples_per_channel,
item_id,
}
}
}

28
codex-rs/app-server/src/bespoke_event_handling.rs
View file

@ -350,6 +350,20 @@ pub(crate) async fn apply_bespoke_event_handling(
if let ApiVersion::V2 = api_version {
match event.payload {
RealtimeEvent::SessionUpdated { .. } => {}
RealtimeEvent::InputAudioSpeechStarted(event) => {
let notification = ThreadRealtimeItemAddedNotification {
thread_id: conversation_id.to_string(),
item: serde_json::json!({
"type": "input_audio_buffer.speech_started",
"item_id": event.item_id,
}),
};
outgoing
.send_server_notification(ServerNotification::ThreadRealtimeItemAdded(
notification,
))
.await;
}
RealtimeEvent::InputTranscriptDelta(_) => {}
RealtimeEvent::OutputTranscriptDelta(_) => {}
RealtimeEvent::AudioOut(audio) => {
@ -363,6 +377,20 @@ pub(crate) async fn apply_bespoke_event_handling(
)
.await;
}
RealtimeEvent::ResponseCancelled(event) => {
let notification = ThreadRealtimeItemAddedNotification {
thread_id: conversation_id.to_string(),
item: serde_json::json!({
"type": "response.cancelled",
"response_id": event.response_id,
}),
};
outgoing
.send_server_notification(ServerNotification::ThreadRealtimeItemAdded(
notification,
))
.await;
}
RealtimeEvent::ConversationItemAdded(item) => {
let notification = ThreadRealtimeItemAddedNotification {
thread_id: conversation_id.to_string(),

9
codex-rs/app-server/tests/suite/v2/realtime_conversation.rs
View file

@ -70,6 +70,7 @@ async fn realtime_conversation_streams_v2_notifications() -> Result<()> {
"message": "upstream boom"
}),
],
vec![],
]])
.await;
@ -135,6 +136,7 @@ async fn realtime_conversation_streams_v2_notifications() -> Result<()> {
sample_rate: 24_000,
num_channels: 1,
samples_per_channel: Some(480),
item_id: None,
},
})
.await?;
@ -191,7 +193,7 @@ async fn realtime_conversation_streams_v2_notifications() -> Result<()> {
let connections = realtime_server.connections();
assert_eq!(connections.len(), 1);
let connection = &connections[0];
assert_eq!(connection.len(), 3);
assert_eq!(connection.len(), 4);
assert_eq!(
connection[0].body_json()["type"].as_str(),
Some("session.update")
@ -211,6 +213,10 @@ async fn realtime_conversation_streams_v2_notifications() -> Result<()> {
.as_str()
.context("expected websocket request type")?
.to_string(),
connection[3].body_json()["type"]
.as_str()
.context("expected websocket request type")?
.to_string(),
];
request_types.sort();
assert_eq!(
@ -218,6 +224,7 @@ async fn realtime_conversation_streams_v2_notifications() -> Result<()> {
[
"conversation.item.create".to_string(),
"input_audio_buffer.append".to_string(),
"response.create".to_string(),
]
);

