diff --git a/docs/models/supported_models.md b/docs/models/supported_models.md
index 7ff9531c5..7f20d2052 100644
--- a/docs/models/supported_models.md
+++ b/docs/models/supported_models.md
@@ -790,6 +790,7 @@ Speech2Text models trained specifically for Automatic Speech Recognition.
 
 | Architecture | Models | Example HF Models | [LoRA](../features/lora.md) | [PP](../serving/parallelism_scaling.md) |
 |--------------|--------|-------------------|----------------------|---------------------------|
+| `FunASRForConditionalGeneration` | FunASR | `allendou/Fun-ASR-Nano-2512-vllm`, etc. | | |
 | `Gemma3nForConditionalGeneration` | Gemma3n | `google/gemma-3n-E2B-it`, `google/gemma-3n-E4B-it`, etc. | | |
 | `GlmAsrForConditionalGeneration` | GLM-ASR | `zai-org/GLM-ASR-Nano-2512` | ✅︎ | ✅︎ |
 | `GraniteSpeechForConditionalGeneration` | Granite Speech | `ibm-granite/granite-speech-3.3-2b`, `ibm-granite/granite-speech-3.3-8b`, etc. | ✅︎ | ✅︎ |
diff --git a/examples/online_serving/openai_transcription_client.py b/examples/online_serving/openai_transcription_client.py
index 966bfd2a4..478a0a7ea 100644
--- a/examples/online_serving/openai_transcription_client.py
+++ b/examples/online_serving/openai_transcription_client.py
@@ -26,7 +26,9 @@ from openai import AsyncOpenAI, OpenAI
 from vllm.assets.audio import AudioAsset
 
 
-def sync_openai(audio_path: str, client: OpenAI, model: str):
+def sync_openai(
+    audio_path: str, client: OpenAI, model: str, *, repetition_penalty: float = 1.3
+):
     """
     Perform synchronous transcription using OpenAI-compatible API.
     """
@@ -40,7 +42,7 @@ def sync_openai(audio_path: str, client: OpenAI, model: str):
             # Additional sampling params not provided by OpenAI API.
             extra_body=dict(
                 seed=4419,
-                repetition_penalty=1.3,
+                repetition_penalty=repetition_penalty,
             ),
         )
         print("transcription result [sync]:", transcription.text)
@@ -129,7 +131,12 @@ def main(args):
     print(f"Using model: {model}")
 
     # Run the synchronous function
-    sync_openai(args.audio_path if args.audio_path else mary_had_lamb, client, model)
+    sync_openai(
+        audio_path=args.audio_path if args.audio_path else mary_had_lamb,
+        client=client,
+        model=model,
+        repetition_penalty=args.repetition_penalty,
+    )
 
     # Run the asynchronous function
     if "openai" in model:
@@ -161,5 +168,11 @@ if __name__ == "__main__":
         default=None,
         help="The path to the audio file to transcribe.",
     )
+    parser.add_argument(
+        "--repetition_penalty",
+        type=float,
+        default=1.3,
+        help="repetition penalty",
+    )
     args = parser.parse_args()
     main(args)
diff --git a/tests/models/registry.py b/tests/models/registry.py
index d2c67cf7e..abc621d8e 100644
--- a/tests/models/registry.py
+++ b/tests/models/registry.py
@@ -713,6 +713,10 @@ _MULTIMODAL_EXAMPLE_MODELS = {
         "baidu/ERNIE-4.5-VL-28B-A3B-PT",
         trust_remote_code=True,
     ),
+    "FunASRForConditionalGeneration": _HfExamplesInfo(
+        "allendou/Fun-ASR-Nano-2512-vllm",
+        is_available_online=False,
+    ),
     "FunAudioChatForConditionalGeneration": _HfExamplesInfo(
         "funaudiochat", is_available_online=False
     ),
diff --git a/vllm/model_executor/models/funasr.py b/vllm/model_executor/models/funasr.py
new file mode 100644
index 000000000..b4d4fb5b7
--- /dev/null
+++ b/vllm/model_executor/models/funasr.py
@@ -0,0 +1,1057 @@
+# SPDX-License-Identifier: Apache-2.0
+# SPDX-FileCopyrightText: Copyright contributors to the vLLM project
+
+import math
+from collections.abc import Iterable, Mapping, Sequence
+from typing import Annotated, Literal, cast
+
+import numpy as np
+import torch
+import torch.nn.functional as F
+from torch import nn
+from transformers import (
+    BatchFeature,
+    Qwen3Config,
+)
+
+from vllm.config import ModelConfig, SpeechToTextConfig, VllmConfig
+from vllm.config.multimodal import BaseDummyOptions
+from vllm.distributed import get_tensor_model_parallel_world_size
+from vllm.inputs.data import PromptType
+from vllm.logger import init_logger
+from vllm.model_executor.layers.activation import _ACTIVATION_REGISTRY
+from vllm.model_executor.layers.attention.mm_encoder_attention import (
+    MMEncoderAttention,
+)
+from vllm.model_executor.layers.linear import (
+    ColumnParallelLinear,
+    QKVParallelLinear,
+    ReplicatedLinear,
+    RowParallelLinear,
+)
+from vllm.model_executor.layers.logits_processor import LogitsProcessor
+from vllm.model_executor.layers.vocab_parallel_embedding import ParallelLMHead
+from vllm.model_executor.model_loader.weight_utils import default_weight_loader
+from vllm.model_executor.models.whisper_utils import (
+    ISO639_1_SUPPORTED_LANGS,
+)
+from vllm.multimodal import MULTIMODAL_REGISTRY
+from vllm.multimodal.inputs import (
+    MultiModalDataDict,
+    MultiModalFieldConfig,
+    MultiModalKwargsItems,
+)
+from vllm.multimodal.parse import MultiModalDataItems, MultiModalDataParser
+from vllm.multimodal.processing import (
+    BaseDummyInputsBuilder,
+    BaseMultiModalProcessor,
+    BaseProcessingInfo,
+    PromptReplacement,
+    PromptUpdate,
+    PromptUpdateDetails,
+)
+from vllm.transformers_utils.processor import cached_processor_from_config
+from vllm.transformers_utils.processors.funasr_processor import FunASRFeatureExtractor
+from vllm.utils.jsontree import json_map_leaves
+from vllm.utils.tensor_schema import TensorSchema, TensorShape
+
+from .interfaces import (
+    MultiModalEmbeddings,
+    SupportsMultiModal,
+    SupportsTranscription,
+    _require_is_multimodal,
+)
+from .qwen3 import Qwen3Model
+from .utils import (
+    AutoWeightsLoader,
+    WeightsMapper,
+    _merge_multimodal_embeddings,
+    maybe_prefix,
+)
+
+logger = init_logger(__name__)
+
+
+def sequence_mask(lengths, maxlen=None, dtype=torch.float32, device=None):
+    if maxlen is None:
+        maxlen = lengths.max()
+    row_vector = torch.arange(0, maxlen, 1).to(lengths.device)
+    matrix = torch.unsqueeze(lengths, dim=-1)
+    mask = row_vector < matrix
+    mask = mask.detach()
+
+    return mask.type(dtype).to(device) if device is not None else mask.type(dtype)
+
+
+class LayerNorm(torch.nn.LayerNorm):
