Switch realtime TUI to WebRTC transport

This commit is contained in:
Justin Uberti
2026-03-20 16:08:45 -07:00
parent ddda43919f
commit fc01075f74
20 changed files with 1183 additions and 3489 deletions

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@@ -20,7 +20,9 @@ tokio-tungstenite = { workspace = true }
tungstenite = { workspace = true }
tracing = { workspace = true }
eventsource-stream = { workspace = true }
libwebrtc = "0.3.26"
regex-lite = { workspace = true }
reqwest = { workspace = true, features = ["json", "multipart"] }
tokio-util = { workspace = true, features = ["codec"] }
url = { workspace = true }
@@ -30,7 +32,6 @@ assert_matches = { workspace = true }
pretty_assertions = { workspace = true }
tokio-test = { workspace = true }
wiremock = { workspace = true }
reqwest = { workspace = true }
[lints]
workspace = true

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@@ -0,0 +1,5 @@
fn main() {
if std::env::var("CARGO_CFG_TARGET_OS").as_deref() == Ok("macos") {
println!("cargo:rustc-link-arg=-ObjC");
}
}

View File

@@ -1,7 +1,8 @@
pub mod compact;
pub mod memories;
pub mod models;
pub mod realtime_websocket;
pub mod realtime_webrtc;
mod realtime_websocket;
pub mod responses;
pub mod responses_websocket;
mod session;

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@@ -0,0 +1,544 @@
use crate::endpoint::realtime_websocket::parse_realtime_event;
use crate::error::ApiError;
use crate::provider::Provider;
use codex_protocol::protocol::RealtimeEvent;
use http::HeaderMap;
use libwebrtc::MediaType;
use libwebrtc::data_channel::DataChannel;
use libwebrtc::data_channel::DataChannelInit;
use libwebrtc::data_channel::DataChannelState;
use libwebrtc::peer_connection::OfferOptions;
use libwebrtc::peer_connection::PeerConnection;
use libwebrtc::peer_connection_factory::PeerConnectionFactory;
use libwebrtc::peer_connection_factory::RtcConfiguration;
use libwebrtc::peer_connection_factory::native::PeerConnectionFactoryExt;
use libwebrtc::rtp_transceiver::RtpTransceiverDirection;
use libwebrtc::rtp_transceiver::RtpTransceiverInit;
use libwebrtc::session_description::SdpType;
use libwebrtc::session_description::SessionDescription;
use reqwest::Client;
use reqwest::multipart::Form;
use serde_json::json;
use std::sync::Arc;
use std::sync::Mutex as StdMutex;
use std::sync::atomic::AtomicBool;
use std::sync::atomic::Ordering;
use tokio::sync::Mutex;
use tokio::sync::mpsc;
use tokio::sync::oneshot;
use tokio::time::Duration;
use tracing::debug;
use tracing::info;
use tracing::warn;
use url::Url;
const REALTIME_CALLS_PATH: &str = "/v1/realtime/calls";
const REALTIME_DATA_CHANNEL_LABEL: &str = "oai-events";
const REALTIME_VOICE: &str = "marin";
#[derive(Debug, Clone, PartialEq, Eq)]
pub struct RealtimeSessionConfig {
pub instructions: String,
pub model: Option<String>,
pub session_id: Option<String>,
}
pub struct RealtimeWebrtcConnection {
writer: RealtimeWebrtcWriter,
events: RealtimeWebrtcEvents,
}
#[derive(Clone)]
pub struct RealtimeWebrtcWriter {
peer_connection: PeerConnection,
data_channel: DataChannel,
is_closed: Arc<AtomicBool>,
}
#[derive(Clone)]
pub struct RealtimeWebrtcEvents {
rx_event: Arc<Mutex<mpsc::UnboundedReceiver<RealtimeEvent>>>,
is_closed: Arc<AtomicBool>,
}
pub struct RealtimeWebrtcClient {
provider: Provider,
}
impl RealtimeWebrtcConnection {
pub async fn send_conversation_item_create(&self, text: String) -> Result<(), ApiError> {
self.writer.send_conversation_item_create(text).await
}
pub async fn send_conversation_handoff_append(
&self,
handoff_id: String,
output_text: String,
) -> Result<(), ApiError> {
self.writer
.send_conversation_handoff_append(handoff_id, output_text)