166
codex-rs/codex-api/src/endpoint/realtime_websocket/methods.rs
View file

@ -272,12 +272,12 @@ impl RealtimeWebsocketConnection {
impl RealtimeWebsocketWriter {
pub async fn send_audio_frame(&self, frame: RealtimeAudioFrame) -> Result<(), ApiError> {
self.send_json(RealtimeOutboundMessage::InputAudioBufferAppend { audio: frame.data })
self.send_json(&RealtimeOutboundMessage::InputAudioBufferAppend { audio: frame.data })
.await
}
pub async fn send_conversation_item_create(&self, text: String) -> Result<(), ApiError> {
self.send_json(conversation_item_create_message(self.event_parser, text))
self.send_json(&conversation_item_create_message(self.event_parser, text))
.await
}
@ -286,7 +286,7 @@ impl RealtimeWebsocketWriter {
handoff_id: String,
output_text: String,
) -> Result<(), ApiError> {
self.send_json(conversation_handoff_append_message(
self.send_json(&conversation_handoff_append_message(
self.event_parser,
handoff_id,
output_text,
@ -294,6 +294,11 @@ impl RealtimeWebsocketWriter {
.await
}
pub async fn send_response_create(&self) -> Result<(), ApiError> {
self.send_json(&RealtimeOutboundMessage::ResponseCreate)
.await
}
pub async fn send_session_update(
&self,
instructions: String,
@ -301,7 +306,7 @@ impl RealtimeWebsocketWriter {
) -> Result<(), ApiError> {
let session_mode = normalized_session_mode(self.event_parser, session_mode);
let session = session_update_session(self.event_parser, instructions, session_mode);
self.send_json(RealtimeOutboundMessage::SessionUpdate { session })
self.send_json(&RealtimeOutboundMessage::SessionUpdate { session })
.await
}
@ -319,11 +324,14 @@ impl RealtimeWebsocketWriter {
Ok(())
}
async fn send_json(&self, message: RealtimeOutboundMessage) -> Result<(), ApiError> {
let payload = serde_json::to_string(&message)
async fn send_json(&self, message: &RealtimeOutboundMessage) -> Result<(), ApiError> {
let payload = serde_json::to_string(message)
.map_err(|err| ApiError::Stream(format!("failed to encode realtime request: {err}")))?;
debug!(?message, "realtime websocket request");
self.send_payload(payload).await
}
pub async fn send_payload(&self, payload: String) -> Result<(), ApiError> {
if self.is_closed.load(Ordering::SeqCst) {
return Err(ApiError::Stream(
"realtime websocket connection is closed".to_string(),
@ -392,6 +400,7 @@ impl RealtimeWebsocketEvents {
async fn update_active_transcript(&self, event: &mut RealtimeEvent) {
let mut active_transcript = self.active_transcript.lock().await;
match event {
RealtimeEvent::InputAudioSpeechStarted(_) => {}
RealtimeEvent::InputTranscriptDelta(RealtimeTranscriptDelta { delta }) => {
append_transcript_delta(&mut active_transcript.entries, "user", delta);
}
@ -403,6 +412,7 @@ impl RealtimeWebsocketEvents {
}
RealtimeEvent::SessionUpdated { .. }
| RealtimeEvent::AudioOut(_)
| RealtimeEvent::ResponseCancelled(_)
| RealtimeEvent::ConversationItemAdded(_)
| RealtimeEvent::ConversationItemDone { .. }
| RealtimeEvent::Error(_) => {}
@ -616,6 +626,8 @@ mod tests {
use crate::endpoint::realtime_websocket::protocol::RealtimeHandoffRequested;
use crate::endpoint::realtime_websocket::protocol::RealtimeTranscriptDelta;
use crate::endpoint::realtime_websocket::protocol::RealtimeTranscriptEntry;
use codex_protocol::protocol::RealtimeInputAudioSpeechStarted;
use codex_protocol::protocol::RealtimeResponseCancelled;
use http::HeaderValue;
use pretty_assertions::assert_eq;
use serde_json::Value;
@ -660,6 +672,7 @@ mod tests {
sample_rate: 48000,
num_channels: 1,
samples_per_channel: Some(960),
item_id: None,
}))
);
}
@ -809,10 +822,112 @@ mod tests {
sample_rate: 24_000,
num_channels: 1,
samples_per_channel: None,
item_id: None,
}))
);
}
#[test]
fn parse_realtime_v2_response_audio_delta_with_item_id() {
let payload = json!({
"type": "response.audio.delta",
"delta": "AQID",
"item_id": "item_audio_1"
})
.to_string();
assert_eq!(
parse_realtime_event(payload.as_str(), RealtimeEventParser::RealtimeV2),
Some(RealtimeEvent::AudioOut(RealtimeAudioFrame {
data: "AQID".to_string(),
sample_rate: 24_000,
num_channels: 1,
samples_per_channel: None,
item_id: Some("item_audio_1".to_string()),
}))
);
}
#[test]
fn parse_realtime_v2_speech_started_event() {
let payload = json!({
"type": "input_audio_buffer.speech_started",
"item_id": "item_input_1"
})
.to_string();
assert_eq!(
parse_realtime_event(payload.as_str(), RealtimeEventParser::RealtimeV2),
Some(RealtimeEvent::InputAudioSpeechStarted(
RealtimeInputAudioSpeechStarted {
item_id: Some("item_input_1".to_string()),
}
))
);
}
#[test]
fn parse_realtime_v2_response_cancelled_event() {
let payload = json!({
"type": "response.cancelled",
"response": {"id": "resp_cancelled_1"}
})
.to_string();
assert_eq!(
parse_realtime_event(payload.as_str(), RealtimeEventParser::RealtimeV2),
Some(RealtimeEvent::ResponseCancelled(
RealtimeResponseCancelled {
response_id: Some("resp_cancelled_1".to_string()),
}
))
);
}
#[test]
fn parse_realtime_v2_response_done_handoff_event() {
let payload = json!({
"type": "response.done",
"response": {
"output": [{
"id": "item_123",
"type": "function_call",
"name": "codex",
"call_id": "call_123",
"arguments": "{\"prompt\":\"delegate from done\"}"
}]
}
})
.to_string();
assert_eq!(
parse_realtime_event(payload.as_str(), RealtimeEventParser::RealtimeV2),
Some(RealtimeEvent::HandoffRequested(RealtimeHandoffRequested {
handoff_id: "call_123".to_string(),
item_id: "item_123".to_string(),
input_transcript: "delegate from done".to_string(),
active_transcript: Vec::new(),
}))
);
}
#[test]
fn parse_realtime_v2_response_created_event() {
let payload = json!({
"type": "response.created",
"response": {"id": "resp_created_1"}
})
.to_string();
assert_eq!(
parse_realtime_event(payload.as_str(), RealtimeEventParser::RealtimeV2),
Some(RealtimeEvent::ConversationItemAdded(json!({
"type": "response.created",
"response": {"id": "resp_created_1"}
})))
);
}
#[test]
fn merge_request_headers_matches_http_precedence() {
let mut provider_headers = HeaderMap::new();
@ -1169,6 +1284,7 @@ mod tests {
sample_rate: 48000,
num_channels: 1,
samples_per_channel: Some(960),
item_id: None,
})
.await
.expect("send audio");
@ -1196,6 +1312,7 @@ mod tests {
sample_rate: 48000,
num_channels: 1,
samples_per_channel: None,
item_id: None,
})
);
@ -1285,9 +1402,38 @@ mod tests {
first_json["session"]["type"],
Value::String("realtime".to_string())
);
assert_eq!(first_json["session"]["output_modalities"], json!(["audio"]));
assert_eq!(
first_json["session"]["audio"]["input"]["format"],
json!({
"type": "audio/pcm",
"rate": 24_000,
})
);
assert_eq!(
first_json["session"]["audio"]["input"]["noise_reduction"],
json!({
"type": "near_field",
})
);
assert_eq!(
first_json["session"]["audio"]["input"]["turn_detection"],
json!({
"type": "server_vad",
"interrupt_response": true,
"create_response": true,
})
);
assert_eq!(
first_json["session"]["audio"]["output"]["format"],
json!({
"type": "audio/pcm",
"rate": 24_000,
})
);
assert_eq!(
first_json["session"]["audio"]["output"]["voice"],
Value::String("alloy".to_string())
Value::String("marin".to_string())
);
assert_eq!(
first_json["session"]["tools"][0]["type"],
@ -1301,6 +1447,10 @@ mod tests {
first_json["session"]["tools"][0]["parameters"]["required"],
json!(["prompt"])
);
assert_eq!(
first_json["session"]["tool_choice"],
Value::String("auto".to_string())
);
ws.send(Message::Text(
json!({
@ -1511,6 +1661,7 @@ mod tests {
sample_rate: 24_000,
num_channels: 1,
samples_per_channel: Some(480),
item_id: None,
})
.await
.expect("send audio");
@ -1690,6 +1841,7 @@ mod tests {
sample_rate: 48000,
num_channels: 1,
samples_per_channel: Some(960),
item_id: None,
}),
)
.await

1
codex-rs/codex-api/src/endpoint/realtime_websocket/methods_common.rs
View file

@ -12,7 +12,6 @@ use crate::endpoint::realtime_websocket::protocol::RealtimeSessionMode;
use crate::endpoint::realtime_websocket::protocol::SessionUpdateSession;
pub(super) const REALTIME_AUDIO_SAMPLE_RATE: u32 = 24_000;
pub(super) const REALTIME_AUDIO_FORMAT: &str = "audio/pcm";
pub(super) fn normalized_session_mode(
event_parser: RealtimeEventParser,

