diff --git a/demos/speech_recognition/run.sh b/demos/speech_recognition/run.sh
deleted file mode 100755
index a9ae937d2a54fe4a4d5663b2f673e9ce49d3ea1c..0000000000000000000000000000000000000000
--- a/demos/speech_recognition/run.sh
+++ /dev/null
@@ -1,11 +0,0 @@
-#!/bin/bash
-
-# wget -c https://paddlespeech.bj.bcebos.com/PaddleAudio/zh.wav https://paddlespeech.bj.bcebos.com/PaddleAudio/en.wav
-
-# asr
-export CUDA_VISIBLE_DEVICES=0
-paddlespeech asr --input audio/119994.wav -v
-
-
-# asr + punc
-# paddlespeech asr --input ./zh.wav | paddlespeech text --task punc
\ No newline at end of file
diff --git a/examples/aishell/asr1/conf/preprocess.yaml b/examples/aishell/asr1/conf/preprocess.yaml
index 6fccd19547b6c51d3bf33721802e42a174081296..d3992cb9fc3e46b2a8779d4e276223e1477570af 100644
--- a/examples/aishell/asr1/conf/preprocess.yaml
+++ b/examples/aishell/asr1/conf/preprocess.yaml
@@ -5,7 +5,7 @@ process:
     n_mels: 80
     n_shift: 160
     win_length: 400
-    dither: 0.0
+    dither: 0.1
   - type: cmvn_json
     cmvn_path: data/mean_std.json
   # these three processes are a.k.a. SpecAugument
diff --git a/examples/aishell/asr1/conf/tuning/decode.yaml b/examples/aishell/asr1/conf/tuning/decode.yaml
index f0a5ba6b5775f5f69e60112f0404a33acfc8d0a4..72ede9272b190289daac57946c7e44d5407ef0b0 100644
--- a/examples/aishell/asr1/conf/tuning/decode.yaml
+++ b/examples/aishell/asr1/conf/tuning/decode.yaml
@@ -3,9 +3,9 @@ decode_batch_size: 128
 error_rate_type: cer 
 decoding_method: attention # 'attention', 'ctc_greedy_search', 'ctc_prefix_beam_search', 'attention_rescoring'
 ctc_weight: 0.5 # ctc weight for attention rescoring decode mode.
-decoding_chunk_size: 1 # decoding chunk size. Defaults to -1.
+decoding_chunk_size: -1 # decoding chunk size. Defaults to -1.
     # <0: for decoding, use full chunk.
     # >0: for decoding, use fixed chunk size as set.
     # 0: used for training, it's prohibited here. 
 num_decoding_left_chunks: -1  # number of left chunks for decoding. Defaults to -1.
-simulate_streaming: True  # simulate streaming inference. Defaults to False.
+simulate_streaming: False  # simulate streaming inference. Defaults to False.
diff --git a/examples/aishell/asr1/run.sh b/examples/aishell/asr1/run.sh
index be7116a750e43f3d9ba84084d11467bfccbd6026..c54dae9cffd7bf3f1ad86833200531d64a4c6397 100644
--- a/examples/aishell/asr1/run.sh
+++ b/examples/aishell/asr1/run.sh
@@ -3,12 +3,12 @@ source path.sh
 set -e
 
 gpus=0,1,2,3
-stage=5
-stop_stage=5
-conf_path=conf/chunk_conformer.yaml
+stage=0
+stop_stage=50
+conf_path=conf/conformer.yaml
 decode_conf_path=conf/tuning/decode.yaml
 avg_num=20
-audio_file=audio/zh.wav
+audio_file=data/demo_01_03.wav
 
 source ${MAIN_ROOT}/utils/parse_options.sh || exit 1;
 
@@ -44,7 +44,7 @@ fi
 # Optionally, you can add LM and test it with runtime.
 if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
     # test a single .wav file
-    CUDA_VISIBLE_DEVICES=0 ./local/test_wav.sh ${conf_path} ${decode_conf_path} exp/chunk_conformer/checkpoints/multi_cn ${audio_file} || exit -1
+    CUDA_VISIBLE_DEVICES=0 ./local/test_wav.sh ${conf_path} ${decode_conf_path} exp/${ckpt}/checkpoints/${avg_ckpt} ${audio_file} || exit -1
 fi
 
 # Not supported at now!!!
diff --git a/paddlespeech/__init__.py b/paddlespeech/__init__.py
index 92c1df7c4085d60ac2ecc25a22b80ef270bd51da..b781c4a8e5cc99590e179faf1c4c3989349d4216 100644
--- a/paddlespeech/__init__.py
+++ b/paddlespeech/__init__.py
@@ -14,6 +14,3 @@
 import _locale
 