.await
}
pub async fn send_function_call_output(
&self,
call_id: String,
output_text: String,
) -> Result<(), ApiError> {
self.writer
.send_function_call_output(call_id, output_text)
.await
}
pub async fn send_response_create(&self) -> Result<(), ApiError> {
self.writer.send_response_create().await
}
pub async fn close(&self) -> Result<(), ApiError> {
self.writer.close().await
}
pub async fn next_event(&self) -> Result<Option<RealtimeEvent>, ApiError> {
self.events.next_event().await
}
pub fn writer(&self) -> RealtimeWebrtcWriter {
self.writer.clone()
}
pub fn events(&self) -> RealtimeWebrtcEvents {
self.events.clone()
}
fn new(
peer_connection: PeerConnection,
data_channel: DataChannel,
rx_event: mpsc::UnboundedReceiver<RealtimeEvent>,
is_closed: Arc<AtomicBool>,
) -> Self {
Self {
writer: RealtimeWebrtcWriter {
peer_connection,
data_channel,
is_closed: Arc::clone(&is_closed),
},
events: RealtimeWebrtcEvents {
rx_event: Arc::new(Mutex::new(rx_event)),
is_closed,
},
}
}
}
impl RealtimeWebrtcWriter {
pub async fn send_conversation_item_create(&self, text: String) -> Result<(), ApiError> {
self.send_json(json!({
"type": "conversation.item.create",
"item": {
"type": "message",
"role": "user",
"content": [{
"type": "input_text",
"text": text,
}],
},
}))
.await
}
pub async fn send_conversation_handoff_append(
&self,
_handoff_id: String,
output_text: String,
) -> Result<(), ApiError> {
self.send_json(json!({
"type": "conversation.item.create",
"item": {
"type": "message",
"role": "assistant",
"content": [{
"type": "output_text",
"text": output_text,
}],
},
}))
.await
}
pub async fn send_function_call_output(
&self,
call_id: String,
output_text: String,
) -> Result<(), ApiError> {
self.send_json(json!({
"type": "conversation.item.create",
"item": {
"type": "function_call_output",
"call_id": call_id,
"output": json!({
"content": output_text,
}).to_string(),
},
}))
.await
}
pub async fn send_response_create(&self) -> Result<(), ApiError> {
self.send_json(json!({ "type": "response.create" })).await
}
pub async fn close(&self) -> Result<(), ApiError> {
if self.is_closed.swap(true, Ordering::SeqCst) {
return Ok(());
}
self.data_channel.close();
self.peer_connection.close();
Ok(())
}
async fn send_json(&self, payload: serde_json::Value) -> Result<(), ApiError> {
if self.is_closed.load(Ordering::SeqCst) {
return Err(ApiError::Stream(
"realtime WebRTC connection is closed".to_string(),
));
}
let serialized = serde_json::to_vec(&payload).map_err(|err| {
ApiError::Stream(format!("failed to serialize realtime event: {err}"))
})?;
self.data_channel.send(&serialized, false).map_err(|err| {
ApiError::Stream(format!("failed to send realtime data channel event: {err}"))
})
}
}
impl RealtimeWebrtcEvents {
pub async fn next_event(&self) -> Result<Option<RealtimeEvent>, ApiError> {
if self.is_closed.load(Ordering::SeqCst) {
return Ok(None);
}
match self.rx_event.lock().await.recv().await {
Some(event) => Ok(Some(event)),
None => {
self.is_closed.store(true, Ordering::SeqCst);
Ok(None)
}
}
}
}
impl RealtimeWebrtcClient {
pub fn new(provider: Provider) -> Self {
Self { provider }
}
pub async fn connect(
&self,
config: RealtimeSessionConfig,
extra_headers: HeaderMap,
default_headers: HeaderMap,
) -> Result<RealtimeWebrtcConnection, ApiError> {
info!("initializing realtime WebRTC peer connection");
let factory = PeerConnectionFactory::with_platform_adm();
let peer_connection = factory
.create_peer_connection(RtcConfiguration::default())
.map_err(|err| {
ApiError::Stream(format!("failed to create WebRTC peer connection: {err}"))
})?;
// Negotiate an audio m-line and attach a local mic track backed by the platform ADM.