23
codex-rs/codex-api/src/endpoint/realtime_websocket/methods_v1.rs
View file

@ -1,25 +1,27 @@
use crate::endpoint::realtime_websocket::methods_common::REALTIME_AUDIO_FORMAT;
use crate::endpoint::realtime_websocket::methods_common::REALTIME_AUDIO_SAMPLE_RATE;
use crate::endpoint::realtime_websocket::protocol::AudioFormatType;
use crate::endpoint::realtime_websocket::protocol::ConversationContentType;
use crate::endpoint::realtime_websocket::protocol::ConversationItemContent;
use crate::endpoint::realtime_websocket::protocol::ConversationItemPayload;
use crate::endpoint::realtime_websocket::protocol::ConversationItemType;
use crate::endpoint::realtime_websocket::protocol::ConversationMessageItem;
use crate::endpoint::realtime_websocket::protocol::ConversationRole;
use crate::endpoint::realtime_websocket::protocol::RealtimeOutboundMessage;
use crate::endpoint::realtime_websocket::protocol::SessionAudio;
use crate::endpoint::realtime_websocket::protocol::SessionAudioFormat;
use crate::endpoint::realtime_websocket::protocol::SessionAudioInput;
use crate::endpoint::realtime_websocket::protocol::SessionAudioOutput;
use crate::endpoint::realtime_websocket::protocol::SessionAudioVoice;
use crate::endpoint::realtime_websocket::protocol::SessionType;
use crate::endpoint::realtime_websocket::protocol::SessionUpdateSession;
const REALTIME_V1_SESSION_TYPE: &str = "quicksilver";
pub(super) fn conversation_item_create_message(text: String) -> RealtimeOutboundMessage {
RealtimeOutboundMessage::ConversationItemCreate {
item: ConversationItemPayload::Message(ConversationMessageItem {
kind: "message".to_string(),
role: "user".to_string(),
r#type: ConversationItemType::Message,
role: ConversationRole::User,
content: vec![ConversationItemContent {
kind: "text".to_string(),
r#type: ConversationContentType::Text,
text,
}],
}),
@ -38,20 +40,25 @@ pub(super) fn conversation_handoff_append_message(
pub(super) fn session_update_session(instructions: String) -> SessionUpdateSession {
SessionUpdateSession {
kind: REALTIME_V1_SESSION_TYPE.to_string(),
r#type: SessionType::Quicksilver,
instructions: Some(instructions),
output_modalities: None,
audio: SessionAudio {
input: SessionAudioInput {
format: SessionAudioFormat {
kind: REALTIME_AUDIO_FORMAT.to_string(),
r#type: AudioFormatType::AudioPcm,
rate: REALTIME_AUDIO_SAMPLE_RATE,
},
noise_reduction: None,
turn_detection: None,
},
output: Some(SessionAudioOutput {
format: None,
voice: SessionAudioVoice::Fathom,
}),
},
tools: None,
tool_choice: None,
}
}

57
codex-rs/codex-api/src/endpoint/realtime_websocket/methods_v2.rs
View file

@ -1,31 +1,42 @@
use crate::endpoint::realtime_websocket::methods_common::REALTIME_AUDIO_FORMAT;
use crate::endpoint::realtime_websocket::methods_common::REALTIME_AUDIO_SAMPLE_RATE;
use crate::endpoint::realtime_websocket::protocol::AudioFormatType;
use crate::endpoint::realtime_websocket::protocol::ConversationContentType;
use crate::endpoint::realtime_websocket::protocol::ConversationFunctionCallOutputItem;
use crate::endpoint::realtime_websocket::protocol::ConversationItemContent;
use crate::endpoint::realtime_websocket::protocol::ConversationItemPayload;
use crate::endpoint::realtime_websocket::protocol::ConversationItemType;
use crate::endpoint::realtime_websocket::protocol::ConversationMessageItem;
use crate::endpoint::realtime_websocket::protocol::ConversationRole;
use crate::endpoint::realtime_websocket::protocol::NoiseReductionType;
use crate::endpoint::realtime_websocket::protocol::RealtimeOutboundMessage;
use crate::endpoint::realtime_websocket::protocol::RealtimeSessionMode;
use crate::endpoint::realtime_websocket::protocol::SessionAudio;
use crate::endpoint::realtime_websocket::protocol::SessionAudioFormat;
use crate::endpoint::realtime_websocket::protocol::SessionAudioInput;
use crate::endpoint::realtime_websocket::protocol::SessionAudioOutput;
use crate::endpoint::realtime_websocket::protocol::SessionAudioOutputFormat;
use crate::endpoint::realtime_websocket::protocol::SessionAudioVoice;
use crate::endpoint::realtime_websocket::protocol::SessionFunctionTool;
use crate::endpoint::realtime_websocket::protocol::SessionNoiseReduction;
use crate::endpoint::realtime_websocket::protocol::SessionToolType;
use crate::endpoint::realtime_websocket::protocol::SessionTurnDetection;
use crate::endpoint::realtime_websocket::protocol::SessionType;
use crate::endpoint::realtime_websocket::protocol::SessionUpdateSession;
use crate::endpoint::realtime_websocket::protocol::TurnDetectionType;
use serde_json::json;
const REALTIME_V2_SESSION_TYPE: &str = "realtime";
const REALTIME_V2_OUTPUT_MODALITY_AUDIO: &str = "audio";
const REALTIME_V2_TOOL_CHOICE: &str = "auto";
const REALTIME_V2_CODEX_TOOL_NAME: &str = "codex";
const REALTIME_V2_CODEX_TOOL_DESCRIPTION: &str = "Delegate work to Codex and return the result.";
const REALTIME_V2_CODEX_TOOL_DESCRIPTION: &str = "Delegate a request to Codex and return the final result to the user. Use this as the default action. If the user asks to do something next, later, after this, or once current work finishes, call this tool so the work is actually queued instead of merely promising to do it later.";
pub(super) fn conversation_item_create_message(text: String) -> RealtimeOutboundMessage {
RealtimeOutboundMessage::ConversationItemCreate {
item: ConversationItemPayload::Message(ConversationMessageItem {
kind: "message".to_string(),
role: "user".to_string(),
r#type: ConversationItemType::Message,
role: ConversationRole::User,
content: vec![ConversationItemContent {
kind: "input_text".to_string(),
r#type: ConversationContentType::InputText,
text,
}],
}),
@ -38,7 +49,7 @@ pub(super) fn conversation_handoff_append_message(
) -> RealtimeOutboundMessage {
RealtimeOutboundMessage::ConversationItemCreate {
item: ConversationItemPayload::FunctionCallOutput(ConversationFunctionCallOutputItem {
kind: "function_call_output".to_string(),
r#type: ConversationItemType::FunctionCallOutput,
call_id: handoff_id,
output: output_text,
}),
@ -51,21 +62,34 @@ pub(super) fn session_update_session(
) -> SessionUpdateSession {
match session_mode {
RealtimeSessionMode::Conversational => SessionUpdateSession {
kind: REALTIME_V2_SESSION_TYPE.to_string(),
r#type: SessionType::Realtime,
instructions: Some(instructions),
output_modalities: Some(vec![REALTIME_V2_OUTPUT_MODALITY_AUDIO.to_string()]),
audio: SessionAudio {
input: SessionAudioInput {
format: SessionAudioFormat {
kind: REALTIME_AUDIO_FORMAT.to_string(),
r#type: AudioFormatType::AudioPcm,
rate: REALTIME_AUDIO_SAMPLE_RATE,
},
noise_reduction: Some(SessionNoiseReduction {
r#type: NoiseReductionType::NearField,
}),
turn_detection: Some(SessionTurnDetection {
r#type: TurnDetectionType::ServerVad,
interrupt_response: true,
create_response: true,
}),
},
output: Some(SessionAudioOutput {
voice: SessionAudioVoice::Alloy,
format: Some(SessionAudioOutputFormat {
r#type: AudioFormatType::AudioPcm,
rate: REALTIME_AUDIO_SAMPLE_RATE,
}),
voice: SessionAudioVoice::Marin,
}),
},
tools: Some(vec![SessionFunctionTool {
kind: "function".to_string(),
r#type: SessionToolType::Function,
name: REALTIME_V2_CODEX_TOOL_NAME.to_string(),
description: REALTIME_V2_CODEX_TOOL_DESCRIPTION.to_string(),
parameters: json!({
@ -73,27 +97,32 @@ pub(super) fn session_update_session(
"properties": {
"prompt": {
"type": "string",
"description": "Prompt text for the delegated Codex task."
"description": "The user request to delegate to Codex."
}
},
"required": ["prompt"],
"additionalProperties": false
}),
}]),
tool_choice: Some(REALTIME_V2_TOOL_CHOICE.to_string()),
},
RealtimeSessionMode::Transcription => SessionUpdateSession {
kind: "transcription".to_string(),
r#type: SessionType::Transcription,
instructions: None,
output_modalities: None,
audio: SessionAudio {
input: SessionAudioInput {
format: SessionAudioFormat {
kind: REALTIME_AUDIO_FORMAT.to_string(),
r#type: AudioFormatType::AudioPcm,
rate: REALTIME_AUDIO_SAMPLE_RATE,
},
noise_reduction: None,
turn_detection: None,
},
output: None,
},
tools: None,
tool_choice: None,
},
}
}

