commit 1385a6f46b485e4a74350a1af532f4b78a012fb5
Author: Lu Xinlong <luxinlong02@4paradigm.com>
Date:   Wed Apr 8 06:41:00 2026 +0000

    initial commit

diff --git a/funasr/Dockerfile.funasr-bi150 b/funasr/Dockerfile.funasr-bi150
new file mode 100644
index 0000000..bf0f46d
--- /dev/null
+++ b/funasr/Dockerfile.funasr-bi150
@@ -0,0 +1,27 @@
+FROM corex:4.3.8
+
+WORKDIR /root
+
+RUN set -eux; \
+  # 1) 把 aliyun 源替换成官方源（避免 403）
+  sed -i -E 's|http://mirrors\.aliyun\.com/ubuntu|http://archive.ubuntu.com/ubuntu|g' /etc/apt/sources.list; \
+  sed -i -E 's|http://mirrors\.aliyun\.com/ubuntu|http://archive.ubuntu.com/ubuntu|g' /etc/apt/sources.list.d/*.list 2>/dev/null || true; \
+  \
+  # 2) 更新并安装
+  apt-get update; \
+  apt-get install -y --no-install-recommends vim net-tools ca-certificates; \
+  rm -rf /var/lib/apt/lists/*
+
+ADD . /root/
+
+COPY requirements.txt /root
+RUN pip install -r /root/requirements.txt -i https://nexus.4pd.io/repository/pypi-all/simple --extra-index-url https://mirror.sjtu.edu.cn/pypi/web/simple
+RUN pip install funasr==1.3.1 'transformers>=4.51.3' openai-whisper \
+    -i https://nexus.4pd.io/repository/pypi-all/simple --extra-index-url https://mirror.sjtu.edu.cn/pypi/web/simple
+# Patch files
+
+COPY ./replaced_files/bi_v150/cif_predictor.py /usr/local/lib/python3.10/site-packages/funasr/models/paraformer/
+COPY ./replaced_files/funasr_nano_model.py /usr/local/lib/python3.10/site-packages/funasr/models/fun_asr_nano/model.py
+
+ENTRYPOINT ["python3"]
+CMD ["main.py"]
\ No newline at end of file
diff --git a/funasr/README.md b/funasr/README.md
new file mode 100644
index 0000000..978c7e4
--- /dev/null
+++ b/funasr/README.md
@@ -0,0 +1,28 @@
+# 天数智芯 天垓150 ASR（FunASR架构）
+
+## 镜像构造
+```shell
+docker build -f ./Dockerfile.funasr-bi150 -t <your_image> .
+```
+其中，基础镜像 corex:4.3.8 通过联系天数智芯智铠100厂商技术支持可获取
+
+## 使用说明
+
+### 使用 FastAPI 启动ASR服务：
+例如：
+```shell
+docker run -dit -v /usr/src:/usr/src -v /lib/modules:/lib/modules --device=/dev/iluvatar0:/dev/iluvatar0 \
+    -v /mnt/contest_ceph/leaderboard/modelHubXC/iic/SenseVoiceSmall:/model \
+    --network=host <your_image> \
+    main.py --model_dir /model --model_type sensevoice --use_gpu --port 1111
+```
+具体参数代码设定可参考代码文件
+
+### 测试ASR服务
+项目根路径`sample_data`目录下附带上了中文的测试音频和附带内容
+
+```shell
+curl -X POST http://localhost:1111/transduce \
+  -F "audio=@../sample_data/lei-jun-test.wav" \
+  -F "lang=zh"
+```
\ No newline at end of file
diff --git a/funasr/fastapi_funasr.py b/funasr/fastapi_funasr.py
new file mode 100644
index 0000000..2db8cf4
--- /dev/null
+++ b/funasr/fastapi_funasr.py
@@ -0,0 +1,271 @@
+import os
+import time
+import argparse
+import torchaudio
+import torch
+import traceback
+
+from fastapi import FastAPI, File, UploadFile, HTTPException, BackgroundTasks, Form
+import uuid
+import uvicorn
+from funasr import AutoModel
+from funasr.utils.postprocess_utils import rich_transcription_postprocess
+# from funasr.models.fun_asr_nano.model import FunASRNano
+
+os.makedirs("./input", exist_ok=True)
+status = "Running"
+model = None
+device = ""
+app = FastAPI()
+
+CUSTOM_DEVICE = os.getenv("CUSTOM_DEVICE", "")
+if CUSTOM_DEVICE.startswith("mlu"):
+    import torch_mlu
+elif CUSTOM_DEVICE.startswith("ascend"):
+    import torch_npu
+elif CUSTOM_DEVICE.startswith("pt"):
+    import torch_dipu
+
+def make_all_dense(module: torch.nn.Module):
+    for name, param in list(module.named_parameters(recurse=True)):
+        if getattr(param, "is_sparse", False) and param.is_sparse:
+            with torch.no_grad():
+                dense = param.to_dense().contiguous()
+            parent = module
+            *mods, leaf = name.split(".")
+            for m in mods:
+                parent = getattr(parent, m)
+            setattr(parent, leaf, torch.nn.Parameter(dense, requires_grad=param.requires_grad))
+
+    # 处理 buffer（如 running_mean 等）
+    for name, buf in list(module.named_buffers(recurse=True)):
+        # PyTorch 稀疏张量 layout 不是 strided
+        if buf.layout != torch.strided:
+            dense = buf.to_dense().contiguous()
+            parent = module
+            *mods, leaf = name.split(".")
+            for m in mods:
+                parent = getattr(parent, m)
+            parent.register_buffer(leaf, dense, persistent=True)
+
+def split_audio(waveform, sample_rate, segment_seconds=20):
+    segment_samples = segment_seconds * sample_rate
+    segments = []
+    for i in range(0, waveform.shape[1], segment_samples):
+        segment = waveform[:, i:i + segment_samples]
+        if segment.shape[1] > 0:
+            segments.append(segment)
+    return segments
+
+# def determine_model_type(model_name):
+#     if "sensevoice" in model_name.lower():
+#         return "sensevoice"
+#     elif "whisper" in model_name.lower():
+#         return "whisper"
+#     elif "paraformer" in model_name.lower():
+#         return "paraformer"
+#     elif "conformer" in model_name.lower():
+#         return "conformer"
+#     elif "uniasr" in model_name.lower():
+#         return "uni_asr"
+#     else:
+#         return "unknown"
+
+@app.on_event("startup")
+def load_model():
+    global status, model, device
+
+    config = app.state.config
+    use_gpu = config.get("use_gpu", True)
+    model_dir = config.get("model_dir", "/model")
+    model_type = config.get("model_type", "sensevoice")
+    warmup = config.get("warmup", False)
+    print(">> Startup config:")
+    print("   model_dir =", model_dir, flush=True)
+    print("   model_type =", model_type, flush=True)
+    print("   use_gpu =", use_gpu, flush=True)
+    print("   warmup =", warmup, flush=True)
+
+    device = "cpu"
+    if use_gpu:
+        if CUSTOM_DEVICE.startswith("mlu"):
+            device = "mlu:0"
+        elif CUSTOM_DEVICE.startswith("ascend"):
+            device = "npu:0"
+        else:
+            device = "cuda:0"
+
+    # 针对加速卡的特殊处理部分
+    if device == "cuda:0" and torch.cuda.get_device_name() == "Iluvatar BI-V100" and model_type == "whisper":
+        # 天垓100情况下的Whisper需要绕过不支持算子
+        torch.backends.cuda.enable_flash_sdp(False)
+        torch.backends.cuda.enable_mem_efficient_sdp(False)
+        torch.backends.cuda.enable_math_sdp(True)
+    print(f"device: {device}", flush=True)
+
+    dense_convert = False
+    if device == "cuda:0" and CUSTOM_DEVICE.startswith("pt") and model_type == "whisper":
+        dense_convert = True
+    if device.startswith("npu") and model_type == "whisper":
+        # Ascend NPU 加载whisper的部分会有Sparse部分device不匹配
+        dense_convert = True
+
+    print(f"dense_convert: {dense_convert}", flush=True)
+    if dense_convert:
+        model = AutoModel(
+            model=model_dir,
+            vad_model=None,
+            disable_update=True,
+            device="cpu"
+        )
+        make_all_dense(model.model)
+        model.model.to(dtype=torch.float32, memory_format=torch.contiguous_format)
+        model.model.to(device)
+        model.kwargs["device"] = device
+    else:
+        # 不使用VAD, punct，spk模型，就测试原始ASR能力
+        model = AutoModel(
+            model=model_dir,
+            # vad_model="fsmn-vad",
+            # vad_kwargs={"max_single_segment_time": 30000},
+            vad_model=None,
+            device=device,
+            disable_update=True
+        )
+
+    if device.startswith("npu") or warmup:
+        # Ascend NPU由于底层设计的不同，初始化卡的调度比其他卡更复杂，要先进行warmup
+        print("Start warmup...", flush=True)
+        res = model.generate(input="warmup.wav")
+        print("warmup complete.", flush=True)
+
+    status = "Success"
+
+
+def test_funasr(audio_file, lang):
+    # 推理部分
+    waveform, sample_rate = torchaudio.load(audio_file)
+    # print(waveform.shape)
+    duration = waveform.shape[1] / sample_rate
+    segments = split_audio(waveform, sample_rate, segment_seconds=20)
+
+    generated_text = ""
+    processing_time = 0
+    model_type = app.state.config.get("model_type", "sensevoice")
+    if model_type == "uni_asr":
+        # uni_asr比较特殊，设计就是处理长音频的（自带VAD），切分的话前20s如果几乎没有人讲话全是音乐直接会报错
+        # 因为可能会被切掉所有音频导致实际编解码输入为0
+        ts1 = time.time()
+        res = model.generate(
+            input=audio_file
+        )
+        generated_text = res[0]["text"]
+        ts2 = time.time()
+        processing_time = ts2 - ts1
+    else:
+        # 按照切分的音频依次输入
+        for i, segment in enumerate(segments):
+            segment_path = f"temp_seg_{i}.wav"
+            torchaudio.save(segment_path, segment, sample_rate)
+            ts1 = time.time()
+            text = None
+            if model_type == "sensevoice":
+                res = model.generate(
+                    input=segment_path,
+                    cache={},
+                    language="auto",  # "zh", "en", "yue", "ja", "ko", "nospeech"
+                    use_itn=True,
+                    batch_size_s=60,
+                    merge_vad=False,
+                    # merge_length_s=15,
+                )
+                text = rich_transcription_postprocess(res[0]["text"])
+            elif model_type == "whisper":
+                DecodingOptions = {
+                    "task": "transcribe",
+                    "language": lang,
+                    "beam_size": None,
+                    "fp16": False,
+                    "without_timestamps": False,
+                    "prompt": None,
+                }
+                res = model.generate(
+                    DecodingOptions=DecodingOptions,
+                    input=segment_path,
+                    batch_size_s=0,
+                )
+                text = res[0]["text"]
+            elif model_type == "paraformer":
+                res = model.generate(
+                    input=segment_path,
+                    batch_size_s=300
+                )
+                text = res[0]["text"]
+                # paraformer模型会一个字一个字输出，中间夹太多空格会影响1-cer的结果
+                if lang == "zh":
+                    text = text.replace(" ", "")
+            elif model_type == "conformer":
+                res = model.generate(
+                    input=segment_path,
+                    batch_size_s=300
+                )
+                text = res[0]["text"]
+            # elif model_type == "uni_asr":
+            #     if i == 0:
+            #         os.remove(segment_path)
+            #         continue
+            #     res = model.generate(
+            #         input=segment_path
+            #     )
+            #     text = res[0]["text"]
+            else:
+                raise RuntimeError("unknown model type")
+            if text is not None:
+                # some models output "▁" (9601, Unicode U+2581) as separator between words, replace them with space for better readability
+                text = text.replace("_", " ")
+                text = text.replace(chr(9601), " ")
+            ts2 = time.time()
+            generated_text += text
+            processing_time += (ts2 - ts1)
+            os.remove(segment_path)
+
+    rtf = processing_time / duration
+    print("Text:", generated_text, flush=True)
+    print(f"Audio duration:\t{duration:.3f} s", flush=True)
+    print(f"Elapsed:\t{processing_time:.3f} s", flush=True)
+    print(f"RTF = {processing_time:.3f}/{duration:.3f} = {rtf:.3f}", flush=True)
+
+    return generated_text
+
+@app.get("/health")
+def health():
+
+    if status=="Running":
+        return {
+            "status":"loading model"
+        }
+    ret = {
+        "status": "ok" if status == "Success" else "failed",
+    }
+    return ret
+
+@app.post("/transduce")
+def transduce(
+    audio: UploadFile = File(...),
+    lang: str = Form("zh"),
+    background_tasks: BackgroundTasks = None
+):
+    try:
+        file_path = f"./input/{uuid.uuid4()}.wav"
+        with open(file_path, "wb") as f:
+            f.write(audio.file.read())
+        background_tasks.add_task(os.remove, file_path)
+        generated_text = test_funasr(file_path, lang)
+
+        return {"generated_text": generated_text}
+    except Exception:
+        raise HTTPException(status_code=500, detail=f"Processing failed: \n{traceback.format_exc()}")
+
+# if __name__ == "__main__":
+
+#     uvicorn.run("fastapi_funasr:app", host="0.0.0.0", port=1111, workers=1)
diff --git a/funasr/main.py b/funasr/main.py
new file mode 100644
index 0000000..ae902f6
--- /dev/null
+++ b/funasr/main.py
@@ -0,0 +1,27 @@
+import argparse
+import uvicorn
+from fastapi_funasr import app
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser()
+    parser.add_argument("--model_dir", type=str, default="/model", help="model directory")
+    parser.add_argument("--model_type", type=str, default="sensevoice", help="model type, e.g. sensevoice")
+    parser.add_argument("--use_gpu", action="store_true", default=True)
+    parser.add_argument("--warmup", action="store_true", help="whether do warmup when first initializing model")
+    parser.add_argument("--port", type=int, default=8000, help="service port")
+
+    args = parser.parse_args()
+
+    # 将参数加到 app.state 中
+    app.state.config = {
+        "model_dir": args.model_dir,
+        "model_type": args.model_type,
+        "use_gpu": args.use_gpu, # True
+        "warmup": args.warmup
+    }
+
+    uvicorn.run("fastapi_funasr:app",
+        host="0.0.0.0",
+        port=args.port,
+        workers=1
+    )
\ No newline at end of file
diff --git a/funasr/replaced_files/bi_v150/cif_predictor.py b/funasr/replaced_files/bi_v150/cif_predictor.py
new file mode 100644
index 0000000..5477804
--- /dev/null
+++ b/funasr/replaced_files/bi_v150/cif_predictor.py
@@ -0,0 +1,762 @@
+#!/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)
+
+import torch
+import logging
+import numpy as np
+
+from funasr.register import tables
+from funasr.train_utils.device_funcs import to_device
+from funasr.models.transformer.utils.nets_utils import make_pad_mask
+from torch.cuda.amp import autocast
+
+
+@tables.register("predictor_classes", "CifPredictor")
+class CifPredictor(torch.nn.Module):
+    def __init__(
+        self,
+        idim,
+        l_order,
+        r_order,
+        threshold=1.0,
+        dropout=0.1,
+        smooth_factor=1.0,
+        noise_threshold=0,
+        tail_threshold=0.45,
+    ):
+        super().__init__()
+
+        self.pad = torch.nn.ConstantPad1d((l_order, r_order), 0)
+        self.cif_conv1d = torch.nn.Conv1d(idim, idim, l_order + r_order + 1, groups=idim)
+        self.cif_output = torch.nn.Linear(idim, 1)
+        self.dropout = torch.nn.Dropout(p=dropout)
+        self.threshold = threshold
+        self.smooth_factor = smooth_factor
+        self.noise_threshold = noise_threshold
+        self.tail_threshold = tail_threshold
+
+    def forward(
+        self,
+        hidden,
+        target_label=None,
+        mask=None,
+        ignore_id=-1,
+        mask_chunk_predictor=None,
+        target_label_length=None,
+    ):
+
+        with autocast(False):
+            h = hidden
+            context = h.transpose(1, 2)
+            queries = self.pad(context)
+            memory = self.cif_conv1d(queries)
+            output = memory + context
+            output = self.dropout(output)
+            output = output.transpose(1, 2)
+            output = torch.relu(output)
+            output = self.cif_output(output)
+            alphas = torch.sigmoid(output)
+            alphas = torch.nn.functional.relu(alphas * self.smooth_factor - self.noise_threshold)