+    def __init__(self, nout, dim=-1):
+        super().__init__(nout, eps=1e-12)
+        self.dim = dim
+
+    def forward(self, x: torch.Tensor):
+        if self.dim == -1:
+            return super().forward(x)
+        return super().forward(x.transpose(self.dim, -1)).transpose(self.dim, -1)
+
+
+class EncoderLayerSANM(nn.Module):
+    def __init__(
+        self,
+        in_size: int,
+        size: int,
+        self_attn: nn.Module,
+        feed_forward: nn.Module,
+        normalize_before=True,
+    ):
+        super().__init__()
+        self.self_attn = self_attn
+        self.feed_forward = feed_forward
+        self.norm1 = LayerNorm(in_size)
+        self.norm2 = LayerNorm(size)
+        self.in_size = in_size
+        self.size = size
+        self.normalize_before = normalize_before
+
+    def forward(
+        self,
+        hidden_states: torch.Tensor,
+        mask: torch.Tensor | None = None,
+        cache=None,
+        mask_shfit_chunk=None,
+        mask_att_chunk_encoder=None,
+    ):
+        residual = hidden_states
+        hidden_states = self.norm1(hidden_states)
+
+        if self.in_size == self.size:
+            hidden_states = residual + self.self_attn(
+                hidden_states,
+                mask,
+                mask_shfit_chunk=mask_shfit_chunk,
+                mask_att_chunk_encoder=mask_att_chunk_encoder,
+            )
+        else:
+            hidden_states = self.self_attn(
+                hidden_states,
+                mask,
+                mask_shfit_chunk=mask_shfit_chunk,
+                mask_att_chunk_encoder=mask_att_chunk_encoder,
+            )
+
+        residual = hidden_states
+        hidden_states = self.norm2(hidden_states)
+        hidden_states = residual + self.feed_forward(hidden_states)
+
+        return hidden_states, mask, cache, mask_shfit_chunk, mask_att_chunk_encoder
+
+
+class MultiHeadedAttentionSANM(nn.Module):
+    def __init__(
+        self,
+        n_head: int,
+        in_feat: int,
+        n_feat: int,
+        kernel_size: int,
+        sanm_shift: int = 0,
+    ):
+        super().__init__()
+        assert n_feat % n_head == 0
+        # We assume d_v always equals d_k
+        self.d_k = n_feat // n_head
+        self.h = n_head
+        self.out_proj = ReplicatedLinear(
+            input_size=n_feat,
+            output_size=n_feat,
+            bias=True,
+        )
+        self.linear_q_k_v = ReplicatedLinear(
+            input_size=in_feat,
+            output_size=n_feat * 3,
+            bias=True,
+        )
+        self.attn = None
+
+        self.fsmn_block = nn.Conv1d(
+            n_feat, n_feat, kernel_size, stride=1, padding=0, groups=n_feat, bias=False
+        )
+        # padding
+        left_padding = (kernel_size - 1) // 2
+        if sanm_shift > 0:
+            left_padding = left_padding + sanm_shift
+        right_padding = kernel_size - 1 - left_padding
+        self.pad_fn = nn.ConstantPad1d((left_padding, right_padding), 0.0)
+
+    def forward_fsmn(
+        self,
+        inputs: torch.Tensor,
+        mask: torch.Tensor,
+        mask_shfit_chunk: torch.Tensor = None,
+    ):
+        b, t, d = inputs.size()
+        if mask is not None:
+            mask = torch.reshape(mask, (b, -1, 1))
+            if mask_shfit_chunk is not None:
+                mask = mask * mask_shfit_chunk
+            inputs = inputs * mask
+
+        x = inputs.transpose(1, 2)
+        x = self.pad_fn(x)
+        x = self.fsmn_block(x)
+        x = x.transpose(1, 2)
+        x += inputs
+        if mask is not None:
+            x = x * mask
+        return x
+
+    def forward_qkv(self, x: torch.Tensor):
+        b, t, d = x.size()
+
+        q_k_v, _ = self.linear_q_k_v(x)
+        q, k, v = torch.split(q_k_v, int(self.h * self.d_k), dim=-1)
+        q_h = torch.reshape(q, (b, t, self.h, self.d_k)).transpose(1, 2)
+        k_h = torch.reshape(k, (b, t, self.h, self.d_k)).transpose(1, 2)
+        v_h = torch.reshape(v, (b, t, self.h, self.d_k)).transpose(1, 2)
+
+        return q_h, k_h, v_h, v
+
+    def forward_attention(
+        self,
+        value: torch.Tensor,
+        scores: torch.Tensor,
+        mask: torch.Tensor,
+        mask_att_chunk_encoder: torch.Tensor = None,
+    ):
+        n_batch = value.size(0)
+        if mask is not None:
+            if mask_att_chunk_encoder is not None:
+                mask = mask * mask_att_chunk_encoder
+
+            mask = mask.unsqueeze(1).eq(0)
+
+            min_value = -float("inf")
+            scores = scores.masked_fill(mask, min_value)
+            attn = torch.softmax(scores, dim=-1).masked_fill(mask, 0.0)
+        else:
+            attn = torch.softmax(scores, dim=-1)
+
+        p_attn = attn
+        x = torch.matmul(p_attn, value)
+        x = x.transpose(1, 2).contiguous().view(n_batch, -1, self.h * self.d_k)
+
+        out, _ = self.out_proj(x)
+        return out
+
+    def forward(
+        self,
+        hidden_states: torch.Tensor,
+        mask: torch.Tensor,
+        mask_shfit_chunk: torch.Tensor = None,
+        mask_att_chunk_encoder: torch.Tensor = None,
+    ):
+        q_h, k_h, v_h, v = self.forward_qkv(hidden_states)
+        fsmn_memory = self.forward_fsmn(v, mask, mask_shfit_chunk)
+        q_h = q_h * self.d_k ** (-0.5)
+        scores = torch.matmul(q_h, k_h.transpose(-2, -1))
+        att_outs = self.forward_attention(v_h, scores, mask, mask_att_chunk_encoder)
+        return att_outs + fsmn_memory
+
+
+class SinusoidalPositionEncoder(torch.nn.Module):
+    def __init__(self, d_model=80):
+        super().__init__()
+
+    def encode(
+        self,
+        positions: torch.Tensor = None,
+        depth: int = None,
+        dtype: torch.dtype = torch.float32,
+    ):
+        batch_size = positions.size(0)
+        positions = positions.type(dtype)
+        device = positions.device
+        log_timescale_increment = torch.log(
+            torch.tensor([10000], dtype=dtype, device=device)
+        ) / (depth / 2 - 1)
+        inv_timescales = torch.exp(
+            torch.arange(depth / 2, device=device).type(dtype)
+            * (-log_timescale_increment)
+        )
+        inv_timescales = torch.reshape(inv_timescales, [batch_size, -1])
+        scaled_time = torch.reshape(positions, [1, -1, 1]) * torch.reshape(