103
codex-rs/codex-api/src/endpoint/realtime_websocket/protocol.rs
View file

@ -39,6 +39,8 @@ pub(super) enum RealtimeOutboundMessage {
handoff_id: String,
output_text: String,
},
#[serde(rename = "response.create")]
ResponseCreate,
#[serde(rename = "session.update")]
SessionUpdate { session: SessionUpdateSession },
#[serde(rename = "conversation.item.create")]
@ -48,12 +50,24 @@ pub(super) enum RealtimeOutboundMessage {
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionUpdateSession {
#[serde(rename = "type")]
pub(super) kind: String,
pub(super) r#type: SessionType,
#[serde(skip_serializing_if = "Option::is_none")]
pub(super) instructions: Option<String>,
#[serde(skip_serializing_if = "Option::is_none")]
pub(super) output_modalities: Option<Vec<String>>,
pub(super) audio: SessionAudio,
#[serde(skip_serializing_if = "Option::is_none")]
pub(super) tools: Option<Vec<SessionFunctionTool>>,
#[serde(skip_serializing_if = "Option::is_none")]
pub(super) tool_choice: Option<String>,
}
#[derive(Debug, Clone, Copy, Serialize)]
#[serde(rename_all = "snake_case")]
pub(super) enum SessionType {
Quicksilver,
Realtime,
Transcription,
}
#[derive(Debug, Clone, Serialize)]
@ -66,17 +80,29 @@ pub(super) struct SessionAudio {
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionAudioInput {
pub(super) format: SessionAudioFormat,
#[serde(skip_serializing_if = "Option::is_none")]
pub(super) noise_reduction: Option<SessionNoiseReduction>,
#[serde(skip_serializing_if = "Option::is_none")]
pub(super) turn_detection: Option<SessionTurnDetection>,
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionAudioFormat {
#[serde(rename = "type")]
pub(super) kind: String,
pub(super) r#type: AudioFormatType,
pub(super) rate: u32,
}
#[derive(Debug, Clone, Copy, Serialize)]
pub(super) enum AudioFormatType {
#[serde(rename = "audio/pcm")]
AudioPcm,
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionAudioOutput {
#[serde(skip_serializing_if = "Option::is_none")]
pub(super) format: Option<SessionAudioOutputFormat>,
pub(super) voice: SessionAudioVoice,
}
@ -84,18 +110,64 @@ pub(super) struct SessionAudioOutput {
pub(super) enum SessionAudioVoice {
#[serde(rename = "fathom")]
Fathom,
#[serde(rename = "alloy")]
Alloy,
#[serde(rename = "marin")]
Marin,
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionNoiseReduction {
#[serde(rename = "type")]
pub(super) r#type: NoiseReductionType,
}
#[derive(Debug, Clone, Copy, Serialize)]
#[serde(rename_all = "snake_case")]
pub(super) enum NoiseReductionType {
NearField,
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionTurnDetection {
#[serde(rename = "type")]
pub(super) r#type: TurnDetectionType,
pub(super) interrupt_response: bool,
pub(super) create_response: bool,
}
#[derive(Debug, Clone, Copy, Serialize)]
#[serde(rename_all = "snake_case")]
pub(super) enum TurnDetectionType {
ServerVad,
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionAudioOutputFormat {
#[serde(rename = "type")]
pub(super) r#type: AudioFormatType,
pub(super) rate: u32,
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct ConversationMessageItem {
#[serde(rename = "type")]
pub(super) kind: String,
pub(super) role: String,
pub(super) r#type: ConversationItemType,
pub(super) role: ConversationRole,
pub(super) content: Vec<ConversationItemContent>,
}
#[derive(Debug, Clone, Copy, Serialize)]
#[serde(rename_all = "snake_case")]
pub(super) enum ConversationItemType {
Message,
FunctionCallOutput,
}
#[derive(Debug, Clone, Copy, Serialize)]
#[serde(rename_all = "snake_case")]
pub(super) enum ConversationRole {
User,
}
#[derive(Debug, Clone, Serialize)]
#[serde(untagged)]
pub(super) enum ConversationItemPayload {
@ -106,7 +178,7 @@ pub(super) enum ConversationItemPayload {
#[derive(Debug, Clone, Serialize)]
pub(super) struct ConversationFunctionCallOutputItem {
#[serde(rename = "type")]
pub(super) kind: String,
pub(super) r#type: ConversationItemType,
pub(super) call_id: String,
pub(super) output: String,
}
@ -114,19 +186,32 @@ pub(super) struct ConversationFunctionCallOutputItem {
#[derive(Debug, Clone, Serialize)]
pub(super) struct ConversationItemContent {
#[serde(rename = "type")]
pub(super) kind: String,
pub(super) r#type: ConversationContentType,
pub(super) text: String,
}
#[derive(Debug, Clone, Copy, Serialize)]
#[serde(rename_all = "snake_case")]
pub(super) enum ConversationContentType {
Text,
InputText,
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionFunctionTool {
#[serde(rename = "type")]
pub(super) kind: String,
pub(super) r#type: SessionToolType,
pub(super) name: String,
pub(super) description: String,
pub(super) parameters: Value,
}
#[derive(Debug, Clone, Copy, Serialize)]
#[serde(rename_all = "snake_case")]
pub(super) enum SessionToolType {
Function,
}
pub(super) fn parse_realtime_event(
payload: &str,
event_parser: RealtimeEventParser,

1
codex-rs/codex-api/src/endpoint/realtime_websocket/protocol_v1.rs
View file

@ -35,6 +35,7 @@ pub(super) fn parse_realtime_event_v1(payload: &str) -> Option<RealtimeEvent> {
.get("samples_per_channel")
.and_then(Value::as_u64)
.and_then(|value| u32::try_from(value).ok()),
item_id: None,
}))
}
"conversation.input_transcript.delta" => {