let audio_transceiver = peer_connection
.add_transceiver_for_media(
MediaType::Audio,
RtpTransceiverInit {
direction: RtpTransceiverDirection::SendRecv,
stream_ids: vec!["realtime".to_string()],
send_encodings: Vec::new(),
},
)
.map_err(|err| ApiError::Stream(format!("failed to add audio transceiver: {err}")))?;
let local_audio_source = factory.create_audio_source();
let local_audio_track = factory.create_audio_track("realtime-mic", local_audio_source);
audio_transceiver
.sender()
.set_track(Some(local_audio_track.into()))
.map_err(|err| {
ApiError::Stream(format!("failed to attach ADM audio track to sender: {err}"))
})?;
let data_channel = peer_connection
.create_data_channel(REALTIME_DATA_CHANNEL_LABEL, DataChannelInit::default())
.map_err(|err| {
ApiError::Stream(format!("failed to create realtime data channel: {err}"))
})?;
let (tx_event, rx_event) = mpsc::unbounded_channel();
let is_closed = Arc::new(AtomicBool::new(false));
let (tx_open, rx_open) = oneshot::channel::<()>();
let tx_open = Arc::new(StdMutex::new(Some(tx_open)));
{
let tx_event = tx_event.clone();
data_channel.on_message(Some(Box::new(move |buffer| {
if buffer.binary {
debug!(
payload_len = buffer.data.len(),
"ignoring binary realtime data channel message"
);
return;
}
let payload = match std::str::from_utf8(buffer.data) {
Ok(payload) => payload,
Err(err) => {
debug!("received non-utf8 realtime data channel message: {err}");
return;
}
};
if let Some(event) = parse_realtime_event(payload)
&& tx_event.send(event).is_err()
{
debug!("dropping realtime event because receiver closed");
}
})));
}
{
let is_closed = Arc::clone(&is_closed);
let tx_open = Arc::clone(&tx_open);
data_channel.on_state_change(Some(Box::new(move |state| match state {
DataChannelState::Connecting => {}
DataChannelState::Open => {
if let Ok(mut tx_open) = tx_open.lock()
&& let Some(tx_open) = tx_open.take()
{
let _ = tx_open.send(());
}
}
DataChannelState::Closing | DataChannelState::Closed => {
is_closed.store(true, Ordering::SeqCst);
}
})));
}
let offer = peer_connection
.create_offer(OfferOptions {
ice_restart: false,
offer_to_receive_audio: true,
offer_to_receive_video: false,
})
.await
.map_err(|err| ApiError::Stream(format!("failed to create WebRTC offer: {err}")))?;
peer_connection
.set_local_description(offer.clone())
.await
.map_err(|err| ApiError::Stream(format!("failed to set local description: {err}")))?;
let url = realtime_calls_url(&self.provider.base_url)?;
let headers = merge_request_headers(&self.provider.headers, extra_headers, default_headers);
info!(url = %url, "posting realtime WebRTC offer");
let http_client = Client::new();
let mut request = http_client
.post(url)
.multipart(session_form(&config, &offer)?);
for (name, value) in &headers {
request = request.header(name, value);
}
let response = request.send().await.map_err(|err| {
ApiError::Stream(format!("failed to post realtime WebRTC offer: {err}"))
})?;
let status = response.status();
let answer_sdp = response.text().await.map_err(|err| {
ApiError::Stream(format!("failed to read realtime WebRTC answer body: {err}"))
})?;
if !status.is_success() {
return Err(ApiError::Stream(format!(
"realtime WebRTC offer failed with HTTP {status}: {answer_sdp}"
)));
}
let answer = SessionDescription::parse(&answer_sdp, SdpType::Answer)
.map_err(|err| ApiError::Stream(format!("failed to parse WebRTC answer SDP: {err}")))?;
peer_connection
.set_remote_description(answer)
.await
.map_err(|err| ApiError::Stream(format!("failed to set remote description: {err}")))?;
if tokio::time::timeout(Duration::from_secs(10), rx_open)
.await
.is_err()
{
warn!("timed out waiting for realtime data channel to open");
}
Ok(RealtimeWebrtcConnection::new(
peer_connection,
data_channel,
rx_event,
is_closed,
))
}
}
fn realtime_calls_url(base_url: &str) -> Result<Url, ApiError> {
let mut url =
Url::parse(base_url).map_err(|err| ApiError::Stream(format!("invalid base URL: {err}")))?;
url.set_path(REALTIME_CALLS_PATH);
Ok(url)
}
fn session_form(
config: &RealtimeSessionConfig,
offer: &SessionDescription,
) -> Result<Form, ApiError> {
let session_json = serde_json::to_string(&session_payload(config))
.map_err(|err| ApiError::Stream(format!("failed to serialize realtime session: {err}")))?;
Ok(Form::new()
.text("sdp", offer.to_string())
.text("session", session_json))
}
fn session_payload(config: &RealtimeSessionConfig) -> serde_json::Value {
let mut session = json!({
"type": "realtime",
"instructions": config.instructions,
"output_modalities": ["audio"],
"audio": {
"output": {
"voice": REALTIME_VOICE,
},
"input": {
"turn_detection": {
"type": "server_vad",
"interrupt_response": true,
"create_response": true,
},
},
},
"tools": [
{
"type": "function",