+            if mask is not None:
+                mask = mask.transpose(-1, -2).float()
+                alphas = alphas * mask
+            if mask_chunk_predictor is not None:
+                alphas = alphas * mask_chunk_predictor
+            alphas = alphas.squeeze(-1)
+            mask = mask.squeeze(-1)
+            if target_label_length is not None:
+                target_length = target_label_length
+            elif target_label is not None:
+                target_length = (target_label != ignore_id).float().sum(-1)
+            else:
+                target_length = None
+            token_num = alphas.sum(-1)
+            if target_length is not None:
+                alphas *= (target_length / token_num)[:, None].repeat(1, alphas.size(1))
+            elif self.tail_threshold > 0.0:
+                hidden, alphas, token_num = self.tail_process_fn(
+                    hidden, alphas, token_num, mask=mask
+                )
+
+            acoustic_embeds, cif_peak = cif(hidden, alphas, self.threshold)
+
+            if target_length is None and self.tail_threshold > 0.0:
+                token_num_int = torch.max(token_num).type(torch.int32).item()
+                acoustic_embeds = acoustic_embeds[:, :token_num_int, :]
+
+        return acoustic_embeds, token_num, alphas, cif_peak
+
+    def tail_process_fn(self, hidden, alphas, token_num=None, mask=None):
+        b, t, d = hidden.size()
+        tail_threshold = self.tail_threshold
+        if mask is not None:
+            zeros_t = torch.zeros((b, 1), dtype=torch.float32, device=alphas.device)
+            ones_t = torch.ones_like(zeros_t)
+            mask_1 = torch.cat([mask, zeros_t], dim=1)
+            mask_2 = torch.cat([ones_t, mask], dim=1)
+            mask = mask_2 - mask_1
+            tail_threshold = mask * tail_threshold
+            alphas = torch.cat([alphas, zeros_t], dim=1)
+            alphas = torch.add(alphas, tail_threshold)
+        else:
+            tail_threshold = torch.tensor([tail_threshold], dtype=alphas.dtype).to(alphas.device)
+            tail_threshold = torch.reshape(tail_threshold, (1, 1))
+            alphas = torch.cat([alphas, tail_threshold], dim=1)
+        zeros = torch.zeros((b, 1, d), dtype=hidden.dtype).to(hidden.device)
+        hidden = torch.cat([hidden, zeros], dim=1)
+        token_num = alphas.sum(dim=-1)
+        token_num_floor = torch.floor(token_num)
+
+        return hidden, alphas, token_num_floor
+
+    def gen_frame_alignments(
+        self, alphas: torch.Tensor = None, encoder_sequence_length: torch.Tensor = None
+    ):
+        batch_size, maximum_length = alphas.size()
+        int_type = torch.int32
+
+        is_training = self.training
+        if is_training:
+            token_num = torch.round(torch.sum(alphas, dim=1)).type(int_type)
+        else:
+            token_num = torch.floor(torch.sum(alphas, dim=1)).type(int_type)
+
+        max_token_num = torch.max(token_num).item()
+
+        alphas_cumsum = torch.cumsum(alphas, dim=1)
+        alphas_cumsum = torch.floor(alphas_cumsum).type(int_type)
+        alphas_cumsum = alphas_cumsum[:, None, :].repeat(1, max_token_num, 1)
+
+        index = torch.ones([batch_size, max_token_num], dtype=int_type)
+        index = torch.cumsum(index, dim=1)
+        index = index[:, :, None].repeat(1, 1, maximum_length).to(alphas_cumsum.device)
+
+        index_div = torch.floor(torch.true_divide(alphas_cumsum, index)).type(int_type)
+        index_div_bool_zeros = index_div.eq(0)
+        index_div_bool_zeros_count = torch.sum(index_div_bool_zeros, dim=-1) + 1
+        index_div_bool_zeros_count = torch.clamp(
+            index_div_bool_zeros_count, 0, encoder_sequence_length.max()
+        )
+        token_num_mask = (~make_pad_mask(token_num, maxlen=max_token_num)).to(token_num.device)
+        index_div_bool_zeros_count *= token_num_mask
+
+        index_div_bool_zeros_count_tile = index_div_bool_zeros_count[:, :, None].repeat(
+            1, 1, maximum_length
+        )
+        ones = torch.ones_like(index_div_bool_zeros_count_tile)
+        zeros = torch.zeros_like(index_div_bool_zeros_count_tile)
+        ones = torch.cumsum(ones, dim=2)
+        cond = index_div_bool_zeros_count_tile == ones
+        index_div_bool_zeros_count_tile = torch.where(cond, zeros, ones)
+
+        index_div_bool_zeros_count_tile_bool = index_div_bool_zeros_count_tile.type(torch.bool)
+        index_div_bool_zeros_count_tile = 1 - index_div_bool_zeros_count_tile_bool.type(int_type)
+        index_div_bool_zeros_count_tile_out = torch.sum(index_div_bool_zeros_count_tile, dim=1)
+        index_div_bool_zeros_count_tile_out = index_div_bool_zeros_count_tile_out.type(int_type)
+        predictor_mask = (
+            (~make_pad_mask(encoder_sequence_length, maxlen=encoder_sequence_length.max()))
+            .type(int_type)
+            .to(encoder_sequence_length.device)
+        )
+        index_div_bool_zeros_count_tile_out = index_div_bool_zeros_count_tile_out * predictor_mask
+
+        predictor_alignments = index_div_bool_zeros_count_tile_out
+        predictor_alignments_length = predictor_alignments.sum(-1).type(
+            encoder_sequence_length.dtype
+        )
+        return predictor_alignments.detach(), predictor_alignments_length.detach()
+
+
+@tables.register("predictor_classes", "CifPredictorV2")
+class CifPredictorV2(torch.nn.Module):
+    def __init__(
+        self,
+        idim,
+        l_order,
+        r_order,
+        threshold=1.0,
+        dropout=0.1,
+        smooth_factor=1.0,
+        noise_threshold=0,
+        tail_threshold=0.0,
+        tf2torch_tensor_name_prefix_torch="predictor",
+        tf2torch_tensor_name_prefix_tf="seq2seq/cif",
+        tail_mask=True,
+    ):
+        super().__init__()
+
+        self.pad = torch.nn.ConstantPad1d((l_order, r_order), 0)
+        self.cif_conv1d = torch.nn.Conv1d(idim, idim, l_order + r_order + 1)
+        self.cif_output = torch.nn.Linear(idim, 1)
+        self.dropout = torch.nn.Dropout(p=dropout)
+        self.threshold = threshold
+        self.smooth_factor = smooth_factor
+        self.noise_threshold = noise_threshold
+        self.tail_threshold = tail_threshold
+        self.tf2torch_tensor_name_prefix_torch = tf2torch_tensor_name_prefix_torch
+        self.tf2torch_tensor_name_prefix_tf = tf2torch_tensor_name_prefix_tf
+        self.tail_mask = tail_mask
+
+    def forward(
+        self,
+        hidden,
+        target_label=None,
+        mask=None,
+        ignore_id=-1,
+        mask_chunk_predictor=None,
+        target_label_length=None,
+    ):
+
+        with autocast(False):
+            h = hidden
+            context = h.transpose(1, 2)
+            queries = self.pad(context)
+            output = torch.relu(self.cif_conv1d(queries))
+            output = output.transpose(1, 2)
+
+            output = self.cif_output(output)
+            alphas = torch.sigmoid(output)
+            alphas = torch.nn.functional.relu(alphas * self.smooth_factor - self.noise_threshold)
+            if mask is not None:
+                mask = mask.transpose(-1, -2).float()
+                alphas = alphas * mask
+            if mask_chunk_predictor is not None:
+                alphas = alphas * mask_chunk_predictor
+            alphas = alphas.squeeze(-1)
+            mask = mask.squeeze(-1)
+            if target_label_length is not None:
+                target_length = target_label_length.squeeze(-1)
+            elif target_label is not None:
+                target_length = (target_label != ignore_id).float().sum(-1)
+            else:
+                target_length = None
+            token_num = alphas.sum(-1)
+            if target_length is not None:
+                alphas *= (target_length / token_num)[:, None].repeat(1, alphas.size(1))
+            elif self.tail_threshold > 0.0:
+                if self.tail_mask:
+                    hidden, alphas, token_num = self.tail_process_fn(
+                        hidden, alphas, token_num, mask=mask
+                    )
+                else:
+                    hidden, alphas, token_num = self.tail_process_fn(
+                        hidden, alphas, token_num, mask=None
+                    )
+
+            acoustic_embeds, cif_peak = cif_v1(hidden, alphas, self.threshold)
+            if target_length is None and self.tail_threshold > 0.0:
+                token_num_int = torch.max(token_num).type(torch.int32).item()
+                acoustic_embeds = acoustic_embeds[:, :token_num_int, :]
+
+        return acoustic_embeds, token_num, alphas, cif_peak
+
+    def forward_chunk(self, hidden, cache=None, **kwargs):
+        is_final = kwargs.get("is_final", False)
+        batch_size, len_time, hidden_size = hidden.shape
+        h = hidden
+        context = h.transpose(1, 2)
+        queries = self.pad(context)
+        output = torch.relu(self.cif_conv1d(queries))
+        output = output.transpose(1, 2)
+        output = self.cif_output(output)
+        alphas = torch.sigmoid(output)
+        alphas = torch.nn.functional.relu(alphas * self.smooth_factor - self.noise_threshold)
+
+        alphas = alphas.squeeze(-1)
+
+        token_length = []
+        list_fires = []
+        list_frames = []
+        cache_alphas = []
+        cache_hiddens = []
+
+        if cache is not None and "chunk_size" in cache:
+            alphas[:, : cache["chunk_size"][0]] = 0.0
+            if not is_final:
+                alphas[:, sum(cache["chunk_size"][:2]) :] = 0.0
+        if cache is not None and "cif_alphas" in cache and "cif_hidden" in cache:
+            cache["cif_hidden"] = to_device(cache["cif_hidden"], device=hidden.device)
+            cache["cif_alphas"] = to_device(cache["cif_alphas"], device=alphas.device)
+            hidden = torch.cat((cache["cif_hidden"], hidden), dim=1)
+            alphas = torch.cat((cache["cif_alphas"], alphas), dim=1)
+        if cache is not None and is_final:
+            tail_hidden = torch.zeros((batch_size, 1, hidden_size), device=hidden.device)
+            tail_alphas = torch.tensor([[self.tail_threshold]], device=alphas.device)
+            tail_alphas = torch.tile(tail_alphas, (batch_size, 1))
+            hidden = torch.cat((hidden, tail_hidden), dim=1)
+            alphas = torch.cat((alphas, tail_alphas), dim=1)
+
+        len_time = alphas.shape[1]
+        for b in range(batch_size):
+            integrate = 0.0
+            frames = torch.zeros((hidden_size), device=hidden.device)
+            list_frame = []
+            list_fire = []
+            for t in range(len_time):
+                alpha = alphas[b][t]
+                if alpha + integrate < self.threshold:
+                    integrate += alpha
+                    list_fire.append(integrate)
+                    frames += alpha * hidden[b][t]
+                else:
+                    frames += (self.threshold - integrate) * hidden[b][t]
+                    list_frame.append(frames)
+                    integrate += alpha
+                    list_fire.append(integrate)
+                    integrate -= self.threshold
+                    frames = integrate * hidden[b][t]
+
+            cache_alphas.append(integrate)
+            if integrate > 0.0:
+                cache_hiddens.append(frames / integrate)
+            else:
+                cache_hiddens.append(frames)
+
+            token_length.append(torch.tensor(len(list_frame), device=alphas.device))
+            list_fires.append(list_fire)
+            list_frames.append(list_frame)
+
+        cache["cif_alphas"] = torch.stack(cache_alphas, axis=0)
+        cache["cif_alphas"] = torch.unsqueeze(cache["cif_alphas"], axis=0)
+        cache["cif_hidden"] = torch.stack(cache_hiddens, axis=0)
+        cache["cif_hidden"] = torch.unsqueeze(cache["cif_hidden"], axis=0)
+
+        max_token_len = max(token_length)
+        if max_token_len == 0:
+            return hidden, torch.stack(token_length, 0), None, None
+        list_ls = []
+        for b in range(batch_size):
+            pad_frames = torch.zeros(
+                (max_token_len - token_length[b], hidden_size), device=alphas.device
+            )
+            if token_length[b] == 0:
+                list_ls.append(pad_frames)
+            else:
+                list_frames[b] = torch.stack(list_frames[b])
+                list_ls.append(torch.cat((list_frames[b], pad_frames), dim=0))
+
+        cache["cif_alphas"] = torch.stack(cache_alphas, axis=0)
+        cache["cif_alphas"] = torch.unsqueeze(cache["cif_alphas"], axis=0)
+        cache["cif_hidden"] = torch.stack(cache_hiddens, axis=0)
+        cache["cif_hidden"] = torch.unsqueeze(cache["cif_hidden"], axis=0)
+        return torch.stack(list_ls, 0), torch.stack(token_length, 0), None, None
+
+    def tail_process_fn(self, hidden, alphas, token_num=None, mask=None):
+        b, t, d = hidden.size()
+        tail_threshold = self.tail_threshold
+        if mask is not None:
+            zeros_t = torch.zeros((b, 1), dtype=torch.float32, device=alphas.device)
+            ones_t = torch.ones_like(zeros_t)
+            mask_1 = torch.cat([mask, zeros_t], dim=1)
+            mask_2 = torch.cat([ones_t, mask], dim=1)
+            mask = mask_2 - mask_1
+            tail_threshold = mask * tail_threshold
+            alphas = torch.cat([alphas, zeros_t], dim=1)
+            alphas = torch.add(alphas, tail_threshold)
+        else:
+            tail_threshold = torch.tensor([tail_threshold], dtype=alphas.dtype).to(alphas.device)
+            tail_threshold = torch.reshape(tail_threshold, (1, 1))
+            if b > 1:
+                alphas = torch.cat([alphas, tail_threshold.repeat(b, 1)], dim=1)
+            else:
+                alphas = torch.cat([alphas, tail_threshold], dim=1)
+        zeros = torch.zeros((b, 1, d), dtype=hidden.dtype).to(hidden.device)
+        hidden = torch.cat([hidden, zeros], dim=1)
+        token_num = alphas.sum(dim=-1)
+        token_num_floor = torch.floor(token_num)
+
+        return hidden, alphas, token_num_floor
+
+    def gen_frame_alignments(
+        self, alphas: torch.Tensor = None, encoder_sequence_length: torch.Tensor = None
+    ):
+        batch_size, maximum_length = alphas.size()
+        int_type = torch.int32
+
+        is_training = self.training
+        if is_training:
+            token_num = torch.round(torch.sum(alphas, dim=1)).type(int_type)
+        else:
+            token_num = torch.floor(torch.sum(alphas, dim=1)).type(int_type)
+
+        max_token_num = torch.max(token_num).item()
+
+        alphas_cumsum = torch.cumsum(alphas, dim=1)
+        alphas_cumsum = torch.floor(alphas_cumsum).type(int_type)
+        alphas_cumsum = alphas_cumsum[:, None, :].repeat(1, max_token_num, 1)
+
+        index = torch.ones([batch_size, max_token_num], dtype=int_type)
+        index = torch.cumsum(index, dim=1)
+        index = index[:, :, None].repeat(1, 1, maximum_length).to(alphas_cumsum.device)
+
+        index_div = torch.floor(torch.true_divide(alphas_cumsum, index)).type(int_type)
+        index_div_bool_zeros = index_div.eq(0)
+        index_div_bool_zeros_count = torch.sum(index_div_bool_zeros, dim=-1) + 1
+        index_div_bool_zeros_count = torch.clamp(
+            index_div_bool_zeros_count, 0, encoder_sequence_length.max()
+        )