60
codex-rs/codex-api/src/endpoint/realtime_websocket/protocol_v2.rs
View file

@ -5,6 +5,8 @@ use crate::endpoint::realtime_websocket::protocol_common::parse_transcript_delta
use codex_protocol::protocol::RealtimeAudioFrame;
use codex_protocol::protocol::RealtimeEvent;
use codex_protocol::protocol::RealtimeHandoffRequested;
use codex_protocol::protocol::RealtimeInputAudioSpeechStarted;
use codex_protocol::protocol::RealtimeResponseCancelled;
use serde_json::Map as JsonMap;
use serde_json::Value;
use tracing::debug;
@ -19,7 +21,9 @@ pub(super) fn parse_realtime_event_v2(payload: &str) -> Option<RealtimeEvent> {
match message_type.as_str() {
"session.updated" => parse_session_updated_event(&parsed),
"response.output_audio.delta" => parse_output_audio_delta_event(&parsed),
"response.output_audio.delta" | "response.audio.delta" => {
parse_output_audio_delta_event(&parsed)
}
"conversation.item.input_audio_transcription.delta" => {
parse_transcript_delta_event(&parsed, "delta").map(RealtimeEvent::InputTranscriptDelta)
}
@ -30,11 +34,37 @@ pub(super) fn parse_realtime_event_v2(payload: &str) -> Option<RealtimeEvent> {
"response.output_text.delta" | "response.output_audio_transcript.delta" => {
parse_transcript_delta_event(&parsed, "delta").map(RealtimeEvent::OutputTranscriptDelta)
}
"input_audio_buffer.speech_started" => Some(RealtimeEvent::InputAudioSpeechStarted(
RealtimeInputAudioSpeechStarted {
item_id: parsed
.get("item_id")
.and_then(Value::as_str)
.map(str::to_string),
},
)),
"conversation.item.added" => parsed
.get("item")
.cloned()
.map(RealtimeEvent::ConversationItemAdded),
"conversation.item.done" => parse_conversation_item_done_event(&parsed),
"response.created" => Some(RealtimeEvent::ConversationItemAdded(parsed)),
"response.done" => parse_response_done_event(parsed),
"response.cancelled" => Some(RealtimeEvent::ResponseCancelled(
RealtimeResponseCancelled {
response_id: parsed
.get("response")
.and_then(Value::as_object)
.and_then(|response| response.get("id"))
.and_then(Value::as_str)
.map(str::to_string)
.or_else(|| {
parsed
.get("response_id")
.and_then(Value::as_str)
.map(str::to_string)
}),
},
)),
"error" => parse_error_event(&parsed),
_ => {
debug!("received unsupported realtime v2 event type: {message_type}, data: {payload}");
@ -67,6 +97,10 @@ fn parse_output_audio_delta_event(parsed: &Value) -> Option<RealtimeEvent> {
.get("samples_per_channel")
.and_then(Value::as_u64)
.and_then(|value| u32::try_from(value).ok()),
item_id: parsed
.get("item_id")
.and_then(Value::as_str)
.map(str::to_string),
}))
}
@ -82,6 +116,30 @@ fn parse_conversation_item_done_event(parsed: &Value) -> Option<RealtimeEvent> {
.map(|item_id| RealtimeEvent::ConversationItemDone { item_id })
}
fn parse_response_done_event(parsed: Value) -> Option<RealtimeEvent> {
if let Some(handoff) = parse_response_done_handoff_requested_event(&parsed) {
return Some(handoff);
}
Some(RealtimeEvent::ConversationItemAdded(parsed))
}
fn parse_response_done_handoff_requested_event(parsed: &Value) -> Option<RealtimeEvent> {
let item = parsed
.get("response")
.and_then(Value::as_object)
.and_then(|response| response.get("output"))
.and_then(Value::as_array)?
.iter()
.find(|item| {
item.get("type").and_then(Value::as_str) == Some("function_call")
&& item.get("name").and_then(Value::as_str) == Some(CODEX_TOOL_NAME)
})?
.as_object()?;
parse_handoff_requested_event(item)
}
fn parse_handoff_requested_event(item: &JsonMap<String, Value>) -> Option<RealtimeEvent> {
let item_type = item.get("type").and_then(Value::as_str);
let item_name = item.get("name").and_then(Value::as_str);

3
codex-rs/codex-api/tests/realtime_websocket_e2e.rs
View file

@ -170,6 +170,7 @@ async fn realtime_ws_e2e_session_create_and_event_flow() {
sample_rate: 48000,
num_channels: 1,
samples_per_channel: Some(960),
item_id: None,
})
.await
.expect("send audio");
@ -186,6 +187,7 @@ async fn realtime_ws_e2e_session_create_and_event_flow() {
sample_rate: 48000,
num_channels: 1,
samples_per_channel: None,
item_id: None,
})
);
@ -254,6 +256,7 @@ async fn realtime_ws_e2e_send_while_next_event_waits() {
sample_rate: 48000,
num_channels: 1,
samples_per_channel: Some(960),
item_id: None,
}),
)
.await

3
codex-rs/core/src/codex.rs
View file

@ -2614,6 +2614,9 @@ impl Session {
if !matches!(msg, EventMsg::TurnComplete(_)) {
return;
}
if let Err(err) = self.conversation.handoff_complete().await {
debug!("failed to finalize realtime handoff output: {err}");
}
self.conversation.clear_active_handoff().await;
}

1
codex-rs/core/src/codex_tests.rs
View file

@ -2735,6 +2735,7 @@ fn submission_dispatch_span_uses_debug_for_realtime_audio() {
sample_rate: 16_000,
num_channels: 1,
samples_per_channel: Some(160),
item_id: None,
},
}),
trace: None,

