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Fix realtime test audio fixtures

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- use valid 24 kHz mono PCM audio in realtime tests
- keep websocket/WebRTC request sequencing aligned with transport behavior

Co-authored-by: Codex <noreply@openai.com>
This commit is contained in:
Ahmed Ibrahim
2026-04-05 01:38:42 -07:00
parent 76397dbdd0
commit 11180cefd9
2 changed files with 19 additions and 6 deletions
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8
codex-rs/app-server/tests/suite/v2/realtime_conversation.rs
View File

@@ -3,6 +3,8 @@ use anyhow::Result;
use app_test_support::McpProcess;
use app_test_support::create_mock_responses_server_sequence_unchecked;
use app_test_support::to_response;
use base64::Engine;
use base64::engine::general_purpose::STANDARD as BASE64_STANDARD;
use codex_app_server_protocol::JSONRPCError;
use codex_app_server_protocol::JSONRPCResponse;
use codex_app_server_protocol::LoginAccountResponse;
@@ -40,6 +42,10 @@ use tokio::time::timeout;
const DEFAULT_TIMEOUT: Duration = Duration::from_secs(10);
const STARTUP_CONTEXT_HEADER: &str = "Startup context from Codex.";
fn realtime_pcm_silence_20ms_base64() -> String {
BASE64_STANDARD.encode([0u8; 960])
}
#[tokio::test]
async fn realtime_conversation_streams_v2_notifications() -> Result<()> {
skip_if_no_network!(Ok(()));
@@ -155,7 +161,7 @@ async fn realtime_conversation_streams_v2_notifications() -> Result<()> {
.send_thread_realtime_append_audio_request(ThreadRealtimeAppendAudioParams {
thread_id: started.thread_id.clone(),
audio: ThreadRealtimeAudioChunk {
data: "BQYH".to_string(),
data: realtime_pcm_silence_20ms_base64(),
sample_rate: 24_000,
num_channels: 1,
samples_per_channel: Some(480),

17
codex-rs/core/tests/suite/realtime_conversation.rs
View File

@@ -1,5 +1,7 @@
use anyhow::Context;
use anyhow::Result;
use base64::Engine;
use base64::engine::general_purpose::STANDARD as BASE64_STANDARD;
use chrono::Utc;
use codex_login::CodexAuth;
use codex_login::OPENAI_API_KEY_ENV_VAR;
@@ -41,6 +43,11 @@ const MEMORY_PROMPT_PHRASE: &str =
"You have access to a memory folder with guidance from prior runs.";
const REALTIME_CONVERSATION_TEST_SUBPROCESS_ENV_VAR: &str =
"CODEX_REALTIME_CONVERSATION_TEST_SUBPROCESS";
fn realtime_pcm_silence_20ms_base64() -> String {
BASE64_STANDARD.encode([0u8; 960])
}
fn websocket_request_text(
request: &core_test_support::responses::WebSocketRequest,
) -> Option<String> {
@@ -207,7 +214,7 @@ async fn conversation_start_audio_text_close_round_trip() -> Result<()> {
test.codex
.submit(Op::RealtimeConversationAudio(ConversationAudioParams {
frame: RealtimeAudioFrame {
data: "AQID".to_string(),
data: realtime_pcm_silence_20ms_base64(),
sample_rate: 24000,
num_channels: 1,
samples_per_channel: Some(480),
@@ -426,7 +433,7 @@ async fn conversation_audio_before_start_emits_error() -> Result<()> {
test.codex
.submit(Op::RealtimeConversationAudio(ConversationAudioParams {
frame: RealtimeAudioFrame {
data: "AQID".to_string(),
data: realtime_pcm_silence_20ms_base64(),
sample_rate: 24000,
num_channels: 1,
samples_per_channel: Some(480),
@@ -625,7 +632,7 @@ async fn conversation_second_start_replaces_runtime() -> Result<()> {
test.codex
.submit(Op::RealtimeConversationAudio(ConversationAudioParams {
frame: RealtimeAudioFrame {
data: "AQID".to_string(),
data: realtime_pcm_silence_20ms_base64(),
sample_rate: 24000,
num_channels: 1,
samples_per_channel: Some(480),
@@ -1585,7 +1592,7 @@ async fn inbound_handoff_request_clears_active_transcript_after_each_handoff() -
test.codex
.submit(Op::RealtimeConversationAudio(ConversationAudioParams {
frame: RealtimeAudioFrame {
data: "AQID".to_string(),
data: realtime_pcm_silence_20ms_base64(),
sample_rate: 24000,
num_channels: 1,
samples_per_channel: Some(480),
@@ -2073,7 +2080,7 @@ async fn inbound_handoff_request_steers_active_turn() -> Result<()> {
test.codex
.submit(Op::RealtimeConversationAudio(ConversationAudioParams {
frame: RealtimeAudioFrame {
data: "AQID".to_string(),
data: realtime_pcm_silence_20ms_base64(),
sample_rate: 24000,
num_channels: 1,
samples_per_channel: Some(480),