+        token_num_mask = (~make_pad_mask(token_num, maxlen=max_token_num)).to(token_num.device)
+        index_div_bool_zeros_count *= token_num_mask
+
+        index_div_bool_zeros_count_tile = index_div_bool_zeros_count[:, :, None].repeat(
+            1, 1, maximum_length
+        )
+        ones = torch.ones_like(index_div_bool_zeros_count_tile)
+        zeros = torch.zeros_like(index_div_bool_zeros_count_tile)
+        ones = torch.cumsum(ones, dim=2)
+        cond = index_div_bool_zeros_count_tile == ones
+        index_div_bool_zeros_count_tile = torch.where(cond, zeros, ones)
+
+        index_div_bool_zeros_count_tile_bool = index_div_bool_zeros_count_tile.type(torch.bool)
+        index_div_bool_zeros_count_tile = 1 - index_div_bool_zeros_count_tile_bool.type(int_type)
+        index_div_bool_zeros_count_tile_out = torch.sum(index_div_bool_zeros_count_tile, dim=1)
+        index_div_bool_zeros_count_tile_out = index_div_bool_zeros_count_tile_out.type(int_type)
+        predictor_mask = (
+            (~make_pad_mask(encoder_sequence_length, maxlen=encoder_sequence_length.max()))
+            .type(int_type)
+            .to(encoder_sequence_length.device)
+        )
+        index_div_bool_zeros_count_tile_out = index_div_bool_zeros_count_tile_out * predictor_mask
+
+        predictor_alignments = index_div_bool_zeros_count_tile_out
+        predictor_alignments_length = predictor_alignments.sum(-1).type(
+            encoder_sequence_length.dtype
+        )
+        return predictor_alignments.detach(), predictor_alignments_length.detach()
+
+
+@tables.register("predictor_classes", "CifPredictorV2Export")
+class CifPredictorV2Export(torch.nn.Module):
+    def __init__(self, model, **kwargs):
+        super().__init__()
+
+        self.pad = model.pad
+        self.cif_conv1d = model.cif_conv1d
+        self.cif_output = model.cif_output
+        self.threshold = model.threshold
+        self.smooth_factor = model.smooth_factor
+        self.noise_threshold = model.noise_threshold
+        self.tail_threshold = model.tail_threshold
+
+    def forward(
+        self,
+        hidden: torch.Tensor,
+        mask: torch.Tensor,
+    ):
+        alphas, token_num = self.forward_cnn(hidden, mask)
+        mask = mask.transpose(-1, -2).float()
+        mask = mask.squeeze(-1)
+        hidden, alphas, token_num = self.tail_process_fn(hidden, alphas, mask=mask)
+        acoustic_embeds, cif_peak = cif_v1_export(hidden, alphas, self.threshold)
+
+        return acoustic_embeds, token_num, alphas, cif_peak
+
+    def forward_cnn(
+        self,
+        hidden: torch.Tensor,
+        mask: torch.Tensor,
+    ):
+        h = hidden
+        context = h.transpose(1, 2)
+        queries = self.pad(context)
+        output = torch.relu(self.cif_conv1d(queries))
+        output = output.transpose(1, 2)
+
+        output = self.cif_output(output)
+        alphas = torch.sigmoid(output)
+        alphas = torch.nn.functional.relu(alphas * self.smooth_factor - self.noise_threshold)
+        mask = mask.transpose(-1, -2).float()
+        alphas = alphas * mask
+        alphas = alphas.squeeze(-1)
+        token_num = alphas.sum(-1)
+
+        return alphas, token_num
+
+    def tail_process_fn(self, hidden, alphas, token_num=None, mask=None):
+        b, t, d = hidden.size()
+        tail_threshold = self.tail_threshold
+
+        zeros_t = torch.zeros((b, 1), dtype=torch.float32, device=alphas.device)
+        ones_t = torch.ones_like(zeros_t)
+
+        mask_1 = torch.cat([mask, zeros_t], dim=1)
+        mask_2 = torch.cat([ones_t, mask], dim=1)
+        mask = mask_2 - mask_1
+        tail_threshold = mask * tail_threshold
+        alphas = torch.cat([alphas, zeros_t], dim=1)
+        alphas = torch.add(alphas, tail_threshold)
+
+        zeros = torch.zeros((b, 1, d), dtype=hidden.dtype).to(hidden.device)
+        hidden = torch.cat([hidden, zeros], dim=1)
+        token_num = alphas.sum(dim=-1)
+        token_num_floor = torch.floor(token_num)
+
+        return hidden, alphas, token_num_floor
+
+
+@torch.jit.script
+def cif_v1_export(hidden, alphas, threshold: float):
+    device = hidden.device
+    dtype = hidden.dtype
+    batch_size, len_time, hidden_size = hidden.size()
+    threshold = torch.tensor([threshold], dtype=alphas.dtype).to(alphas.device)
+
+    frames = torch.zeros(batch_size, len_time, hidden_size, dtype=dtype, device=device)
+    fires = torch.zeros(batch_size, len_time, dtype=dtype, device=device)
+
+    # prefix_sum = torch.cumsum(alphas, dim=1)
+    prefix_sum = torch.cumsum(alphas, dim=1, dtype=torch.float64).to(
+        torch.float32
+    )  # cumsum precision degradation cause wrong result in extreme
+    prefix_sum_floor = torch.floor(prefix_sum)
+    dislocation_prefix_sum = torch.roll(prefix_sum, 1, dims=1)
+    dislocation_prefix_sum_floor = torch.floor(dislocation_prefix_sum)
+
+    dislocation_prefix_sum_floor[:, 0] = 0
+    dislocation_diff = prefix_sum_floor - dislocation_prefix_sum_floor
+
+    fire_idxs = dislocation_diff > 0
+    fires[fire_idxs] = 1
+    fires = fires + prefix_sum - prefix_sum_floor
+
+    # prefix_sum_hidden = torch.cumsum(alphas.unsqueeze(-1).tile((1, 1, hidden_size)) * hidden, dim=1)
+    prefix_sum_hidden = torch.cumsum(alphas.unsqueeze(-1).repeat((1, 1, hidden_size)) * hidden, dim=1)
+    frames = prefix_sum_hidden[fire_idxs]
+    shift_frames = torch.roll(frames, 1, dims=0)
+
+    batch_len = fire_idxs.sum(1)
+    batch_idxs = torch.cumsum(batch_len, dim=0)
+    shift_batch_idxs = torch.roll(batch_idxs, 1, dims=0)
+    shift_batch_idxs[0] = 0
+    shift_frames[shift_batch_idxs] = 0
+
+    remains = fires - torch.floor(fires)
+    # remain_frames = remains[fire_idxs].unsqueeze(-1).tile((1, hidden_size)) * hidden[fire_idxs]
+    remain_frames = remains[fire_idxs].unsqueeze(-1).repeat((1, hidden_size)) * hidden[fire_idxs]
+
+    shift_remain_frames = torch.roll(remain_frames, 1, dims=0)
+    shift_remain_frames[shift_batch_idxs] = 0
+
+    frames = frames - shift_frames + shift_remain_frames - remain_frames
+
+    # max_label_len = batch_len.max()
+    max_label_len = alphas.sum(dim=-1)
+    max_label_len = torch.floor(max_label_len).max().to(dtype=torch.int64)
+
+    # frame_fires = torch.zeros(batch_size, max_label_len, hidden_size, dtype=dtype, device=device)
+    frame_fires = torch.zeros(batch_size, max_label_len, hidden_size, dtype=dtype, device=device)
+    indices = torch.arange(max_label_len, device=device).expand(batch_size, -1)
+    frame_fires_idxs = indices < batch_len.unsqueeze(1)
+    frame_fires[frame_fires_idxs] = frames
+    return frame_fires, fires
+
+
+@torch.jit.script
+def cif_export(hidden, alphas, threshold: float):
+    batch_size, len_time, hidden_size = hidden.size()
+    threshold = torch.tensor([threshold], dtype=alphas.dtype).to(alphas.device)
+
+    # loop varss
+    integrate = torch.zeros([batch_size], dtype=alphas.dtype, device=hidden.device)
+    frame = torch.zeros([batch_size, hidden_size], dtype=hidden.dtype, device=hidden.device)
+    # intermediate vars along time
+    list_fires = []
+    list_frames = []
+
+    for t in range(len_time):
+        alpha = alphas[:, t]
+        distribution_completion = (
+            torch.ones([batch_size], dtype=alphas.dtype, device=hidden.device) - integrate
+        )
+
+        integrate += alpha
+        list_fires.append(integrate)
+
+        fire_place = integrate >= threshold
+        integrate = torch.where(
+            fire_place,
+            integrate - torch.ones([batch_size], dtype=alphas.dtype, device=hidden.device),
+            integrate,
+        )
+        cur = torch.where(fire_place, distribution_completion, alpha)
+        remainds = alpha - cur
+
+        frame += cur[:, None] * hidden[:, t, :]
+        list_frames.append(frame)
+        frame = torch.where(
+            fire_place[:, None].repeat(1, hidden_size), remainds[:, None] * hidden[:, t, :], frame
+        )
+
+    fires = torch.stack(list_fires, 1)
+    frames = torch.stack(list_frames, 1)
+
+    fire_idxs = fires >= threshold
+    frame_fires = torch.zeros_like(hidden)
+    max_label_len = frames[0, fire_idxs[0]].size(0)
+    for b in range(batch_size):
+        frame_fire = frames[b, fire_idxs[b]]
+        frame_len = frame_fire.size(0)
+        frame_fires[b, :frame_len, :] = frame_fire
+
+        if frame_len >= max_label_len:
+            max_label_len = frame_len
+    frame_fires = frame_fires[:, :max_label_len, :]
+    return frame_fires, fires
+
+
+class mae_loss(torch.nn.Module):
+
+    def __init__(self, normalize_length=False):
+        super(mae_loss, self).__init__()
+        self.normalize_length = normalize_length
+        self.criterion = torch.nn.L1Loss(reduction="sum")
+
+    def forward(self, token_length, pre_token_length):
+        loss_token_normalizer = token_length.size(0)
+        if self.normalize_length:
+            loss_token_normalizer = token_length.sum().type(torch.float32)
+        loss = self.criterion(token_length, pre_token_length)
+        loss = loss / loss_token_normalizer
+        return loss
+
+
+def cif(hidden, alphas, threshold):
+    batch_size, len_time, hidden_size = hidden.size()
+
+    # loop varss
+    integrate = torch.zeros([batch_size], device=hidden.device)
+    frame = torch.zeros([batch_size, hidden_size], device=hidden.device)
+    # intermediate vars along time
+    list_fires = []
+    list_frames = []
+
+    for t in range(len_time):
+        alpha = alphas[:, t]
+        distribution_completion = torch.ones([batch_size], device=hidden.device) - integrate
+
+        integrate += alpha
+        list_fires.append(integrate)
+
+        fire_place = integrate >= threshold
+        integrate = torch.where(
+            fire_place, integrate - torch.ones([batch_size], device=hidden.device), integrate
+        )
+        cur = torch.where(fire_place, distribution_completion, alpha)
+        remainds = alpha - cur
+
+        frame += cur[:, None] * hidden[:, t, :]
+        list_frames.append(frame)
+        frame = torch.where(
+            fire_place[:, None].repeat(1, hidden_size), remainds[:, None] * hidden[:, t, :], frame
+        )
+
+    fires = torch.stack(list_fires, 1)
+    frames = torch.stack(list_frames, 1)
+    list_ls = []
+    len_labels = torch.round(alphas.sum(-1)).int()
+    max_label_len = len_labels.max()
+    for b in range(batch_size):
+        fire = fires[b, :]
+        l = torch.index_select(frames[b, :, :], 0, torch.nonzero(fire >= threshold).squeeze())
+        pad_l = torch.zeros([max_label_len - l.size(0), hidden_size], device=hidden.device)
+        list_ls.append(torch.cat([l, pad_l], 0))
+    return torch.stack(list_ls, 0), fires
+
+
+def cif_wo_hidden_v1(alphas, threshold, return_fire_idxs=False):
+    batch_size, len_time = alphas.size()
+    device = alphas.device
+    dtype = alphas.dtype
+
+    threshold = torch.tensor([threshold], dtype=alphas.dtype).to(alphas.device)
+
+    fires = torch.zeros(batch_size, len_time, dtype=dtype, device=device)
+
+    if torch.cuda.get_device_name() == "Iluvatar BI-V150":
+        prefix_sum = torch.cumsum(alphas, dim=1)
+    else:
+        prefix_sum = torch.cumsum(alphas, dim=1, dtype=torch.float64).to(
+            torch.float32
+        )  # cumsum precision degradation cause wrong result in extreme
+    prefix_sum_floor = torch.floor(prefix_sum)
+    dislocation_prefix_sum = torch.roll(prefix_sum, 1, dims=1)
+    dislocation_prefix_sum_floor = torch.floor(dislocation_prefix_sum)
+
+    dislocation_prefix_sum_floor[:, 0] = 0
+    dislocation_diff = prefix_sum_floor - dislocation_prefix_sum_floor
+
+    fire_idxs = dislocation_diff > 0
+    fires[fire_idxs] = 1
+    fires = fires + prefix_sum - prefix_sum_floor
+    if return_fire_idxs:
+        return fires, fire_idxs
+    return fires
+
+
+def cif_v1(hidden, alphas, threshold):
+    fires, fire_idxs = cif_wo_hidden_v1(alphas, threshold, return_fire_idxs=True)
+
+    device = hidden.device
+    dtype = hidden.dtype
+    batch_size, len_time, hidden_size = hidden.size()
+    # frames = torch.zeros(batch_size, len_time, hidden_size, dtype=dtype, device=device)
+    # prefix_sum_hidden = torch.cumsum(alphas.unsqueeze(-1).tile((1, 1, hidden_size)) * hidden, dim=1)
+    frames = torch.zeros(batch_size, len_time, hidden_size, dtype=dtype, device=device)
+    prefix_sum_hidden = torch.cumsum(alphas.unsqueeze(-1).repeat((1, 1, hidden_size)) * hidden, dim=1)
+
+    frames = prefix_sum_hidden[fire_idxs]
+    shift_frames = torch.roll(frames, 1, dims=0)
+
+    batch_len = fire_idxs.sum(1)
+    batch_idxs = torch.cumsum(batch_len, dim=0)
+    shift_batch_idxs = torch.roll(batch_idxs, 1, dims=0)
+    shift_batch_idxs[0] = 0
+    shift_frames[shift_batch_idxs] = 0
+
+    remains = fires - torch.floor(fires)
+    # remain_frames = remains[fire_idxs].unsqueeze(-1).tile((1, hidden_size)) * hidden[fire_idxs]
+    remain_frames = remains[fire_idxs].unsqueeze(-1).repeat((1, hidden_size)) * hidden[fire_idxs]
+
+    shift_remain_frames = torch.roll(remain_frames, 1, dims=0)
+    shift_remain_frames[shift_batch_idxs] = 0
+
+    frames = frames - shift_frames + shift_remain_frames - remain_frames
+
+    # max_label_len = batch_len.max()
+    max_label_len = (
+        torch.round(alphas.sum(-1)).int().max()
+    )  # torch.round to calculate the max length
+
+    # frame_fires = torch.zeros(batch_size, max_label_len, hidden_size, dtype=dtype, device=device)
+    frame_fires = torch.zeros(batch_size, max_label_len, hidden_size, dtype=dtype, device=device)
+    indices = torch.arange(max_label_len, device=device).expand(batch_size, -1)
+    frame_fires_idxs = indices < batch_len.unsqueeze(1)
+    frame_fires[frame_fires_idxs] = frames
+    return frame_fires, fires
+
+
+def cif_wo_hidden(alphas, threshold):
+    batch_size, len_time = alphas.size()
+
+    # loop varss
+    integrate = torch.zeros([batch_size], device=alphas.device)
+    # intermediate vars along time
+    list_fires = []
+
+    for t in range(len_time):
+        alpha = alphas[:, t]
+
+        integrate += alpha
+        list_fires.append(integrate)
+
+        fire_place = integrate >= threshold
+        integrate = torch.where(
+            fire_place,
+            integrate - torch.ones([batch_size], device=alphas.device) * threshold,
+            integrate,
+        )
+
+    fires = torch.stack(list_fires, 1)
+    return fires
diff --git a/funasr/replaced_files/funasr_nano_model.py b/funasr/replaced_files/funasr_nano_model.py
new file mode 100644
index 0000000..73da9f1
--- /dev/null
+++ b/funasr/replaced_files/funasr_nano_model.py
@@ -0,0 +1,746 @@
+import logging
+import os
+import random
+import re
+import string
+import time
+import traceback
+from typing import Union
+
+import torch
+import torch.nn as nn
+
+from funasr.metrics.compute_acc import compute_accuracy
+from funasr.register import tables
+from funasr.train_utils.device_funcs import force_gatherable, to_device
+from funasr.utils.datadir_writer import DatadirWriter
+from funasr.utils.load_utils import extract_fbank, load_audio_text_image_video
+from transformers import AutoConfig, AutoModelForCausalLM
+
+from funasr.models.fun_asr_nano.ctc import CTC
+from funasr.models.fun_asr_nano.tools.utils import forced_align
+
+dtype_map = {"bf16": torch.bfloat16, "fp16": torch.float16, "fp32": torch.float32}
+
+