"name": "codex",
"description": "Delegate the user's request to Codex.",
"parameters": {
"type": "object",
"properties": {
"input_transcript": {
"type": "string",
"description": "Transcript of the user's request.",
},
"send_immediately": {
"type": "boolean",
"description": "Whether Codex should receive the request immediately.",
},
},
"required": ["input_transcript"],
},
},
{
"type": "function",
"name": "cancel_current_operation",
"description": "Cancel the current Codex operation.",
"parameters": {
"type": "object",
"properties": {},
"required": [],
},
},
{
"type": "function",
"name": "turn_off_realtime_mode",
"description": "Turn off realtime voice mode.",
"parameters": {
"type": "object",
"properties": {},
"required": [],
},
},
],
"tool_choice": "auto",
});
if let Some(model) = &config.model {
session["model"] = json!(model);
}
session
}
fn merge_request_headers(
provider_headers: &HeaderMap,
extra_headers: HeaderMap,
default_headers: HeaderMap,
) -> HeaderMap {
let mut headers = provider_headers.clone();
headers.extend(extra_headers);
for (name, value) in &default_headers {
if let http::header::Entry::Vacant(entry) = headers.entry(name) {
entry.insert(value.clone());
}
}
headers
}
#[cfg(test)]
mod tests {
use super::*;
use pretty_assertions::assert_eq;
#[test]
fn realtime_calls_url_uses_webrtc_calls_path() {
assert_eq!(
realtime_calls_url("https://api.openai.com")
.expect("url")
.as_str(),
"https://api.openai.com/v1/realtime/calls"
);
}
#[test]
fn session_form_contains_sdp_and_session_payload() {
let offer = SessionDescription::parse(
"v=0\r\no=- 0 0 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\n",
SdpType::Offer,
)
.expect("offer");
let form = session_form(
&RealtimeSessionConfig {
instructions: "backend prompt".to_string(),
model: Some("gpt-realtime-1.5".to_string()),
session_id: Some("sess_123".to_string()),
},
&offer,
)
.expect("form");
let body = form.boundary().to_string();
assert!(!body.is_empty());
}
#[test]
fn session_payload_omits_session_id_from_body() {
let payload = session_payload(&RealtimeSessionConfig {
instructions: "backend prompt".to_string(),
model: Some("gpt-realtime-1.5".to_string()),
session_id: Some("sess_123".to_string()),
});
assert_eq!(payload.get("id"), None);
assert_eq!(payload.get("model"), Some(&json!("gpt-realtime-1.5")));
}
#[test]
fn peer_connection_factory_smoke() {
let factory = PeerConnectionFactory::default();
let _pc = factory
.create_peer_connection(RtcConfiguration::default())
.expect("pc");
}
}

File diff suppressed because it is too large Load Diff

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@@ -1,10 +1,3 @@
pub mod methods;
pub mod protocol;
mod protocol;
pub use codex_protocol::protocol::RealtimeAudioFrame;
pub use codex_protocol::protocol::RealtimeEvent;
pub use methods::RealtimeWebsocketClient;
pub use methods::RealtimeWebsocketConnection;
pub use methods::RealtimeWebsocketEvents;
pub use methods::RealtimeWebsocketWriter;
pub use protocol::RealtimeSessionConfig;
pub(crate) use protocol::parse_realtime_event;

View File

@@ -1,150 +1,18 @@
pub use codex_protocol::protocol::RealtimeAudioFrame;
pub use codex_protocol::protocol::RealtimeCloseRequested;
pub use codex_protocol::protocol::RealtimeEvent;
pub use codex_protocol::protocol::RealtimeHandoffRequested;
pub use codex_protocol::protocol::RealtimeInputAudioSpeechStarted;
pub use codex_protocol::protocol::RealtimeInterruptRequested;
pub use codex_protocol::protocol::RealtimeOutputAudioDelta;
pub use codex_protocol::protocol::RealtimeResponseCancelled;
pub use codex_protocol::protocol::RealtimeToolAction;
pub use codex_protocol::protocol::RealtimeToolActionRequested;
pub use codex_protocol::protocol::RealtimeTranscriptDelta;
pub use codex_protocol::protocol::RealtimeTranscriptEntry;
use codex_protocol::protocol::RealtimeCloseRequested;
use codex_protocol::protocol::RealtimeEvent;
use codex_protocol::protocol::RealtimeHandoffRequested;
use codex_protocol::protocol::RealtimeInputAudioSpeechStarted;
use codex_protocol::protocol::RealtimeInterruptRequested;
use codex_protocol::protocol::RealtimeResponseCancelled;
use codex_protocol::protocol::RealtimeToolAction;
use codex_protocol::protocol::RealtimeToolActionRequested;
use codex_protocol::protocol::RealtimeTranscriptDelta;
use serde::Deserialize;
use serde::Serialize;
use serde_json::Value;
use std::collections::BTreeMap;
use std::string::ToString;
use tracing::debug;
#[derive(Debug, Clone, PartialEq, Eq)]
pub struct RealtimeSessionConfig {
pub instructions: String,
pub model: Option<String>,
pub session_id: Option<String>,
}
#[derive(Debug, Clone, Serialize)]
#[serde(tag = "type")]
pub(super) enum RealtimeOutboundMessage {
#[serde(rename = "input_audio_buffer.append")]
