diff --git a/paddlespeech/cli/asr/infer.py b/paddlespeech/cli/asr/infer.py
index 53f71a70..f1e46ca1 100644
--- a/paddlespeech/cli/asr/infer.py
+++ b/paddlespeech/cli/asr/infer.py
@@ -129,7 +129,7 @@ model_alias = {
     "paddlespeech.s2t.models.ds2_online:DeepSpeech2ModelOnline",
     "conformer":
     "paddlespeech.s2t.models.u2:U2Model",
-    "conformer2online":
+    "conformer_online":
     "paddlespeech.s2t.models.u2:U2Model",
     "transformer":
     "paddlespeech.s2t.models.u2:U2Model",
diff --git a/paddlespeech/server/conf/ws_conformer_application.yaml b/paddlespeech/server/conf/ws_conformer_application.yaml
index 1a775f85..89a861ef 100644
--- a/paddlespeech/server/conf/ws_conformer_application.yaml
+++ b/paddlespeech/server/conf/ws_conformer_application.yaml
@@ -21,7 +21,7 @@ engine_list: ['asr_online']
 ################################### ASR #########################################
 ################### speech task: asr; engine_type: online #######################
 asr_online:
-    model_type: 'conformer2online_aishell'
+    model_type: 'conformer_online_multi-cn'
     am_model: # the pdmodel file of am static model [optional]
     am_params:  # the pdiparams file of am static model [optional]
     lang: 'zh'
diff --git a/paddlespeech/server/engine/asr/online/asr_engine.py b/paddlespeech/server/engine/asr/online/asr_engine.py
index 77eb5a21..3c2b066c 100644
--- a/paddlespeech/server/engine/asr/online/asr_engine.py
+++ b/paddlespeech/server/engine/asr/online/asr_engine.py
@@ -11,9 +11,10 @@
 # WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
 # See the License for the specific language governing permissions and
 # limitations under the License.
+import copy
 import os
 from typing import Optional
-import copy
+
 import numpy as np
 import paddle
 from numpy import float32
@@ -58,7 +59,7 @@ pretrained_models = {
         'lm_md5':
         '29e02312deb2e59b3c8686c7966d4fe3'
     },
-    "conformer2online_aishell-zh-16k": {
+    "conformer_online_multi-cn-zh-16k": {
         'url':
         'https://paddlespeech.bj.bcebos.com/s2t/multi_cn/asr1/asr1_chunk_conformer_multi_cn_ckpt_0.2.3.model.tar.gz',
         'md5':
@@ -93,19 +94,22 @@ class PaddleASRConnectionHanddler:
         )
         self.config = asr_engine.config
         self.model_config = asr_engine.executor.config
-        # self.model = asr_engine.executor.model
         self.asr_engine = asr_engine
 
         self.init()
         self.reset()
 
     def init(self):
+        # model_type, sample_rate and text_feature is shared for deepspeech2 and conformer
         self.model_type = self.asr_engine.executor.model_type
+        self.sample_rate = self.asr_engine.executor.sample_rate
+        # tokens to text
+        self.text_feature = self.asr_engine.executor.text_feature
+
         if "deepspeech2online" in self.model_type or "deepspeech2offline" in self.model_type:
             from paddlespeech.s2t.io.collator import SpeechCollator
-            self.sample_rate = self.asr_engine.executor.sample_rate
             self.am_predictor = self.asr_engine.executor.am_predictor
-            self.text_feature = self.asr_engine.executor.text_feature
+
             self.collate_fn_test = SpeechCollator.from_config(self.model_config)
             self.decoder = CTCDecoder(
                 odim=self.model_config.output_dim,  # <blank> is in  vocab
@@ -114,7 +118,8 @@ class PaddleASRConnectionHanddler:
                 dropout_rate=0.0,
                 reduction=True,  # sum
                 batch_average=True,  # sum / batch_size
-                grad_norm_type=self.model_config.get('ctc_grad_norm_type', None)) 
+                grad_norm_type=self.model_config.get('ctc_grad_norm_type',
+                                                     None))
 