+@tables.register("model_classes", "FunASRNano")
+class FunASRNano(nn.Module):
+    def __init__(
+        self,
+        audio_encoder: str = None,
+        audio_encoder_conf: dict = None,
+        audio_adaptor: str = None,
+        audio_adaptor_conf: dict = None,
+        llm: str = None,
+        llm_conf: dict = None,
+        input_size: int = 80,
+        length_normalized_loss: bool = False,
+        **kwargs,
+    ):
+        super().__init__()
+
+        # audio encoder
+        hub = audio_encoder_conf.get("hub", None)
+        self.audio_encoder_activation_checkpoint = audio_encoder_conf.get(
+            "activation_checkpoint", False
+        )
+        if hub == "ms":
+            from funasr import AutoModel
+
+            model = AutoModel(model=audio_encoder, model_revision="master")
+            audio_encoder_output_size = (
+                model.model.encoder_output_size
+                if hasattr(model.model, "encoder_output_size")
+                else -1
+            )
+            audio_encoder = (
+                model.model.model.encoder if hasattr(model.model, "model") else model.model.encoder
+            )
+        else:
+            encoder_class = tables.encoder_classes.get(audio_encoder)
+            audio_encoder = encoder_class(input_size=input_size, **audio_encoder_conf)
+            audio_encoder_output_size = audio_encoder.output_size()
+        freeze = audio_encoder_conf.get("freeze", True)
+
+        if freeze:
+            for _, param in audio_encoder.named_parameters():
+                param.requires_grad = False
+            audio_encoder.eval()
+        self.audio_encoder = audio_encoder
+
+        # llm
+        self.llm = None
+        init_param_path = llm_conf.get("init_param_path", None)
+        llm_dim = None
+
+        llm_load_kwargs = llm_conf.get("load_kwargs", {})
+        config = AutoConfig.from_pretrained(init_param_path)
+        model = AutoModelForCausalLM.from_config(config, **llm_load_kwargs)
+
+        freeze = llm_conf.get("freeze", True)
+        if freeze:
+            for _, param in model.named_parameters():
+                param.requires_grad = False
+            model.eval()
+        if llm_conf.get("activation_checkpoint", False):
+            model.gradient_checkpointing_enable()
+
+        self.llm_dtype = llm_conf.get("llm_dtype", "fp32")
+        self.llm = model.to(dtype_map[self.llm_dtype])
+        llm_dim = model.get_input_embeddings().weight.shape[-1]
+
+        # adaptor
+        adaptor_class = tables.adaptor_classes.get(audio_adaptor)
+        if audio_encoder_output_size > 0:
+            audio_adaptor_conf["encoder_dim"] = audio_encoder_output_size
+        audio_adaptor_conf["llm_dim"] = (
+            llm_dim if llm_dim is not None else audio_adaptor_conf["llm_dim"]
+        )
+        audio_adaptor = adaptor_class(**audio_adaptor_conf)
+        freeze = audio_adaptor_conf.get("freeze", False)
+        if freeze:
+            for _, param in audio_adaptor.named_parameters():
+                param.requires_grad = False
+            audio_adaptor.eval()
+        self.audio_adaptor = audio_adaptor
+        self.use_low_frame_rate = audio_adaptor_conf.get("use_low_frame_rate", False)
+
+        # ctc decoder
+        self.ctc_decoder = None
+        # TODO: fix table name
+        ctc_decoder_class = tables.adaptor_classes.get(kwargs.get("ctc_decoder", None))
+        if ctc_decoder_class is not None:
+            ctc_tokenizer = (
+                kwargs.get("ctc_tokenizer", None)
+                if "ctc_tokenizer" in kwargs
+                else kwargs["dataset_conf"]["ctc_tokenizer"]
+            )
+            ctc_tokenizer_conf = (
+                kwargs.get("ctc_tokenizer_conf", None)
+                if "ctc_tokenizer_conf" in kwargs
+                else kwargs["dataset_conf"]["ctc_tokenizer_conf"]
+            )
+            if ctc_tokenizer is not None and ctc_tokenizer_conf is not None:
+                ctc_tokenizer_class = tables.tokenizer_classes.get(ctc_tokenizer)
+                ctc_tokenizer = ctc_tokenizer_class(**ctc_tokenizer_conf)
+                self.ctc_tokenizer = ctc_tokenizer
+            assert ctc_tokenizer is not None, f"ctc_tokenizer must be set"
+
+            ctc_vocab_size = kwargs.get("ctc_vocab_size", 60515)
+            ctc_decoder_conf = kwargs.get("ctc_decoder_conf", {})
+            if audio_encoder_output_size > 0:
+                ctc_decoder_conf["encoder_dim"] = audio_encoder_output_size
+            self.ctc_decoder = ctc_decoder_class(**ctc_decoder_conf)
+            init_param_path = ctc_decoder_conf.get("init_param_path", None)
+            if init_param_path is not None:
+                src_state = torch.load(init_param_path, map_location="cpu")
+                flag = self.ctc_decoder.load_state_dict(src_state, strict=False)
+                logging.info(f"Loading ctc_decoder ckpt: {init_param_path}, status: {flag}")
+            freeze = ctc_decoder_conf.get("freeze", False)
+            if freeze:
+                for _, param in self.ctc_decoder.named_parameters():
+                    param.requires_grad = False
+                self.ctc_decoder.eval()
+
+            ctc_conf = kwargs.get("ctc_conf", {})
+            self.blank_id = ctc_conf.get("blank_id", ctc_vocab_size - 1)
+            self.ctc_weight = kwargs.get("ctc_weight", 0.3)
+            self.ctc = CTC(
+                odim=ctc_vocab_size,
+                encoder_output_size=audio_encoder_output_size,
+                blank_id=self.blank_id,
+                **ctc_conf,
+            )
+            self.detach_ctc_decoder = kwargs.get("detach_ctc_decoder", True)
+            self.error_calculator = None
+
+        self.length_normalized_loss = length_normalized_loss
+        rank = int(os.environ.get("RANK", 0))
+        logging.info(f"rank: {rank}, model is builded.")
+
+    def forward(
+        self,
+        speech: torch.Tensor = None,
+        speech_lengths: torch.Tensor = None,
+        input_ids: torch.Tensor = None,
+        attention_mask: torch.Tensor = None,
+        labels_ids: torch.Tensor = None,
+        fbank_beg: torch.Tensor = None,
+        fbank_mask: torch.Tensor = None,
+        **kwargs,
+    ):
+        batch_size, token_num = input_ids.shape
+        stats = {}
+        input_ids[input_ids < 0] = 0
+        inputs_embeds = self.llm.model.get_input_embeddings()(input_ids)
+        if speech is not None:
+            if len(speech_lengths.size()) > 1:
+                speech_lengths = speech_lengths[:, 0]
+            batch_size_speech, frames, _ = speech.shape
+
+            # audio encoder
+            if self.audio_encoder_activation_checkpoint:
+                from torch.utils.checkpoint import checkpoint
+
+                encoder_out, encoder_out_lens = checkpoint(
+                    self.encode, speech, speech_lengths, use_reentrant=False
+                )
+            else:
+                encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
+
+            # audio_adaptor
+            encoder_out, encoder_out_lens = self.audio_adaptor(encoder_out, encoder_out_lens)
+
+            batch_size, token_num, dims = inputs_embeds.shape
+            fake_token_len = kwargs.get("fake_token_len")
+            fake_token_len[fake_token_len < 0] = 0
+            fbank_beg[fbank_beg < 0] = 0
+
+            speech_idx = 0
+            for batch_idx in range(batch_size):
+                for turn_id in range(fbank_beg.shape[1]):
+                    fbank_beg_idx = fbank_beg[batch_idx, turn_id].item()
+                    if fbank_beg_idx > 0:
+                        speech_token_len = fake_token_len[batch_idx, turn_id]
+                        speech_token = encoder_out[speech_idx, :speech_token_len, :]
+
+                        try:
+                            inputs_embeds[
+                                batch_idx,
+                                fbank_beg_idx : fbank_beg_idx + speech_token_len,
+                                :,
+                            ] = speech_token
+                        except Exception as e:
+                            logging.error(f"{str(e)}, {traceback.format_exc()}")
+                            logging.info(
+                                f"batch_idx: {batch_idx}, inputs_embeds: {inputs_embeds.shape}, fbank_beg_idx: {fbank_beg_idx}, speech_token_len: {speech_token_len}, encoder_out: {encoder_out.shape}, encoder_out_lens: {encoder_out_lens}, fake_token_len: {fake_token_len}, speech_lengths: {speech_lengths}"
+                            )
+                            speech_token_len = encoder_out_lens[speech_idx].item()
+                            speech_token = encoder_out[speech_idx, :speech_token_len, :]
+                            inputs_embeds[
+                                batch_idx,
+                                fbank_beg_idx : fbank_beg_idx + speech_token_len,
+                                :,
+                            ] = speech_token
+
+                        speech_idx += 1
+
+            stats["batch_size_speech"] = batch_size_speech
+            stats["batch_size_x_frames"] = frames * batch_size_speech
+            stats["batch_size_real_frames"] = speech_lengths.sum().item()
+            stats["padding_frames"] = stats["batch_size_x_frames"] - stats["batch_size_real_frames"]
+
+        device_type = next(self.parameters()).device.type
+        with torch.autocast(
+            device_type=device_type if device_type in ["cuda", "xpu", "mps"] else "cpu",
+            enabled=True if self.llm_dtype != "fp32" else False,
+            dtype=dtype_map[self.llm_dtype],
+        ):
+            labels_ids[labels_ids == -1] = -100
+            attention_mask[attention_mask < 0] = 0
+            model_outputs = self.llm(
+                inputs_embeds=inputs_embeds.to(dtype_map[self.llm_dtype]),
+                attention_mask=attention_mask,
+                labels=labels_ids,
+            )
+            loss = model_outputs.loss
+
+        with torch.no_grad():
+            preds = torch.argmax(model_outputs.logits, -1)
+            acc_att = compute_accuracy(preds[:, :-1], labels_ids[:, 1:], ignore_label=-100)
+            stats["acc"] = acc_att
+
+        stats["loss"] = torch.clone(loss.detach())
+        stats["batch_size"] = batch_size
+
+        stats["batch_size_x_tokens"] = token_num * batch_size
+        stats["batch_size_real_tokens"] = attention_mask.sum().item()
+        stats["padding_tokens"] = stats["batch_size_x_tokens"] - stats["batch_size_real_tokens"]
+
+        dialog_turns = (fbank_beg > 0).sum(-1)
+        dialog_turns_max = torch.max(dialog_turns).int().item()
+        dialog_turns_avg = dialog_turns.sum().item() / batch_size
+        stats["dialog_turns_max"] = dialog_turns_max
+        stats["dialog_turns_avg"] = dialog_turns_avg
+
+        # force_gatherable: to-device and to-tensor if scalar for DataParallel
+        if self.length_normalized_loss:
+            batch_size = int((labels_ids > 0 + 1).sum())
+        loss, stats, weight = force_gatherable((loss, stats, batch_size), loss.device)
+        return loss, stats, weight
+
+    def forward_export(self, speech, speech_lengths, **kwargs):
+        x, olens = self.audio_encoder(speech, speech_lengths)
+        encoder_out, encoder_out_lens = self.audio_adaptor(x, olens)
+        return encoder_out, encoder_out_lens
+
+    def encode(self, speech, speech_lengths):
+        # audio encoder
+        encoder_out, encoder_out_lens = self.audio_encoder(speech, speech_lengths)
+
+        return encoder_out, encoder_out_lens
+
+    def data_template(self, data):
+        system, user, assistant = [], [], []
+        for i, item in enumerate(data):
+            role = item["role"]
+            content = item["content"]
+            if role == "system":
+                system.append(content)
+            elif role == "user":
+                if "audio" in item:
+                    audio = item["audio"]
+                    content = [content, audio]
+                user.append(content)
+            elif role == "assistant":
+                assistant.append(content)
+
+        system = system * len(user)
+
+        contents = {
+            "system": system,
+            "user": user,
+            "assistant": assistant,
+        }
+
+        return contents
+
+    def data_load_speech(self, contents: dict, tokenizer, frontend, meta_data={}, **kwargs):
+        system = contents["system"]
+        user = contents["user"]
+        assistant = contents["assistant"]
+        pattern = re.compile(r"(<\|startofspeech\|>.*?<\|endofspeech\|>)")
+        do_think = True
+        sys_prompt = True
+        if "dataset_conf" in kwargs:
+            do_think = kwargs["dataset_conf"].get("do_think", True)
+            sys_prompt = kwargs["dataset_conf"].get("sys_prompt", True)
+
+        input_ids, labels, fbank, fbank_lens, fbank_mask, fbank_beg, fake_token_len = (
+            [],
+            [],
+            [],
+            [],
+            [],
+            [],
+            [],
+        )
+        input_source_ids = []
+        for i, (system_prompt, user_prompt, target_out) in enumerate(zip(system, user, assistant)):
+            if i >= kwargs.get("multiturn_num_max", 5):
+                break
+            if len(input_ids) > kwargs.get("max_token_length", 1500):
+                break
+            if isinstance(user_prompt, (list, tuple)):
+                user_prompt, audio = user_prompt
+            if i == 0:
+                if kwargs.get("infer_with_assistant_input", False):
+                    source_input = f"<|im_start|>system\n{system_prompt}<|im_end|>\n<|im_start|>user\n{user_prompt}"
+                    if not sys_prompt:
+                        source_input = f"<|im_start|>user\n{user_prompt}"
+                else:
+                    source_input = f"<|im_start|>system\n{system_prompt}<|im_end|>\n<|im_start|>user\n{user_prompt}<|im_end|>\n<|im_start|>assistant\n"
+                    if not sys_prompt:
+                        source_input = (
+                            f"<|im_start|>user\n{user_prompt}<|im_end|>\n<|im_start|>assistant\n"
+                        )
+            else:
+                if kwargs.get("infer_with_assistant_input", False):
+                    source_input = f"<|im_start|>user\n{user_prompt}"
+                else:
+                    source_input = (
+                        f"<|im_start|>user\n{user_prompt}<|im_end|>\n<|im_start|>assistant\n"
+                    )
+            if not do_think:
+                source_input += "<think>\n\n</think>\n\n"
+            if kwargs.get("prev_text", None) is not None:
+                source_input += kwargs["prev_text"]
+
+            splits = pattern.split(source_input)
+            source_ids = []
+            fbank_mask_i = []
+            fake_token_len_i = 0
+            fbank_beg_i = -1
+            speech, speech_lengths = [], []
+            for k, sub_str in enumerate(splits):
+                if not sub_str.startswith("<|startofspeech|>"):
+                    sub_token = tokenizer.encode(sub_str)
+                    source_ids += sub_token
+                    fbank_mask_i += [0] * len(sub_token)
+                else:
+                    sub_str = sub_str.replace("<|startofspeech|>", "").replace(
+                        "<|endofspeech|>", ""
