diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/README.md b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/README.md
new file mode 100644
index 000000000..dfd509dd4
--- /dev/null
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/README.md
@@ -0,0 +1,53 @@
+# ModelScope Model
+
+## How to finetune and infer using a pretrained Paraformer-large Model
+
+### Finetune
+
+- Modify finetune training related parameters in `finetune.py`
+    - <strong>output_dir:</strong> # result dir
+    - <strong>data_dir:</strong> # the dataset dir needs to include files: `train/wav.scp`, `train/text`; `validation/wav.scp`, `validation/text`
+    - <strong>dataset_type:</strong> # for dataset larger than 1000 hours, set as `large`, otherwise set as `small`
+    - <strong>batch_bins:</strong> # batch size. For dataset_type is `small`, `batch_bins` indicates the feature frames. For dataset_type is `large`, `batch_bins` indicates the duration in ms
+    - <strong>max_epoch:</strong> # number of training epoch
+    - <strong>lr:</strong> # learning rate
+
+- Then you can run the pipeline to finetune with:
+```python
+    python finetune.py
+```
+
+### Inference
+
+Or you can use the finetuned model for inference directly.
+
+- Setting parameters in `infer.py`
+    - <strong>data_dir:</strong> # the dataset dir needs to include `test/wav.scp`. If `test/text` is also exists, CER will be computed
+    - <strong>output_dir:</strong> # result dir
+    - <strong>ngpu:</strong> # the number of GPUs for decoding
+    - <strong>njob:</strong> # the number of jobs for each GPU
+
+- Then you can run the pipeline to infer with:
+```python
+    python infer.py
+```
+
+- Results
+
+The decoding results can be found in `$output_dir/1best_recog/text.cer`, which includes recognition results of each sample and the CER metric of the whole test set.
+
+### Inference using local finetuned model
+
+- Modify inference related parameters in `infer_after_finetune.py`
+    - <strong>output_dir:</strong> # result dir
+    - <strong>data_dir:</strong> # the dataset dir needs to include `test/wav.scp`. If `test/text` is also exists, CER will be computed
+    - <strong>decoding_model_name:</strong> # set the checkpoint name for decoding, e.g., `valid.cer_ctc.ave.pth`
+
+- Then you can run the pipeline to finetune with:
+```python
+    python infer_after_finetune.py
+```
+
+- Results
+
+The decoding results can be found in `$output_dir/decoding_results/text.cer`, which includes recognition results of each sample and the CER metric of the whole test set.
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/finetune.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/finetune.py
index 1aef9c660..2ecc22917 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/finetune.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/finetune.py
@@ -1,35 +1,36 @@
 import os
+
 from modelscope.metainfo import Trainers
 from modelscope.trainers import build_trainer
+
 from funasr.datasets.ms_dataset import MsDataset
+from funasr.utils.modelscope_param import modelscope_args
 
 
 def modelscope_finetune(params):
-    if not os.path.exists(params["output_dir"]):
-        os.makedirs(params["output_dir"], exist_ok=True)
+    if not os.path.exists(params.output_dir):
+        os.makedirs(params.output_dir, exist_ok=True)
     # dataset split ["train", "validation"]
-    ds_dict = MsDataset.load(params["data_dir"])
+    ds_dict = MsDataset.load(params.data_path)
     kwargs = dict(
-        model=params["model"],
-        model_revision=params["model_revision"],
+        model=params.model,
         data_dir=ds_dict,
-        dataset_type=params["dataset_type"],
-        work_dir=params["output_dir"],
-        batch_bins=params["batch_bins"],
-        max_epoch=params["max_epoch"],
-        lr=params["lr"])
+        dataset_type=params.dataset_type,
+        work_dir=params.output_dir,
+        batch_bins=params.batch_bins,
+        max_epoch=params.max_epoch,
+        lr=params.lr)
     trainer = build_trainer(Trainers.speech_asr_trainer, default_args=kwargs)
     trainer.train()
 
 
 if __name__ == '__main__':
-    params = {}
-    params["output_dir"] = "./checkpoint"
-    params["data_dir"] = "./data"
-    params["batch_bins"] = 2000
-    params["dataset_type"] = "small"
-    params["max_epoch"] = 50
-    params["lr"] = 0.00005
-    params["model"] = "damo/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline"
-    params["model_revision"] = None
+    params = modelscope_args(model="damo/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline", data_path="./data")