InputAudioBufferAppend { audio: String },
#[serde(rename = "response.create")]
ResponseCreate,
#[serde(rename = "conversation.item.truncate")]
ConversationItemTruncate {
item_id: String,
content_index: u32,
audio_end_ms: u32,
},
#[serde(rename = "session.update")]
SessionUpdate { session: Box<SessionUpdateSession> },
#[serde(rename = "conversation.item.create")]
ConversationItemCreate { item: ConversationItem },
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionUpdateSession {
#[serde(rename = "type")]
pub(super) kind: String,
pub(super) instructions: String,
pub(super) output_modalities: Vec<String>,
pub(super) audio: SessionAudio,
pub(super) tools: Vec<SessionTool>,
pub(super) tool_choice: String,
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionAudio {
pub(super) input: SessionAudioInput,
pub(super) output: SessionAudioOutput,
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionAudioInput {
pub(super) format: SessionAudioFormat,
pub(super) noise_reduction: SessionNoiseReduction,
pub(super) turn_detection: SessionTurnDetection,
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionAudioFormat {
#[serde(rename = "type")]
pub(super) kind: String,
pub(super) rate: u32,
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionNoiseReduction {
#[serde(rename = "type")]
pub(super) kind: String,
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionTurnDetection {
#[serde(rename = "type")]
pub(super) kind: String,
pub(super) interrupt_response: bool,
pub(super) create_response: bool,
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionAudioOutput {
pub(super) format: SessionAudioOutputFormat,
pub(super) voice: String,
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionAudioOutputFormat {
#[serde(rename = "type")]
pub(super) kind: String,
pub(super) rate: u32,
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionTool {
#[serde(rename = "type")]
pub(super) kind: String,
pub(super) name: String,
pub(super) description: String,
pub(super) parameters: SessionToolParameters,
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionToolParameters {
#[serde(rename = "type")]
pub(super) kind: String,
pub(super) properties: BTreeMap<String, SessionToolProperty>,
pub(super) required: Vec<String>,
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct SessionToolProperty {
#[serde(rename = "type")]
pub(super) kind: String,
pub(super) description: String,
}
#[derive(Debug, Clone, Serialize)]
#[serde(tag = "type")]
pub(super) enum ConversationItem {
#[serde(rename = "message")]
Message {
role: String,
content: Vec<ConversationItemContent>,
},
#[serde(rename = "function_call_output")]
FunctionCallOutput { call_id: String, output: String },
}
#[derive(Debug, Clone, Serialize)]
pub(super) struct ConversationItemContent {
#[serde(rename = "type")]
pub(super) kind: String,
pub(super) text: String,
}
pub(super) fn parse_realtime_event(payload: &str) -> Option<RealtimeEvent> {
pub(crate) fn parse_realtime_event(payload: &str) -> Option<RealtimeEvent> {
let parsed: Value = match serde_json::from_str(payload) {
Ok(msg) => msg,
Err(err) => {
@@ -160,6 +28,7 @@ pub(super) fn parse_realtime_event(payload: &str) -> Option<RealtimeEvent> {
return None;
}
};
match message_type {
"session.created" | "session.updated" => {
let session_id = parsed
@@ -179,42 +48,6 @@ pub(super) fn parse_realtime_event(payload: &str) -> Option<RealtimeEvent> {
instructions,
})
}
"conversation.output_audio.delta"
| "response.output_audio.delta"
| "response.audio.delta" => {
let data = parsed
.get("delta")
.and_then(Value::as_str)
.or_else(|| parsed.get("data").and_then(Value::as_str))
.map(str::to_string)?;
let sample_rate = parsed
.get("sample_rate")
.and_then(Value::as_u64)
.and_then(|v| u32::try_from(v).ok())
.unwrap_or(24_000);
let num_channels = parsed
.get("channels")
.or_else(|| parsed.get("num_channels"))
.and_then(Value::as_u64)
.and_then(|v| u16::try_from(v).ok())
.unwrap_or(1);
Some(RealtimeEvent::AudioOut(RealtimeOutputAudioDelta {
frame: RealtimeAudioFrame {
data,
sample_rate,
num_channels,
samples_per_channel: parsed
.get("samples_per_channel")
.and_then(Value::as_u64)
.and_then(|v| u32::try_from(v).ok()),
},
item_id: parsed
.get("item_id")
.and_then(Value::as_str)
.map(str::to_string),
}))
}
"input_audio_buffer.speech_started" => Some(RealtimeEvent::InputAudioSpeechStarted(
RealtimeInputAudioSpeechStarted {
item_id: parsed
@@ -525,13 +358,6 @@ fn parse_tool_action_requested(parsed: &Value) -> Option<RealtimeToolActionReque
});
}
if let Some(function_call) = find_function_call(parsed, "compact_conversation") {
return Some(RealtimeToolActionRequested {
call_id: parse_call_id(function_call)?,