+                    )
+                    if sub_str.startswith("!"):
+                        sub_str = sub_str[1:]
+                        if sub_str.startswith("!"):  # !!: audio sample point
+                            sub_str = audio
+                        try:
+                            time1 = time.perf_counter()
+                            data_src = load_audio_text_image_video(
+                                sub_str, fs=frontend.fs, **kwargs
+                            )
+                            time2 = time.perf_counter()
+                            meta_data["load_data"] = f"{time2 - time1:0.3f}"
+                        except Exception as e:
+                            logging.error(f"Loading wav failed! {str(e)}, {traceback.format_exc()}")
+
+                        speech, speech_lengths = extract_fbank(
+                            data_src,
+                            data_type=kwargs.get("data_type", "sound"),
+                            frontend=frontend,
+                            is_final=True,
+                        )  # speech: [b, T, d]
+
+                        time3 = time.perf_counter()
+                        meta_data["extract_feat"] = f"{time3 - time2:0.3f}"
+                        meta_data["batch_data_time"] = (
+                            speech_lengths.sum().item()
+                            * frontend.frame_shift
+                            * frontend.lfr_n
+                            / 1000
+                        )
+
+                        if self.use_low_frame_rate:
+                            olens = 1 + (speech_lengths[0].item() - 3 + 2 * 1) // 2
+                            olens = 1 + (olens - 3 + 2 * 1) // 2
+                            fake_token_len_i = (olens - 1) // 2 + 1
+                        else:
+                            fake_token_len_i = speech_lengths[0].item()
+                        fake_token = [0] * fake_token_len_i
+                        fbank_beg_i = len(source_ids)
+                        source_ids += fake_token
+                        fbank_mask_i += [1] * len(fake_token)
+
+            fbank_beg += [fbank_beg_i + len(input_ids)]
+            fake_token_len += [fake_token_len_i]
+            source_mask = [-100] * len(source_ids)
+            target_out = f"{target_out}<|im_end|>"
+            target_ids = tokenizer.encode(target_out)
+            input_source_ids = input_ids + source_ids
+            input_ids += source_ids + target_ids
+            labels += source_mask + target_ids
+            fbank_mask += fbank_mask_i
+            if len(speech) > 0:
+                fbank.append(speech[0, :, :])
+                fbank_lens.append(speech_lengths)
+
+        input_ids = torch.tensor(input_ids, dtype=torch.int64)  # [: self.max_token_length]
+        attention_mask = torch.tensor([1] * len(input_ids), dtype=torch.int32)
+        labels = torch.tensor(labels, dtype=torch.int64)  # [: self.max_token_length]
+
+        fbank_mask = torch.tensor(fbank_mask, dtype=torch.float32)
+        fbank_beg = torch.tensor(fbank_beg, dtype=torch.int32)
+        fake_token_len = torch.tensor(fake_token_len, dtype=torch.int32)
+        source_ids = torch.tensor(input_source_ids, dtype=torch.int64)
+        target_ids = torch.tensor(target_ids, dtype=torch.int64)
+
+        if len(fbank) > 0:
+            speech = torch.nn.utils.rnn.pad_sequence(fbank, batch_first=True, padding_value=0.0)
+            speech_lengths = torch.nn.utils.rnn.pad_sequence(
+                fbank_lens, batch_first=True, padding_value=-1
+            )
+        else:
+            speech = []
+            speech_lengths = []
+        output = {
+            "speech": speech,
+            "speech_lengths": speech_lengths,
+            "fbank_mask": fbank_mask[None, :],
+            "fbank_beg": fbank_beg[None,],
+            "fake_token_len": fake_token_len[None, :],
+            "input_ids": input_ids[None,],
+            "attention_mask": attention_mask[None,],
+            "labels_ids": labels,
+            "source_ids": source_ids[None, :],
+            "target_ids": target_ids[None, :],
+        }
+
+        return output
+
+    def inference_prepare(
+        self,
+        data_in,
+        data_lengths=None,
+        key: list = None,
+        tokenizer=None,
+        frontend=None,
+        **kwargs,
+    ):
+        meta_data = {}
+
+        if kwargs.get("batch_size", 1) > 1:
+            raise NotImplementedError("batch decoding is not implemented")
+
+        contents = self.data_template(data_in[0])
+        output = self.data_load_speech(contents, tokenizer, frontend, meta_data=meta_data, **kwargs)
+        batch = to_device(output, kwargs["device"])
+
+        # audio encoder
+        speech = batch["speech"]
+
+        if len(speech) > 0:
+            if "audio_embedding" in kwargs and "audio_embedding_lens" in kwargs:
+                encoder_out = kwargs["audio_embedding"]
+                encoder_out_lens = kwargs["audio_embedding_lens"]
+            else:
+                speech_lengths = batch["speech_lengths"][:, 0]
+                # fp16
+                if kwargs.get("fp16", False):
+                    speech = speech.to(torch.float16)
+                elif kwargs.get("bf16", False):
+                    speech = speech.to(torch.bfloat16)
+                # audio encoder
+                encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
+
+                # audio_adaptor
+                adaptor_out, adaptor_out_lens = self.audio_adaptor(encoder_out, encoder_out_lens)
+                meta_data["encoder_out"] = encoder_out
+                meta_data["encoder_out_lens"] = encoder_out_lens
+                meta_data["audio_adaptor_out"] = adaptor_out
+                meta_data["audio_adaptor_out_lens"] = adaptor_out_lens
+
+        input_ids = batch["input_ids"]
+        source_ids = batch["source_ids"]
+        fbank_beg = batch["fbank_beg"]
+        fake_token_len = batch["fake_token_len"]
+
+        if not kwargs.get("teacherforcing", False):
+            input_ids = source_ids
+
+        input_ids[input_ids < 0] = 0
+        inputs_embeds = self.llm.model.get_input_embeddings()(input_ids)
+
+        batch_size, token_num, dims = inputs_embeds.shape
+
+        fake_token_len[fake_token_len < 0] = 0
+        fbank_beg[fbank_beg < 0] = 0
+
+        speech_idx = 0
+        for batch_idx in range(batch_size):
+            for turn_id in range(fbank_beg.shape[1]):
+                fbank_beg_idx = fbank_beg[batch_idx, turn_id].item()
+                if fbank_beg_idx > 0:
+                    speech_token_len = fake_token_len[batch_idx, turn_id]
+                    speech_token = adaptor_out[speech_idx, :speech_token_len, :]
+
+                    try:
+                        inputs_embeds[
+                            batch_idx,
+                            fbank_beg_idx : fbank_beg_idx + speech_token_len,
+                            :,
+                        ] = speech_token
+                    except Exception as e:
+                        #
+                        logging.error(f"{str(e)}, {traceback.format_exc()}")
+                        logging.info(
+                            f"batch_idx: {batch_idx}, inputs_embeds: {inputs_embeds.shape}, fbank_beg_idx: {fbank_beg_idx}, speech_token_len: {speech_token_len}, adaptor_out: {adaptor_out.shape}, adaptor_out_lens: {adaptor_out_lens}, fake_token_len: {fake_token_len}, speech_lengths: {speech_lengths}"
+                        )
+                        speech_token_len = adaptor_out_lens[speech_idx].item()
+                        speech_token = adaptor_out[speech_idx, :speech_token_len, :]
+                        inputs_embeds[
+                            batch_idx,
+                            fbank_beg_idx : fbank_beg_idx + speech_token_len,
+                            :,
+                        ] = speech_token
+
+                    speech_idx += 1
+        return inputs_embeds, contents, batch, source_ids, meta_data
+
+    def get_prompt(self, hotwords: list[str], language: str = None, itn: bool = True):
+        if len(hotwords) > 0:
+            hotwords = ", ".join(hotwords)
+            prompt = f"请结合上下文信息，更加准确地完成语音转写任务。如果没有相关信息，我们会留空。\n\n\n**上下文信息：**\n\n\n"
+            prompt += f"热词列表：[{hotwords}]\n"
+        else:
+            prompt = ""
+        if language is None:
+            prompt += "语音转写"
+        else:
+            prompt += f"语音转写成{language}"
+        if not itn:
+            prompt += "，不进行文本规整"
+        return prompt + "："
+
+    def generate_chatml(self, prompt: str, data: Union[str, torch.Tensor]):
+        if isinstance(data, str):
+            return [
+                {"role": "system", "content": "You are a helpful assistant."},
+                {"role": "user", "content": f"{prompt}<|startofspeech|>!{data}<|endofspeech|>"},
+                {"role": "assistant", "content": "null"},
+            ]
+        elif isinstance(data, torch.Tensor):
+            return [
+                {"role": "system", "content": "You are a helpful assistant."},
+                {
+                    "role": "user",
+                    "content": f"{prompt}<|startofspeech|>!!<|endofspeech|>",
+                    "audio": data,
+                },
+                {"role": "assistant", "content": "null"},
+            ]
+
+    def inference(
+        self,
+        data_in,
+        data_lengths=None,
+        key: list = None,
+        tokenizer=None,
+        frontend=None,
+        **kwargs,
+    ):
+        prompt = self.get_prompt(
+            kwargs.get("hotwords", []), kwargs.get("language", None), kwargs.get("itn", True)
+        )
+        data_in = [self.generate_chatml(prompt, data) for data in data_in]
+
+        if key is None:
+            key = []
+            for _ in data_in:
+                chars = string.ascii_letters + string.digits
+                key.append("rand_key_" + "".join(random.choice(chars) for _ in range(13)))
+
+        return self.inference_llm(
+            data_in,
+            data_lengths=data_lengths,
+            key=key,
+            tokenizer=tokenizer,
+            frontend=frontend,
+            **kwargs,
+        )
+
+    def inference_llm(
+        self,
+        data_in,
+        data_lengths=None,
+        key: list = None,
+        tokenizer=None,
+        frontend=None,
+        **kwargs,
+    ):
+        inputs_embeds, contents, batch, source_ids, meta_data = self.inference_prepare(
+            data_in, data_lengths, key, tokenizer, frontend, **kwargs
+        )
+
+        ctc_results = []
+        if self.ctc_decoder is not None:
+            encoder_out = meta_data["encoder_out"]
+            encoder_out_lens = meta_data["encoder_out_lens"]
+            decoder_out, decoder_out_lens = self.ctc_decoder(encoder_out, encoder_out_lens)
+            ctc_logits = self.ctc.log_softmax(decoder_out)
+
+            b, n, d = encoder_out.size()
+            if isinstance(key[0], (list, tuple)):
+                key = key[0]
+            if len(key) < b:
+                key = key * b
+            for i in range(b):
+                x = ctc_logits[i, : encoder_out_lens[i].item(), :]
+                yseq = x.argmax(dim=-1)
+                yseq = torch.unique_consecutive(yseq, dim=-1)
+                mask = yseq != self.blank_id
+                token_int = yseq[mask].tolist()
+                # Change integer-ids to tokens
+                text = self.ctc_tokenizer.decode(token_int)
+                ctc_results.append({"key": key[i], "text": text, "ctc_logits": x})
+
+        llm_dtype = kwargs.get("llm_dtype", "fp32")
+        if llm_dtype == "fp32":
+            llm_dtype = "fp16" if kwargs.get("fp16", False) else llm_dtype
+            llm_dtype = "bf16" if kwargs.get("bf16", False) else llm_dtype
+
+        device_type = torch.device(kwargs.get("device", "cuda")).type
+        with torch.autocast(
+            device_type=device_type if device_type in ["cuda", "xpu", "mps"] else "cpu",
+            enabled=True if llm_dtype != "fp32" else False,
+            dtype=dtype_map[llm_dtype],
+        ):
+            label = contents["assistant"][-1]
+            self.llm = self.llm.to(dtype_map[llm_dtype])
+            inputs_embeds = inputs_embeds.to(dtype_map[llm_dtype])
+            llm_kwargs = kwargs.get("llm_kwargs", {})
+            if not kwargs.get("teacherforcing", False):
+                attention_mask = batch.get("attention_mask", None)
+                generated_ids = self.llm.generate(
+                    inputs_embeds=inputs_embeds,
+                    attention_mask=attention_mask,
+                    max_new_tokens=kwargs.get("max_length", 512),
+                    pad_token_id=self.llm.config.pad_token_id or self.llm.config.eos_token_id,
+                    **llm_kwargs,
+                )
+
+                response = tokenizer.batch_decode(
+                    generated_ids,
+                    skip_special_tokens=kwargs.get("skip_special_tokens", True),
+                )[0]
+
+                loss = None
+            else:
+                labels_ids = batch["labels_ids"]
+                labels_ids[labels_ids == -1] = -100
+                attention_mask = batch.get("attention_mask", None)
+                model_outputs = self.llm(
+                    inputs_embeds=inputs_embeds,
+                    attention_mask=attention_mask,
+                    labels=labels_ids,
+                    pad_token_id=self.llm.config.pad_token_id or self.llm.config.eos_token_id,
+                    **llm_kwargs,
+                )
+
+                preds = torch.argmax(model_outputs.logits, -1)[:, source_ids.shape[1] :]
+                response = tokenizer.batch_decode(
+                    preds,
+                    add_special_tokens=False,
+                    skip_special_tokens=kwargs.get("skip_special_tokens", True),
+                )[0]
+                loss = model_outputs.loss.item()
+        response = kwargs.get("prev_text", "") + response
+
+        ibest_writer = None
+        if kwargs.get("output_dir") is not None:
+            if not hasattr(self, "writer"):
+                self.writer = DatadirWriter(kwargs.get("output_dir"))
+            ibest_writer = self.writer[f"{0 + 1}best_recog"]
+
+        results = []
+        response_clean = re.sub(r"[^\w\s\u3000\u4e00-\u9fff]+", "", response)
+        result_i = {
+            "key": key[0],
+            "text": re.sub(r"\s+", " ", response.replace("/sil", " ")),
+            "text_tn": response_clean,
+            "label": label,
+        }
+        if loss is not None:
+            result_i["loss"] = loss
+        results.append(result_i)
+
+        for ctc_result, result in zip(ctc_results, results):
+            result["ctc_text"] = ctc_result["text"].replace("<|nospeech|>", "")
+            target_ids = torch.tensor(
+                self.ctc_tokenizer.encode(result["ctc_text"]), dtype=torch.int64
+            )
+            result["ctc_timestamps"] = forced_align(
+                ctc_result["ctc_logits"], target_ids, self.blank_id
+            )
+            target_ids = torch.tensor(self.ctc_tokenizer.encode(result["text"]), dtype=torch.int64)
+            result["timestamps"] = forced_align(ctc_result["ctc_logits"], target_ids, self.blank_id)