+    params.output_dir = "./checkpoint"              # m模型保存路径
+    params.data_path = "./example_data/"            # 数据路径
+    params.dataset_type = "small"                   # 小数据量设置small，若数据量大于1000小时，请使用large
+    params.batch_bins = 2000                       # batch size，如果dataset_type="small"，batch_bins单位为fbank特征帧数，如果dataset_type="large"，batch_bins单位为毫秒，
+    params.max_epoch = 20                           # 最大训练轮数
+    params.lr = 0.00005                             # 设置学习率
+    
     modelscope_finetune(params)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/infer.py
index 85ddeeea1..3a8954640 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/infer.py
@@ -1,13 +1,89 @@
+import os
+import shutil
+from multiprocessing import Pool
+
 from modelscope.pipelines import pipeline
 from modelscope.utils.constant import Tasks
 
-if __name__ == "__main__":
-    audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_fa.wav"
-    output_dir = "./results"
+from funasr.utils.compute_wer import compute_wer
+
+
+def modelscope_infer_core(output_dir, split_dir, njob, idx):
+    output_dir_job = os.path.join(output_dir, "output.{}".format(idx))
+    gpu_id = (int(idx) - 1) // njob
+    if "CUDA_VISIBLE_DEVICES" in os.environ.keys():
+        gpu_list = os.environ['CUDA_VISIBLE_DEVICES'].split(",")
+        os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_list[gpu_id])
+    else:
+        os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_id)
     inference_pipline = pipeline(
         task=Tasks.auto_speech_recognition,
         model="damo/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline",
-        output_dir=output_dir,
+        output_dir=output_dir_job,
+        batch_size=1
     )
-    rec_result = inference_pipline(audio_in=audio_in)
-    print(rec_result)
+    audio_in = os.path.join(split_dir, "wav.{}.scp".format(idx))
+    inference_pipline(audio_in=audio_in)
+
+
+def modelscope_infer(params):
+    # prepare for multi-GPU decoding
+    ngpu = params["ngpu"]
+    njob = params["njob"]
+    output_dir = params["output_dir"]
+    if os.path.exists(output_dir):
+        shutil.rmtree(output_dir)
+    os.mkdir(output_dir)
+    split_dir = os.path.join(output_dir, "split")
+    os.mkdir(split_dir)
+    nj = ngpu * njob
+    wav_scp_file = os.path.join(params["data_dir"], "wav.scp")
+    with open(wav_scp_file) as f:
+        lines = f.readlines()
+        num_lines = len(lines)
+        num_job_lines = num_lines // nj
+    start = 0
+    for i in range(nj):
+        end = start + num_job_lines
+        file = os.path.join(split_dir, "wav.{}.scp".format(str(i + 1)))
+        with open(file, "w") as f:
+            if i == nj - 1:
+                f.writelines(lines[start:])
+            else:
+                f.writelines(lines[start:end])
+        start = end
+
+    p = Pool(nj)
+    for i in range(nj):
+        p.apply_async(modelscope_infer_core,
+                      args=(output_dir, split_dir, njob, str(i + 1)))
+    p.close()
+    p.join()
+
+    # combine decoding results
+    best_recog_path = os.path.join(output_dir, "1best_recog")
+    os.mkdir(best_recog_path)
+    files = ["text", "token", "score"]
+    for file in files:
+        with open(os.path.join(best_recog_path, file), "w") as f:
+            for i in range(nj):
+                job_file = os.path.join(output_dir, "output.{}/1best_recog".format(str(i + 1)), file)
+                with open(job_file) as f_job:
+                    lines = f_job.readlines()
+                f.writelines(lines)
+
+    # If text exists, compute CER
+    text_in = os.path.join(params["data_dir"], "text")
+    if os.path.exists(text_in):
+        text_proc_file = os.path.join(best_recog_path, "token")
+        compute_wer(text_in, text_proc_file, os.path.join(best_recog_path, "text.cer"))
+        os.system("tail -n 3 {}".format(os.path.join(best_recog_path, "text.cer")))
+
+
+if __name__ == "__main__":
+    params = {}
+    params["data_dir"] = "./data/test"
+    params["output_dir"] = "./results"
+    params["ngpu"] = 1
+    params["njob"] = 8
+    modelscope_infer(params)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/infer_after_finetune.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/infer_after_finetune.py
new file mode 100644
index 000000000..d91a40a6c
--- /dev/null
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/infer_after_finetune.py
@@ -0,0 +1,54 @@
+import json
+import os
+import shutil
+
+from modelscope.pipelines import pipeline
+from modelscope.utils.constant import Tasks
+
+from funasr.utils.compute_wer import compute_wer
+
+
+def modelscope_infer_after_finetune(params):