+            inv_timescales, [1, 1, -1]
+        )
+        encoding = torch.cat([torch.sin(scaled_time), torch.cos(scaled_time)], dim=2)
+        return encoding.type(dtype)
+
+    def forward(self, hidden_states: torch.Tensor):
+        batch_size, timesteps, input_dim = hidden_states.size()
+        positions = torch.arange(1, timesteps + 1, device=hidden_states.device)[None, :]
+        position_encoding = self.encode(positions, input_dim, hidden_states.dtype).to(
+            hidden_states.device
+        )
+
+        return hidden_states + position_encoding
+
+
+class SenseVoiceEncoderSmall(nn.Module):
+    def __init__(
+        self,
+        input_size: int,
+        output_size: int = 256,
+        attention_heads: int = 4,
+        linear_units: int = 2048,
+        num_blocks: int = 6,
+        tp_blocks: int = 0,
+        attention_dropout_rate: float = 0.0,
+        normalize_before: bool = True,
+        kernel_size: int = 11,
+        sanm_shift: int = 0,
+        **kwargs,
+    ):
+        super().__init__()
+        self._output_size = output_size
+        self.embed = SinusoidalPositionEncoder()
+
+        self.normalize_before = normalize_before
+
+        positionwise_layer = PositionwiseFeedForward
+        positionwise_layer_args = (
+            output_size,
+            linear_units,
+        )
+
+        encoder_selfattn_layer = MultiHeadedAttentionSANM
+        encoder_selfattn_layer_args0 = (
+            attention_heads,
+            input_size,
+            output_size,
+            kernel_size,
+            sanm_shift,
+        )
+        encoder_selfattn_layer_args = (
+            attention_heads,
+            output_size,
+            output_size,
+            kernel_size,
+            sanm_shift,
+        )
+
+        self.encoders0 = nn.ModuleList(
+            [
+                EncoderLayerSANM(
+                    input_size,
+                    output_size,
+                    encoder_selfattn_layer(*encoder_selfattn_layer_args0),
+                    positionwise_layer(*positionwise_layer_args),
+                )
+                for i in range(1)
+            ]
+        )
+        self.encoders = nn.ModuleList(
+            [
+                EncoderLayerSANM(
+                    output_size,
+                    output_size,
+                    encoder_selfattn_layer(*encoder_selfattn_layer_args),
+                    positionwise_layer(*positionwise_layer_args),
+                )
+                for i in range(num_blocks - 1)
+            ]
+        )
+
+        self.tp_encoders = nn.ModuleList(
+            [
+                EncoderLayerSANM(
+                    output_size,
+                    output_size,
+                    encoder_selfattn_layer(*encoder_selfattn_layer_args),
+                    positionwise_layer(*positionwise_layer_args),
+                )
+                for i in range(tp_blocks)
+            ]
+        )
+
+        self.after_norm = LayerNorm(output_size)
+
+        self.tp_norm = LayerNorm(output_size)
+
+    def output_size(self) -> int:
+        return self._output_size
+
+    def forward(
+        self,
+        xs_pad: torch.Tensor,
+        ilens: torch.Tensor,
+    ):
+        maxlen = xs_pad.shape[1]
+        masks = sequence_mask(
+            ilens, maxlen=maxlen, dtype=ilens.dtype, device=ilens.device
+        )[:, None, :]
+
+        xs_pad *= self.output_size() ** 0.5
+
+        xs_pad = self.embed(xs_pad)
+
+        for layer_idx, encoder_layer in enumerate(self.encoders0):
+            encoder_outs = encoder_layer(xs_pad, masks)
+            xs_pad, masks = encoder_outs[0], encoder_outs[1]
+
+        for layer_idx, encoder_layer in enumerate(self.encoders):
+            encoder_outs = encoder_layer(xs_pad, masks)
+            xs_pad, masks = encoder_outs[0], encoder_outs[1]
+
+        xs_pad = self.after_norm(xs_pad)
+
+        olens = masks.squeeze(1).sum(1).int()
+
+        for layer_idx, encoder_layer in enumerate(self.tp_encoders):
+            encoder_outs = encoder_layer(xs_pad, masks)
+            xs_pad, masks = encoder_outs[0], encoder_outs[1]
+
+        xs_pad = self.tp_norm(xs_pad)
+        return xs_pad, olens
+
+
+class PositionwiseFeedForward(nn.Module):
+    def __init__(self, idim: int, hidden_units: int):
+        super().__init__()
+        self.w_1 = ColumnParallelLinear(
+            input_size=idim,
+            output_size=hidden_units,
+            bias=True,
+        )
+        self.w_2 = RowParallelLinear(
+            input_size=hidden_units,
+            output_size=idim,
+            bias=True,
+        )
+        self.activation = _ACTIVATION_REGISTRY["relu"]
+
+    def forward(self, hidden_states: torch.Tensor):
+        hidden_states, _ = self.w_1(hidden_states)
+        hidden_states = self.activation(hidden_states)
+        hidden_states, _ = self.w_2(hidden_states)
+        return hidden_states
+
+
+class EncoderLayer(nn.Module):
+    def __init__(
+        self,
+        size: int,
+        self_attn: nn.Module,
+        feed_forward: nn.Module,
+    ):
+        super().__init__()
+        self.self_attn = self_attn
+        self.feed_forward = feed_forward
+        self.norm1 = LayerNorm(size)
+        self.norm2 = LayerNorm(size)
+
+    def forward(self, hidden_states: torch.Tensor):
+        residual = hidden_states
+        hidden_states = self.norm1(hidden_states)
+        hidden_states = residual + self.self_attn(hidden_states, None, None)
+        residual = hidden_states
+        hidden_states = self.norm2(hidden_states)
+        hidden_states = residual + self.feed_forward(hidden_states)
+
+        return hidden_states
+
+
+class FunASRAudioAttention(nn.Module):
+    def __init__(
+        self,
+        num_heads: int,
+        embed_dim: int,
+        prefix: str = "",
+    ):
+        super().__init__()
+        self.embed_dim = embed_dim
+        self.num_heads = num_heads
+        self.head_dim = self.embed_dim // self.num_heads
+        tp_size = get_tensor_model_parallel_world_size()
+        self.num_local_heads = self.num_heads // tp_size
+
+        if (self.head_dim * self.num_heads) != self.embed_dim:
+            raise ValueError(
+                f"embed_dim must be divisible by num_heads (got `embed_dim`: "
+                f"{self.embed_dim} and `num_heads`: {self.num_heads})."
