diff --git a/demos/streaming_asr_server/local/rtf_from_log.py b/demos/streaming_asr_server/local/rtf_from_log.py
index a5634388bbdfbe88475f0ec4520b47ad25a9457c..4f30d640067d513bb3d700845c48bdcae392d18b 100755
--- a/demos/streaming_asr_server/local/rtf_from_log.py
+++ b/demos/streaming_asr_server/local/rtf_from_log.py
@@ -33,8 +33,9 @@ if __name__ == '__main__':
     P = 0.0
     n = 0
     for m in rtfs:
-        n += 1
+        # not accurate, may have duplicate log
+        n += 1  
         T += m['T']
         P += m['P']
 
-    print(f"RTF: {P/T}, utts: {n}")
+    print(f"RTF: {P/T}")
diff --git a/demos/streaming_asr_server/local/websocket_client.py b/demos/streaming_asr_server/local/websocket_client.py
index 51ae7a2f45591b60c28bd77d611401995682b909..8b70eb2d68ef4cfbecdb3ba545b45c854924e51f 100644
--- a/demos/streaming_asr_server/local/websocket_client.py
+++ b/demos/streaming_asr_server/local/websocket_client.py
@@ -18,7 +18,6 @@
 import argparse
 import asyncio
 import codecs
-import logging
 import os
 
 from paddlespeech.cli.log import logger
@@ -44,7 +43,7 @@ def main(args):
 
     # support to process batch audios from wav.scp
     if args.wavscp and os.path.exists(args.wavscp):
-        logging.info(f"start to process the wavscp: {args.wavscp}")
+        logger.info(f"start to process the wavscp: {args.wavscp}")
         with codecs.open(args.wavscp, 'r', encoding='utf-8') as f,\
              codecs.open("result.txt", 'w', encoding='utf-8') as w:
             for line in f: