[Feature] WebSocket streaming audio input for ASR

python/sglang/srt/entrypoints/http_server.py

@@ -54,6 +54,7 @@
     Query,
     Request,
     UploadFile,
+    WebSocket,
 )
 from fastapi.exceptions import RequestValidationError
 from fastapi.middleware.cors import CORSMiddleware
@@ -1580,6 +1581,12 @@ async def openai_v1_audio_transcriptions(
     )
 
 
+@app.websocket("/v1/audio/transcriptions/stream")
+async def openai_v1_audio_transcriptions_ws(ws: WebSocket):
+    """WebSocket endpoint for real-time streaming audio transcription."""
+    await ws.app.state.openai_serving_transcription.handle_websocket(ws)
+
+
 @app.get("/v1/models", response_class=ORJSONResponse)
 async def available_models():
     """Show available models. OpenAI-compatible endpoint."""

python/sglang/srt/entrypoints/openai/serving_transcription.py

@@ -29,7 +29,7 @@
 import uuid
 from typing import TYPE_CHECKING, AsyncGenerator, List, Optional, Union
 
-from fastapi import Request
+from fastapi import Request, WebSocket
 from fastapi.responses import ORJSONResponse, Response, StreamingResponse
 
 from sglang.srt.entrypoints.openai.protocol import (
@@ -45,6 +45,7 @@
 from sglang.srt.entrypoints.openai.serving_base import OpenAIServingBase
 from sglang.srt.entrypoints.openai.streaming_asr import (
     StreamingASRState,
+    process_asr_chunk,
     split_audio_chunks,
 )
 from sglang.srt.entrypoints.openai.transcription_adapters import resolve_adapter
@@ -272,7 +273,6 @@ async def _generate_chunked_asr_stream(
         - Token-level streaming within chunks (stream=True)
         - Encoder window caching across chunks
         - Cross-chunk KV cache reuse
-        - WebSocket endpoint for real-time audio input
         """
         created_time = int(time.time())
         request_id = f"{self._request_id_prefix()}{uuid.uuid4().hex}"
@@ -288,43 +288,18 @@ async def _generate_chunked_asr_stream(
                     logger.info("[streaming_asr] client disconnected, stopping")
                     break
                 is_last = i == len(chunks) - 1
-                prompt = self._adapter.prompt_template + state.get_prefix_text()
 
-                chunk_request = GenerateReqInput(
-                    text=prompt,
+                delta = await process_asr_chunk(
+                    tokenizer_manager=self.tokenizer_manager,
+                    adapter=self._adapter,
+                    state=state,
                     audio_data=chunk_audio,
                     sampling_params=adapted_request.sampling_params,
-                    stream=False,
-                    modalities=["audio"],
+                    is_last=is_last,
+                    raw_request=raw_request,
                     routing_key=self.extract_routing_key(raw_request),
                 )
 
-                try:
-                    ret = None
-                    async for ret in self.tokenizer_manager.generate_request(
-                        chunk_request, raw_request
-                    ):
-                        break
-                except asyncio.CancelledError:
-                    raise
-                except ValueError as e:
-                    logger.warning(
-                        "[streaming_asr] chunk %d failed with ValueError: %s", i, e
-                    )
-                    continue
-
-                if ret is None:
-                    logger.warning("[streaming_asr] empty response for chunk %d", i)
-                    continue
-
-                text = self._adapter.postprocess_text(ret.get("text", ""))
-
-                if is_last:
-                    state.full_transcript = text
-                    delta = state.finalize()
-                else:
-                    delta = state.update(text)
-
                 if delta:
                     for word in delta.split(" "):
                         if not word:
@@ -366,3 +341,11 @@ async def _generate_chunked_asr_stream(
             yield f"data: {error}\n\n"
 
         yield "data: [DONE]\n\n"
+
+    async def handle_websocket(self, websocket: WebSocket) -> None:
+        """Handle a Realtime transcription session over WebSocket."""
+        from sglang.srt.entrypoints.openai.serving_transcription_websocket import (
+            handle_realtime_transcription,
+        )
+
+        await handle_realtime_transcription(websocket, self)