+            )
+
+        self.scaling = self.head_dim**-0.5
+
+        self.qkv = QKVParallelLinear(
+            hidden_size=self.embed_dim,
+            head_size=self.head_dim,
+            total_num_heads=self.num_heads,
+            total_num_kv_heads=self.num_heads,
+            bias=True,
+            prefix=f"{prefix}.qkv",
+        )
+
+        self.out_proj = RowParallelLinear(
+            input_size=self.embed_dim,
+            output_size=self.embed_dim,
+            bias=True,
+            prefix=f"{prefix}.out_proj",
+        )
+
+        self.attn = MMEncoderAttention(
+            num_heads=self.num_local_heads,
+            head_size=self.head_dim,
+            scale=self.scaling,
+        )
+
+    def forward(
+        self,
+        hidden_states: torch.Tensor,
+        cu_seqlens: torch.Tensor,
+        max_seqlen: torch.Tensor | None,
+    ) -> torch.Tensor:
+        bs, seq_length, _ = hidden_states.size()
+        qkv, _ = self.qkv(hidden_states)
+        q, k, v = qkv.chunk(3, dim=-1)
+        q = q.view(bs, seq_length, -1, self.head_dim)
+        k = k.view(bs, seq_length, -1, self.head_dim)
+        v = v.view(bs, seq_length, -1, self.head_dim)
+
+        attn_output = self.attn(
+            query=q,
+            key=k,
+            value=v,
+            cu_seqlens=cu_seqlens,
+            max_seqlen=max_seqlen,
+        )
+
+        attn_output = attn_output.view(bs, seq_length, -1)
+        output, _ = self.out_proj(attn_output)
+        return output
+
+
+class Transformer(nn.Module):
+    def __init__(
+        self,
+        downsample_rate=2,
+        encoder_dim=1280,
+        llm_dim=4096,
+        ffn_dim: int = 2048,
+        prefix: str = "",
+        **kwargs,
+    ):
+        super().__init__()
+        self.k = downsample_rate
+        self.encoder_dim = encoder_dim
+        self.llm_dim = llm_dim
+        self.linear1 = ColumnParallelLinear(
+            input_size=self.encoder_dim * self.k,
+            output_size=ffn_dim,
+            bias=True,
+        )
+        self.relu = nn.ReLU()
+        self.linear2 = RowParallelLinear(
+            input_size=ffn_dim,
+            output_size=self.llm_dim,
+            bias=True,
+        )
+
+        self.blocks = None
+        if kwargs.get("n_layer", 2) > 0:
+            self.blocks = nn.ModuleList(
+                [
+                    EncoderLayer(
+                        llm_dim,
+                        FunASRAudioAttention(
+                            kwargs.get("attention_heads", 8),
+                            llm_dim,
+                            prefix=f"{prefix}.self_attn",
+                        ),
+                        PositionwiseFeedForward(
+                            llm_dim,
+                            llm_dim // 4,
+                        ),
+                    )
+                    for _ in range(kwargs.get("n_layer", 2))
+                ]
+            )
+
+    def forward(self, hidden_states: torch.Tensor, ilens: int = 0):
+        batch_size, seq_len, dim = hidden_states.size()
+        chunk_num = (seq_len - 1) // self.k + 1
+        pad_num = chunk_num * self.k - seq_len
+        hidden_states = F.pad(hidden_states, (0, 0, 0, pad_num, 0, 0), value=0.0)
+        seq_len = hidden_states.size(1)
+
+        hidden_states = hidden_states.contiguous()
+        hidden_states = hidden_states.view(batch_size, chunk_num, dim * self.k)
+        hidden_states, _ = self.linear1(hidden_states)
+        hidden_states = self.relu(hidden_states)
+        hidden_states, _ = self.linear2(hidden_states)
+
+        olens = None
+        olens = (ilens - 1) // self.k + 1
+
+        if self.blocks is not None:
+            for layer, block in enumerate(self.blocks):
+                hidden_states = block(hidden_states)
+        return hidden_states, olens
+
+
+class FunASRAudioInputs(TensorSchema):
+    """
+    Dimensions:
+        - b: Batch size
+        - nmb: Number of mel bins
+        - t: Time frames (M)
+    """
+
+    input_features: Annotated[
+        list[torch.Tensor] | None,
+        TensorShape("b", "nmb", "t"),
+    ]
+    speech_lengths: Annotated[
+        list[torch.Tensor] | None,
+        TensorShape("b"),
+    ]
+
+
+class FunASREncoder(nn.Module):
+    def __init__(
+        self, *, vllm_config: VllmConfig, prefix: str = "", init_in_fp32: bool = False
+    ):
+        super().__init__()
+        self.audio_encoder = SenseVoiceEncoderSmall(
+            input_size=560, **vllm_config.model_config.hf_config.audio_encoder_conf
+        )
+        self.audio_adaptor = Transformer(
+            downsample_rate=1,
+            use_low_frame_rate=True,
+            ffn_dim=2048,
+            llm_dim=1024,
+            encoder_dim=512,
+            n_layer=2,
+            freeze=True,
+            prefix=maybe_prefix(prefix, "audio_encoder"),
+        )
+
+    def load_weights(self, weights: Iterable[tuple[str, torch.Tensor]]) -> set[str]:
+        """Load weights with mapping from HuggingFace format."""
