diff --git a/README.md b/README.md
index 9cbd6a6..d090827 100644
--- a/README.md
+++ b/README.md
@@ -98,7 +98,6 @@ VOICEPRINT_STORE=/data/asr_voiceprints.json
 VOICEPRINT_MATCH_THRESHOLD=0.45
 REALTIME_PARTIAL_MIN_SECONDS=1.5
 REALTIME_PARTIAL_INTERVAL_SECONDS=2.0
-REALTIME_PARTIAL_WINDOW_SECONDS=6.0
 ```
 
 说明：
diff --git a/asr/main.py b/asr/main.py
index 8c17c54..ac18d4c 100644
--- a/asr/main.py
+++ b/asr/main.py
@@ -103,7 +103,6 @@ VOICEPRINT_STORE = os.environ.get("VOICEPRINT_STORE", "/tmp/asr_voiceprints.json
 VOICEPRINT_MATCH_THRESHOLD = float(os.environ.get("VOICEPRINT_MATCH_THRESHOLD", "0.45"))
 REALTIME_PARTIAL_INTERVAL_SECONDS = float(os.environ.get("REALTIME_PARTIAL_INTERVAL_SECONDS", "2.0"))
 REALTIME_PARTIAL_MIN_SECONDS = float(os.environ.get("REALTIME_PARTIAL_MIN_SECONDS", "1.5"))
-REALTIME_PARTIAL_WINDOW_SECONDS = float(os.environ.get("REALTIME_PARTIAL_WINDOW_SECONDS", "6.0"))
 
 
 def _voiceprint_key(database_name: str, collection_name: str) -> str:
@@ -262,12 +261,6 @@ def _write_pcm_wav(audio_bytes: bytes) -> str:
     return tmp_path
 
 
-def _recent_pcm(audio_chunks: list[bytes], window_seconds: float) -> bytes:
-    max_bytes = max(int(window_seconds * 32000), 32000)
-    audio_bytes = b"".join(audio_chunks)
-    return audio_bytes[-max_bytes:]
-
-
 def _text_delta(previous: str, current: str) -> str:
     previous = previous or ""
     current = current or ""
@@ -286,8 +279,8 @@ def _text_delta(previous: str, current: str) -> str:
 async def run_asr_websocket(websocket: WebSocket):
     await websocket.accept(subprotocol="binary")
     audio_chunks = []
+    partial_chunks = []
     last_partial_at = 0.0
-    last_partial_text = ""
 
     try:
         while True:
@@ -295,28 +288,30 @@ async def run_asr_websocket(websocket: WebSocket):
 
             if "bytes" in message and message["bytes"]:
                 audio_chunks.append(message["bytes"])
+                partial_chunks.append(message["bytes"])
                 audio_seconds = sum(len(chunk) for chunk in audio_chunks) / 32000.0
+                partial_seconds = sum(len(chunk) for chunk in partial_chunks) / 32000.0
                 now = asyncio.get_running_loop().time()
 
                 if (
-                    audio_seconds >= REALTIME_PARTIAL_MIN_SECONDS
+                    partial_seconds >= REALTIME_PARTIAL_MIN_SECONDS
                     and now - last_partial_at >= REALTIME_PARTIAL_INTERVAL_SECONDS
                 ):
                     last_partial_at = now
+                    partial_audio = b"".join(partial_chunks)
+                    partial_chunks = []
                     partial_path = None
                     try:
-                        partial_path = _write_pcm_wav(_recent_pcm(audio_chunks, REALTIME_PARTIAL_WINDOW_SECONDS))
+                        partial_path = _write_pcm_wav(partial_audio)
                         result = await run_asr(partial_path, language="zh", response_format="json")
                         partial_text = result.get("text", "") if isinstance(result, dict) else ""
-                        if partial_text and partial_text != last_partial_text:
+                        if partial_text:
                             await websocket.send_json({
                                 "mode": "2pass-online",
                                 "text": partial_text,
                                 "timestamp": str(int(audio_seconds * 1000)),
-                                "is_full_text": True,
                                 "is_final": False,
                             })
-                            last_partial_text = partial_text
                     except Exception as e:
                         logger.warning(f"实时部分识别失败: {e}")
                     finally:
diff --git a/config.env b/config.env
index b535128..6c35916 100644
--- a/config.env
+++ b/config.env
@@ -31,7 +31,6 @@ VOICEPRINT_STORE=/data/asr_voiceprints.json
 VOICEPRINT_MATCH_THRESHOLD=0.45
 REALTIME_PARTIAL_MIN_SECONDS=1.5
 REALTIME_PARTIAL_INTERVAL_SECONDS=2.0
-REALTIME_PARTIAL_WINDOW_SECONDS=6.0
 
 # fastapi_wss legacy/local ASR-with-speaker helper
 ASR_LOCAL_URL=http://asr-test:59805/asr