clone/codex
1
0
Fork 0
mirror of https://github.com/openai/codex.git synced 2026-04-24 06:35:50 +00:00
Code Issues Packages Projects Releases Wiki Activity

Use non-silent realtime test audio fixtures

Browse Source 
- replace silent PCM realtime test fixtures with a deterministic tone
- avoid codec paths optimizing away audio in WebRTC test flows

Co-authored-by: Codex <noreply@openai.com>
This commit is contained in:
Ahmed Ibrahim
2026-04-05 09:48:50 -07:00
parent 11180cefd9
commit 236891f5d3
2 changed files with 22 additions and 10 deletions
Download Patch File Download Diff File Expand all files Collapse all files

12
codex-rs/app-server/tests/suite/v2/realtime_conversation.rs
View File

@@ -42,8 +42,14 @@ 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])
fn realtime_pcm_test_tone_20ms_base64() -> String {
let pcm_bytes: Vec<u8> = (0..480)
.flat_map(|index| {
let sample = if index % 2 == 0 { 1024_i16 } else { -1024_i16 };
sample.to_le_bytes()
})
.collect();
BASE64_STANDARD.encode(pcm_bytes)
}
#[tokio::test]
@@ -161,7 +167,7 @@ async fn realtime_conversation_streams_v2_notifications() -> Result<()> {
.send_thread_realtime_append_audio_request(ThreadRealtimeAppendAudioParams {
thread_id: started.thread_id.clone(),
audio: ThreadRealtimeAudioChunk {
data: realtime_pcm_silence_20ms_base64(),
data: realtime_pcm_test_tone_20ms_base64(),
sample_rate: 24_000,
num_channels: 1,
samples_per_channel: Some(480),

20
codex-rs/core/tests/suite/realtime_conversation.rs
View File

@@ -44,8 +44,14 @@ const MEMORY_PROMPT_PHRASE: &str =
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 realtime_pcm_test_tone_20ms_base64() -> String {
let pcm_bytes: Vec<u8> = (0..480)
.flat_map(|index| {
let sample = if index % 2 == 0 { 1024_i16 } else { -1024_i16 };
sample.to_le_bytes()
})
.collect();
BASE64_STANDARD.encode(pcm_bytes)
}
fn websocket_request_text(
@@ -214,7 +220,7 @@ async fn conversation_start_audio_text_close_round_trip() -> Result<()> {
test.codex
.submit(Op::RealtimeConversationAudio(ConversationAudioParams {
frame: RealtimeAudioFrame {
data: realtime_pcm_silence_20ms_base64(),
data: realtime_pcm_test_tone_20ms_base64(),
sample_rate: 24000,
num_channels: 1,
samples_per_channel: Some(480),
@@ -433,7 +439,7 @@ async fn conversation_audio_before_start_emits_error() -> Result<()> {
test.codex
.submit(Op::RealtimeConversationAudio(ConversationAudioParams {
frame: RealtimeAudioFrame {
data: realtime_pcm_silence_20ms_base64(),
data: realtime_pcm_test_tone_20ms_base64(),
sample_rate: 24000,
num_channels: 1,
samples_per_channel: Some(480),
@@ -632,7 +638,7 @@ async fn conversation_second_start_replaces_runtime() -> Result<()> {
test.codex
.submit(Op::RealtimeConversationAudio(ConversationAudioParams {
frame: RealtimeAudioFrame {
data: realtime_pcm_silence_20ms_base64(),
data: realtime_pcm_test_tone_20ms_base64(),
sample_rate: 24000,
num_channels: 1,
samples_per_channel: Some(480),
@@ -1592,7 +1598,7 @@ async fn inbound_handoff_request_clears_active_transcript_after_each_handoff() -
test.codex
.submit(Op::RealtimeConversationAudio(ConversationAudioParams {
frame: RealtimeAudioFrame {
data: realtime_pcm_silence_20ms_base64(),
data: realtime_pcm_test_tone_20ms_base64(),
sample_rate: 24000,
num_channels: 1,
samples_per_channel: Some(480),
@@ -2080,7 +2086,7 @@ async fn inbound_handoff_request_steers_active_turn() -> Result<()> {
test.codex
.submit(Op::RealtimeConversationAudio(ConversationAudioParams {
frame: RealtimeAudioFrame {
data: realtime_pcm_silence_20ms_base64(),
data: realtime_pcm_test_tone_20ms_base64(),
sample_rate: 24000,
num_channels: 1,
samples_per_channel: Some(480),