+        stacked_params_mapping = [
+            # (param_name, shard_name, shard_id)
+            ("self_attn.qkv.", "self_attn.q_proj.", "q"),
+            ("self_attn.qkv.", "self_attn.k_proj.", "k"),
+            ("self_attn.qkv.", "self_attn.v_proj.", "v"),
+        ]
+        params_dict = dict(self.named_parameters(remove_duplicate=False))
+        loaded_params: set[str] = set()
+
+        for name, loaded_weight in weights:
+            for param_name, weight_name, shard_id in stacked_params_mapping:
+                if weight_name not in name:
+                    continue
+                name = name.replace(weight_name, param_name)
+
+                param = params_dict[name]
+                weight_loader = param.weight_loader
+                weight_loader(param, loaded_weight, shard_id)
+                break
+            else:
+                param = params_dict.get(name)
+                if param is not None:
+                    weight_loader = getattr(
+                        param, "weight_loader", default_weight_loader
+                    )
+                    weight_loader(param, loaded_weight)
+            loaded_params.add(name)
+        return loaded_params
+
+
+class FunASRModel(nn.Module):
+    def __init__(self, *, vllm_config: VllmConfig, prefix: str = ""):
+        super().__init__()
+        self.encoder = FunASREncoder(
+            vllm_config=vllm_config, prefix=maybe_prefix(prefix, "encoder")
+        )
+        self.decoder = Qwen3Model(
+            vllm_config=vllm_config, prefix=maybe_prefix(prefix, "decoder")
+        )
+
+    def forward(
+        self,
+        input_ids: torch.Tensor | None,
+        positions: torch.Tensor,
+        inputs_embeds: torch.Tensor | None = None,
+    ) -> torch.Tensor:
+        decoder_outputs = self.decoder(
+            input_ids=input_ids,
+            positions=positions,
+            inputs_embeds=inputs_embeds,
+        )
+        return decoder_outputs
+
+    def get_encoder_outputs(
+        self,
+        speech: torch.Tensor | list[torch.Tensor] | None,
+        speech_lengths: torch.Tensor | list[torch.Tensor] | None,
+    ) -> torch.Tensor | None:
+        self.feat_permute = False
+
+        if self.feat_permute:
+            encoder_out, encoder_out_lens = self.encoder.audio_encoder(
+                speech.permute(0, 2, 1), speech_lengths
+            )
+        else:
+            encoder_out, encoder_out_lens = self.encoder.audio_encoder(
+                speech, speech_lengths
+            )
+
+        encoder_out, encoder_out_lens = self.encoder.audio_adaptor(
+            encoder_out, encoder_out_lens
+        )
+        return encoder_out
+
+
+class FunASRProcessingInfo(BaseProcessingInfo):
+    def get_hf_config(self) -> Qwen3Config:
+        return self.ctx.get_hf_config(Qwen3Config)
+
+    @property
+    def skip_prompt_length_check(self) -> bool:
+        return True  # Because the encoder prompt is padded
+
+    def get_supported_mm_limits(self) -> Mapping[str, int | None]:
+        return {"audio": 1}
+
+    def get_feature_extractor(self, **kwargs: object) -> FunASRFeatureExtractor:
+        hf_processor = self.get_hf_processor(**kwargs)
+        feature_extractor = hf_processor.feature_extractor  # type: ignore
+        assert isinstance(feature_extractor, FunASRFeatureExtractor)
+        return feature_extractor
+
+    def get_target_channels(self) -> int:
+        return 1
+
+    def get_num_audio_tokens(self) -> int:
+        return self.get_hf_config().max_source_positions
+
+
+class FunASRDummyInputsBuilder(BaseDummyInputsBuilder[FunASRProcessingInfo]):
+    def get_dummy_text(self, mm_counts: Mapping[str, int]) -> str:
+        num_audios = mm_counts.get("audio", 0)
+
+        return "<|AUDIO|>" * num_audios
+
+    def get_dummy_mm_data(
+        self,
+        seq_len: int,
+        mm_counts: Mapping[str, int],
+        mm_options: Mapping[str, BaseDummyOptions] | None = None,
+    ) -> MultiModalDataDict:
+        feature_extractor = self.info.get_feature_extractor()
+
+        sampling_rate = feature_extractor.sampling_rate
+        audio_len = feature_extractor.chunk_length * sampling_rate
+        num_audios = mm_counts.get("audio", 0)
+
+        audio_overrides = mm_options.get("audio") if mm_options else None
+
+        return {
+            "audio": self._get_dummy_audios(
+                length=audio_len, num_audios=num_audios, overrides=audio_overrides
+            )
+        }
+
+
+class FunASRMultiModalProcessor(BaseMultiModalProcessor[FunASRProcessingInfo]):
+    def _get_data_parser(self) -> MultiModalDataParser:
+        feature_extractor = self.info.get_feature_extractor()
+        return MultiModalDataParser(
+            target_sr=feature_extractor.sampling_rate,
+            target_channels=self.info.get_target_channels(),
+        )
+
+    def _call_hf_processor(
+        self,
+        prompt: str,
+        mm_data: Mapping[str, object],
+        mm_kwargs: Mapping[str, object],
+        tok_kwargs: Mapping[str, object],
+    ) -> BatchFeature:
+        if mm_data:
+            feature_extractor = self.info.get_feature_extractor(**mm_kwargs)
+            mm_data = dict(audio=mm_data.pop("audios"))
+            mm_kwargs = dict(
+                **mm_kwargs,
+                sampling_rate=feature_extractor.sampling_rate,
+            )
+        processed_outputs = super()._call_hf_processor(
+            prompt=prompt,
+            mm_data=mm_data,
+            mm_kwargs=mm_kwargs,
+            tok_kwargs=tok_kwargs,
+        )
+        if "labels" in processed_outputs:
+            processed_outputs["input_ids"] = processed_outputs.pop("labels")
+        return processed_outputs
+
+    def _get_mm_fields_config(
+        self,
+        hf_inputs: BatchFeature,
+        hf_processor_mm_kwargs: Mapping[str, object],
+    ) -> Mapping[str, MultiModalFieldConfig]:
+        return dict(
+            input_features=MultiModalFieldConfig.batched("audio"),
+            speech_lengths=MultiModalFieldConfig.batched("audio"),
+            fake_token_len=MultiModalFieldConfig.batched("audio"),
+        )
+
+    def _get_prompt_updates(
+        self,
+        mm_items: MultiModalDataItems,
+        hf_processor_mm_kwargs: Mapping[str, object],
+        out_mm_kwargs: MultiModalKwargsItems,
+    ) -> Sequence[PromptUpdate]:
+        processor = self.info.get_hf_processor(**hf_processor_mm_kwargs)
+        tokenizer = self.info.get_tokenizer()
+        vocab = tokenizer.get_vocab()
+
+        # Use getattr with default to be compatible with transformers<4.48
+        audio_token = getattr(processor, "audio_token", "<|AUDIO|>")
+
+        audio_token_id = vocab[audio_token]
+
+        out_mm_data = out_mm_kwargs.get_data()
+
+        fake_token_len = out_mm_data.get("fake_token_len")
+        if fake_token_len is None:
+            audio_output_lengths = []
+        else:
+            assert isinstance(fake_token_len, torch.Tensor)
+
+            audio_output_lengths = fake_token_len.tolist()
+
+        def get_replacement_qwen2_audio(item_idx: int):
+            if audio_output_lengths:
+                num_features = audio_output_lengths[item_idx]
+            else:
+                audio_embeds = out_mm_data["audio_embeds"][item_idx]
+                assert len(audio_embeds.shape) == 2, "audio_embeds must be a 2D tensor"
+                num_features = audio_embeds.shape[0]
+
+            audio_tokens = [audio_token_id] * num_features
+
+            return PromptUpdateDetails.select_token_id(
+                audio_tokens,
+                embed_token_id=audio_token_id,
+            )
+
+        return [
+            PromptReplacement(
+                modality="audio",
+                target=audio_token,
+                replacement=get_replacement_qwen2_audio,
+            )
+        ]
+
+
+@MULTIMODAL_REGISTRY.register_processor(
+    FunASRMultiModalProcessor,
+    info=FunASRProcessingInfo,
+    dummy_inputs=FunASRDummyInputsBuilder,
+)
+class FunASRForConditionalGeneration(
+    nn.Module, SupportsTranscription, SupportsMultiModal
+):
+    packed_modules_mapping = {
+        "self_attn.qkv_proj": [
+            "self_attn.q_proj",
+            "self_attn.k_proj",
+            "self_attn.v_proj",
+        ],
+        "encoder_attn.kv_proj": ["encoder_attn.k_proj", "encoder_attn.v_proj"],
+    }
+
+    hf_to_vllm_mapper = WeightsMapper(
+        orig_to_new_substr={
+            "linear_q.": "q_proj.",
+            "linear_k.": "k_proj.",
+            "linear_v.": "v_proj.",
+            "linear_out.": "out_proj.",
+        }
+    )
+
+    supports_transcription_only = True
+    supports_segment_timestamp = True
+    supported_languages = ISO639_1_SUPPORTED_LANGS
+
+    @classmethod
+    def validate_language(cls, language: str | None) -> str | None:
+        if language is None:
+            # TODO language should be optional and can be guessed.