action: RealtimeToolAction::CompactConversation,
});
}
None
}
@@ -551,80 +377,39 @@ fn parse_response_cancelled(parsed: &Value) -> Option<RealtimeResponseCancelled>
}
fn find_function_call<'a>(parsed: &'a Value, name: &str) -> Option<&'a Value> {
parsed
let output = 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(name)
})
.and_then(Value::as_array)?;
output.iter().find(|item| {
item.get("type").and_then(Value::as_str) == Some("function_call")
&& item.get("name").and_then(Value::as_str) == Some(name)
})
}
#[derive(Debug, Default)]
struct ParsedHandoffArguments {
input_transcript: String,
send_immediately: bool,
}
fn parse_handoff_arguments(arguments: &str) -> ParsedHandoffArguments {
#[derive(Debug, Deserialize)]
struct HandoffArguments {
#[derive(Debug, Deserialize, Default)]
struct RawHandoffArguments {
#[serde(default)]
prompt: Option<String>,
#[serde(default)]
text: Option<String>,
#[serde(default)]
input: Option<String>,
#[serde(default)]
message: Option<String>,
#[serde(default)]
input_transcript: Option<String>,
input_transcript: String,
#[serde(default)]
send_immediately: bool,
#[serde(default)]
messages: Vec<RealtimeTranscriptEntry>,
}
let Some(parsed) = serde_json::from_str::<HandoffArguments>(arguments).ok() else {
return ParsedHandoffArguments {
serde_json::from_str::<RawHandoffArguments>(arguments)
.map(|raw| ParsedHandoffArguments {
input_transcript: raw.input_transcript,
send_immediately: raw.send_immediately,
})
.unwrap_or_else(|_| ParsedHandoffArguments {
input_transcript: arguments.to_string(),
send_immediately: false,
};
};
for value in [
parsed.prompt,
parsed.text,
parsed.input,
parsed.message,
parsed.input_transcript,
]
.into_iter()
.flatten()
{
if !value.is_empty() {
return ParsedHandoffArguments {
input_transcript: value,
send_immediately: parsed.send_immediately,
};
}
}
if let Some(message) = parsed
.messages
.into_iter()
.find(|message| message.role == "user" && !message.text.is_empty())
{
return ParsedHandoffArguments {
input_transcript: message.text,
send_immediately: parsed.send_immediately,
};
}
ParsedHandoffArguments {
input_transcript: String::new(),
send_immediately: parsed.send_immediately,
}
})
}

View File

@@ -27,9 +27,9 @@ pub use crate::common::create_text_param_for_request;
pub use crate::endpoint::compact::CompactClient;
pub use crate::endpoint::memories::MemoriesClient;
pub use crate::endpoint::models::ModelsClient;
pub use crate::endpoint::realtime_websocket::RealtimeSessionConfig;
pub use crate::endpoint::realtime_websocket::RealtimeWebsocketClient;
pub use crate::endpoint::realtime_websocket::RealtimeWebsocketConnection;
pub use crate::endpoint::realtime_webrtc::RealtimeSessionConfig;
pub use crate::endpoint::realtime_webrtc::RealtimeWebrtcClient;
pub use crate::endpoint::realtime_webrtc::RealtimeWebrtcConnection;
pub use crate::endpoint::responses::ResponsesClient;
pub use crate::endpoint::responses::ResponsesOptions;
pub use crate::endpoint::responses_websocket::ResponsesWebsocketClient;

View File

@@ -1,482 +0,0 @@
use std::collections::HashMap;
use std::future::Future;
use std::time::Duration;
use codex_api::RealtimeAudioFrame;
use codex_api::RealtimeEvent;
use codex_api::RealtimeSessionConfig;
use codex_api::RealtimeWebsocketClient;
use codex_api::provider::Provider;
use codex_api::provider::RetryConfig;
use codex_protocol::protocol::RealtimeOutputAudioDelta;
use futures::SinkExt;
use futures::StreamExt;
use http::HeaderMap;
use serde_json::Value;
use serde_json::json;
use tokio::net::TcpListener;
use tokio_tungstenite::accept_async;
use tokio_tungstenite::tungstenite::Message;
type RealtimeWsStream = tokio_tungstenite::WebSocketStream<tokio::net::TcpStream>;
async fn spawn_realtime_ws_server<Handler, Fut>(
handler: Handler,
) -> (String, tokio::task::JoinHandle<()>)
where
Handler: FnOnce(RealtimeWsStream) -> Fut + Send + 'static,
Fut: Future<Output = ()> + Send + 'static,
{
let listener = match TcpListener::bind("127.0.0.1:0").await {
Ok(listener) => listener,
Err(err) => panic!("failed to bind test websocket listener: {err}"),
};
let addr = match listener.local_addr() {
Ok(addr) => addr.to_string(),
Err(err) => panic!("failed to read local websocket listener address: {err}"),
};
let server = tokio::spawn(async move {
let (stream, _) = match listener.accept().await {
Ok(stream) => stream,
Err(err) => panic!("failed to accept test websocket connection: {err}"),
};
let ws = match accept_async(stream).await {
Ok(ws) => ws,
Err(err) => panic!("failed to complete websocket handshake: {err}"),
};
handler(ws).await;
});
(addr, server)
}
fn test_provider(base_url: String) -> Provider {
Provider {
name: "test".to_string(),
base_url,
query_params: Some(HashMap::new()),
headers: HeaderMap::new(),