366
codex-rs/core/src/realtime_conversation.rs
View file

@ -11,6 +11,8 @@ use crate::realtime_context::build_realtime_startup_context;
use async_channel::Receiver;
use async_channel::Sender;
use async_channel::TrySendError;
use base64::Engine;
use base64::engine::general_purpose::STANDARD as BASE64_STANDARD;
use codex_api::Provider as ApiProvider;
use codex_api::RealtimeAudioFrame;
use codex_api::RealtimeEvent;
@ -34,6 +36,8 @@ use codex_protocol::protocol::RealtimeHandoffRequested;
use http::HeaderMap;
use http::HeaderValue;
use http::header::AUTHORIZATION;
use serde_json::Value;
use serde_json::json;
use std::sync::Arc;
use std::sync::atomic::AtomicBool;
use std::sync::atomic::Ordering;
@ -49,51 +53,72 @@ const USER_TEXT_IN_QUEUE_CAPACITY: usize = 64;
const HANDOFF_OUT_QUEUE_CAPACITY: usize = 64;
const OUTPUT_EVENTS_QUEUE_CAPACITY: usize = 256;
const REALTIME_STARTUP_CONTEXT_TOKEN_BUDGET: usize = 5_000;
const ACTIVE_RESPONSE_CONFLICT_ERROR_PREFIX: &str =
"Conversation already has an active response in progress:";
pub(crate) struct RealtimeConversationManager {
state: Mutex<Option<ConversationState>>,
}
#[derive(Clone, Copy, Debug, PartialEq, Eq)]
enum RealtimeSessionKind {
V1,
V2,
}
#[derive(Clone, Debug)]
struct RealtimeHandoffState {
output_tx: Sender<HandoffOutput>,
active_handoff: Arc<Mutex<Option<String>>>,
last_output_text: Arc<Mutex<Option<String>>>,
session_kind: RealtimeSessionKind,
}
#[derive(Debug, PartialEq, Eq)]
struct HandoffOutput {
handoff_id: String,
output_text: String,
enum HandoffOutput {
ImmediateAppend {
handoff_id: String,
output_text: String,
},
FinalToolCall {
handoff_id: String,
output_text: String,
},
}
#[derive(Debug, PartialEq, Eq)]
struct OutputAudioState {
item_id: String,
audio_end_ms: u32,
}
struct RealtimeInputTask {
writer: RealtimeWebsocketWriter,
events: RealtimeWebsocketEvents,
user_text_rx: Receiver<String>,
handoff_output_rx: Receiver<HandoffOutput>,
audio_rx: Receiver<RealtimeAudioFrame>,
events_tx: Sender<RealtimeEvent>,
handoff_state: RealtimeHandoffState,
session_kind: RealtimeSessionKind,
}
impl RealtimeHandoffState {
fn new(output_tx: Sender<HandoffOutput>) -> Self {
fn new(output_tx: Sender<HandoffOutput>, session_kind: RealtimeSessionKind) -> Self {
Self {
output_tx,
active_handoff: Arc::new(Mutex::new(None)),
last_output_text: Arc::new(Mutex::new(None)),
session_kind,
}
}
async fn send_output(&self, output_text: String) -> CodexResult<()> {
let Some(handoff_id) = self.active_handoff.lock().await.clone() else {
return Ok(());
};
self.output_tx
.send(HandoffOutput {
handoff_id,
output_text,
})
.await
.map_err(|_| CodexErr::InvalidRequest("conversation is not running".to_string()))?;
Ok(())
}
}
#[allow(dead_code)]
struct ConversationState {
audio_tx: Sender<RealtimeAudioFrame>,
user_text_tx: Sender<String>,
writer: RealtimeWebsocketWriter,
handoff: RealtimeHandoffState,
task: JoinHandle<()>,
realtime_active: Arc<AtomicBool>,
@ -129,6 +154,10 @@ impl RealtimeConversationManager {
state.task.abort();
let _ = state.task.await;
}
let session_kind = match session_config.event_parser {
RealtimeEventParser::V1 => RealtimeSessionKind::V1,
RealtimeEventParser::RealtimeV2 => RealtimeSessionKind::V2,
};
let client = RealtimeWebsocketClient::new(api_provider);
let connection = client
@ -152,21 +181,23 @@ impl RealtimeConversationManager {
async_channel::bounded::<RealtimeEvent>(OUTPUT_EVENTS_QUEUE_CAPACITY);
let realtime_active = Arc::new(AtomicBool::new(true));
let handoff = RealtimeHandoffState::new(handoff_output_tx);
let task = spawn_realtime_input_task(
writer,
let handoff = RealtimeHandoffState::new(handoff_output_tx, session_kind);
let task = spawn_realtime_input_task(RealtimeInputTask {
writer: writer.clone(),
events,
user_text_rx,
handoff_output_rx,
audio_rx,
events_tx,
handoff.clone(),
);
handoff_state: handoff.clone(),
session_kind,
});
let mut guard = self.state.lock().await;
*guard = Some(ConversationState {
audio_tx,
user_text_tx,
writer,
handoff,
task,
realtime_active: Arc::clone(&realtime_active),
@ -228,7 +259,51 @@ impl RealtimeConversationManager {
state.handoff.clone()
};
handoff.send_output(output_text).await
let Some(handoff_id) = handoff.active_handoff.lock().await.clone() else {
return Ok(());
};
*handoff.last_output_text.lock().await = Some(output_text.clone());
if matches!(handoff.session_kind, RealtimeSessionKind::V1) {
handoff
.output_tx
.send(HandoffOutput::ImmediateAppend {
handoff_id,
output_text,
})
.await
.map_err(|_| CodexErr::InvalidRequest("conversation is not running".to_string()))?;
}
Ok(())
}
pub(crate) async fn handoff_complete(&self) -> CodexResult<()> {
let handoff = {
let guard = self.state.lock().await;
guard.as_ref().map(|state| state.handoff.clone())
};
let Some(handoff) = handoff else {
return Ok(());
};
if matches!(handoff.session_kind, RealtimeSessionKind::V1) {
return Ok(());
}
let Some(handoff_id) = handoff.active_handoff.lock().await.clone() else {
return Ok(());
};
let Some(output_text) = handoff.last_output_text.lock().await.clone() else {
return Ok(());
};
handoff
.output_tx
.send(HandoffOutput::FinalToolCall {
handoff_id,
output_text,
})
.await
.map_err(|_| CodexErr::InvalidRequest("conversation is not running".to_string()))
}
pub(crate) async fn active_handoff_id(&self) -> Option<String> {
@ -246,6 +321,7 @@ impl RealtimeConversationManager {
};
if let Some(handoff) = handoff {
*handoff.active_handoff.lock().await = None;
*handoff.last_output_text.lock().await = None;
}
}
@ -467,7 +543,6 @@ pub(crate) async fn handle_text(
params: ConversationTextParams,
) {
debug!(text = %params.text, "[realtime-text] appending realtime conversation text input");
if let Err(err) = sess.conversation.text_in(params.text).await {
error!("failed to append realtime text: {err}");
send_conversation_error(sess, sub_id, err.to_string(), CodexErrorInfo::BadRequest).await;
@ -491,16 +566,23 @@ pub(crate) async fn handle_close(sess: &Arc<Session>, sub_id: String) {
}
}
fn spawn_realtime_input_task(
writer: RealtimeWebsocketWriter,
events: RealtimeWebsocketEvents,
user_text_rx: Receiver<String>,
handoff_output_rx: Receiver<HandoffOutput>,
audio_rx: Receiver<RealtimeAudioFrame>,
events_tx: Sender<RealtimeEvent>,
handoff_state: RealtimeHandoffState,
) -> JoinHandle<()> {
fn spawn_realtime_input_task(input: RealtimeInputTask) -> JoinHandle<()> {
let RealtimeInputTask {
writer,
events,
user_text_rx,
handoff_output_rx,
audio_rx,
events_tx,
handoff_state,
session_kind,
} = input;
tokio::spawn(async move {
let mut pending_response_create = false;
let mut response_in_progress = false;
let mut output_audio_state: Option<OutputAudioState> = None;
loop {
tokio::select! {
text = user_text_rx.recv() => {
@ -511,23 +593,66 @@ fn spawn_realtime_input_task(
warn!("failed to send input text: {mapped_error}");
break;
}
if matches!(session_kind, RealtimeSessionKind::V2) {
if response_in_progress {
pending_response_create = true;
} else if let Err(err) = writer.send_response_create().await {
let mapped_error = map_api_error(err);
warn!("failed to send text response.create: {mapped_error}");
break;
} else {
pending_response_create = false;
response_in_progress = true;
}
}
}
Err(_) => break,
}
}
handoff_output = handoff_output_rx.recv() => {
match handoff_output {
Ok(HandoffOutput {
handoff_id,
output_text,
}) => {
if let Err(err) = writer
.send_conversation_handoff_append(handoff_id, output_text)
.await
{
let mapped_error = map_api_error(err);
warn!("failed to send handoff output: {mapped_error}");
break;
Ok(handoff_output) => {
match handoff_output {
HandoffOutput::ImmediateAppend {
handoff_id,
output_text,
} => {
if let Err(err) = writer
.send_conversation_handoff_append(handoff_id, output_text)
.await
{
let mapped_error = map_api_error(err);
warn!("failed to send handoff output: {mapped_error}");
break;
}
}
HandoffOutput::FinalToolCall {
handoff_id,
output_text,
} => {
if let Err(err) = writer
.send_conversation_handoff_append(handoff_id, output_text)
.await
{
let mapped_error = map_api_error(err);
warn!("failed to send handoff output: {mapped_error}");
break;
}
if matches!(session_kind, RealtimeSessionKind::V2) {
if response_in_progress {
pending_response_create = true;
} else if let Err(err) = writer.send_response_create().await {
let mapped_error = map_api_error(err);
warn!(
"failed to send handoff response.create: {mapped_error}"
);
break;
} else {
pending_response_create = false;
response_in_progress = true;
}
}
}
}
}
Err(_) => break,
@ -536,12 +661,108 @@ fn spawn_realtime_input_task(
event = events.next_event() => {
match event {
Ok(Some(event)) => {
if let RealtimeEvent::HandoffRequested(handoff) = &event {
*handoff_state.active_handoff.lock().await =
Some(handoff.handoff_id.clone());
let mut should_stop = false;
let mut forward_event = true;
match &event {
RealtimeEvent::ConversationItemAdded(item) => {
match item.get("type").and_then(Value::as_str) {
Some("response.created")
if matches!(session_kind, RealtimeSessionKind::V2) =>
{
response_in_progress = true;
}
Some("response.done")
if matches!(session_kind, RealtimeSessionKind::V2) =>
{
response_in_progress = false;
output_audio_state = None;
if pending_response_create {
if let Err(err) = writer.send_response_create().await {
let mapped_error = map_api_error(err);
warn!(
"failed to send deferred response.create: {mapped_error}"
);
break;
}
pending_response_create = false;
response_in_progress = true;
}
}
_ => {}
}
}
RealtimeEvent::AudioOut(frame) => {
if matches!(session_kind, RealtimeSessionKind::V2) {
update_output_audio_state(&mut output_audio_state, frame);
}
}
RealtimeEvent::InputAudioSpeechStarted(event) => {
if matches!(session_kind, RealtimeSessionKind::V2)
&& let Some(output_audio_state) =
output_audio_state.take()
&& event
.item_id
.as_deref()
.is_none_or(|item_id| item_id == output_audio_state.item_id)
&& let Err(err) = writer
.send_payload(json!({
"type": "conversation.item.truncate",
"item_id": output_audio_state.item_id,
"content_index": 0,
"audio_end_ms": output_audio_state.audio_end_ms,
})
.to_string())
.await
{
let mapped_error = map_api_error(err);
warn!("failed to truncate realtime audio: {mapped_error}");
}
}
RealtimeEvent::ResponseCancelled(_) => {
response_in_progress = false;
output_audio_state = None;
if matches!(session_kind, RealtimeSessionKind::V2)
&& pending_response_create
{
if let Err(err) = writer.send_response_create().await {
let mapped_error = map_api_error(err);
warn!(
"failed to send deferred response.create after cancellation: {mapped_error}"
);
break;
}
pending_response_create = false;
response_in_progress = true;
}
}
RealtimeEvent::HandoffRequested(handoff) => {
*handoff_state.active_handoff.lock().await =
Some(handoff.handoff_id.clone());
*handoff_state.last_output_text.lock().await = None;
response_in_progress = false;
output_audio_state = None;
}
RealtimeEvent::Error(message)
if matches!(session_kind, RealtimeSessionKind::V2)
&& message.starts_with(ACTIVE_RESPONSE_CONFLICT_ERROR_PREFIX) =>
{
warn!(
"realtime rejected response.create because a response is already in progress; deferring follow-up response.create"
);
pending_response_create = true;
response_in_progress = true;
forward_event = false;
}
RealtimeEvent::Error(_) => {
should_stop = true;
}
RealtimeEvent::SessionUpdated { .. }
| RealtimeEvent::InputTranscriptDelta(_)
| RealtimeEvent::OutputTranscriptDelta(_)
| RealtimeEvent::ConversationItemDone { .. } => {}
}
let should_stop = matches!(&event, RealtimeEvent::Error(_));
if events_tx.send(event).await.is_err() {
if forward_event && events_tx.send(event).await.is_err() {
break;
}
if should_stop {
@ -588,6 +809,49 @@ fn spawn_realtime_input_task(
})
}
fn update_output_audio_state(
output_audio_state: &mut Option<OutputAudioState>,
frame: &RealtimeAudioFrame,
) {
let Some(item_id) = frame.item_id.clone() else {
return;
};
let audio_end_ms = audio_duration_ms(frame);
if audio_end_ms == 0 {
return;
}
if let Some(current) = output_audio_state.as_mut()
&& current.item_id == item_id
{
current.audio_end_ms = current.audio_end_ms.saturating_add(audio_end_ms);
return;
}
*output_audio_state = Some(OutputAudioState {
item_id,
audio_end_ms,
});
}
fn audio_duration_ms(frame: &RealtimeAudioFrame) -> u32 {
let Some(samples_per_channel) = frame
.samples_per_channel
.or_else(|| decoded_samples_per_channel(frame))
else {
return 0;
};
let sample_rate = u64::from(frame.sample_rate.max(1));
((u64::from(samples_per_channel) * 1_000) / sample_rate) as u32
}
fn decoded_samples_per_channel(frame: &RealtimeAudioFrame) -> Option<u32> {
let bytes = BASE64_STANDARD.decode(&frame.data).ok()?;
let channels = usize::from(frame.num_channels.max(1));
let samples = bytes.len().checked_div(2)?.checked_div(channels)?;
u32::try_from(samples).ok()
}
async fn send_conversation_error(
sess: &Arc<Session>,
sub_id: String,