+            # For now we default to en. See
+            # https://github.com/huggingface/transformers/blob/main/src/transformers/models/whisper/generation_whisper.py#L1520
+            logger.warning(
+                "Defaulting to language='en'. If you wish to transcribe "
+                "audio in a different language, pass the `language` field "
+                "in the TranscriptionRequest."
+            )
+            language = "en"
+        return super().validate_language(language)
+
+    @classmethod
+    def get_generation_prompt(
+        cls,
+        audio: np.ndarray,
+        model_config: ModelConfig,  # not needed here
+        stt_config: SpeechToTextConfig,
+        language: str | None,
+        task_type: Literal["transcribe", "translate"],
+        request_prompt: str,
+        to_language: str | None,
+    ) -> PromptType:
+        if language is None:
+            raise ValueError(
+                "Language must be specified when creating the funasr prompt"
+            )
+
+        funasr_prompt = "<|im_start|>system\nYou are a helpful assistant.<|im_end|>\n<|im_start|>user\n语音转写：<|AUDIO|><|im_end|>\n<|im_start|>assistant\n"  # noqa: E501
+        prompt = {
+            "prompt": funasr_prompt,
+            "multi_modal_data": {
+                "audio": (audio, stt_config.sample_rate),
+            },
+        }
+        return cast(PromptType, prompt)
+
+    @classmethod
+    def get_speech_to_text_config(
+        cls, model_config: ModelConfig, task_type: str
+    ) -> SpeechToTextConfig:
+        processor = cached_processor_from_config(model_config)
+
+        return SpeechToTextConfig(
+            max_audio_clip_s=processor.feature_extractor.chunk_length,
+            sample_rate=processor.feature_extractor.sampling_rate,
+        )
+
+    @classmethod
+    def get_num_audio_tokens(
+        cls,
+        audio_duration_s: float,
+        stt_config: SpeechToTextConfig,
+        model_config: ModelConfig,
+    ) -> int | None:
+        processor = cached_processor_from_config(model_config)
+        hop_length = processor.feature_extractor.hop_length
+        assert hop_length is not None
+        return math.ceil(audio_duration_s * stt_config.sample_rate / hop_length)
+
+    def __init__(self, *, vllm_config: VllmConfig, prefix: str = ""):
+        super().__init__()
+        config = vllm_config.model_config.hf_config
+        quant_config = vllm_config.quant_config
+        self.config = config
+        self.dtype = vllm_config.model_config.dtype
+
+        self.model = FunASRModel(
+            vllm_config=vllm_config,
+            prefix=maybe_prefix(prefix, "model"),
+        )
+        logit_scale = getattr(config, "logit_scale", 1.0)
+
+        if config.tie_word_embeddings:
+            self.lm_head = self.model.decoder.embed_tokens
+        else:
+            self.lm_head = ParallelLMHead(
+                config.vocab_size,
+                config.hidden_size,
+                quant_config=quant_config,
+                prefix=maybe_prefix(prefix, "lm_head"),
+            )
+        self.logits_processor = LogitsProcessor(config.vocab_size, scale=logit_scale)
+
+    def forward(
+        self,
+        input_ids: torch.Tensor,
+        positions: torch.Tensor,
+        inputs_embeds: torch.Tensor | None = None,
+        **kwargs,
+    ) -> torch.Tensor:
+        decoder_outputs = self.model(
+            input_ids=input_ids,
+            positions=positions,
+            inputs_embeds=inputs_embeds,
+        )
+        return decoder_outputs
+
+    def get_language_model(self) -> torch.nn.Module:
+        return self.model.decoder
+
+    def embed_multimodal(self, **kwargs: object) -> MultiModalEmbeddings:
+        audio_input = self._parse_and_validate_audio_input(**kwargs)
+
+        speech = audio_input["input_features"]
+        speech_lengths = audio_input["speech_lengths"]
+        enc_output = self.model.get_encoder_outputs(
+            speech=speech, speech_lengths=speech_lengths
+        )
+
+        return enc_output
+
+    def embed_input_ids(
+        self,
+        input_ids: torch.Tensor,
+        multimodal_embeddings: MultiModalEmbeddings | None = None,
+        *,
+        is_multimodal: torch.Tensor | None = None,
+        handle_oov_mm_token: bool = False,
+    ) -> torch.Tensor:
+        inputs_embeds = self.model.decoder.embed_input_ids(input_ids)
+
+        return _merge_multimodal_embeddings(
+            inputs_embeds=inputs_embeds,
+            multimodal_embeddings=multimodal_embeddings,
+            is_multimodal=_require_is_multimodal(is_multimodal),
+        )
+
+    def _parse_and_validate_audio_input(self, **kwargs: object) -> FunASRAudioInputs:
+        input_features = kwargs.pop("input_features", None)
+        speech_lengths = kwargs.pop("speech_lengths", None)
+
+        if input_features is not None:
+            input_features = json_map_leaves(lambda x: x.to(self.dtype), input_features)
+
+        if speech_lengths is not None:
+            speech_lengths = json_map_leaves(lambda x: x.to(self.dtype), speech_lengths)
+
+        return FunASRAudioInputs(
+            input_features=input_features, speech_lengths=speech_lengths
+        )
+
+    def compute_logits(self, hidden_states: torch.Tensor) -> torch.Tensor:
+        logits = self.logits_processor(self.lm_head, hidden_states)
+        return logits
+
+    def load_weights(self, weights: Iterable[tuple[str, torch.Tensor]]) -> set[str]:
+        loader = AutoWeightsLoader(
+            self,
+        )
+
+        # add fake zeros bias for k_proj to state_dict
+        weights = _create_fake_bias_for_k_proj(weights)
+        return loader.load_weights(weights, mapper=self.hf_to_vllm_mapper)
+
+
+def _create_fake_bias_for_k_proj(
+    weights: Iterable[tuple[str, torch.Tensor]],
+) -> Iterable[tuple[str, torch.Tensor]]:
+    """
+    Create full zeros bias for k_proj weight in self-attn and x-attn layers.
+    So that the bias for k_proj in qkv_proj can be initialized with zeros.