+            for timestamps in [result["timestamps"], result["ctc_timestamps"]]:
+                for timestamp in timestamps:
+                    timestamp["token"] = self.ctc_tokenizer.decode([timestamp["token"]])
+                    timestamp["start_time"] = timestamp["start_time"] * 6 * 10 / 1000
+                    timestamp["end_time"] = timestamp["end_time"] * 6 * 10 / 1000
+
+        if ibest_writer is not None:
+            ibest_writer["text"][key[0]] = response.replace("\n", " ")
+            ibest_writer["label"][key[0]] = label.replace("\n", " ")
+            ibest_writer["text_tn"][key[0]] = response_clean
+
+        return results, meta_data
+
+    @staticmethod
+    def from_pretrained(model: str = None, **kwargs):
+        from funasr import AutoModel
+
+        model, kwargs = AutoModel.build_model(model=model, trust_remote_code=True, **kwargs)
+
+        return model, kwargs
diff --git a/funasr/requirements.txt b/funasr/requirements.txt
new file mode 100644
index 0000000..38952e9
--- /dev/null
+++ b/funasr/requirements.txt
@@ -0,0 +1,14 @@
+requests
+wheel
+websocket-client
+pydantic>=2.0.0
+numpy<2.0
+PYYaml
+Levenshtein
+ruamel.yaml
+nltk==3.7
+pynini==2.1.6
+soundfile
+fastapi
+uvicorn
+python-multipart
\ No newline at end of file
diff --git a/funasr/warmup.wav b/funasr/warmup.wav
new file mode 100644
index 0000000..645d314
Binary files /dev/null and b/funasr/warmup.wav differ
diff --git a/sample_data/lei-jun-test.wav b/sample_data/lei-jun-test.wav
new file mode 100644
index 0000000..84f4482
Binary files /dev/null and b/sample_data/lei-jun-test.wav differ
diff --git a/sample_data/lei-jun.txt b/sample_data/lei-jun.txt
new file mode 100644
index 0000000..fc8a560
--- /dev/null
+++ b/sample_data/lei-jun.txt
@@ -0,0 +1 @@
+朋友们晚上好，欢迎大家来参加今天晚上的活动，谢谢大家。这是我第四次办年度演讲，前三次呢因为疫情的原因，都在小米科技园内举办。现场呢人很少。这是第四次，我们仔细想了想，我们还是想办一个比较大的聚会。然后呢让我们的新朋友老朋友一起聚一聚。今天的话呢我们就在北京的国家会议中心呢举办了这么一个活动。现场呢来了很多人大概有3500人。还有很多很多的朋友呢，通过观看直播的方式来参与。再一次呢对大家的参加，表示感谢，谢谢大家。两个月前我参加了今年武汉大学的毕业典礼。今年呢是武汉大学建校130周年，作为校友被母校邀请在毕业典礼上致辞，这对我来说，是至高无上的荣誉。站在讲台的那一刻，面对全校师生，关于武大的所有的记忆一下子涌现在脑海里。今天呢我就先和大家聊聊武大往事。那还是36年前，1987年我呢考上了武汉大学的计算机系。在武汉大学的图书馆里看了一本书《硅谷之火》，建立了我一生的梦想。看完书以后，热血沸腾，激动得睡不着觉。我还记得那天晚上星光很亮。我就在武大的操场上，就是屏幕上这个操场，走了一圈又一圈，走了整整一个晚上。我心里有团火，我也想办一个伟大的公司，就是这样，梦想之火，在我心里彻底点燃了。但是一个大一的新生，但是一个大一的新生，一个从县城里出来的年轻人，什么也不会，什么也没有，就想创办一家伟大的公司，这不就是天方夜谭吗？这么离谱的一个梦想，该如何实现呢？那天晚上我想了一整晚上，说实话，越想越糊涂，完全理不清头绪，后来我在想，哎干脆别想了，把书念好是正事，所以呢我就下定决心认认真真读书，那么我怎么能够把书读得不同凡响呢？
\ No newline at end of file
diff --git a/sherpa-onnx/Dockerfile.sherpa-onnx-bi150 b/sherpa-onnx/Dockerfile.sherpa-onnx-bi150
new file mode 100644
index 0000000..4a199a7
--- /dev/null
+++ b/sherpa-onnx/Dockerfile.sherpa-onnx-bi150
@@ -0,0 +1,25 @@
+FROM corex:4.3.8
+
+WORKDIR /root
+
+RUN set -eux; \
+  # 1) 把 aliyun 源替换成官方源（避免 403）
+  sed -i -E 's|http://mirrors\.aliyun\.com/ubuntu|http://archive.ubuntu.com/ubuntu|g' /etc/apt/sources.list; \
+  sed -i -E 's|http://mirrors\.aliyun\.com/ubuntu|http://archive.ubuntu.com/ubuntu|g' /etc/apt/sources.list.d/*.list 2>/dev/null || true; \
+  \
+  # 2) 更新并安装
+  apt-get update; \
+  apt-get install -y --no-install-recommends vim net-tools ca-certificates libasound2-dev patchelf; \
+  rm -rf /var/lib/apt/lists/*
+
+ADD . /root/
+
+COPY requirements.txt /root
+RUN pip install -r requirements.txt -i https://nexus.4pd.io/repository/pypi-all/simple --extra-index-url https://mirror.sjtu.edu.cn/pypi/web/simple
+COPY sherpa_onnx-1.12.5+corex4.3.8-cp310-cp310-linux_x86_64.whl /root
+RUN pip install ./sherpa_onnx-1.12.5+corex4.3.8-cp310-cp310-linux_x86_64.whl
+
+ENV LD_LIBRARY_PATH=/usr/local/corex-4.3.8/lib64/python3/dist-packages/tvm/:$LD_LIBRARY_PATH
+
+ENTRYPOINT ["python3"]
+CMD ["./main_sherpa.py"]
diff --git a/sherpa-onnx/README.md b/sherpa-onnx/README.md
new file mode 100644
index 0000000..9fdaa94
--- /dev/null
+++ b/sherpa-onnx/README.md
@@ -0,0 +1,28 @@
+# 天数智芯 天垓150 ASR（Sherpa-ONNX架构）
+
+## 镜像构造
+```shell
+docker build -f ./Dockerfile.sherpa-onnx-bi150 -t <your_image> .
+```
+其中，基础镜像 corex:4.3.8 通过联系天数智芯智铠100厂商技术支持可获取
+
+## 使用说明
+
+### 使用 FastAPI 启动ASR服务：
+例如：
+```shell
+docker run -dit -v /usr/src:/usr/src -v /lib/modules:/lib/modules --device=/dev/iluvatar0:/dev/iluvatar0 \
+    -v /mnt/contest_ceph/leaderboard/modelHubXC/mariolux/sherpa-onnx-dolphin-small-ctc-multi-lang-2025-04-02:/model \
+    --network=host <your_image> \
+    main_sherpa.py --model_dir /model --model_type dolphon_ctc --offline_model --use_gpu --port 1111
+```
+具体参数代码设定可参考代码文件
+
+### 测试ASR服务
+项目根路径`sample_data`目录下附带上了中文的测试音频和附带内容
+
+```shell
+curl -X POST http://localhost:1111/transduce \
+  -F "audio=@../sample_data/lei-jun-test.wav" \
+  -F "lang=zh"
+```
\ No newline at end of file
diff --git a/sherpa-onnx/fastapi_sherpa.py b/sherpa-onnx/fastapi_sherpa.py
new file mode 100644
index 0000000..633ceeb
--- /dev/null
+++ b/sherpa-onnx/fastapi_sherpa.py
@@ -0,0 +1,520 @@
+import os
+import sys
+import time
+import uuid
+import datetime
+import tempfile
+import soundfile as sf
+import sherpa_onnx
+import traceback
+
+from fastapi import FastAPI, File, UploadFile, HTTPException, BackgroundTasks, Form
+
+os.makedirs("./input", exist_ok=True)
+status = "Running"
+recognizer = None
+device = ""
+model_type = ""
+app = FastAPI()
+
+CUSTOM_DEVICE = os.getenv("CUSTOM_DEVICE", "")
+
+# 根据名称判断模型类型，比较杂，一共种类的自定义类型包括(针对OfflineRecognizer)
+# moonshine
+# fire_red
+# dolphin_ctc
+# paraformer
+# telespeech_ctc
+# whisper
+# sensevoice
+# zipformer_ctc
+# transducer
+# nemo_ctc
+# nemo_canary
+# wenet_ctc
+# 针对OnlineRecognizer只有 zipformer_ctc transducer paraformer nemo_ctc wenet_ctc 四种
+def get_asr_model_type(model_name):
+    # 根据名称判断模型类型以及需要检测的语种任务
+    # nemo_ctc, nemo_canary, moonshine 目前sherpa-onnx没有中文模型，执行英文ASR任务，其余模型执行中文ASR
+    # 所有nemo模型(nemo_ctc, nemo_canary以及transuducer中的nemo模型)均无中文模型
+    # 英文模型也并非全部大类都支持
+    
+    # 特殊规则
+    # zipformer带ctc的才属于zipformer_ctc那一类，否则属于transducer类
+    # nemo也是带上ctc或者canary才属于单独类别，否则属于transducer类
+    # conformer均为transducer类,但是得在nemo之后判断
+    # wenet 由于同时wenetspeech为数据集名称，各种类型都有可能，这个逻辑需放在后面
+    model_type = "unknown"
+    model_name_lower = model_name.lower()
+    if "tdnn" in model_name_lower:
+        model_type = "tdnn" # tdnn类别不适用，目前仅有一个模型只能识别希伯来语中的yes/no两种词语
+    elif "moonshine" in model_name_lower:
+        model_type = "moonshine"
+    elif "fire-red" in model_name_lower:
+        model_type = "fire_red"
+    elif "dolphin" in model_name_lower:
+        model_type = "dolphin_ctc"
+    elif "paraformer" in model_name_lower:
+        model_type = "paraformer"
+    elif "telespeech" in model_name_lower:
+        model_type = "telespeech_ctc"
+    elif "whisper" in model_name_lower:
+        model_type = "whisper"
+    elif "sense-voice" in model_name_lower:
+        model_type = "sensevoice"
+    elif "zipformer" in model_name_lower:
+        if "ctc" in model_name_lower:
+            model_type = "zipformer_ctc"
+        else:
+            model_type = "transducer"
+    elif "nemo" in model_name_lower:
+        if "ctc" in model_name_lower:
+            model_type = "nemo_ctc"
+        elif "canary" in model_name_lower:
+            model_type = "nemo_canary"
+        else:
+            model_type = "transducer"
+    elif "conformer" in model_name_lower or "lstm" in model_name_lower:
+        model_type = "transducer"
+    elif "wenet" in model_name_lower:
+        model_type = "wenet_ctc"
+    else:
+        model_type = "unknown"
+    return model_type
+    
+@app.on_event("startup")
+def load_model():
+    global status, recognizer, device, model_type
+    config = app.state.config
+    use_gpu = config.get("use_gpu", True)
+    model_dir = config.get("model_dir", "/model")
+    _model_type = config.get("model_type", None)
+    _model_name = config.get("model_name", None)
+    warmup = config.get("warmup", False)
+    isOffline = config.get("offline_model", True)
+    num_threads = config.get("num_threads", 2)
+    
+    device = "cpu"
+    if use_gpu:
+        if CUSTOM_DEVICE.startswith("mlu"):
+            device = "mlu:0"
+        elif CUSTOM_DEVICE.startswith("ascend"):
+            device = "npu:0"
+        else:
+            device = "cuda:0"
+    
+    # sherpa-onnx类型繁杂，当用户清楚的时候可提供model_type参数，抑或是提供完整的模型名称也行
+    # 因为挂载进入镜像的时候镜像内的文件路径不一定包含了模型名称
+    if _model_type:
+        model_type = _model_type
+    elif _model_name:
+        model_type = get_asr_model_type(_model_name)
+    else:
+        print("model_name and model_type both not provided, start guessing using model_dir", flush=True)
+        model_name = os.path.basename(model_dir)
+        model_type = get_asr_model_type(model_name)
+        
+    print(">> Startup config:")
+    print("   model_dir =", model_dir, flush=True)
+    print("   model_type =", model_type, flush=True)
+    print("   use_gpu =", use_gpu, flush=True)
+    print("   warmup =", warmup, flush=True)
+    print("   isOffline =", isOffline, flush=True)
+    print("   num_threads =", num_threads, flush=True)
+    
+    try:
+        recognizer = None
+        provider = "cuda" if use_gpu else "cpu"
+        file_list = os.listdir(model_dir)
+        # 目录内的模型文件可能会有多套（例如量化和不带量化版），选取大小最大的那一套
+        if model_type == "whisper":
+            encoder_list, decoder_list = [], []
+            tokens = ""
+            for file in file_list:
+                if "encode" in file and file.endswith(".onnx"):
+                    encoder_list.append(file)
+                elif "decode" in file and file.endswith(".onnx"):
+                    decoder_list.append(file)
+                elif "token" in file and file.endswith(".txt"):
+                    tokens = file
+            encoder = sorted(encoder_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            decoder = sorted(decoder_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            recognizer = sherpa_onnx.OfflineRecognizer.from_whisper(
+                encoder=model_dir + "/" + encoder,
+                decoder=model_dir + "/" + decoder,
+                tokens=model_dir + "/" + tokens,
+                language="zh",
+                debug=False,
+                provider=provider,
+                num_threads=num_threads
+            )
+            
+        elif model_type == "sensevoice":
+            model_list = []
+            tokens = ""
+            for file in file_list:
+                if file.endswith(".onnx"):
+                    model_list.append(file)
+                elif file.endswith(".txt") and "token" in file:
+                    tokens = file
+            model = sorted(model_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            recognizer = sherpa_onnx.OfflineRecognizer.from_sense_voice(
+                model=model_dir + "/" + model,
+                tokens=model_dir + "/" + tokens,
+                debug=False,
+                use_itn=True,
+                language="zh",
+                provider=provider,
+                num_threads=num_threads
+            )
+        elif model_type == "paraformer":
+            model_list = []
+            for file in file_list:
+                if file.endswith(".onnx"):
+                    model_list.append(file)
+                elif file.endswith(".txt") and "token" in file:
+                    tokens = file
+            model = sorted(model_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            if isOffline:
+                recognizer = sherpa_onnx.OfflineRecognizer.from_paraformer(
+                    paraformer=model_dir + "/" + model,
+                    tokens=model_dir + "/" + tokens,
+                    debug=False,
+                    provider=provider,
+                    num_threads=num_threads
+                )
+            else:
+                recognizer = sherpa_onnx.OnlineRecognizer.from_paraformer(
+                    paraformer=model_dir + "/" + model,
+                    tokens=model_dir + "/" + tokens,
+                    debug=False,
+                    provider=provider,
+                    num_threads=num_threads
+                )
+        elif model_type == "zipformer_ctc":
+            model_list = []
+            for file in file_list:
+                if file.endswith(".onnx"):
+                    model_list.append(file)
+                elif file.endswith(".txt") and "token" in file:
+                    tokens = file
+            model = sorted(model_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            if isOffline:
+                recognizer = sherpa_onnx.OfflineRecognizer.from_zipformer_ctc(
+                    model=model_dir + "/" + model,
+                    tokens=model_dir + "/" + tokens,
+                    debug=False,
+                    provider=provider,
+                    num_threads=num_threads
+                )
+            else:
+                recognizer = sherpa_onnx.OnlineRecognizer.from_zipformer2_ctc(
+                    model=model_dir + "/" + model,
+                    tokens=model_dir + "/" + tokens,
+                    debug=False,
+                    provider=provider,
+                    num_threads=num_threads
+                )
+        elif model_type == "telespeech_ctc":
+            model_list = []
+            for file in file_list:
+                if file.endswith(".onnx"):
+                    model_list.append(file)
+                elif file.endswith(".txt") and "token" in file:
+                    tokens = file
+            model = sorted(model_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            recognizer = sherpa_onnx.OfflineRecognizer.from_telespeech_ctc(
+                model=model_dir + "/" + model,
+                tokens=model_dir + "/" + tokens,
+                debug=False,
+                provider=provider,
+                num_threads=num_threads
+            )
+        elif model_type == "fire_red":
+            encoder_list, decoder_list = [], []
+            tokens = ""
+            for file in file_list:
+                if "encode" in file and file.endswith(".onnx"):
+                    encoder_list.append(file)
+                elif "decode" in file and file.endswith(".onnx"):
+                    decoder_list.append(file)
+                elif "token" in file and file.endswith(".txt"):
+                    tokens = file
+            encoder = sorted(encoder_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            decoder = sorted(decoder_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            recognizer = sherpa_onnx.OfflineRecognizer.from_fire_red_asr(
+                encoder=model_dir + "/" + encoder,
+                decoder=model_dir + "/" + decoder,
+                tokens=model_dir + "/" + tokens,
+                debug=False,
+                provider=provider,
+                num_threads=num_threads
+            )
+        elif model_type == "wenet_ctc":
+            model_list = []
+            for file in file_list:
+                if file.endswith(".onnx"):
+                    model_list.append(file)
+                elif file.endswith(".txt") and "token" in file:
+                    tokens = file
+            model = sorted(model_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            if isOffline:
+                recognizer = sherpa_onnx.OfflineRecognizer.from_wenet_ctc(
+                    model=model_dir + "/" + model,
+                    tokens=model_dir + "/" + tokens,
+                    debug=False,