 _locale._getdefaultlocale = (lambda *args: ['en_US', 'utf8'])
-
-
-
diff --git a/paddlespeech/s2t/exps/u2/bin/test_wav.py b/paddlespeech/s2t/exps/u2/bin/test_wav.py
index 6bc86d8f8756a9b1b4f044abeeb7e8a16bf1f87f..86c3db89f82878e093d71ec9a56740fb05767b7b 100644
--- a/paddlespeech/s2t/exps/u2/bin/test_wav.py
+++ b/paddlespeech/s2t/exps/u2/bin/test_wav.py
@@ -128,12 +128,10 @@ if __name__ == "__main__":
     args = parser.parse_args()
 
     config = CfgNode(new_allowed=True)
-    
+
     if args.config:
-        print(f"load config: {args.config}")
         config.merge_from_file(args.config)
     if args.decode_cfg:
-        print(f"load decode cfg: {args.decode_cfg}")
         decode_confs = CfgNode(new_allowed=True)
         decode_confs.merge_from_file(args.decode_cfg)
         config.decode = decode_confs
diff --git a/paddlespeech/server/conf/ws_application.yaml b/paddlespeech/server/conf/ws_application.yaml
index b2eaf50019baef1bec53fabb16646c3b1e02a878..b958bdf69668382adceb10ea90aaf7cde7e0fe4f 100644
--- a/paddlespeech/server/conf/ws_application.yaml
+++ b/paddlespeech/server/conf/ws_application.yaml
@@ -4,7 +4,7 @@
 #                             SERVER SETTING                                    #
 #################################################################################
 host: 0.0.0.0
-port: 8096
+port: 8090
 
 # The task format in the engin_list is: <speech task>_<engine type>
 # task choices = ['asr_online', 'tts_online']
diff --git a/paddlespeech/server/tests/asr/online/websocket_client.py b/paddlespeech/server/tests/asr/online/websocket_client.py
index 58506606edce48d635ec84e3108fb83320bf7480..661eb4dd9eb2c62a300343d19368b91629c43f31 100644
--- a/paddlespeech/server/tests/asr/online/websocket_client.py
+++ b/paddlespeech/server/tests/asr/online/websocket_client.py
@@ -105,7 +105,7 @@ class ASRAudioHandler:
 def main(args):
     logging.basicConfig(level=logging.INFO)
     logging.info("asr websocket client start")
-    handler = ASRAudioHandler("127.0.0.1", 8096)
+    handler = ASRAudioHandler("127.0.0.1", 8090)
     loop = asyncio.get_event_loop()
 
     # support to process single audio file
diff --git a/paddlespeech/server/ws/asr_socket.py b/paddlespeech/server/ws/asr_socket.py
index 65c04f67fc9524315558d46625446c7976725911..ad4a1124ec01698245ee83803c8573835ab06214 100644
--- a/paddlespeech/server/ws/asr_socket.py
+++ b/paddlespeech/server/ws/asr_socket.py
@@ -93,7 +93,6 @@ async def websocket_endpoint(websocket: WebSocket):
                     sample_rate = asr_engine.config.sample_rate
                     x_chunk, x_chunk_lens = asr_engine.preprocess(samples,
                                                                   sample_rate)
-                    print(x_chunk_lens)
                     asr_engine.run(x_chunk, x_chunk_lens)
                     asr_results = asr_engine.postprocess()
                 asr_results = asr_engine.postprocess()
diff --git a/paddlespeech/vector/cluster/diarization.py b/paddlespeech/vector/cluster/diarization.py
index 5b215725761849398ff70235b48fd55d5be8c699..816ab0dee68bca949051c390051750f739a9c413 100644
--- a/paddlespeech/vector/cluster/diarization.py
+++ b/paddlespeech/vector/cluster/diarization.py
@@ -18,11 +18,11 @@ A few sklearn functions are modified in this script as per requirement.
 """
 import argparse
 import warnings
+from distutils.util import strtobool
 
 import numpy as np
 import scipy
 import sklearn
-from distutils.util import strtobool
 from scipy import sparse
 from scipy.sparse.csgraph import connected_components
 from scipy.sparse.csgraph import laplacian as csgraph_laplacian
diff --git a/setup.py b/setup.py
index 9a8bb66bb0d93d24ee172841512e53ad2edb1ee5..82ff6341265a407c62b2599e6e73493c8b9087e1 100644
--- a/setup.py
+++ b/setup.py
@@ -168,7 +168,7 @@ class DevelopCommand(develop):
     def run(self):
         develop.run(self)
         # must after develop.run, or pkg install by shell will not see
-    #    self.execute(_post_install, (self.install_lib, ), msg="Post Install...")
+        self.execute(_post_install, (self.install_lib, ), msg="Post Install...")
 
 
 class InstallCommand(install):
diff --git a/utils/DER.py b/utils/DER.py
index 59bcbec473489e0f7930caed5838c152d3d0f874..d6ab695d8f498dd9aafebe6b43b645cc5de709e3 100755
--- a/utils/DER.py
+++ b/utils/DER.py
@@ -26,9 +26,9 @@ import argparse
 import os
 import re
 import subprocess
+from distutils.util import strtobool
 
 import numpy as np
-from distutils.util import strtobool
 
 FILE_IDS = re.compile(r"(?<=Speaker Diarization for).+(?=\*\*\*)")
 SCORED_SPEAKER_TIME = re.compile(r"(?<=SCORED SPEAKER TIME =)[\d.]+")