Add vLLM v0.18.1 source tree with KV transfer abort fix

third_party/vllm/ now tracked in git for direct patch management.
Based on vLLM v0.18.1 release with one patch applied:

  vllm/v1/core/sched/scheduler.py:
    Replace fatal assert with graceful skip when KV transfer callback
    arrives for an already-aborted request during PD disaggregated serving.

Future vLLM modifications should be made directly in third_party/vllm/
and committed normally. The patches/ directory is kept as documentation
of what changed from upstream.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>

third_party/vllm/examples/online_serving/openai_realtime_client.py

@@ -0,0 +1,151 @@
+# SPDX-License-Identifier: Apache-2.0
+# SPDX-FileCopyrightText: Copyright contributors to the vLLM project
+"""
+This script demonstrates how to use the vLLM Realtime WebSocket API to perform
+audio transcription by uploading an audio file.
+
+Before running this script, you must start the vLLM server with a realtime-capable
+model, for example:
+
+    vllm serve mistralai/Voxtral-Mini-4B-Realtime-2602 --enforce-eager
+
+Requirements:
+- vllm with audio support
+- websockets
+- librosa
+- numpy
+
+The script:
+1. Connects to the Realtime WebSocket endpoint
+2. Converts an audio file to PCM16 @ 16kHz
+3. Sends audio chunks to the server
+4. Receives and prints transcription as it streams
+"""
+
+import argparse
+import asyncio
+import base64
+import json
+
+import librosa
+import numpy as np
+import websockets
+
+from vllm.assets.audio import AudioAsset
+
+
+def audio_to_pcm16_base64(audio_path: str) -> str:
+    """
+    Load an audio file and convert it to base64-encoded PCM16 @ 16kHz.
+    """
+    # Load audio and resample to 16kHz mono
+    audio, _ = librosa.load(audio_path, sr=16000, mono=True)
+    # Convert to PCM16
+    pcm16 = (audio * 32767).astype(np.int16)
+    # Encode as base64
+    return base64.b64encode(pcm16.tobytes()).decode("utf-8")
+
+
+async def realtime_transcribe(audio_path: str, host: str, port: int, model: str):
+    """
+    Connect to the Realtime API and transcribe an audio file.
+    """
+    uri = f"ws://{host}:{port}/v1/realtime"
+
+    async with websockets.connect(uri) as ws:
+        # Wait for session.created
+        response = json.loads(await ws.recv())
+        if response["type"] == "session.created":
+            print(f"Session created: {response['id']}")
+        else:
+            print(f"Unexpected response: {response}")
+            return
+
+        # Validate model
+        await ws.send(json.dumps({"type": "session.update", "model": model}))
+
+        # Signal ready to start
+        await ws.send(json.dumps({"type": "input_audio_buffer.commit"}))
+
+        # Convert audio file to base64 PCM16
+        print(f"Loading audio from: {audio_path}")
+        audio_base64 = audio_to_pcm16_base64(audio_path)
+
+        # Send audio in chunks (4KB of raw audio = ~8KB base64)
+        chunk_size = 4096
+        audio_bytes = base64.b64decode(audio_base64)
+        total_chunks = (len(audio_bytes) + chunk_size - 1) // chunk_size
+
+        print(f"Sending {total_chunks} audio chunks...")
+        for i in range(0, len(audio_bytes), chunk_size):
+            chunk = audio_bytes[i : i + chunk_size]
+            await ws.send(
+                json.dumps(
+                    {
+                        "type": "input_audio_buffer.append",
+                        "audio": base64.b64encode(chunk).decode("utf-8"),
+                    }
+                )
+            )
+
+        # Signal all audio is sent
+        await ws.send(json.dumps({"type": "input_audio_buffer.commit", "final": True}))
+        print("Audio sent. Waiting for transcription...\n")
+
+        # Receive transcription
+        print("Transcription: ", end="", flush=True)
+        while True:
+            response = json.loads(await ws.recv())
+            if response["type"] == "transcription.delta":
+                print(response["delta"], end="", flush=True)
+            elif response["type"] == "transcription.done":
+                print(f"\n\nFinal transcription: {response['text']}")
+                if response.get("usage"):
+                    print(f"Usage: {response['usage']}")
+                break
+            elif response["type"] == "error":
+                print(f"\nError: {response['error']}")
+                break
+
+
+def main(args):
+    if args.audio_path:
+        audio_path = args.audio_path
+    else:
+        # Use default audio asset
+        audio_path = str(AudioAsset("mary_had_lamb").get_local_path())
+        print(f"No audio path provided, using default: {audio_path}")
+
+    asyncio.run(realtime_transcribe(audio_path, args.host, args.port, args.model))
+
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser(
+        description="Realtime WebSocket Transcription Client"
+    )
+    parser.add_argument(
+        "--model",
+        type=str,
+        default="mistralai/Voxtral-Mini-4B-Realtime-2602",
+        help="Model that is served and should be pinged.",
+    )
+    parser.add_argument(
+        "--audio_path",
+        type=str,
+        default=None,
+        help="Path to the audio file to transcribe.",
+    )
+    parser.add_argument(
+        "--host",
+        type=str,
+        default="localhost",
+        help="vLLM server host (default: localhost)",
+    )
+    parser.add_argument(
+        "--port",
+        type=int,
+        default=8000,
+        help="vLLM server port (default: 8000)",
+    )
+    args = parser.parse_args()
+    main(args)