diff --git a/examples/industrial_data_pretraining/ct-transformer/infer.sh b/examples/industrial_data_pretraining/ct-transformer/infer.sh
index f6c5c2354..bd8ac0567 100644
--- a/examples/industrial_data_pretraining/ct-transformer/infer.sh
+++ b/examples/industrial_data_pretraining/ct-transformer/infer.sh
@@ -1,14 +1,13 @@
 
 # download model
-local_path_root=./modelscope_models
+local_path_root=../modelscope_models
 mkdir -p ${local_path_root}
 local_path=${local_path_root}/punc_ct-transformer_zh-cn-common-vocab272727-pytorch
-git clone https://www.modelscope.cn/damo/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404.git ${local_path}
+git clone https://www.modelscope.cn/damo/punc_ct-transformer_zh-cn-common-vocab272727-pytorch.git ${local_path}
 
 
 python funasr/bin/inference.py \
 +model="${local_path}" \
 +input="${local_path}/example/punc_example.txt" \
 +output_dir="./outputs/debug" \
-+device="cpu" \
-+debug="true"
++device="cpu"
diff --git a/examples/industrial_data_pretraining/fsmn-vad/demo.py b/examples/industrial_data_pretraining/fsmn-vad/demo.py
index b3e9bb6ea..6c112e2fd 100644
--- a/examples/industrial_data_pretraining/fsmn-vad/demo.py
+++ b/examples/industrial_data_pretraining/fsmn-vad/demo.py
@@ -5,7 +5,7 @@
 
 from funasr import AutoModel
 
-model = AutoModel(model="/Users/zhifu/Downloads/modelscope_models/speech_fsmn_vad_zh-cn-16k-common-pytorch")
+model = AutoModel(model="../modelscope_models/speech_fsmn_vad_zh-cn-16k-common-pytorch")
 
-res = model(input="/Users/zhifu/Downloads/modelscope_models/speech_fsmn_vad_zh-cn-16k-common-pytorch/example/vad_example.wav")
+res = model(input="../modelscope_models/speech_fsmn_vad_zh-cn-16k-common-pytorch/example/vad_example.wav")
 print(res)
\ No newline at end of file
diff --git a/examples/industrial_data_pretraining/fsmn-vad/infer.sh b/examples/industrial_data_pretraining/fsmn-vad/infer.sh
index 7662a5314..94e1b3db7 100644
--- a/examples/industrial_data_pretraining/fsmn-vad/infer.sh
+++ b/examples/industrial_data_pretraining/fsmn-vad/infer.sh
@@ -1,6 +1,6 @@
 
 # download model
-local_path_root=./modelscope_models
+local_path_root=../modelscope_models
 mkdir -p ${local_path_root}
 local_path=${local_path_root}/speech_fsmn_vad_zh-cn-16k-common-pytorch
 git clone https://www.modelscope.cn/damo/speech_fsmn_vad_zh-cn-16k-common-pytorch.git ${local_path}
diff --git a/examples/industrial_data_pretraining/paraformer-long/demo.py b/examples/industrial_data_pretraining/paraformer-long/demo.py
index ca61ee328..83e024e08 100644
--- a/examples/industrial_data_pretraining/paraformer-long/demo.py
+++ b/examples/industrial_data_pretraining/paraformer-long/demo.py
@@ -5,10 +5,10 @@
 
 from funasr import AutoModel
 
-model = AutoModel(model="../modelscope_models/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
+model = AutoModel(model="../modelscope_models/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
                   vad_model="../modelscope_models/speech_fsmn_vad_zh-cn-16k-common-pytorch",
                   punc_model="../modelscope_models/punc_ct-transformer_zh-cn-common-vocab272727-pytorch",
                   )
 
-res = model(input="../modelscope_models/speech_fsmn_vad_zh-cn-16k-common-pytorch/example/asr_example.wav", batch_size_s=300, batch_size_threshold_s=60)
+res = model(input="../modelscope_models/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch/example/asr_example.wav", batch_size_s=300, batch_size_threshold_s=60)
 print(res)
\ No newline at end of file
diff --git a/examples/industrial_data_pretraining/paraformer-long/infer.sh b/examples/industrial_data_pretraining/paraformer-long/infer.sh
index 39942195d..2ec237b23 100644
--- a/examples/industrial_data_pretraining/paraformer-long/infer.sh
+++ b/examples/industrial_data_pretraining/paraformer-long/infer.sh
@@ -3,8 +3,8 @@
 local_path_root=../modelscope_models
 mkdir -p ${local_path_root}
 
