[Tests] Fix Qwen3-Omni E2E LiveKit Test (#8294)
Signed-off-by: vraiti <vraiti@redhat.com> Co-authored-by: Claude Opus 4.6 <noreply@anthropic.com>
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vraitiandClaude Opus 4.6 authored and GitHub committed
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@@ -109,14 +109,14 @@ realtime_server_vad_server_params = [
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]
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def _pcm16_mono_16k_from_wav_bytes(wav_bytes: bytes) -> bytes:
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def _pcm16_mono_from_wav_bytes(wav_bytes: bytes, sample_rate_hz: int = 16000) -> bytes:
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with wave.open(io.BytesIO(wav_bytes), "rb") as wf:
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if wf.getnchannels() != 1:
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raise ValueError(f"Expected mono WAV, got {wf.getnchannels()} channels")
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if wf.getsampwidth() != 2:
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raise ValueError(f"Expected 16-bit PCM, sampwidth={wf.getsampwidth()}")
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if wf.getframerate() != 16000:
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raise ValueError(f"Expected 16 kHz input for /v1/realtime, got {wf.getframerate()} Hz")
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if wf.getframerate() != sample_rate_hz:
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raise ValueError(f"Expected {sample_rate_hz} Hz input for /v1/realtime, got {wf.getframerate()} Hz")
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if wf.getcomptype() != "NONE":
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raise ValueError(f"Expected uncompressed PCM, comptype={wf.getcomptype()!r}")
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return wf.readframes(wf.getnframes())
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@@ -148,13 +148,13 @@ async def _run_realtime_audio_roundtrip(
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text_chunks: list[str] = []
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final_text = ""
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delta_events = 0
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audio_done = False
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bytes_per_ms = 16000 * 2 // 1000
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bytes_per_ms = 24000 * 2 // 1000
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chunk_bytes = max(bytes_per_ms * chunk_ms, 2)
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async with websockets.connect(uri, max_size=64 * 1024 * 1024) as ws:
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await ws.send(json.dumps({"type": "session.update", "model": model}))
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await ws.send(json.dumps({"type": "input_audio_buffer.commit", "final": False}))
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await ws.send(json.dumps({"type": "session.update", "session": {"type": "realtime", "model": model}}))
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for i in range(0, len(pcm16), chunk_bytes):
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chunk = pcm16[i : i + chunk_bytes]
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@@ -169,7 +169,8 @@ async def _run_realtime_audio_roundtrip(
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if send_delay_ms > 0:
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await asyncio.sleep(send_delay_ms / 1000.0)
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await ws.send(json.dumps({"type": "input_audio_buffer.commit", "final": True}))
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await ws.send(json.dumps({"type": "input_audio_buffer.commit"}))
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await ws.send(json.dumps({"type": "response.create"}))
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while True:
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message = await asyncio.wait_for(ws.recv(), timeout=completion_timeout_s)
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@@ -179,7 +180,19 @@ async def _run_realtime_audio_roundtrip(
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event = json.loads(message)
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event_type = event.get("type")
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if event_type == "session.created":
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if event_type in {
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"session.created",
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"conversation.created",
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"session.updated",
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"input_audio_buffer.committed",
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"conversation.item.added",
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"conversation.item.done",
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"response.created",
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"response.output_item.added",
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"response.content_part.added",
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"response.content_part.done",
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"response.output_item.done",
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}:
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continue
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if event_type == "response.output_audio.delta":
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@@ -187,22 +200,26 @@ async def _run_realtime_audio_roundtrip(
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sr = event.get("sample_rate_hz")
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if isinstance(sr, int) and sr > 0:
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output_sr = sr
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audio_b64 = event.get("audio", "")
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audio_b64 = event.get("delta", "")
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if audio_b64:
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incremental.append(base64.b64decode(audio_b64))
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continue
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if event_type == "transcription.delta":
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if event_type == "response.output_audio_transcript.delta":
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d = event.get("delta", "")
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if d:
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text_chunks.append(d)
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continue
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if event_type == "transcription.done":
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final_text = event.get("text", "") or "".join(text_chunks)
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if event_type == "response.output_audio_transcript.done":
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final_text = event.get("transcript") or "".join(text_chunks)
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continue
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if event_type == "response.output_audio.done":
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audio_done = True
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continue
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if event_type == "response.done":
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break
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if event_type == "error":
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@@ -214,8 +231,9 @@ async def _run_realtime_audio_roundtrip(
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return {
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"output_pcm": out_pcm,
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"output_sample_rate": output_sr,
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"transcription_text": final_text if final_text else "".join(text_chunks),
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"transcription_text": final_text or "".join(text_chunks),
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"delta_events": delta_events,
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"audio_done": audio_done,
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}
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@@ -263,6 +281,7 @@ async def _run_server_vad_audio_roundtrips(
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{
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"type": "session.update",
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"session": {
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"type": "realtime",
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"model": model,
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"audio": {
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"input": {
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@@ -394,20 +413,21 @@ def _synthetic_pcm16_input(
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syn = generate_synthetic_audio(
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duration_s,
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1,
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sample_rate=16000,
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sample_rate=24000,
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phrase_text=phrase_text,
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)
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wav_bytes = base64.b64decode(syn["base64"])
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return _pcm16_mono_16k_from_wav_bytes(wav_bytes)
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return _pcm16_mono_from_wav_bytes(wav_bytes, sample_rate_hz=24000)
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def _server_vad_pcm16_input() -> bytes:
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"""Load the fixed single-turn speech fixture used by the Server VAD E2E."""
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return _pcm16_mono_16k_from_wav_bytes(validated_input_wav().read_bytes())
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return _pcm16_mono_from_wav_bytes(validated_input_wav().read_bytes())
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def _assert_realtime_smoke(result: dict) -> None:
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out_pcm = result["output_pcm"]
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assert result["audio_done"], "No response.output_audio.done event"
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assert result["delta_events"] >= 1
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assert out_pcm, "No output PCM from response.output_audio.delta"
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assert len(out_pcm) % 2 == 0
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@@ -480,7 +500,7 @@ class TestQwen3OmniRealtimeWebSocket:
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)
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assert len(answers) == 3, answers
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assert "assistant" in answers[0].lower(), answers
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assert "qwen" in answers[0].lower(), answers
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assert "paris" in answers[1].lower(), answers
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assert "paris" in answers[2].lower(), answers
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@@ -690,7 +710,7 @@ def qwen_duplex_server(tmp_path_factory, cached_silero_vad_artifact, run_level):
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def _duplex_fixture_pcm(path: Path, *, duration_s: float | None = None) -> bytes:
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"""Replay checked-in speech at the web client's negotiated 24 kHz PCM16 rate."""
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pcm = np.frombuffer(_pcm16_mono_16k_from_wav_bytes(path.read_bytes()), dtype="<i2")
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pcm = np.frombuffer(_pcm16_mono_from_wav_bytes(path.read_bytes()), dtype="<i2")
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if duration_s is not None:
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pcm = pcm[: int(duration_s * 16000)]
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positions = np.arange(len(pcm) * 3 // 2) * (2.0 / 3.0)
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@@ -775,6 +795,7 @@ async def _duplex_recorded_session(runtime, log_path: Path, *, instructions: str
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{
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"type": "session.update",
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"session": {
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"type": "realtime",
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"model": server.model,
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"instructions": instructions,
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"temperature": 0,
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