+    """
+    for name, weight in weights:
+        if name.endswith(".k_proj.weight"):
+            bias = torch.zeros(weight.size(0))
+            bias_name = name.replace("weight", "bias")
+            yield from [(name, weight), (bias_name, bias)]
+        else:
+            yield name, weight
diff --git a/vllm/model_executor/models/registry.py b/vllm/model_executor/models/registry.py
index 1871591c9..59fcd9117 100644
--- a/vllm/model_executor/models/registry.py
+++ b/vllm/model_executor/models/registry.py
@@ -325,6 +325,7 @@ _MULTIMODAL_MODELS = {
         "ernie45_vl",
         "Ernie4_5_VLMoeForConditionalGeneration",
     ),
+    "FunASRForConditionalGeneration": ("funasr", "FunASRForConditionalGeneration"),  # noqa: E501
     "FunAudioChatForConditionalGeneration": (
         "funaudiochat",
         "FunAudioChatForConditionalGeneration",
diff --git a/vllm/transformers_utils/processors/__init__.py b/vllm/transformers_utils/processors/__init__.py
index af25dbe4c..d726fd39a 100644
--- a/vllm/transformers_utils/processors/__init__.py
+++ b/vllm/transformers_utils/processors/__init__.py
@@ -10,6 +10,7 @@ reasons:
 
 from vllm.transformers_utils.processors.bagel import BagelProcessor
 from vllm.transformers_utils.processors.deepseek_vl2 import DeepseekVLV2Processor
+from vllm.transformers_utils.processors.funasr_processor import FunASRProcessor
 from vllm.transformers_utils.processors.hunyuan_vl import HunYuanVLProcessor
 from vllm.transformers_utils.processors.hunyuan_vl_image import HunYuanVLImageProcessor
 from vllm.transformers_utils.processors.ovis import OvisProcessor
@@ -18,6 +19,7 @@ from vllm.transformers_utils.processors.ovis2_5 import Ovis2_5Processor
 __all__ = [
     "BagelProcessor",
     "DeepseekVLV2Processor",
+    "FunASRProcessor",
     "HunYuanVLProcessor",
     "HunYuanVLImageProcessor",
     "OvisProcessor",
diff --git a/vllm/transformers_utils/processors/funasr_processor.py b/vllm/transformers_utils/processors/funasr_processor.py
new file mode 100644
index 000000000..4807c87d3
--- /dev/null
+++ b/vllm/transformers_utils/processors/funasr_processor.py
@@ -0,0 +1,504 @@
+# SPDX-License-Identifier: Apache-2.0
+# SPDX-FileCopyrightText: Copyright contributors to the vLLM project
+
+
+import numpy as np
+import torch
+import torch.nn as nn
+import torchaudio.compliance.kaldi as kaldi
+from torch.nn.utils.rnn import pad_sequence
+from transformers import (
+    AutoFeatureExtractor,
+    AutoProcessor,
+    BatchFeature,
+)
+from transformers.feature_extraction_sequence_utils import SequenceFeatureExtractor
+from transformers.processing_utils import ProcessorMixin
+from transformers.utils import TensorType
+
+from vllm.logger import init_logger
+
+logger = init_logger(__name__)
+
+
+def apply_cmvn(inputs, cmvn):  # noqa
+    """
+    Apply CMVN with mvn data
+    """
+
+    device = inputs.device
+    # dtype = inputs.dtype
+    frame, dim = inputs.shape
+
+    means = cmvn[0:1, :dim]
+    vars = cmvn[1:2, :dim]
+    inputs += means.to(device)
+    inputs *= vars.to(device)
+
+    return inputs.type(torch.float32)
+
+
+def apply_lfr(inputs, lfr_m, lfr_n):
+    # LFR_inputs = []
+    T = inputs.shape[0]
+    T_lfr = int(np.ceil(T / lfr_n))
+    left_padding = inputs[0].repeat((lfr_m - 1) // 2, 1)
+    inputs = torch.vstack((left_padding, inputs))
+    T = T + (lfr_m - 1) // 2
+    feat_dim = inputs.shape[-1]
+    strides = (lfr_n * feat_dim, 1)
+    sizes = (T_lfr, lfr_m * feat_dim)
+    last_idx = (T - lfr_m) // lfr_n + 1
+    num_padding = lfr_m - (T - last_idx * lfr_n)
+    if num_padding > 0:
+        num_padding = (
+            (2 * lfr_m - 2 * T + (T_lfr - 1 + last_idx) * lfr_n)
+            / 2
+            * (T_lfr - last_idx)
+        )
+        inputs = torch.vstack([inputs] + [inputs[-1:]] * int(num_padding))
+    LFR_outputs = inputs.as_strided(sizes, strides)
+    return LFR_outputs.clone().type(torch.float32)
+
+
+def load_cmvn(cmvn_file):
+    with open(cmvn_file, encoding="utf-8") as f:
+        lines = f.readlines()
+    means_list = []
+    vars_list = []
+    for i in range(len(lines)):
+        line_item = lines[i].split()
+        if line_item[0] == "<AddShift>":
+            line_item = lines[i + 1].split()
+            if line_item[0] == "<LearnRateCoef>":
+                add_shift_line = line_item[3 : (len(line_item) - 1)]
+                means_list = list(add_shift_line)
+                continue
+        elif line_item[0] == "<Rescale>":
+            line_item = lines[i + 1].split()
+            if line_item[0] == "<LearnRateCoef>":
+                rescale_line = line_item[3 : (len(line_item) - 1)]
+                vars_list = list(rescale_line)
+                continue
+    means = np.array(means_list).astype(np.float32)
+    vars = np.array(vars_list).astype(np.float32)
+    cmvn = np.array([means, vars])
+    cmvn = torch.as_tensor(cmvn, dtype=torch.float32)
+    return cmvn
+
+
+class WavFrontend(nn.Module):
+    """Conventional frontend structure for ASR."""