-local_path=${local_path_root}/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch
-git clone https://www.modelscope.cn/damo/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404.git ${local_path}
+local_path=${local_path_root}/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch
+git clone https://www.modelscope.cn/damo/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch.git ${local_path}
 
 local_path_vad=${local_path_root}/speech_fsmn_vad_zh-cn-16k-common-pytorch
 git clone https://www.modelscope.cn/damo/speech_fsmn_vad_zh-cn-16k-common-pytorch.git ${local_path_vad}
diff --git a/examples/industrial_data_pretraining/seaco_paraformer/demo.py b/examples/industrial_data_pretraining/seaco_paraformer/demo.py
index 039d3f3ac..9aec94a08 100644
--- a/examples/industrial_data_pretraining/seaco_paraformer/demo.py
+++ b/examples/industrial_data_pretraining/seaco_paraformer/demo.py
@@ -5,12 +5,12 @@
 
 from funasr import AutoModel
 
-model = AutoModel(model="../modelscope_models/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404",
+model = AutoModel(model="../modelscope_models/speech_seaco_paraformer_large_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
                   )
 
 #vad_model="../modelscope_models/speech_fsmn_vad_zh-cn-16k-common-pytorch",
 #punc_model="../modelscope_models/punc_ct-transformer_zh-cn-common-vocab272727-pytorch"
 