retry: RetryConfig {
max_attempts: 1,
base_delay: Duration::from_millis(1),
retry_429: false,
retry_5xx: false,
retry_transport: false,
},
stream_idle_timeout: Duration::from_secs(5),
}
}
#[tokio::test]
async fn realtime_ws_e2e_session_create_and_event_flow() {
let (addr, server) = spawn_realtime_ws_server(|mut ws: RealtimeWsStream| async move {
let first = ws
.next()
.await
.expect("first msg")
.expect("first msg ok")
.into_text()
.expect("text");
let first_json: Value = serde_json::from_str(&first).expect("json");
assert_eq!(first_json["type"], "session.update");
assert_eq!(
first_json["session"]["type"],
Value::String("realtime".to_string())
);
assert_eq!(
first_json["session"]["instructions"],
Value::String("backend prompt".to_string())
);
assert_eq!(
first_json["session"]["audio"]["input"]["format"]["type"],
Value::String("audio/pcm".to_string())
);
assert_eq!(
first_json["session"]["audio"]["input"]["format"]["rate"],
Value::from(24_000)
);
assert_eq!(
first_json["session"]["audio"]["input"]["noise_reduction"]["type"],
Value::String("near_field".to_string())
);
assert_eq!(
first_json["session"]["audio"]["input"]["turn_detection"]["type"],
Value::String("server_vad".to_string())
);
assert_eq!(
first_json["session"]["audio"]["input"]["turn_detection"]["interrupt_response"],
Value::Bool(true)
);
assert_eq!(
first_json["session"]["audio"]["input"]["turn_detection"]["create_response"],
Value::Bool(true)
);
assert_eq!(
first_json["session"]["audio"]["output"]["format"]["type"],
Value::String("audio/pcm".to_string())
);
assert_eq!(
first_json["session"]["audio"]["output"]["format"]["rate"],
Value::from(24_000)
);
assert_eq!(
first_json["session"]["audio"]["output"]["voice"],
Value::String("marin".to_string())
);
assert_eq!(
first_json["session"]["tool_choice"],
Value::String("auto".to_string())
);
assert_eq!(
first_json["session"]["tools"][0]["type"],
Value::String("function".to_string())
);
assert_eq!(
first_json["session"]["tools"][0]["name"],
Value::String("codex".to_string())
);
assert_eq!(
first_json["session"]["tools"][0]["parameters"]["properties"]["send_immediately"]
["type"],
Value::String("boolean".to_string())
);
assert_eq!(
first_json["session"]["tools"][1]["type"],
Value::String("function".to_string())
);
assert_eq!(
first_json["session"]["tools"][1]["name"],
Value::String("cancel_current_operation".to_string())
);
assert_eq!(
first_json["session"]["tools"][1]["parameters"]["type"],
Value::String("object".to_string())
);
assert_eq!(
first_json["session"]["tools"][1]["parameters"]["required"],
Value::Array(Vec::new())
);
assert_eq!(
first_json["session"]["tools"][1]["parameters"]["properties"],
json!({})
);
assert_eq!(
first_json["session"]["tools"]
.as_array()
.expect("tools array")
.iter()
.map(|tool| tool["name"].as_str().expect("tool name"))
.collect::<Vec<_>>(),
vec![
"codex",
"cancel_current_operation",
"turn_off_realtime_mode",
"manage_message_queue",
"manage_runtime_settings",
"run_tui_command",
]
);
assert_eq!(
first_json["session"]["tools"][4]["parameters"]["properties"]["working_directory"]
["type"],
Value::String("string".to_string())
);
ws.send(Message::Text(
json!({
"type": "session.updated",
"session": {"id": "sess_mock", "instructions": "backend prompt"}
})
.to_string()
.into(),
))
.await
.expect("send session.updated");
let second = ws
.next()
.await
.expect("second msg")
.expect("second msg ok")
.into_text()
.expect("text");
let second_json: Value = serde_json::from_str(&second).expect("json");
assert_eq!(second_json["type"], "input_audio_buffer.append");
ws.send(Message::Text(
json!({
"type": "response.output_audio.delta",
"delta": "AQID",
"sample_rate": 48000,
"channels": 1
})
.to_string()
.into(),
))
.await
.expect("send audio out");
})
.await;
let client = RealtimeWebsocketClient::new(test_provider(format!("http://{addr}")));
let connection = client
.connect(
RealtimeSessionConfig {
instructions: "backend prompt".to_string(),
model: Some("realtime-test-model".to_string()),
session_id: Some("conv_123".to_string()),
},
HeaderMap::new(),
HeaderMap::new(),
)
.await
.expect("connect");
let created = connection
.next_event()
.await
.expect("next event")
.expect("event");
assert_eq!(
created,
RealtimeEvent::SessionUpdated {
session_id: "sess_mock".to_string(),
instructions: Some("backend prompt".to_string()),
}
);
connection
.send_audio_frame(RealtimeAudioFrame {
data: "AQID".to_string(),
sample_rate: 48000,
num_channels: 1,
samples_per_channel: Some(960),
})
.await
.expect("send audio");
let audio_event = connection
.next_event()
.await
.expect("next event")
.expect("event");
assert_eq!(
audio_event,
RealtimeEvent::AudioOut(RealtimeOutputAudioDelta {
frame: RealtimeAudioFrame {