+    # prepare for decoding
+    pretrained_model_path = os.path.join(os.environ["HOME"], ".cache/modelscope/hub", params["modelscope_model_name"])
+    for file_name in params["required_files"]:
+        if file_name == "configuration.json":
+            with open(os.path.join(pretrained_model_path, file_name)) as f:
+                config_dict = json.load(f)
+                config_dict["model"]["am_model_name"] = params["decoding_model_name"]
+            with open(os.path.join(params["output_dir"], "configuration.json"), "w") as f:
+                json.dump(config_dict, f, indent=4, separators=(',', ': '))
+        else:
+            shutil.copy(os.path.join(pretrained_model_path, file_name),
+                        os.path.join(params["output_dir"], file_name))
+    decoding_path = os.path.join(params["output_dir"], "decode_results")
+    if os.path.exists(decoding_path):
+        shutil.rmtree(decoding_path)
+    os.mkdir(decoding_path)
+
+    # decoding
+    inference_pipeline = pipeline(
+        task=Tasks.auto_speech_recognition,
+        model=params["output_dir"],
+        output_dir=decoding_path,
+        batch_size=1
+    )
+    audio_in = os.path.join(params["data_dir"], "wav.scp")
+    inference_pipeline(audio_in=audio_in)
+
+    # computer CER if GT text is set
+    text_in = os.path.join(params["data_dir"], "text")
+    if os.path.exists(text_in):
+        text_proc_file = os.path.join(decoding_path, "1best_recog/token")
+        compute_wer(text_in, text_proc_file, os.path.join(decoding_path, "text.cer"))
+        os.system("tail -n 3 {}".format(os.path.join(decoding_path, "text.cer")))
+
+
+if __name__ == '__main__':
+    params = {}
+    params["modelscope_model_name"] = "damo/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline"
+    params["required_files"] = ["am.mvn", "decoding.yaml", "configuration.json"]
+    params["output_dir"] = "./checkpoint"
+    params["data_dir"] = "./data/test"
+    params["decoding_model_name"] = "20epoch.pth"
+    modelscope_infer_after_finetune(params)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/README.md b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/README.md
new file mode 100644
index 000000000..dfd509dd4
--- /dev/null
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/README.md
@@ -0,0 +1,53 @@
+# ModelScope Model
+
+## How to finetune and infer using a pretrained Paraformer-large Model
+
+### Finetune
+
+- Modify finetune training related parameters in `finetune.py`
+    - <strong>output_dir:</strong> # result dir
+    - <strong>data_dir:</strong> # the dataset dir needs to include files: `train/wav.scp`, `train/text`; `validation/wav.scp`, `validation/text`
+    - <strong>dataset_type:</strong> # for dataset larger than 1000 hours, set as `large`, otherwise set as `small`
+    - <strong>batch_bins:</strong> # batch size. For dataset_type is `small`, `batch_bins` indicates the feature frames. For dataset_type is `large`, `batch_bins` indicates the duration in ms
+    - <strong>max_epoch:</strong> # number of training epoch
+    - <strong>lr:</strong> # learning rate
+
+- Then you can run the pipeline to finetune with:
+```python
+    python finetune.py
+```
+
+### Inference
+
+Or you can use the finetuned model for inference directly.
+
+- Setting parameters in `infer.py`
+    - <strong>data_dir:</strong> # the dataset dir needs to include `test/wav.scp`. If `test/text` is also exists, CER will be computed
+    - <strong>output_dir:</strong> # result dir
+    - <strong>ngpu:</strong> # the number of GPUs for decoding
+    - <strong>njob:</strong> # the number of jobs for each GPU
+
+- Then you can run the pipeline to infer with:
+```python
+    python infer.py
+```
+
+- Results
+
+The decoding results can be found in `$output_dir/1best_recog/text.cer`, which includes recognition results of each sample and the CER metric of the whole test set.
+
+### Inference using local finetuned model
+
+- Modify inference related parameters in `infer_after_finetune.py`
+    - <strong>output_dir:</strong> # result dir
+    - <strong>data_dir:</strong> # the dataset dir needs to include `test/wav.scp`. If `test/text` is also exists, CER will be computed
+    - <strong>decoding_model_name:</strong> # set the checkpoint name for decoding, e.g., `valid.cer_ctc.ave.pth`
+
+- Then you can run the pipeline to finetune with:
+```python
+    python infer_after_finetune.py
+```
+
+- Results
+
+The decoding results can be found in `$output_dir/decoding_results/text.cer`, which includes recognition results of each sample and the CER metric of the whole test set.