48
codex-rs/core/src/realtime_conversation_tests.rs
View file

@ -1,5 +1,5 @@
use super::HandoffOutput;
use super::RealtimeHandoffState;
use super::RealtimeSessionKind;
use super::realtime_text_from_handoff_request;
use async_channel::bounded;
use codex_protocol::protocol::RealtimeHandoffRequested;
@ -57,7 +57,7 @@ fn ignores_empty_handoff_request_input_transcript() {
#[tokio::test]
async fn clears_active_handoff_explicitly() {
let (tx, _rx) = bounded(1);
let state = RealtimeHandoffState::new(tx);
let state = RealtimeHandoffState::new(tx, RealtimeSessionKind::V1);
*state.active_handoff.lock().await = Some("handoff_1".to_string());
assert_eq!(
@ -68,47 +68,3 @@ async fn clears_active_handoff_explicitly() {
*state.active_handoff.lock().await = None;
assert_eq!(state.active_handoff.lock().await.clone(), None);
}
#[tokio::test]
async fn sends_multiple_handoff_outputs_until_cleared() {
let (tx, rx) = bounded(4);
let state = RealtimeHandoffState::new(tx);
state
.send_output("ignored".to_string())
.await
.expect("send");
assert!(rx.is_empty());
*state.active_handoff.lock().await = Some("handoff_1".to_string());
state.send_output("result".to_string()).await.expect("send");
state
.send_output("result 2".to_string())
.await
.expect("send");
let output_1 = rx.recv().await.expect("recv");
assert_eq!(
output_1,
HandoffOutput {
handoff_id: "handoff_1".to_string(),
output_text: "result".to_string(),
}
);
let output_2 = rx.recv().await.expect("recv");
assert_eq!(
output_2,
HandoffOutput {
handoff_id: "handoff_1".to_string(),
output_text: "result 2".to_string(),
}
);
*state.active_handoff.lock().await = None;
state
.send_output("ignored after clear".to_string())
.await
.expect("send");
assert!(rx.is_empty());
}