+                    provider=provider,
+                    num_threads=num_threads
+                )
+            else:
+                recognizer = sherpa_onnx.OnlineRecognizer.from_wenet_ctc(
+                    model=model_dir + "/" + model,
+                    tokens=model_dir + "/" + tokens,
+                    debug=False,
+                    provider=provider,
+                    num_threads=num_threads
+                )
+        elif model_type == "dolphin_ctc":
+            model_list = []
+            for file in file_list:
+                if file.endswith(".onnx"):
+                    model_list.append(file)
+                elif file.endswith(".txt") and "token" in file:
+                    tokens = file
+            model = sorted(model_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            recognizer = sherpa_onnx.OfflineRecognizer.from_dolphin_ctc(
+                model=model_dir + "/" + model,
+                tokens=model_dir + "/" + tokens,
+                debug=False,
+                provider=provider,
+                num_threads=num_threads
+            )
+        elif model_type == "transducer":
+            encoder_list, decoder_list, joiner_list = [], [], []
+            tokens = ""
+            for file in file_list:
+                if "encode" in file and file.endswith(".onnx"):
+                    encoder_list.append(file)
+                elif "decode" in file and file.endswith(".onnx"):
+                    decoder_list.append(file)
+                elif "joiner" in file and file.endswith(".onnx"):
+                    joiner_list.append(file)
+                elif "token" in file and file.endswith(".txt"):
+                    tokens = file
+            encoder = sorted(encoder_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            decoder = sorted(decoder_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            joiner = sorted(joiner_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            # 特殊情况，zipformer,conformer都是icefall导出，默认类型即可，nemo-transducer需要专门区分
+            transducer_type = "nemo_transducer" if "nemo" in model_name.lower() else "transducer"
+            if isOffline:
+                recognizer = sherpa_onnx.OfflineRecognizer.from_transducer(
+                    encoder=model_dir + "/" + encoder,
+                    decoder=model_dir + "/" + decoder,
+                    joiner=model_dir + "/" + joiner,
+                    tokens=model_dir + "/" + tokens,
+                    model_type=transducer_type,
+                    debug=False,
+                    provider=provider,
+                    num_threads=num_threads
+                )
+            else:
+                recognizer = sherpa_onnx.OnlineRecognizer.from_transducer(
+                    encoder=model_dir + "/" + encoder,
+                    decoder=model_dir + "/" + decoder,
+                    joiner=model_dir + "/" + joiner,
+                    tokens=model_dir + "/" + tokens,
+                    debug=False,
+                    provider=provider,
+                    num_threads=num_threads
+                )
+        elif model_type == "nemo_ctc":
+            model_list = []
+            for file in file_list:
+                if file.endswith(".onnx"):
+                    model_list.append(file)
+                elif file.endswith(".txt") and "token" in file:
+                    tokens = file
+            model = sorted(model_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            if isOffline:
+                recognizer = sherpa_onnx.OfflineRecognizer.from_nemo_ctc(
+                    model=model_dir + "/" + model,
+                    tokens=model_dir + "/" + tokens,
+                    debug=False,
+                    provider=provider,
+                    num_threads=num_threads
+                )
+            else:
+                recognizer = sherpa_onnx.OnlineRecognizer.from_nemo_ctc(
+                    model=model_dir + "/" + model,
+                    tokens=model_dir + "/" + tokens,
+                    debug=False,
+                    provider=provider,
+                    num_threads=num_threads
+                )
+        elif model_type == "nemo_canary":
+            encoder_list, decoder_list = [], []
+            tokens = ""
+            for file in file_list:
+                if "encode" in file and file.endswith(".onnx"):
+                    encoder_list.append(file)
+                elif "decode" in file and file.endswith(".onnx"):
+                    decoder_list.append(file)
+                elif "token" in file and file.endswith(".txt"):
+                    tokens = file
+            encoder = sorted(encoder_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            decoder = sorted(decoder_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            recognizer = sherpa_onnx.OfflineRecognizer.from_nemo_canary(
+                encoder=model_dir + "/" + encoder,
+                decoder=model_dir + "/" + decoder,
+                tokens=model_dir + "/" + tokens,
+                debug=False,
+                provider=provider,
+                num_threads=num_threads
+            )
+        elif model_type == "moonshine":
+            preprocessor_list, encoder_list, cached_decoder_list, uncached_decoder_list = [], [], [], []
+            tokens = ""
+            for file in file_list:
+                if "preprocess" in file and file.endswith(".onnx"):
+                    preprocessor_list.append(file)
+                elif "encode" in file and file.endswith(".onnx"):
+                    encoder_list.append(file)
+                elif "uncached_decode" in file and file.endswith(".onnx"):
+                    uncached_decoder_list.append(file)
+                elif "cached_decode" in file and file.endswith(".onnx"):
+                    cached_decoder_list.append(file)
+                elif "token" in file and file.endswith(".txt"):
+                    tokens = file
+            preprocessor = sorted(preprocessor_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            encoder = sorted(encoder_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            cached_decoder = sorted(cached_decoder_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            uncached_decoder = sorted(uncached_decoder_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            recognizer = sherpa_onnx.OfflineRecognizer.from_moonshine(
+                preprocessor=model_dir + "/" + preprocessor,
+                encoder=model_dir + "/" + encoder,
+                cached_decoder=model_dir + "/" + cached_decoder,
+                uncached_decoder=model_dir + "/" + uncached_decoder,
+                tokens=model_dir + "/" + tokens,
+                debug=False,
+                provider=provider,
+                num_threads=num_threads
+            )
+        elif model_type == "tdnn_ctc":
+            model_list = []
+            for file in file_list:
+                if file.endswith(".onnx"):
+                    model_list.append(file)
+                elif file.endswith(".txt") and "token" in file:
+                    tokens = file
+            model = sorted(model_list, key=lambda x: os.path.getsize(model_dir + "/" + x), reverse=True)[0]
+            recognizer = sherpa_onnx.OfflineRecognizer.from_tdnn_ctc(
+                model=model_dir + "/" + model,
+                tokens=model_dir + "/" + tokens,
+                debug=False,
+                provider=provider,
+                num_threads=num_threads
+            )
+        else:
+            raise RuntimeError("Cannot recognize model_type")
+    except Exception as e:
+        raise RuntimeError(f"Failed to initial cuda model: {e}")
+    
+    if warmup:
+        print("Start warmup...", flush=True)
+        stream = recognizer.create_stream()
+        audio, sample_rate = sf.read("warmup.wav", dtype="float32", always_2d=True)
+        stream.accept_waveform(sample_rate, audio)
+        recognizer.decode_stream(stream)
+        print("warmup complete.", flush=True)
+
+    status = "Success"
+    
+def test_sherpa(wavefile):
+    isOffline = app.state.config.get("offline_model", True)
+    audio, sample_rate = sf.read(wavefile, dtype="float32", always_2d=True)
+    audio = audio[:, 0]
+    generated_text = ""
+    
+    start_t = datetime.datetime.now()
+    if isOffline:
+        # OfflineRecognizer非流式模型推理
+        if model_type in ["sensevoice"]:
+            stream = recognizer.create_stream()
+            stream.accept_waveform(sample_rate, audio)
+            recognizer.decode_stream(stream)
+            generated_text = stream.result.text
+        else:
+            # offline-asr model 大多对长音频支持不佳，模型训练音频不长以及导出onnx结构中对一些中间态维度可能有上限
+            # 哪怕原版CPU推理中间可能都会崩溃，采取小段切分形式测试
+            start_index = 0
+            internal = int(sample_rate * 29)
+            while start_index < len(audio):
+                stream = recognizer.create_stream()
+                stream.accept_waveform(sample_rate, audio[start_index:start_index + internal])
+                recognizer.decode_stream(stream)
+                generated_text += stream.result.text
+                start_index += internal
+    else:
+        # OnlineRecognizer流式模型推理,统一每一次只投喂2s音频数据
+        stream = recognizer.create_stream()
+        start_index = 0
+        chunk_size = int(sample_rate * 2)
+
+        while start_index < len(audio):
+            chunk = audio[start_index:start_index + chunk_size]
+            stream.accept_waveform(sample_rate, chunk)
+
+            while recognizer.is_ready(stream):
+                recognizer.decode_stream(stream)
+                # mid_text = recognizer.get_result(stream)
+                # print("partial result: " + mid_text, flush=True)
+            start_index += chunk_size
+
+        while recognizer.is_ready(stream):
+            recognizer.decode_stream(stream)
+        generated_text = recognizer.get_result(stream)
+
+    end_t = datetime.datetime.now()
+    elapsed_seconds = (end_t - start_t).total_seconds()
+    duration = audio.shape[-1] / sample_rate
+    rtf = elapsed_seconds / duration
+    
+    print("Text:", generated_text, flush=True)
+    print(f"Audio duration:\t{duration:.3f} s", flush=True)
+    print(f"Elapsed:\t{elapsed_seconds:.3f} s", flush=True)
+    print(f"RTF = {elapsed_seconds:.3f}/{duration:.3f} = {rtf:.3f}", flush=True)
+    
+    return generated_text
+
+@app.get("/health")
+def health():
+
+    if status=="Running":
+        return {
+            "status":"loading model"
+        }
+    ret = {
+        "status": "ok" if status == "Success" else "failed",
+    }
+    return ret
+
+@app.post("/transduce")
+def transduce(
+    audio: UploadFile = File(...),
+    lang: str = Form("zh"),
+    background_tasks: BackgroundTasks = None
+):
+    try:
+        file_path = f"./input/{uuid.uuid4()}.wav"
+        with open(file_path, "wb") as f:
+            f.write(audio.file.read())
+        background_tasks.add_task(os.remove, file_path)
+        generated_text = test_sherpa(file_path)
+
+        return {"generated_text": generated_text}
+    except Exception:
+        raise HTTPException(status_code=500, detail=f"Processing failed: \n{traceback.format_exc()}")
+
+# if __name__ == "__main__":
+
+#     uvicorn.run("fastapi_sherpa:app", host="0.0.0.0", port=1111, workers=1)
\ No newline at end of file
diff --git a/sherpa-onnx/main_sherpa.py b/sherpa-onnx/main_sherpa.py
new file mode 100644
index 0000000..6cece6a
--- /dev/null
+++ b/sherpa-onnx/main_sherpa.py
@@ -0,0 +1,33 @@
+import argparse
+import uvicorn
+from fastapi_sherpa import app
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser()
+    parser.add_argument("--model_dir", type=str, default="/model", required=True, help="model directory")
+    parser.add_argument("--model_type", type=str, default=None, help="model type, e.g. sensevoice")
+    parser.add_argument("--use_gpu", action="store_true", default=True)
+    parser.add_argument("--warmup", action="store_true", help="whether do warmup when first initializing model")
+    parser.add_argument("--model_name", type=str, default=None, help="model's full name(optional) to determine model type")
+    parser.add_argument("--num_threads", type=int, default=2, help="number of threads with model inference")
+    parser.add_argument("--offline_model", action="store_true", help="indicating a non-streaming model when this flag is set")
+    parser.add_argument("--port", type=int, default=8000, help="service port")
+
+    args = parser.parse_args()
+
+    # 将参数加到 app.state 中
+    app.state.config = {
+        "model_dir": args.model_dir,
+        "model_type": args.model_type,
+        "model_name": args.model_name,
+        "num_threads": args.num_threads,
+        "offline_model": args.offline_model,
+        "use_gpu": args.use_gpu, # True
+        "warmup": args.warmup,
+    }
+
+    uvicorn.run("fastapi_sherpa:app",
+        host="0.0.0.0",
+        port=args.port,
+        workers=1
+    )
\ No newline at end of file
diff --git a/sherpa-onnx/requirements.txt b/sherpa-onnx/requirements.txt
new file mode 100644
index 0000000..38952e9
--- /dev/null
+++ b/sherpa-onnx/requirements.txt
@@ -0,0 +1,14 @@
+requests
+wheel
+websocket-client
+pydantic>=2.0.0
+numpy<2.0
+PYYaml
+Levenshtein
+ruamel.yaml
+nltk==3.7
+pynini==2.1.6
+soundfile
+fastapi
+uvicorn
+python-multipart
\ No newline at end of file
diff --git a/sherpa-onnx/sherpa_onnx-1.12.5+corex4.3.8-cp310-cp310-linux_x86_64.whl b/sherpa-onnx/sherpa_onnx-1.12.5+corex4.3.8-cp310-cp310-linux_x86_64.whl
new file mode 100644
index 0000000..87a06a4
Binary files /dev/null and b/sherpa-onnx/sherpa_onnx-1.12.5+corex4.3.8-cp310-cp310-linux_x86_64.whl differ
diff --git a/sherpa-onnx/warmup.wav b/sherpa-onnx/warmup.wav
new file mode 100644
index 0000000..645d314
Binary files /dev/null and b/sherpa-onnx/warmup.wav differ
diff --git a/transformers/Dockerfile.transformers-bi150 b/transformers/Dockerfile.transformers-bi150
new file mode 100644
index 0000000..cc1bf1a
--- /dev/null
+++ b/transformers/Dockerfile.transformers-bi150
@@ -0,0 +1,23 @@
+FROM corex:4.3.8
+
+WORKDIR /root
+
+RUN set -eux; \
+  # 1) 把 aliyun 源替换成官方源（避免 403）
+  sed -i -E 's|http://mirrors\.aliyun\.com/ubuntu|http://archive.ubuntu.com/ubuntu|g' /etc/apt/sources.list; \
+  sed -i -E 's|http://mirrors\.aliyun\.com/ubuntu|http://archive.ubuntu.com/ubuntu|g' /etc/apt/sources.list.d/*.list 2>/dev/null || true; \
+  \
+  # 2) 更新并安装
+  apt-get update; \
+  apt-get install -y --no-install-recommends vim net-tools ca-certificates libasound2-dev patchelf; \
+  rm -rf /var/lib/apt/lists/*
+
+ADD . /root/
+
+COPY requirements.txt /root
+RUN pip install -r requirements.txt -i https://nexus.4pd.io/repository/pypi-all/simple --extra-index-url https://mirror.sjtu.edu.cn/pypi/web/simple
+
+RUN pip install transformers==4.51.3 -i https://nexus.4pd.io/repository/pypi-all/simple --extra-index-url https://mirror.sjtu.edu.cn/pypi/web/simple
+
+ENTRYPOINT ["python3"]
+CMD ["./main_transformers.py"]
\ No newline at end of file
diff --git a/transformers/README.md b/transformers/README.md
new file mode 100644
index 0000000..f6be08d
--- /dev/null
+++ b/transformers/README.md
@@ -0,0 +1,28 @@
+# 天数智芯 天垓150 ASR（Transformers架构）
+
+## 镜像构造
+```shell
+docker build -f ./Dockerfile.transformers-bi150 -t <your_image> .