-res = model(input="../modelscope_models/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404/example/asr_example.wav",
+res = model(input="../modelscope_models/speech_seaco_paraformer_large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/example/asr_example.wav",
             hotword='达摩院 磨搭')
 print(res)
\ No newline at end of file
diff --git a/examples/industrial_data_pretraining/tp_aligner/README_zh.md b/examples/industrial_data_pretraining/tp_aligner/README_zh.md
new file mode 100644
index 000000000..8ddb202a2
--- /dev/null
+++ b/examples/industrial_data_pretraining/tp_aligner/README_zh.md
@@ -0,0 +1,42 @@
+(简体中文|[English](./README.md))
+
+# 语音识别
+
+> **注意**:
+> pipeline 支持 [modelscope模型仓库](https://alibaba-damo-academy.github.io/FunASR/en/model_zoo/modelscope_models.html#pretrained-models-on-modelscope) 中的所有模型进行推理和微调。这里我们以典型模型作为示例来演示使用方法。
+
+## 推理
+
+### 快速使用
+#### [Paraformer 模型](https://www.modelscope.cn/models/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/summary)
+```python
+from funasr import AutoModel
+
+model = AutoModel(model="/Users/zhifu/Downloads/modelscope_models/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch")
+
+res = model(input="/Users/zhifu/Downloads/modelscope_models/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/example/asr_example.wav")
+print(res)
+```
+
+### API接口说明
+#### AutoModel 定义
+- `model`: [模型仓库](https://alibaba-damo-academy.github.io/FunASR/en/model_zoo/modelscope_models.html#pretrained-models-on-modelscope) 中的模型名称，或本地磁盘中的模型路径
+- `device`: `cuda`（默认），使用 GPU 进行推理。如果为`cpu`，则使用 CPU 进行推理
+- `ncpu`: `None` （默认），设置用于 CPU 内部操作并行性的线程数
+- `output_dir`: `None` （默认），如果设置，输出结果的输出路径
+- `batch_size`: `1` （默认），解码时的批处理大小
+#### AutoModel 推理
+- `input`: 要解码的输入，可以是：
+  - wav文件路径, 例如: asr_example.wav
+  - pcm文件路径, 例如: asr_example.pcm，此时需要指定音频采样率fs（默认为16000）
+  - 音频字节数流，例如：麦克风的字节数数据
+  - wav.scp，kaldi 样式的 wav 列表 (`wav_id \t wav_path`), 例如:
+  ```text
+  asr_example1  ./audios/asr_example1.wav
+  asr_example2  ./audios/asr_example2.wav
+  ```
+  在这种输入 `wav.scp` 的情况下，必须设置 `output_dir` 以保存输出结果
+  - 音频采样点，例如：`audio, rate = soundfile.read("asr_example_zh.wav")`, 数据类型为 numpy.ndarray。支持batch输入，类型为list：
+  ```[audio_sample1, audio_sample2, ..., audio_sampleN]```
+  - fbank输入，支持组batch。shape为[batch, frames, dim]，类型为torch.Tensor，例如
+- `output_dir`: None （默认），如果设置，输出结果的输出路径
diff --git a/examples/industrial_data_pretraining/tp_aligner/demo.py b/examples/industrial_data_pretraining/tp_aligner/demo.py
new file mode 100644
index 000000000..0d4be5b99
--- /dev/null
+++ b/examples/industrial_data_pretraining/tp_aligner/demo.py
@@ -0,0 +1,15 @@
+#!/usr/bin/env python3
+# -*- encoding: utf-8 -*-
+# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
+#  MIT License  (https://opensource.org/licenses/MIT)
+
+from funasr import AutoModel
+
+model = AutoModel(model="/Users/zhifu/modelscope_models/speech_timestamp_prediction-v1-16k-offline")
+
+res = model(input=("/Users/zhifu/funasr_github/test_local/wav.scp",
+                   "/Users/zhifu/funasr_github/test_local/text.txt"),
+            data_type=("sound", "text"),
+            batch_size=2,
+            )
+print(res)
diff --git a/examples/industrial_data_pretraining/tp_aligner/infer.sh b/examples/industrial_data_pretraining/tp_aligner/infer.sh
new file mode 100644
index 000000000..ded296f74
--- /dev/null
+++ b/examples/industrial_data_pretraining/tp_aligner/infer.sh
@@ -0,0 +1,17 @@
+
+# download model
+local_path_root=../modelscope_models
+mkdir -p ${local_path_root}
+local_path=${local_path_root}/speech_timestamp_prediction-v1-16k-offline
+git clone https://www.modelscope.cn/damo/speech_timestamp_prediction-v1-16k-offline.git ${local_path}
+
+
+python funasr/bin/inference.py \
++model="${local_path}" \
++input='["/Users/zhifu/funasr_github/test_local/wav.scp", "/Users/zhifu/funasr_github/test_local/text.txt"]' \
++data_type='["sound", "text"]' \
++output_dir="./outputs/debug" \
++device="cpu" \
++batch_size=2 \
++debug="true"
+
diff --git a/funasr/bin/inference.py b/funasr/bin/inference.py
index 77b46f7f6..1fac92e02 100644
--- a/funasr/bin/inference.py
+++ b/funasr/bin/inference.py
@@ -4,11 +4,11 @@ import torch
 import numpy as np
 import hydra
 import json
-from omegaconf import DictConfig, OmegaConf
+from omegaconf import DictConfig, OmegaConf, ListConfig
 import logging
 from funasr.download.download_from_hub import download_model
 from funasr.train_utils.set_all_random_seed import set_all_random_seed
-from funasr.datasets.audio_datasets.load_audio_extract_fbank import load_bytes