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/finetune.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/finetune.py
index 3bdf1cca2..2469e5318 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/finetune.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/finetune.py
@@ -1,35 +1,36 @@
 import os
+
 from modelscope.metainfo import Trainers
 from modelscope.trainers import build_trainer
+
 from funasr.datasets.ms_dataset import MsDataset
+from funasr.utils.modelscope_param import modelscope_args
 
 
 def modelscope_finetune(params):
-    if not os.path.exists(params["output_dir"]):
-        os.makedirs(params["output_dir"], exist_ok=True)
+    if not os.path.exists(params.output_dir):
+        os.makedirs(params.output_dir, exist_ok=True)
     # dataset split ["train", "validation"]
-    ds_dict = MsDataset.load(params["data_dir"])
+    ds_dict = MsDataset.load(params.data_path)
     kwargs = dict(
-        model=params["model"],
-        model_revision=params["model_revision"],
+        model=params.model,
         data_dir=ds_dict,
-        dataset_type=params["dataset_type"],
-        work_dir=params["output_dir"],
-        batch_bins=params["batch_bins"],
-        max_epoch=params["max_epoch"],
-        lr=params["lr"])
+        dataset_type=params.dataset_type,
+        work_dir=params.output_dir,
+        batch_bins=params.batch_bins,
+        max_epoch=params.max_epoch,
+        lr=params.lr)
     trainer = build_trainer(Trainers.speech_asr_trainer, default_args=kwargs)
     trainer.train()
 
 
 if __name__ == '__main__':
-    params = {}
-    params["output_dir"] = "./checkpoint"
-    params["data_dir"] = "./data"
-    params["batch_bins"] = 2000
-    params["dataset_type"] = "small"
-    params["max_epoch"] = 50
-    params["lr"] = 0.00005
-    params["model"] = "damo/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online"
-    params["model_revision"] = None
+    params = modelscope_args(model="damo/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online", data_path="./data")
+    params.output_dir = "./checkpoint"              # m模型保存路径
+    params.data_path = "./example_data/"            # 数据路径
+    params.dataset_type = "small"                   # 小数据量设置small，若数据量大于1000小时，请使用large
+    params.batch_bins = 2000                       # batch size，如果dataset_type="small"，batch_bins单位为fbank特征帧数，如果dataset_type="large"，batch_bins单位为毫秒，
+    params.max_epoch = 20                           # 最大训练轮数
+    params.lr = 0.00005                             # 设置学习率
+    
     modelscope_finetune(params)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/infer.py
index 960c39331..ecb138181 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/infer.py
@@ -1,13 +1,89 @@
+import os
+import shutil
+from multiprocessing import Pool
+
 from modelscope.pipelines import pipeline
 from modelscope.utils.constant import Tasks
 
-if __name__ == "__main__":
-    audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_fa.wav"
-    output_dir = "./results"
+from funasr.utils.compute_wer import compute_wer
+
+
+def modelscope_infer_core(output_dir, split_dir, njob, idx):
+    output_dir_job = os.path.join(output_dir, "output.{}".format(idx))
+    gpu_id = (int(idx) - 1) // njob
+    if "CUDA_VISIBLE_DEVICES" in os.environ.keys():
+        gpu_list = os.environ['CUDA_VISIBLE_DEVICES'].split(",")
+        os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_list[gpu_id])
+    else:
+        os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_id)
     inference_pipline = pipeline(
         task=Tasks.auto_speech_recognition,
         model="damo/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online",
-        output_dir=output_dir,
+        output_dir=output_dir_job,
+        batch_size=1
     )
-    rec_result = inference_pipline(audio_in=audio_in)
-    print(rec_result)
+    audio_in = os.path.join(split_dir, "wav.{}.scp".format(idx))
+    inference_pipline(audio_in=audio_in)
+
+
+def modelscope_infer(params):
+    # prepare for multi-GPU decoding
+    ngpu = params["ngpu"]
+    njob = params["njob"]
+    output_dir = params["output_dir"]
+    if os.path.exists(output_dir):
+        shutil.rmtree(output_dir)
+    os.mkdir(output_dir)
+    split_dir = os.path.join(output_dir, "split")
+    os.mkdir(split_dir)
+    nj = ngpu * njob
+    wav_scp_file = os.path.join(params["data_dir"], "wav.scp")