+```
+其中，基础镜像 corex:4.3.8 通过联系天数智芯智铠100厂商技术支持可获取
+
+## 使用说明
+
+### 使用 FastAPI 启动ASR服务：
+例如：
+```shell
+docker run -dit -v /usr/src:/usr/src -v /lib/modules:/lib/modules --device=/dev/iluvatar0:/dev/iluvatar0 \
+    -v /mnt/contest_ceph/leaderboard/modelHubXC/openai-mirror/whisper-small:/model \
+    --network=host <your_image> \
+    main_transformers.py --model_dir /model --use_gpu --port 1111
+```
+具体参数代码设定可参考代码文件
+
+### 测试ASR服务
+项目根路径`sample_data`目录下附带上了中文的测试音频和附带内容
+
+```shell
+curl -X POST http://localhost:1111/transduce \
+  -F "audio=@../sample_data/lei-jun-test.wav" \
+  -F "lang=zh"
+```
\ No newline at end of file
diff --git a/transformers/fastapi_transformers.py b/transformers/fastapi_transformers.py
new file mode 100644
index 0000000..1ac7a7b
--- /dev/null
+++ b/transformers/fastapi_transformers.py
@@ -0,0 +1,232 @@
+import os
+import time
+import uuid
+import json
+import inspect
+import traceback
+import numpy as np
+import torch
+import torchaudio
+
+from fastapi import FastAPI, File, UploadFile, HTTPException, BackgroundTasks, Form
+import uvicorn
+from transformers import pipeline as hf_pipeline
+
+os.makedirs("./input", exist_ok=True)
+status = "Running"
+asr_pipeline = None
+is_whisper = False  # 唯一需要区分的分支：Whisper(seq2seq) vs 其余所有 CTC 类模型
+device = ""
+app = FastAPI()
+
+CUSTOM_DEVICE = os.getenv("CUSTOM_DEVICE", "")
+if CUSTOM_DEVICE.startswith("mlu"):
+    import torch_mlu
+elif CUSTOM_DEVICE.startswith("ascend"):
+    import torch_npu
+elif CUSTOM_DEVICE.startswith("pt"):
+    import torch_dipu
+
+
+class _SamplingRateCompatProxy:
+    """为非标准 FeatureExtractor 提供兼容性包装。
+
+    transformers pipeline 的 preprocess 固定会向 feature_extractor 传 sampling_rate、
+    return_tensors 等标准 kwargs，但部分模型（如 GraniteSpeech）的 FeatureExtractor
+    没有实现这些参数。此代理在初始化时检查签名，调用时只转发 FeatureExtractor 实际接受的参数。
+    调用前须确保音频已按模型期望采样率重采样完毕（run_asr 中已完成）。
+    """
+    def __init__(self, fe):
+        object.__setattr__(self, "_fe", fe)
+        # 初始化时检查一次签名，确定接受哪些参数
+        try:
+            sig = inspect.signature(fe.__call__)
+            has_var_kw = any(
+                p.kind == inspect.Parameter.VAR_KEYWORD
+                for p in sig.parameters.values()
+            )
+            accepted = None if has_var_kw else set(sig.parameters.keys()) - {"self"}
+        except Exception:
+            accepted = None  # 无法检测时不过滤
+        object.__setattr__(self, "_accepted", accepted)
+
+    def __call__(self, *args, **kwargs):
+        accepted = object.__getattribute__(self, "_accepted")
+        if accepted is not None:
+            kwargs = {k: v for k, v in kwargs.items() if k in accepted}
+        return object.__getattribute__(self, "_fe")(*args, **kwargs)
+
+    def __getattr__(self, name):
+        return getattr(object.__getattribute__(self, "_fe"), name)
+
+    def __setattr__(self, name, value):
+        setattr(object.__getattribute__(self, "_fe"), name, value)
+
+
+def _check_is_whisper(model_dir: str, model_type_override: str = None) -> bool:
+    """判断是否为 Whisper 架构。
+    优先使用用户显式传入的 model_type_override，
+    否则读 config.json 中的 model_type 字段（所有 whisper fine-tuned 模型均有此字段）。
+    """
+    if model_type_override:
+        return model_type_override.lower() == "whisper"
+    config_path = os.path.join(model_dir, "config.json")
+    if os.path.exists(config_path):
+        with open(config_path, "r") as f:
+            cfg = json.load(f)
+        return cfg.get("model_type", "").lower() == "whisper"
+    # config.json 不存在时，从目录名做最后兜底
+    return "whisper" in os.path.basename(model_dir).lower()
+
+
+@app.on_event("startup")
+def load_model():
+    global status, asr_pipeline, is_whisper, device
+
+    config = app.state.config
+    use_gpu = config.get("use_gpu", True)
+    model_dir = config.get("model_dir", "/model")
+    model_type_override = config.get("model_type", None)  # 可选，仅用于覆盖自动判断
+    warmup = config.get("warmup", False)
+    use_fp16 = config.get("fp16", False)  # 默认 fp32，需要用户显式开启
+
+    # 与 fastapi_funasr.py 保持一致的设备字符串逻辑，直接传字符串给 pipeline
+    device = "cpu"
+    if use_gpu:
+        if CUSTOM_DEVICE.startswith("mlu"):
+            device = "mlu:0"
+        elif CUSTOM_DEVICE.startswith("ascend"):
+            device = "npu:0"
+        else:
+            device = "cuda:0"
+
+    is_whisper = _check_is_whisper(model_dir, model_type_override)
+    # 默认 fp32，跨平台兼容性最好且不影响精度对比
+    # fp16 需要用户显式开启（--fp16），且应确认当前硬件支持
+    torch_dtype = torch.float16 if use_fp16 else torch.float32
+
+    print(">> Startup config:")
+    print("   model_dir =", model_dir, flush=True)
+    print("   is_whisper =", is_whisper, flush=True)
+    print("   device =", device, flush=True)
+    print("   torch_dtype =", torch_dtype, flush=True)
+    print("   chunk_length_s =", app.state.config.get("chunk_length_s", 30), flush=True)
+    print("   warmup =", warmup, flush=True)
+
+    # transformers pipeline 直接接受设备字符串（"cpu"/"cuda:0"/"mlu:0"/"npu:0"）
+    # 会自动读取 config.json 实例化正确的模型类，无需手动指定架构
+    # 注意：不在 pipeline 构建时传 chunk_length_s，由 run_asr 自行分片后逐段调用
+    # 原因：部分模型（如 GraniteSpeech）的 FeatureExtractor 不接受 sampling_rate 参数，
+    #       而 pipeline 内部的 chunk_iter 固定会传该参数，导致报错
+    asr_pipeline = hf_pipeline(
+        task="automatic-speech-recognition",
+        model=model_dir,
+        device=device,
+        torch_dtype=torch_dtype,
+    )
+
+    # 检查 feature extractor 是否接受 sampling_rate 参数
+    # pipeline 的 preprocess 固定会传此参数（硬编码行为），不接受的模型需要代理包装
+    try:
+        sig = inspect.signature(asr_pipeline.feature_extractor.__call__)
+        accepts_sr = "sampling_rate" in sig.parameters or any(
+            p.kind == inspect.Parameter.VAR_KEYWORD for p in sig.parameters.values()
+        )
+    except Exception:
+        accepts_sr = True  # 无法检测时保守假设接受
+    if not accepts_sr:
+        asr_pipeline.feature_extractor = _SamplingRateCompatProxy(asr_pipeline.feature_extractor)
+        print("   Note: FeatureExtractor does not accept sampling_rate, applied compat proxy", flush=True)
+
+    if warmup:
+        print("Start warmup...", flush=True)
+        # 获取模型期望的采样率，绝大多数模型的 feature extractor 都有此属性
+        # 极少数非标准模型可能没有，兜底用 16000（ASR 领域最通用的标准采样率）
+        target_sr = getattr(asr_pipeline.feature_extractor, "sampling_rate", 16000)
+        dummy = np.zeros(target_sr, dtype=np.float32)  # 1 秒静音
+        asr_pipeline(dummy, **_build_infer_kwargs("zh"))
+        print("warmup complete.", flush=True)
+
+    status = "Success"
+
+
+def _build_infer_kwargs(lang: str) -> dict:
+    """Whisper 推理时需要额外传语言参数；CTC 类无需额外参数。
+    不再传 return_timestamps，因为我们自行分片后逐段调用 pipeline，无需 pipeline 内部拼接。
+    """
+    if is_whisper:
+        return {"generate_kwargs": {"language": lang, "task": "transcribe"}}
+    return {}
+
+
+def run_asr(audio_file: str, lang: str) -> str:
+    waveform, sample_rate = torchaudio.load(audio_file)
+    duration = waveform.shape[1] / sample_rate
+
+    # 多声道转单声道
+    if waveform.shape[0] > 1:
+        waveform = waveform.mean(dim=0, keepdim=True)
+
+    # 提前重采样到模型期望的采样率
+    # 传 numpy array（非 dict）给 pipeline，跳过 pipeline 内部的 sampling_rate 传递逻辑，
+    # 规避部分模型（如 GraniteSpeech）的 FeatureExtractor 不接受 sampling_rate 参数的问题
+    # 获取模型期望的采样率，绝大多数模型的 feature extractor 都有此属性
+    # 极少数非标准模型可能没有，兜底用 16000（ASR 领域最通用的标准采样率）
+    target_sr = getattr(asr_pipeline.feature_extractor, "sampling_rate", 16000)
+    if sample_rate != target_sr:
+        resampler = torchaudio.transforms.Resample(sample_rate, target_sr)
+        waveform = resampler(waveform)
+
+    audio_array = waveform.squeeze(0).numpy().astype(np.float32)
+
+    chunk_length_s = app.state.config.get("chunk_length_s", 30)
+    chunk_samples = chunk_length_s * target_sr
+    infer_kwargs = _build_infer_kwargs(lang)
+
+    ts1 = time.time()
+    texts = []
+    for i in range(0, len(audio_array), chunk_samples):
+        chunk = audio_array[i : i + chunk_samples]
+        result = asr_pipeline(chunk, **infer_kwargs)
+        texts.append(result["text"])
+    ts2 = time.time()
+
+    generated_text = "".join(texts)
+    # wav2vec2 系列模型会用 U+2581 (▁) 作为词间分隔符，替换为空格
+    generated_text = generated_text.replace("▁", " ").replace(chr(9601), " ").strip()
+
+    processing_time = ts2 - ts1
+    rtf = processing_time / duration
+    print("Text:", generated_text, flush=True)
+    print(f"Audio duration:\t{duration:.3f} s", flush=True)
+    print(f"Elapsed:\t{processing_time:.3f} s", flush=True)
+    print(f"RTF = {processing_time:.3f}/{duration:.3f} = {rtf:.3f}", flush=True)
+
+    return generated_text
+
+
+@app.get("/health")
+def health():
+    if status == "Running":
+        return {"status": "loading model"}
+    return {"status": "ok" if status == "Success" else "failed"}
+
+
+@app.post("/transduce")
+def transduce(
+    audio: UploadFile = File(...),
+    lang: str = Form("zh"),
+    background_tasks: BackgroundTasks = None,
+):
+    try:
+        file_path = f"./input/{uuid.uuid4()}.wav"
+        with open(file_path, "wb") as f:
+            f.write(audio.file.read())
+        background_tasks.add_task(os.remove, file_path)
+        generated_text = run_asr(file_path, lang)
+        return {"generated_text": generated_text}
+    except Exception:
+        raise HTTPException(status_code=500, detail=f"Processing failed: \n{traceback.format_exc()}")
+
+# if __name__ == "__main__":
+#     uvicorn.run("fastapi_transformers:app", host="0.0.0.0", port=8000, workers=1)
diff --git a/transformers/main_transformers.py b/transformers/main_transformers.py
new file mode 100644
index 0000000..c7c3aa0
--- /dev/null
+++ b/transformers/main_transformers.py
@@ -0,0 +1,38 @@
+import argparse
+import uvicorn
+from fastapi_transformers import app
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser()
+    parser.add_argument("--model_dir", type=str, default="/model",
+                        help="模型目录（挂载到容器内的路径）")
+    parser.add_argument("--model_type", type=str, default=None,
+                        help="可选，仅在自动推断失败时手动指定: whisper 或不填（CTC 类均不需要填）")
+    parser.add_argument("--use_gpu", action="store_true", default=True,
+                        help="是否使用 GPU（CUDA）")
+    parser.add_argument("--warmup", action="store_true",
+                        help="启动时用静音片段执行一次 warmup 推理")
+    parser.add_argument("--chunk_length_s", type=int, default=30,
+                        help="长音频切片长度（秒），逐段推理，默认 30")
+    parser.add_argument("--fp16", action="store_true", default=False,
+                        help="使用 float16 推理（默认 float32）。仅在确认硬件支持时开启，"
+                             "注意 fp16/fp32 之间存在精度差异，跨卡对比时建议保持默认 fp32")
+    parser.add_argument("--port", type=int, default=8000,
+                        help="FastAPI 服务端口，默认 8000")
+
+    args = parser.parse_args()
+
+    app.state.config = {
+        "model_dir": args.model_dir,
+        "model_type": args.model_type,
+        "use_gpu": args.use_gpu,
+        "warmup": args.warmup,
+        "chunk_length_s": args.chunk_length_s,
+        "fp16": args.fp16,
+    }
+
+    uvicorn.run("fastapi_transformers:app",
+        host="0.0.0.0",
+        port=args.port,
+        workers=1
+    )
diff --git a/transformers/requirements.txt b/transformers/requirements.txt
new file mode 100644
index 0000000..38952e9
--- /dev/null
+++ b/transformers/requirements.txt
@@ -0,0 +1,14 @@
+requests
+wheel
+websocket-client
+pydantic>=2.0.0
+numpy<2.0
+PYYaml
+Levenshtein
+ruamel.yaml
+nltk==3.7
+pynini==2.1.6
+soundfile
+fastapi
+uvicorn
+python-multipart
\ No newline at end of file
diff --git a/transformers/warmup.wav b/transformers/warmup.wav
new file mode 100644
index 0000000..645d314
Binary files /dev/null and b/transformers/warmup.wav differ