+from funasr.utils.load_utils import load_bytes
 from funasr.train_utils.device_funcs import to_device
 from tqdm import tqdm
 from funasr.train_utils.load_pretrained_model import load_pretrained_model
@@ -17,11 +17,11 @@ import random
 import string
 from funasr.register import tables
 
-from funasr.datasets.audio_datasets.load_audio_extract_fbank import load_audio, extract_fbank
+from funasr.utils.load_utils import load_audio_and_text_image_video, extract_fbank
 from funasr.utils.vad_utils import slice_padding_audio_samples
 from funasr.utils.timestamp_tools import time_stamp_sentence
 
-def build_iter_for_infer(data_in, input_len=None, data_type="sound", key=None):
+def build_iter_for_infer(data_in, input_len=None, data_type=None, key=None):
 	"""
 	
 	:param input:
@@ -58,9 +58,19 @@ def build_iter_for_infer(data_in, input_len=None, data_type="sound", key=None):
 			key = "rand_key_" + ''.join(random.choice(chars) for _ in range(13))
 			data_list = [data_in]
 			key_list = [key]
-	elif isinstance(data_in, (list, tuple)): # [audio sample point, fbank]
-		data_list = data_in
-		key_list = ["rand_key_" + ''.join(random.choice(chars) for _ in range(13)) for _ in range(len(data_in))]
+	elif isinstance(data_in, (list, tuple)):
+		if data_type is not None and isinstance(data_type, (list, tuple)):
+			data_list_tmp = []
+			for data_in_i, data_type_i in zip(data_in, data_type):
+				key_list, data_list_i = build_iter_for_infer(data_in=data_in_i, data_type=data_type_i)
+				data_list_tmp.append(data_list_i)
+			data_list = []
+			for item in zip(*data_list_tmp):
+				data_list.append(item)
+		else:
+			# [audio sample point, fbank]
+			data_list = data_in
+			key_list = ["rand_key_" + ''.join(random.choice(chars) for _ in range(13)) for _ in range(len(data_in))]
 	else: # raw text; audio sample point, fbank; bytes
 		if isinstance(data_in, bytes): # audio bytes
 			data_in = load_bytes(data_in)
@@ -72,7 +82,16 @@ def build_iter_for_infer(data_in, input_len=None, data_type="sound", key=None):
 	return key_list, data_list
 
 @hydra.main(config_name=None, version_base=None)
-def main_hydra(kwargs: DictConfig):
+def main_hydra(cfg: DictConfig):
+	def to_plain_list(cfg_item):
+		if isinstance(cfg_item, ListConfig):
+			return OmegaConf.to_container(cfg_item, resolve=True)
+		elif isinstance(cfg_item, DictConfig):
+			return {k: to_plain_list(v) for k, v in cfg_item.items()}
+		else:
+			return cfg_item
+	
+	kwargs = to_plain_list(cfg)
 	log_level = getattr(logging, kwargs.get("log_level", "INFO").upper())
 
 	logging.basicConfig(level=log_level)
@@ -125,7 +144,7 @@ class AutoModel:
 		device = kwargs.get("device", "cuda")
 		if not torch.cuda.is_available() or kwargs.get("ngpu", 0):
 			device = "cpu"
-			kwargs["batch_size"] = 1
+			# kwargs["batch_size"] = 1
 		kwargs["device"] = device
 		
 		if kwargs.get("ncpu", None):
@@ -182,8 +201,8 @@ class AutoModel:
 		
 		data_type = kwargs.get("data_type", "sound")
 		batch_size = kwargs.get("batch_size", 1)
-		if kwargs.get("device", "cpu") == "cpu":
-			batch_size = 1
+		# if kwargs.get("device", "cpu") == "cpu":
+		# 	batch_size = 1
 		
 		key_list, data_list = build_iter_for_infer(input, input_len=input_len, data_type=data_type, key=key)
 		
@@ -259,7 +278,7 @@ class AutoModel:
 			key = res[i]["key"]
 			vadsegments = res[i]["value"]
 			input_i = data_list[i]
-			speech = load_audio(input_i, fs=kwargs["frontend"].fs, audio_fs=kwargs.get("fs", 16000))
+			speech = load_audio_and_text_image_video(input_i, fs=kwargs["frontend"].fs, audio_fs=kwargs.get("fs", 16000))
 			speech_lengths = len(speech)
 			n = len(vadsegments)
 			data_with_index = [(vadsegments[i], i) for i in range(n)]
@@ -398,7 +417,7 @@ class AutoFrontend:
 
 			# extract fbank feats
 			time1 = time.perf_counter()
-			audio_sample_list = load_audio(data_batch, fs=self.frontend.fs, audio_fs=kwargs.get("fs", 16000))
+			audio_sample_list = load_audio_and_text_image_video(data_batch, fs=self.frontend.fs, audio_fs=kwargs.get("fs", 16000))
 			time2 = time.perf_counter()
 			meta_data["load_data"] = f"{time2 - time1:0.3f}"
 			speech, speech_lengths = extract_fbank(audio_sample_list, data_type=kwargs.get("data_type", "sound"),
diff --git a/funasr/datasets/audio_datasets/datasets.py b/funasr/datasets/audio_datasets/datasets.py
index bfdf86a4f..0139c93ac 100644
--- a/funasr/datasets/audio_datasets/datasets.py
+++ b/funasr/datasets/audio_datasets/datasets.py
@@ -8,7 +8,7 @@ import torchaudio
 import time
 import logging
 
-from funasr.datasets.audio_datasets.load_audio_extract_fbank import load_audio, extract_fbank