data: "AQID".to_string(),
sample_rate: 48000,
num_channels: 1,
samples_per_channel: None,
},
item_id: None,
})
);
connection.close().await.expect("close");
server.await.expect("server task");
}
#[tokio::test]
async fn realtime_ws_e2e_send_while_next_event_waits() {
let (addr, server) = spawn_realtime_ws_server(|mut ws: RealtimeWsStream| async move {
let first = ws
.next()
.await
.expect("first msg")
.expect("first msg ok")
.into_text()
.expect("text");
let first_json: Value = serde_json::from_str(&first).expect("json");
assert_eq!(first_json["type"], "session.update");
let second = ws
.next()
.await
.expect("second msg")
.expect("second msg ok")
.into_text()
.expect("text");
let second_json: Value = serde_json::from_str(&second).expect("json");
assert_eq!(second_json["type"], "input_audio_buffer.append");
ws.send(Message::Text(
json!({
"type": "session.updated",
"session": {"id": "sess_after_send", "instructions": "backend prompt"}
})
.to_string()
.into(),
))
.await
.expect("send session.updated");
})
.await;
let client = RealtimeWebsocketClient::new(test_provider(format!("http://{addr}")));
let connection = client
.connect(
RealtimeSessionConfig {
instructions: "backend prompt".to_string(),
model: Some("realtime-test-model".to_string()),
session_id: Some("conv_123".to_string()),
},
HeaderMap::new(),
HeaderMap::new(),
)
.await
.expect("connect");
let (send_result, next_result) = tokio::join!(
async {
tokio::time::timeout(
Duration::from_millis(200),
connection.send_audio_frame(RealtimeAudioFrame {
data: "AQID".to_string(),
sample_rate: 48000,
num_channels: 1,
samples_per_channel: Some(960),
}),
)
.await
},
connection.next_event()
);
send_result
.expect("send should not block on next_event")
.expect("send audio");
let next_event = next_result.expect("next event").expect("event");
assert_eq!(
next_event,
RealtimeEvent::SessionUpdated {
session_id: "sess_after_send".to_string(),
instructions: Some("backend prompt".to_string()),
}
);
connection.close().await.expect("close");
server.await.expect("server task");
}
#[tokio::test]
async fn realtime_ws_e2e_disconnected_emitted_once() {
let (addr, server) = spawn_realtime_ws_server(|mut ws: RealtimeWsStream| async move {
let first = ws
.next()
.await
.expect("first msg")
.expect("first msg ok")
.into_text()
.expect("text");
let first_json: Value = serde_json::from_str(&first).expect("json");
assert_eq!(first_json["type"], "session.update");
ws.send(Message::Close(None)).await.expect("send close");
})
.await;
let client = RealtimeWebsocketClient::new(test_provider(format!("http://{addr}")));
let connection = client
.connect(
RealtimeSessionConfig {
instructions: "backend prompt".to_string(),
model: Some("realtime-test-model".to_string()),
session_id: Some("conv_123".to_string()),
},
HeaderMap::new(),
HeaderMap::new(),
)
.await
.expect("connect");
let first = connection.next_event().await.expect("next event");
assert_eq!(first, None);
let second = connection.next_event().await.expect("next event");
assert_eq!(second, None);
server.await.expect("server task");
}
#[tokio::test]
async fn realtime_ws_e2e_forwards_unknown_text_events() {
let (addr, server) = spawn_realtime_ws_server(|mut ws: RealtimeWsStream| async move {
let first = ws
.next()
.await
.expect("first msg")
.expect("first msg ok")
.into_text()
.expect("text");
let first_json: Value = serde_json::from_str(&first).expect("json");
assert_eq!(first_json["type"], "session.update");
ws.send(Message::Text(
json!({
"type": "response.created",
"response": {"id": "resp_unknown"}
})
.to_string()
.into(),
))
.await
.expect("send unknown event");
ws.send(Message::Text(
json!({
"type": "session.updated",
"session": {"id": "sess_after_unknown", "instructions": "backend prompt"}
})
.to_string()
.into(),
))
.await
.expect("send session.updated");
})
.await;
let client = RealtimeWebsocketClient::new(test_provider(format!("http://{addr}")));
let connection = client
.connect(
RealtimeSessionConfig {
instructions: "backend prompt".to_string(),
model: Some("realtime-test-model".to_string()),
session_id: Some("conv_123".to_string()),
},
HeaderMap::new(),
HeaderMap::new(),
)
.await
.expect("connect");
let first_event = connection
.next_event()
.await
.expect("next event")
.expect("event");
assert_eq!(
first_event,
RealtimeEvent::ConversationItemAdded(json!({
"type": "response.created",
"response": {"id": "resp_unknown"}
}))
);
let second_event = connection
.next_event()
.await
.expect("next event")
.expect("event");
assert_eq!(
second_event,
RealtimeEvent::SessionUpdated {
session_id: "sess_after_unknown".to_string(),
instructions: Some("backend prompt".to_string()),
}
);
connection.close().await.expect("close");
server.await.expect("server task");
}