             cfg = self.model_config.decode
             decode_batch_size = 1  # for online
@@ -123,20 +128,16 @@ class PaddleASRConnectionHanddler:
                 cfg.decoding_method, cfg.lang_model_path, cfg.alpha, cfg.beta,
                 cfg.beam_size, cfg.cutoff_prob, cfg.cutoff_top_n,
                 cfg.num_proc_bsearch)
-                # frame window samples length and frame shift samples length
-                
-            self.win_length = int(self.model_config.window_ms * self.sample_rate)
-            self.n_shift = int(self.model_config.stride_ms * self.sample_rate)
+            # frame window samples length and frame shift samples length
 
-        elif "conformer" in self.model_type or "transformer" in self.model_type or "wenetspeech" in self.model_type:
-            self.sample_rate = self.asr_engine.executor.sample_rate
+            self.win_length = int(self.model_config.window_ms *
+                                  self.sample_rate)
+            self.n_shift = int(self.model_config.stride_ms * self.sample_rate)
 
+        elif "conformer" in self.model_type or "transformer" in self.model_type:
             # acoustic model
             self.model = self.asr_engine.executor.model
 
-            # tokens to text
-            self.text_feature = self.asr_engine.executor.text_feature
-
             # ctc decoding config
             self.ctc_decode_config = self.asr_engine.executor.config.decode
             self.searcher = CTCPrefixBeamSearch(self.ctc_decode_config)
@@ -189,7 +190,7 @@ class PaddleASRConnectionHanddler:
             audio = paddle.to_tensor(audio, dtype='float32')
             # audio_len = paddle.to_tensor(audio_len)
             audio = paddle.unsqueeze(audio, axis=0)
-            
+
             if self.cached_feat is None:
                 self.cached_feat = audio
             else:
@@ -211,7 +212,7 @@ class PaddleASRConnectionHanddler:
             logger.info(
                 f"After extract feat, the connection remain the audio samples: {self.remained_wav.shape}"
             )
-        elif "conformer2online" in self.model_type:
+        elif "conformer_online" in self.model_type:
             logger.info("Online ASR extract the feat")
             samples = np.frombuffer(samples, dtype=np.int16)
             assert samples.ndim == 1
@@ -264,41 +265,43 @@ class PaddleASRConnectionHanddler:
     def reset(self):
         if "deepspeech2online" in self.model_type or "deepspeech2offline" in self.model_type:
             # for deepspeech2 
-            self.chunk_state_h_box = copy.deepcopy(self.asr_engine.executor.chunk_state_h_box)
-            self.chunk_state_c_box = copy.deepcopy(self.asr_engine.executor.chunk_state_c_box)
+            self.chunk_state_h_box = copy.deepcopy(
+                self.asr_engine.executor.chunk_state_h_box)
+            self.chunk_state_c_box = copy.deepcopy(
+                self.asr_engine.executor.chunk_state_c_box)
             self.decoder.reset_decoder(batch_size=1)
-        elif "conformer" in self.model_type or "transformer" in self.model_type or "wenetspeech" in self.model_type:
-            # for conformer online
-            self.subsampling_cache = None
-            self.elayers_output_cache = None
-            self.conformer_cnn_cache = None
-            self.encoder_out = None
-            self.cached_feat = None
-            self.remained_wav = None
-            self.offset = 0
-            self.num_samples = 0
-            self.device = None
-            self.hyps = []
-            self.num_frames = 0
-            self.chunk_num = 0
-            self.global_frame_offset = 0
-            self.result_transcripts = ['']
+
+        # for conformer online
+        self.subsampling_cache = None
+        self.elayers_output_cache = None
+        self.conformer_cnn_cache = None
+        self.encoder_out = None
+        self.cached_feat = None
+        self.remained_wav = None
+        self.offset = 0
+        self.num_samples = 0
+        self.device = None
+        self.hyps = []
+        self.num_frames = 0
+        self.chunk_num = 0
+        self.global_frame_offset = 0
+        self.result_transcripts = ['']
 