+from funasr.utils.load_utils import load_audio_and_text_image_video, extract_fbank
 from funasr.register import tables
 
 @tables.register("dataset_classes", "AudioDataset")
diff --git a/funasr/datasets/audio_datasets/load_audio_extract_fbank.py b/funasr/datasets/audio_datasets/load_audio_extract_fbank.py
deleted file mode 100644
index c8883ee14..000000000
--- a/funasr/datasets/audio_datasets/load_audio_extract_fbank.py
+++ /dev/null
@@ -1,76 +0,0 @@
-import os
-import torch
-import json
-import torch.distributed as dist
-import numpy as np
-import kaldiio
-import librosa
-import torchaudio
-import time
-import logging
-from torch.nn.utils.rnn import pad_sequence
-
-def load_audio(audio_or_path_or_list, fs: int=16000, audio_fs: int=16000):
-
-	if isinstance(audio_or_path_or_list, (list, tuple)):
-		return [load_audio(audio, fs=fs, audio_fs=audio_fs) for audio in audio_or_path_or_list]
-	
-	if isinstance(audio_or_path_or_list, str) and os.path.exists(audio_or_path_or_list):
-		audio_or_path_or_list, audio_fs = torchaudio.load(audio_or_path_or_list)
-		audio_or_path_or_list = audio_or_path_or_list[0, :]
-	elif isinstance(audio_or_path_or_list, np.ndarray): # audio sample point
-		audio_or_path_or_list = np.squeeze(audio_or_path_or_list) #[n_samples,]
-		
-	if audio_fs != fs:
-		resampler = torchaudio.transforms.Resample(audio_fs, fs)
-		resampled_waveform = resampler(audio_or_path_or_list[None, :])[0, :]
-	return audio_or_path_or_list
-#
-# def load_audio_from_list(audio_list, fs: int=16000, audio_fs: int=16000):
-# 	if isinstance(audio_list, (list, tuple)):
-# 		return [load_audio(audio_or_path, fs=fs, audio_fs=audio_fs) for audio_or_path in audio_list]
-
-
-def load_bytes(input):
-	middle_data = np.frombuffer(input, dtype=np.int16)
-	middle_data = np.asarray(middle_data)
-	if middle_data.dtype.kind not in 'iu':
-		raise TypeError("'middle_data' must be an array of integers")
-	dtype = np.dtype('float32')
-	if dtype.kind != 'f':
-		raise TypeError("'dtype' must be a floating point type")
-	
-	i = np.iinfo(middle_data.dtype)
-	abs_max = 2 ** (i.bits - 1)
-	offset = i.min + abs_max
-	array = np.frombuffer((middle_data.astype(dtype) - offset) / abs_max, dtype=np.float32)
-	return array
-
-def extract_fbank(data, data_len = None, data_type: str="sound", frontend=None):
-	# import pdb;
-	# pdb.set_trace()
-	if isinstance(data, np.ndarray):
-		data = torch.from_numpy(data)
-		if len(data.shape) < 2:
-			data = data[None, :] # data: [batch, N]
-		data_len = [data.shape[1]] if data_len is None else data_len
-	elif isinstance(data, torch.Tensor):
-		if len(data.shape) < 2:
-			data = data[None, :] # data: [batch, N]
-		data_len = [data.shape[1]] if data_len is None else data_len
-	elif isinstance(data, (list, tuple)):
-		data_list, data_len = [], []
-		for data_i in data:
-			if isinstance(data, np.ndarray):
-				data_i = torch.from_numpy(data_i)
-			data_list.append(data_i)
-			data_len.append(data_i.shape[0])
-		data = pad_sequence(data_list, batch_first=True) # data: [batch, N]
-	# import pdb;
-	# pdb.set_trace()
-	if data_type == "sound":
-		data, data_len = frontend(data, data_len)
-	
-	if isinstance(data_len, (list, tuple)):
-		data_len = torch.tensor([data_len])
-	return data.to(torch.float32), data_len.to(torch.int32)
\ No newline at end of file
diff --git a/funasr/models/bicif_paraformer/model.py b/funasr/models/bicif_paraformer/model.py
index 25b0462fb..aced0883f 100644
--- a/funasr/models/bicif_paraformer/model.py
+++ b/funasr/models/bicif_paraformer/model.py
@@ -23,7 +23,7 @@ from funasr.train_utils.device_funcs import force_gatherable
 
 from funasr.models.paraformer.search import Hypothesis
 
-from funasr.datasets.audio_datasets.load_audio_extract_fbank import load_audio, extract_fbank
+from funasr.utils.load_utils import load_audio_and_text_image_video, extract_fbank
 from funasr.utils import postprocess_utils
 from funasr.utils.datadir_writer import DatadirWriter
 from funasr.utils.timestamp_tools import ts_prediction_lfr6_standard
@@ -243,7 +243,7 @@ class BiCifParaformer(Paraformer):
         else:
             # extract fbank feats
             time1 = time.perf_counter()
-            audio_sample_list = load_audio(data_in, fs=frontend.fs, audio_fs=kwargs.get("fs", 16000))
+            audio_sample_list = load_audio_and_text_image_video(data_in, fs=frontend.fs, audio_fs=kwargs.get("fs", 16000))
             time2 = time.perf_counter()
             meta_data["load_data"] = f"{time2 - time1:0.3f}"