+    with open(wav_scp_file) as f:
+        lines = f.readlines()
+        num_lines = len(lines)
+        num_job_lines = num_lines // nj
+    start = 0
+    for i in range(nj):
+        end = start + num_job_lines
+        file = os.path.join(split_dir, "wav.{}.scp".format(str(i + 1)))
+        with open(file, "w") as f:
+            if i == nj - 1:
+                f.writelines(lines[start:])
+            else:
+                f.writelines(lines[start:end])
+        start = end
+
+    p = Pool(nj)
+    for i in range(nj):
+        p.apply_async(modelscope_infer_core,
+                      args=(output_dir, split_dir, njob, str(i + 1)))
+    p.close()
+    p.join()
+
+    # combine decoding results
+    best_recog_path = os.path.join(output_dir, "1best_recog")
+    os.mkdir(best_recog_path)
+    files = ["text", "token", "score"]
+    for file in files:
+        with open(os.path.join(best_recog_path, file), "w") as f:
+            for i in range(nj):
+                job_file = os.path.join(output_dir, "output.{}/1best_recog".format(str(i + 1)), file)
+                with open(job_file) as f_job:
+                    lines = f_job.readlines()
+                f.writelines(lines)
+
+    # If text exists, compute CER
+    text_in = os.path.join(params["data_dir"], "text")
+    if os.path.exists(text_in):
+        text_proc_file = os.path.join(best_recog_path, "token")
+        compute_wer(text_in, text_proc_file, os.path.join(best_recog_path, "text.cer"))
+        os.system("tail -n 3 {}".format(os.path.join(best_recog_path, "text.cer")))
+
+
+if __name__ == "__main__":
+    params = {}
+    params["data_dir"] = "./data/test"
+    params["output_dir"] = "./results"
+    params["ngpu"] = 1
+    params["njob"] = 8
+    modelscope_infer(params)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/infer_after_finetune.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/infer_after_finetune.py
new file mode 100644
index 000000000..f9fb0db8a
--- /dev/null
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/infer_after_finetune.py
@@ -0,0 +1,54 @@
+import json
+import os
+import shutil
+
+from modelscope.pipelines import pipeline
+from modelscope.utils.constant import Tasks
+
+from funasr.utils.compute_wer import compute_wer
+
+
+def modelscope_infer_after_finetune(params):
+    # prepare for decoding
+    pretrained_model_path = os.path.join(os.environ["HOME"], ".cache/modelscope/hub", params["modelscope_model_name"])
+    for file_name in params["required_files"]:
+        if file_name == "configuration.json":
+            with open(os.path.join(pretrained_model_path, file_name)) as f:
+                config_dict = json.load(f)
+                config_dict["model"]["am_model_name"] = params["decoding_model_name"]
+            with open(os.path.join(params["output_dir"], "configuration.json"), "w") as f:
+                json.dump(config_dict, f, indent=4, separators=(',', ': '))
+        else:
+            shutil.copy(os.path.join(pretrained_model_path, file_name),
+                        os.path.join(params["output_dir"], file_name))
+    decoding_path = os.path.join(params["output_dir"], "decode_results")
+    if os.path.exists(decoding_path):
+        shutil.rmtree(decoding_path)
+    os.mkdir(decoding_path)
+
+    # decoding
+    inference_pipeline = pipeline(
+        task=Tasks.auto_speech_recognition,
+        model=params["output_dir"],
+        output_dir=decoding_path,
+        batch_size=1
+    )
+    audio_in = os.path.join(params["data_dir"], "wav.scp")
+    inference_pipeline(audio_in=audio_in)
+
+    # computer CER if GT text is set
+    text_in = os.path.join(params["data_dir"], "text")
+    if os.path.exists(text_in):
+        text_proc_file = os.path.join(decoding_path, "1best_recog/token")
+        compute_wer(text_in, text_proc_file, os.path.join(decoding_path, "text.cer"))
+        os.system("tail -n 3 {}".format(os.path.join(decoding_path, "text.cer")))
+
+
+if __name__ == '__main__':
+    params = {}
+    params["modelscope_model_name"] = "damo/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online"
+    params["required_files"] = ["am.mvn", "decoding.yaml", "configuration.json"]
+    params["output_dir"] = "./checkpoint"
+    params["data_dir"] = "./data/test"
+    params["decoding_model_name"] = "20epoch.pth"
+    modelscope_infer_after_finetune(params)