     def decode(self, is_finished=False):
         if "deepspeech2online" in self.model_type:
             # x_chunk 是特征数据
-            decoding_chunk_size = 1 # decoding_chunk_size=1 in deepspeech2 model
-            context = 7             # context=7 in deepspeech2 model
-            subsampling = 4         # subsampling=4 in deepspeech2 model
+            decoding_chunk_size = 1  # decoding_chunk_size=1 in deepspeech2 model
+            context = 7  # context=7 in deepspeech2 model
+            subsampling = 4  # subsampling=4 in deepspeech2 model
             stride = subsampling * decoding_chunk_size
             cached_feature_num = context - subsampling
             # decoding window for model
-            decoding_window = (decoding_chunk_size - 1) * subsampling + context        
-            
+            decoding_window = (decoding_chunk_size - 1) * subsampling + context
+
             if self.cached_feat is None:
                 logger.info("no audio feat, please input more pcm data")
-                return            
-            
+                return
+
             num_frames = self.cached_feat.shape[1]
             logger.info(
                 f"Required decoding window {decoding_window} frames, and the connection has {num_frames} frames"
@@ -306,14 +309,14 @@ class PaddleASRConnectionHanddler:
             # the cached feat must be larger decoding_window
             if num_frames < decoding_window and not is_finished:
                 logger.info(
-                f"frame feat num is less than {decoding_window}, please input more pcm data"
+                    f"frame feat num is less than {decoding_window}, please input more pcm data"
                 )
                 return None, None
 
             # if is_finished=True, we need at least context frames
             if num_frames < context:
                 logger.info(
-                "flast {num_frames} is less than context {context} frames, and we cannot do model forward"
+                    "flast {num_frames} is less than context {context} frames, and we cannot do model forward"
                 )
                 return None, None
             logger.info("start to do model forward")
@@ -334,8 +337,7 @@ class PaddleASRConnectionHanddler:
 
             self.result_transcripts = [trans_best]
 
-            self.cached_feat = self.cached_feat[:, end -
-                                            cached_feature_num:, :]
+            self.cached_feat = self.cached_feat[:, end - cached_feature_num:, :]
             # return trans_best[0]            
         elif "conformer" in self.model_type or "transformer" in self.model_type:
             try:
@@ -354,8 +356,7 @@ class PaddleASRConnectionHanddler:
         logger.info("start to decoce one chunk with deepspeech2 model")
         input_names = self.am_predictor.get_input_names()
         audio_handle = self.am_predictor.get_input_handle(input_names[0])
-        audio_len_handle = self.am_predictor.get_input_handle(
-                input_names[1])
+        audio_len_handle = self.am_predictor.get_input_handle(input_names[1])
         h_box_handle = self.am_predictor.get_input_handle(input_names[2])
         c_box_handle = self.am_predictor.get_input_handle(input_names[3])
 
@@ -374,11 +375,11 @@ class PaddleASRConnectionHanddler:
         output_names = self.am_predictor.get_output_names()
         output_handle = self.am_predictor.get_output_handle(output_names[0])
         output_lens_handle = self.am_predictor.get_output_handle(
-                output_names[1])
+            output_names[1])
         output_state_h_handle = self.am_predictor.get_output_handle(
-                output_names[2])
+            output_names[2])
         output_state_c_handle = self.am_predictor.get_output_handle(
-                output_names[3])
+            output_names[3])
 
         self.am_predictor.run()
 