             speech, speech_lengths = extract_fbank(audio_sample_list, data_type=kwargs.get("data_type", "sound"),
diff --git a/funasr/models/contextual_paraformer/model.py b/funasr/models/contextual_paraformer/model.py
index 1bb2d3262..c277ffc16 100644
--- a/funasr/models/contextual_paraformer/model.py
+++ b/funasr/models/contextual_paraformer/model.py
@@ -46,7 +46,7 @@ else:
 	@contextmanager
 	def autocast(enabled=True):
 		yield
-from funasr.datasets.audio_datasets.load_audio_extract_fbank import load_audio, extract_fbank
+from funasr.utils.load_utils import load_audio_and_text_image_video, extract_fbank
 from funasr.utils import postprocess_utils
 from funasr.utils.datadir_writer import DatadirWriter
 
@@ -337,7 +337,7 @@ class ContextualParaformer(Paraformer):
 		
 		# extract fbank feats
 		time1 = time.perf_counter()
-		audio_sample_list = load_audio(data_in, fs=frontend.fs, audio_fs=kwargs.get("fs", 16000))
+		audio_sample_list = load_audio_and_text_image_video(data_in, fs=frontend.fs, audio_fs=kwargs.get("fs", 16000))
 		time2 = time.perf_counter()
 		meta_data["load_data"] = f"{time2 - time1:0.3f}"
 		speech, speech_lengths = extract_fbank(audio_sample_list, data_type=kwargs.get("data_type", "sound"),
diff --git a/funasr/models/fsmn_vad/model.py b/funasr/models/fsmn_vad/model.py
index 488e05e9a..ee1955856 100644
--- a/funasr/models/fsmn_vad/model.py
+++ b/funasr/models/fsmn_vad/model.py
@@ -9,7 +9,7 @@ import math
 from typing import Optional
 import time
 from funasr.register import tables
-from funasr.datasets.audio_datasets.load_audio_extract_fbank import load_audio,extract_fbank
+from funasr.utils.load_utils import load_audio_and_text_image_video,extract_fbank
 from funasr.utils.datadir_writer import DatadirWriter
 from torch.nn.utils.rnn import pad_sequence
 
@@ -544,7 +544,7 @@ class FsmnVAD(nn.Module):
         else:
             # extract fbank feats
             time1 = time.perf_counter()
-            audio_sample_list = load_audio(data_in, fs=frontend.fs, audio_fs=kwargs.get("fs", 16000))
+            audio_sample_list = load_audio_and_text_image_video(data_in, fs=frontend.fs, audio_fs=kwargs.get("fs", 16000))
             time2 = time.perf_counter()
             meta_data["load_data"] = f"{time2 - time1:0.3f}"
             speech, speech_lengths = extract_fbank(audio_sample_list, data_type=kwargs.get("data_type", "sound"),
diff --git a/funasr/models/paraformer/model.py b/funasr/models/paraformer/model.py
index 4ce2c3213..54924205f 100644
--- a/funasr/models/paraformer/model.py
+++ b/funasr/models/paraformer/model.py
@@ -22,7 +22,7 @@ from funasr.models.paraformer.search import Hypothesis
 
 from torch.cuda.amp import autocast
 
-from funasr.datasets.audio_datasets.load_audio_extract_fbank import load_audio, extract_fbank
+from funasr.utils.load_utils import load_audio_and_text_image_video, extract_fbank, load_audio_and_text_image_video
 from funasr.utils import postprocess_utils
 from funasr.utils.datadir_writer import DatadirWriter
 from funasr.register import tables
@@ -466,7 +466,7 @@ class Paraformer(nn.Module):
 		else:
 			# extract fbank feats
 			time1 = time.perf_counter()
-			audio_sample_list = load_audio(data_in, fs=frontend.fs, audio_fs=kwargs.get("fs", 16000))
+			audio_sample_list = load_audio_and_text_image_video(data_in, fs=frontend.fs, audio_fs=kwargs.get("fs", 16000), data_type=kwargs.get("data_type", "sound"), tokenizer=tokenizer)
 			time2 = time.perf_counter()
 			meta_data["load_data"] = f"{time2 - time1:0.3f}"
 			speech, speech_lengths = extract_fbank(audio_sample_list, data_type=kwargs.get("data_type", "sound"), frontend=frontend)
diff --git a/funasr/models/paraformer_streaming/model.py b/funasr/models/paraformer_streaming/model.py
index bb244699f..a57c92764 100644
--- a/funasr/models/paraformer_streaming/model.py
+++ b/funasr/models/paraformer_streaming/model.py
@@ -40,7 +40,7 @@ else:
 	@contextmanager
 	def autocast(enabled=True):
 		yield
-from funasr.datasets.audio_datasets.load_audio_extract_fbank import load_audio, extract_fbank
+from funasr.utils.load_utils import load_audio_and_text_image_video, extract_fbank
 from funasr.utils import postprocess_utils
 from funasr.utils.datadir_writer import DatadirWriter
 from funasr.utils.timestamp_tools import ts_prediction_lfr6_standard
@@ -483,7 +483,7 @@ class Paraformer(nn.Module):
 		meta_data = {}
 		# extract fbank feats
 		time1 = time.perf_counter()
-		audio_sample_list = load_audio(data_in, fs=self.frontend.fs, audio_fs=kwargs.get("fs", 16000))