5
codex-rs/core/tests/suite/realtime_conversation.rs
View file

@ -176,6 +176,7 @@ async fn conversation_start_audio_text_close_round_trip() -> Result<()> {
sample_rate: 24000,
num_channels: 1,
samples_per_channel: Some(480),
item_id: None,
},
}))
.await?;
@ -409,6 +410,7 @@ async fn conversation_audio_before_start_emits_error() -> Result<()> {
sample_rate: 24000,
num_channels: 1,
samples_per_channel: Some(480),
item_id: None,
},
}))
.await?;
@ -518,6 +520,7 @@ async fn conversation_second_start_replaces_runtime() -> Result<()> {
sample_rate: 24000,
num_channels: 1,
samples_per_channel: Some(480),
item_id: None,
},
}))
.await?;
@ -1469,6 +1472,7 @@ async fn inbound_handoff_request_clears_active_transcript_after_each_handoff() -
sample_rate: 24000,
num_channels: 1,
samples_per_channel: Some(480),
item_id: None,
},
}))
.await?;
@ -1954,6 +1958,7 @@ async fn inbound_handoff_request_steers_active_turn() -> Result<()> {
sample_rate: 24000,
num_channels: 1,
samples_per_channel: Some(480),
item_id: None,
},
}))
.await?;

15
codex-rs/protocol/src/protocol.rs
View file

@ -139,6 +139,8 @@ pub struct RealtimeAudioFrame {
pub num_channels: u16,
#[serde(skip_serializing_if = "Option::is_none")]
pub samples_per_channel: Option<u32>,
#[serde(skip_serializing_if = "Option::is_none")]
pub item_id: Option<String>,
}
#[derive(Debug, Clone, Deserialize, Serialize, PartialEq, Eq, JsonSchema, TS)]
@ -160,15 +162,27 @@ pub struct RealtimeHandoffRequested {
pub active_transcript: Vec<RealtimeTranscriptEntry>,
}
#[derive(Debug, Clone, Deserialize, Serialize, PartialEq, Eq, JsonSchema, TS)]
pub struct RealtimeInputAudioSpeechStarted {
pub item_id: Option<String>,
}
#[derive(Debug, Clone, Deserialize, Serialize, PartialEq, Eq, JsonSchema, TS)]
pub struct RealtimeResponseCancelled {
pub response_id: Option<String>,
}
#[derive(Debug, Clone, Deserialize, Serialize, PartialEq, Eq, JsonSchema, TS)]
pub enum RealtimeEvent {
SessionUpdated {
session_id: String,
instructions: Option<String>,
},
InputAudioSpeechStarted(RealtimeInputAudioSpeechStarted),
InputTranscriptDelta(RealtimeTranscriptDelta),
OutputTranscriptDelta(RealtimeTranscriptDelta),
AudioOut(RealtimeAudioFrame),
ResponseCancelled(RealtimeResponseCancelled),
ConversationItemAdded(Value),
ConversationItemDone {
item_id: String,
@ -4078,6 +4092,7 @@ mod tests {
sample_rate: 24_000,
num_channels: 1,
samples_per_channel: Some(480),
item_id: None,
},
});
let start = Op::RealtimeConversationStart(ConversationStartParams {

2
codex-rs/tui/src/chatwidget/realtime.rs
View file

@ -264,9 +264,11 @@ impl ChatWidget {
RealtimeEvent::SessionUpdated { session_id, .. } => {
self.realtime_conversation.session_id = Some(session_id);
}
RealtimeEvent::InputAudioSpeechStarted(_) => {}
RealtimeEvent::InputTranscriptDelta(_) => {}
RealtimeEvent::OutputTranscriptDelta(_) => {}
RealtimeEvent::AudioOut(frame) => self.enqueue_realtime_audio_out(&frame),
RealtimeEvent::ResponseCancelled(_) => {}
RealtimeEvent::ConversationItemAdded(_item) => {}
RealtimeEvent::ConversationItemDone { .. } => {}
RealtimeEvent::HandoffRequested(_) => {}

1
codex-rs/tui/src/voice.rs
View file

@ -428,6 +428,7 @@ fn send_realtime_audio_chunk(
sample_rate: MODEL_AUDIO_SAMPLE_RATE,
num_channels: MODEL_AUDIO_CHANNELS,
samples_per_channel: Some(samples_per_channel),
item_id: None,
},
},
)));

12
codex-rs/tui_app_server/src/chatwidget/realtime.rs
View file

@ -268,9 +268,11 @@ impl ChatWidget {
RealtimeEvent::SessionUpdated { session_id, .. } => {
self.realtime_conversation.session_id = Some(session_id);
}
RealtimeEvent::InputAudioSpeechStarted(_) => self.interrupt_realtime_audio_playback(),
RealtimeEvent::InputTranscriptDelta(_) => {}
RealtimeEvent::OutputTranscriptDelta(_) => {}
RealtimeEvent::AudioOut(frame) => self.enqueue_realtime_audio_out(&frame),
RealtimeEvent::ResponseCancelled(_) => self.interrupt_realtime_audio_playback(),
RealtimeEvent::ConversationItemAdded(_item) => {}
RealtimeEvent::ConversationItemDone { .. } => {}
RealtimeEvent::HandoffRequested(_) => {}
@ -313,6 +315,16 @@ impl ChatWidget {
}
}
#[cfg(not(target_os = "linux"))]
fn interrupt_realtime_audio_playback(&mut self) {
if let Some(player) = &self.realtime_conversation.audio_player {
player.clear();
}
}
#[cfg(target_os = "linux")]
fn interrupt_realtime_audio_playback(&mut self) {}
#[cfg(not(target_os = "linux"))]
fn start_realtime_local_audio(&mut self) {
if self.realtime_conversation.capture_stop_flag.is_some() {

1
codex-rs/tui_app_server/src/voice.rs
View file

@ -426,6 +426,7 @@ fn send_realtime_audio_chunk(
sample_rate: MODEL_AUDIO_SAMPLE_RATE,
num_channels: MODEL_AUDIO_CHANNELS,
samples_per_channel: Some(samples_per_channel),
item_id: None,
},
});
}