@@ -389,7 +390,7 @@ class PaddleASRConnectionHanddler:
 
         self.decoder.next(output_chunk_probs, output_chunk_lens)
         trans_best, trans_beam = self.decoder.decode()
-        logger.info(f"decode one one best result: {trans_best[0]}")
+        logger.info(f"decode one best result: {trans_best[0]}")
         return trans_best[0]
 
     def advance_decoding(self, is_finished=False):
@@ -500,7 +501,7 @@ class PaddleASRConnectionHanddler:
     def rescoring(self):
         if "deepspeech2online" in self.model_type or "deepspeech2offline" in self.model_type:
             return
-        
+
         logger.info("rescoring the final result")
         if "attention_rescoring" != self.ctc_decode_config.decoding_method:
             return
@@ -587,7 +588,7 @@ class ASRServerExecutor(ASRExecutor):
         return decompressed_path
 
     def _init_from_path(self,
-                        model_type: str='wenetspeech',
+                        model_type: str='deepspeech2online_aishell',
                         am_model: Optional[os.PathLike]=None,
                         am_params: Optional[os.PathLike]=None,
                         lang: str='zh',
@@ -647,7 +648,7 @@ class ASRServerExecutor(ASRExecutor):
                 self.download_lm(
                     lm_url,
                     os.path.dirname(self.config.decode.lang_model_path), lm_md5)
-            elif "conformer" in model_type or "transformer" in model_type or "wenetspeech" in model_type:
+            elif "conformer" in model_type or "transformer" in model_type:
                 logger.info("start to create the stream conformer asr engine")
                 if self.config.spm_model_prefix:
                     self.config.spm_model_prefix = os.path.join(
@@ -711,7 +712,7 @@ class ASRServerExecutor(ASRExecutor):
             self.chunk_state_c_box = np.zeros(
                 (self.config.num_rnn_layers, 1, self.config.rnn_layer_size),
                 dtype=float32)
-        elif "conformer" in model_type or "transformer" in model_type or "wenetspeech" in model_type:
+        elif "conformer" in model_type or "transformer" in model_type:
             model_name = model_type[:model_type.rindex(
                 '_')]  # model_type: {model_name}_{dataset}
             logger.info(f"model name: {model_name}")
@@ -742,7 +743,7 @@ class ASRServerExecutor(ASRExecutor):
             self.chunk_state_c_box = np.zeros(
                 (self.config.num_rnn_layers, 1, self.config.rnn_layer_size),
                 dtype=float32)
-        elif "conformer" in self.model_type or "transformer" in self.model_type or "wenetspeech" in self.model_type:
+        elif "conformer" in self.model_type or "transformer" in self.model_type:
             self.transformer_decode_reset()
 
     def decode_one_chunk(self, x_chunk, x_chunk_lens, model_type: str):
@@ -754,7 +755,7 @@ class ASRServerExecutor(ASRExecutor):
             model_type (str): online model type
 
         Returns:
-            [type]: [description]
+            str: one best result
         """
         logger.info("start to decoce chunk by chunk")
         if "deepspeech2online" in model_type:
@@ -795,7 +796,7 @@ class ASRServerExecutor(ASRExecutor):
 
             self.decoder.next(output_chunk_probs, output_chunk_lens)
             trans_best, trans_beam = self.decoder.decode()
-            logger.info(f"decode one one best result: {trans_best[0]}")
+            logger.info(f"decode one best result: {trans_best[0]}")
             return trans_best[0]
 
         elif "conformer" in model_type or "transformer" in model_type:
@@ -972,7 +973,7 @@ class ASRServerExecutor(ASRExecutor):
             x_chunk_lens = np.array([audio_len])
 
             return x_chunk, x_chunk_lens
-        elif "conformer2online" in self.model_type:
+        elif "conformer_online" in self.model_type:
 
             if sample_rate != self.sample_rate:
                 logger.info(f"audio sample rate {sample_rate} is not match,"
@@ -1005,7 +1006,7 @@ class ASREngine(BaseEngine):
 
     def __init__(self):
         super(ASREngine, self).__init__()
-        logger.info("create the online asr engine instache")
+        logger.info("create the online asr engine instance")
 
     def init(self, config: dict) -> bool:
         """init engine resource