+		audio_sample_list = load_audio_and_text_image_video(data_in, fs=self.frontend.fs, audio_fs=kwargs.get("fs", 16000))
 		time2 = time.perf_counter()
 		meta_data["load_data"] = f"{time2 - time1:0.3f}"
 		speech, speech_lengths = extract_fbank(audio_sample_list, data_type=kwargs.get("data_type", "sound"), frontend=self.frontend)
@@ -761,7 +761,7 @@ class BiCifParaformer(Paraformer):
 		meta_data = {}
 		# extract fbank feats
 		time1 = time.perf_counter()
-		audio_sample_list = load_audio(data_in, fs=self.frontend.fs, audio_fs=kwargs.get("fs", 16000))
+		audio_sample_list = load_audio_and_text_image_video(data_in, fs=self.frontend.fs, audio_fs=kwargs.get("fs", 16000))
 		time2 = time.perf_counter()
 		meta_data["load_data"] = f"{time2 - time1:0.3f}"
 		speech, speech_lengths = extract_fbank(audio_sample_list, data_type=kwargs.get("data_type", "sound"),
diff --git a/funasr/models/seaco_paraformer/model.py b/funasr/models/seaco_paraformer/model.py
index d107a57e4..2de125a61 100644
--- a/funasr/models/seaco_paraformer/model.py
+++ b/funasr/models/seaco_paraformer/model.py
@@ -35,7 +35,7 @@ else:
     @contextmanager
     def autocast(enabled=True):
         yield
-from funasr.datasets.audio_datasets.load_audio_extract_fbank import load_audio, extract_fbank
+from funasr.utils.load_utils import load_audio_and_text_image_video, extract_fbank
 from funasr.utils import postprocess_utils
 from funasr.utils.datadir_writer import DatadirWriter
 
@@ -327,7 +327,7 @@ class SeacoParaformer(BiCifParaformer, Paraformer):
         
         # extract fbank feats
         time1 = time.perf_counter()
-        audio_sample_list = load_audio(data_in, fs=frontend.fs, audio_fs=kwargs.get("fs", 16000))
+        audio_sample_list = load_audio_and_text_image_video(data_in, fs=frontend.fs, audio_fs=kwargs.get("fs", 16000))
         time2 = time.perf_counter()
         meta_data["load_data"] = f"{time2 - time1:0.3f}"
         speech, speech_lengths = extract_fbank(audio_sample_list, data_type=kwargs.get("data_type", "sound"),
diff --git a/funasr/models/transducer/model.py b/funasr/models/transducer/model.py
index 21de4ba69..9d9ae4bae 100644
--- a/funasr/models/transducer/model.py
+++ b/funasr/models/transducer/model.py
@@ -45,7 +45,7 @@ else:
 	@contextmanager
 	def autocast(enabled=True):
 		yield
-from funasr.datasets.audio_datasets.load_audio_extract_fbank import load_audio, extract_fbank
+from funasr.utils.load_utils import load_audio_and_text_image_video, extract_fbank
 from funasr.utils import postprocess_utils
 from funasr.utils.datadir_writer import DatadirWriter
 from funasr.models.transformer.utils.nets_utils import get_transducer_task_io
@@ -517,7 +517,7 @@ class Transducer(nn.Module):
 		meta_data = {}
 		# extract fbank feats
 		time1 = time.perf_counter()
-		audio_sample_list = load_audio(data_in, fs=self.frontend.fs, audio_fs=kwargs.get("fs", 16000))
+		audio_sample_list = load_audio_and_text_image_video(data_in, fs=self.frontend.fs, audio_fs=kwargs.get("fs", 16000))
 		time2 = time.perf_counter()
 		meta_data["load_data"] = f"{time2 - time1:0.3f}"
 		speech, speech_lengths = extract_fbank(audio_sample_list, data_type=kwargs.get("data_type", "sound"), frontend=self.frontend)
diff --git a/funasr/models/transformer/model.py b/funasr/models/transformer/model.py
index b710c9732..f09f460ec 100644
--- a/funasr/models/transformer/model.py
+++ b/funasr/models/transformer/model.py
@@ -12,7 +12,7 @@ from funasr.models.transformer.utils.add_sos_eos import add_sos_eos
 from funasr.metrics.compute_acc import th_accuracy
 # from funasr.models.e2e_asr_common import ErrorCalculator
 from funasr.train_utils.device_funcs import force_gatherable
-from funasr.datasets.audio_datasets.load_audio_extract_fbank import load_audio, extract_fbank
+from funasr.utils.load_utils import load_audio_and_text_image_video, extract_fbank
 from funasr.utils import postprocess_utils
 from funasr.utils.datadir_writer import DatadirWriter
 from funasr.register import tables
@@ -392,7 +392,7 @@ class Transformer(nn.Module):
 		meta_data = {}
 		# extract fbank feats
 		time1 = time.perf_counter()
-		audio_sample_list = load_audio(data_in, fs=self.frontend.fs, audio_fs=kwargs.get("fs", 16000))
+		audio_sample_list = load_audio_and_text_image_video(data_in, fs=self.frontend.fs, audio_fs=kwargs.get("fs", 16000))
 		time2 = time.perf_counter()
 		meta_data["load_data"] = f"{time2 - time1:0.3f}"
 		speech, speech_lengths = extract_fbank(audio_sample_list, data_type=kwargs.get("data_type", "sound"), frontend=self.frontend)
diff --git a/funasr/utils/load_utils.py b/funasr/utils/load_utils.py
new file mode 100644
index 000000000..7f1b850fd
--- /dev/null
+++ b/funasr/utils/load_utils.py
@@ -0,0 +1,102 @@
+import os
+import torch
+import json
+import torch.distributed as dist
+import numpy as np
+import kaldiio
+import librosa
+import torchaudio
+import time
+import logging
+from torch.nn.utils.rnn import pad_sequence
+
+# def load_audio(audio_or_path_or_list, fs: int=16000, audio_fs: int=16000):
+#
+# 	if isinstance(audio_or_path_or_list, (list, tuple)):
+# 		return [load_audio(audio, fs=fs, audio_fs=audio_fs) for audio in audio_or_path_or_list]
+#
+# 	if isinstance(audio_or_path_or_list, str) and os.path.exists(audio_or_path_or_list):
+# 		audio_or_path_or_list, audio_fs = torchaudio.load(audio_or_path_or_list)
+# 		audio_or_path_or_list = audio_or_path_or_list[0, :]
+# 	elif isinstance(audio_or_path_or_list, np.ndarray): # audio sample point
+# 		audio_or_path_or_list = np.squeeze(audio_or_path_or_list) #[n_samples,]
+#
+# 	if audio_fs != fs:
+# 		resampler = torchaudio.transforms.Resample(audio_fs, fs)
+# 		audio_or_path_or_list = resampler(audio_or_path_or_list[None, :])[0, :]
+# 	return audio_or_path_or_list
+
+
+def load_audio_and_text_image_video(audio_or_path_or_list, fs: int = 16000, audio_fs: int = 16000, data_type=None, tokenizer=None):
+	if isinstance(audio_or_path_or_list, (list, tuple)):
+		if data_type is not None and isinstance(data_type, (list, tuple)):
+
+			data_types = [data_type] * len(audio_or_path_or_list)
+			audio_or_path_or_list_ret = [[] for d in data_type]
+			for i, (data_type_i, audio_or_path_or_list_i) in enumerate(zip(data_types, audio_or_path_or_list)):
+				
+				for j, (data_type_j, audio_or_path_or_list_j) in enumerate(zip(data_type_i, audio_or_path_or_list_i)):
+					
+					audio_or_path_or_list_j = load_audio_and_text_image_video(audio_or_path_or_list_j, fs=fs, audio_fs=audio_fs, data_type=data_type_j, tokenizer=tokenizer)
+					audio_or_path_or_list_ret[j].append(audio_or_path_or_list_j)
+
+			return audio_or_path_or_list_ret
+		else:
+			return [load_audio_and_text_image_video(audio, fs=fs, audio_fs=audio_fs) for audio in audio_or_path_or_list]
+	
+	if isinstance(audio_or_path_or_list, str) and os.path.exists(audio_or_path_or_list):
+		audio_or_path_or_list, audio_fs = torchaudio.load(audio_or_path_or_list)
+		audio_or_path_or_list = audio_or_path_or_list[0, :]
+	elif isinstance(audio_or_path_or_list, np.ndarray):  # audio sample point
+		audio_or_path_or_list = np.squeeze(audio_or_path_or_list)  # [n_samples,]
+	elif isinstance(audio_or_path_or_list, str) and data_type is not None and data_type == "text" and tokenizer is not None:
+		audio_or_path_or_list = tokenizer.encode(audio_or_path_or_list)
+		
+	if audio_fs != fs and data_type != "text":
+		resampler = torchaudio.transforms.Resample(audio_fs, fs)
+		audio_or_path_or_list = resampler(audio_or_path_or_list[None, :])[0, :]
+	return audio_or_path_or_list
+
+def load_bytes(input):
+	middle_data = np.frombuffer(input, dtype=np.int16)
+	middle_data = np.asarray(middle_data)
+	if middle_data.dtype.kind not in 'iu':
+		raise TypeError("'middle_data' must be an array of integers")
+	dtype = np.dtype('float32')
+	if dtype.kind != 'f':
+		raise TypeError("'dtype' must be a floating point type")
+	
+	i = np.iinfo(middle_data.dtype)
+	abs_max = 2 ** (i.bits - 1)
+	offset = i.min + abs_max
+	array = np.frombuffer((middle_data.astype(dtype) - offset) / abs_max, dtype=np.float32)
+	return array
+
+def extract_fbank(data, data_len = None, data_type: str="sound", frontend=None):
+	# import pdb;
+	# pdb.set_trace()
+	if isinstance(data, np.ndarray):
+		data = torch.from_numpy(data)
+		if len(data.shape) < 2:
+			data = data[None, :] # data: [batch, N]
+		data_len = [data.shape[1]] if data_len is None else data_len
+	elif isinstance(data, torch.Tensor):
+		if len(data.shape) < 2:
+			data = data[None, :] # data: [batch, N]
+		data_len = [data.shape[1]] if data_len is None else data_len
+	elif isinstance(data, (list, tuple)):
+		data_list, data_len = [], []
+		for data_i in data:
+			if isinstance(data, np.ndarray):
+				data_i = torch.from_numpy(data_i)
+			data_list.append(data_i)
+			data_len.append(data_i.shape[0])
+		data = pad_sequence(data_list, batch_first=True) # data: [batch, N]
+	# import pdb;
+	# pdb.set_trace()
+	# if data_type == "sound":
+	data, data_len = frontend(data, data_len)
+	
+	if isinstance(data_len, (list, tuple)):
+		data_len = torch.tensor([data_len])
+	return data.to(torch.float32), data_len.to(torch.int32)
\ No newline at end of file