+
+    def __init__(
+        self,
+        cmvn_file: str = "null",
+        fs: int = 16000,
+        window: str = "hamming",
+        n_mels: int = 80,
+        frame_length: int = 25,
+        frame_shift: int = 10,
+        filter_length_min: int = -1,
+        filter_length_max: int = -1,
+        lfr_m: int = 1,
+        lfr_n: int = 1,
+        dither: float = 1.0,
+        snip_edges: bool = True,
+        upsacle_samples: bool = True,
+        **kwargs,
+    ):
+        super().__init__()
+        self.fs = fs
+        self.window = window
+        self.n_mels = n_mels
+        self.frame_length = frame_length
+        self.frame_shift = frame_shift
+        self.filter_length_min = filter_length_min
+        self.filter_length_max = filter_length_max
+        self.lfr_m = lfr_m
+        self.lfr_n = lfr_n
+        self.cmvn_file = cmvn_file
+        self.dither = dither
+        self.snip_edges = snip_edges
+        self.upsacle_samples = upsacle_samples
+        self.cmvn = None if self.cmvn_file is None else load_cmvn(self.cmvn_file)
+
+    def output_size(self) -> int:
+        return self.n_mels * self.lfr_m
+
+    def forward(
+        self,
+        input: torch.Tensor,
+        input_lengths,
+        **kwargs,
+    ) -> tuple[torch.Tensor, torch.Tensor]:
+        batch_size = input.size(0)
+        feats = []
+        feats_lens = []
+        for i in range(batch_size):
+            waveform_length = input_lengths[i]
+            waveform = input[i][:waveform_length]
+            if self.upsacle_samples:
+                waveform = waveform * (1 << 15)
+            waveform = waveform.unsqueeze(0)
+            mat = kaldi.fbank(
+                waveform,
+                num_mel_bins=self.n_mels,
+                frame_length=min(self.frame_length, waveform_length / self.fs * 1000),
+                frame_shift=self.frame_shift,
+                dither=self.dither,
+                energy_floor=0.0,
+                window_type=self.window,
+                sample_frequency=self.fs,
+                snip_edges=self.snip_edges,
+            )
+
+            if self.lfr_m != 1 or self.lfr_n != 1:
+                mat = apply_lfr(mat, self.lfr_m, self.lfr_n)
+            if self.cmvn is not None:
+                mat = apply_cmvn(mat, self.cmvn)
+            feat_length = mat.size(0)
+            feats.append(mat)
+            feats_lens.append(feat_length)
+
+        feats_lens = torch.as_tensor(feats_lens)
+        if batch_size == 1:
+            feats_pad = feats[0][None, :, :]
+        else:
+            feats_pad = pad_sequence(feats, batch_first=True, padding_value=0.0)
+        return feats_pad, feats_lens
+
+    def forward_fbank(
+        self, input: torch.Tensor, input_lengths: torch.Tensor
+    ) -> tuple[torch.Tensor, torch.Tensor]:
+        batch_size = input.size(0)
+        feats = []
+        feats_lens = []
+        for i in range(batch_size):
+            waveform_length = input_lengths[i]
+            waveform = input[i][:waveform_length]
+            waveform = waveform * (1 << 15)
+            waveform = waveform.unsqueeze(0)
+            mat = kaldi.fbank(
+                waveform,
+                num_mel_bins=self.n_mels,
+                frame_length=self.frame_length,
+                frame_shift=self.frame_shift,
+                dither=self.dither,
+                energy_floor=0.0,
+                window_type=self.window,
+                sample_frequency=self.fs,
+            )
+
+            feat_length = mat.size(0)
+            feats.append(mat)
+            feats_lens.append(feat_length)
+
+        feats_lens = torch.as_tensor(feats_lens)
+        feats_pad = pad_sequence(feats, batch_first=True, padding_value=0.0)
+        return feats_pad, feats_lens
+
+    def forward_lfr_cmvn(
+        self, input: torch.Tensor, input_lengths: torch.Tensor
+    ) -> tuple[torch.Tensor, torch.Tensor]:
+        batch_size = input.size(0)
+        feats = []
+        feats_lens = []
+        for i in range(batch_size):
+            mat = input[i, : input_lengths[i], :]
+            if self.lfr_m != 1 or self.lfr_n != 1:
+                mat = apply_lfr(mat, self.lfr_m, self.lfr_n)
+            if self.cmvn is not None:
+                mat = apply_cmvn(mat, self.cmvn)
+            feat_length = mat.size(0)
+            feats.append(mat)
+            feats_lens.append(feat_length)
+
+        feats_lens = torch.as_tensor(feats_lens)
+        feats_pad = pad_sequence(feats, batch_first=True, padding_value=0.0)
+        return feats_pad, feats_lens
+
+
+class FunASRFeatureExtractor(SequenceFeatureExtractor):
+    r"""
+    Constructs a FunASR feature extractor.
+
+    This feature extractor inherits from [`~feature_extraction_sequence_
+        utils.SequenceFeatureExtractor`] which contains most of the main
+        methods. Users should refer to this superclass for more information
+        regarding those methods.
+
+    This class extracts mel-filter bank features from raw speech using a custom
+    numpy implementation of the `Short Time Fourier Transform` which should
+    match pytorch's `torch.stft` equivalent.
+
+    Args:
+        feature_size (`int`, *optional*, defaults to 80):
+            The feature dimension of the extracted features.
+        sampling_rate (`int`, *optional*, defaults to 16000):
+            The sampling rate at which the audio files should be digitalized
+            expressed in hertz (Hz).
+        hop_length (`int`, *optional*, defaults to 160):
+            Length of the overlapping windows for the STFT used to obtain the
+            Mel Frequency coefficients.
+        chunk_length (`int`, *optional*, defaults to 30):
+            The maximum number of chunks of `sampling_rate` samples used to
+            trim and pad longer or shorter audio sequences.
+        n_fft (`int`, *optional*, defaults to 400):
+            Size of the Fourier transform.
+        padding_value (`float`, *optional*, defaults to 0.0):
+            Padding value used to pad the audio. Should correspond to silences.
+        dither (`float`, *optional*, defaults to 0.0):
+            Adds dithering. In other words, adds a small Gaussian noise to each frame.
+            E.g. use 0.0001 to add dithering with a normal distribution centered
+            around 0.0 with standard deviation 0.0001 (assuming [-1,+1] range
+            of raw_speech). The value 0.0 means no dithering.
+            Dithering has similar effect as `spectrogram(mel_floor=...)`. It reduces
+            the high log_mel_fbank values for signals with hard-zero sections,
+            when VAD cutoff is present in the signal.
+    """
+
+    model_input_names = ["input_features"]
+
+    def __init__(
+        self,
+        feature_size=80,
+        sampling_rate=16000,
+        hop_length=160,
+        chunk_length=30,
+        n_fft=400,
+        padding_value=0.0,
+        dither=0.0,
+        return_attention_mask=False,
+        **kwargs,
+    ):
+        super().__init__(
+            feature_size=feature_size,
+            sampling_rate=sampling_rate,
+            padding_value=padding_value,
+            return_attention_mask=return_attention_mask,
+            **kwargs,
+        )
+        self.frontend_conf = kwargs.get("frontend_conf", {})
+        self.n_fft = n_fft
+        self.hop_length = hop_length
+        self.chunk_length = chunk_length
+        self.n_samples = chunk_length * sampling_rate
+        self.nb_max_frames = self.n_samples // hop_length
+        self.sampling_rate = sampling_rate
+        self.dither = dither
+
+    def extract_fbank(
+        self, data, data_len=None, data_type: str = "sound", frontend=None, **kwargs
+    ):
+        if isinstance(data, np.ndarray):
+            data = torch.from_numpy(data)
+            if len(data.shape) < 2:
+                data = data[None, :]  # data: [batch, N]
+            data_len = [data.shape[1]] if data_len is None else data_len
+        elif isinstance(data, torch.Tensor):
+            if len(data.shape) < 2:
+                data = data[None, :]  # data: [batch, N]
+            data_len = [data.shape[1]] if data_len is None else data_len
+        elif isinstance(data, (list, tuple)):
+            data_list, data_len = [], []
+            for data_i in data:
+                if isinstance(data_i, np.ndarray):
+                    data_i = torch.from_numpy(data_i)
+                data_list.append(data_i)
+                data_len.append(data_i.shape[0])
+            data = pad_sequence(data_list, batch_first=True)
+
+        data, data_len = frontend(data, data_len, **kwargs)
+
+        if isinstance(data_len, (list, tuple)):
+            data_len = torch.tensor([data_len])
+        return data.to(torch.float32), data_len.to(torch.int32)
+
+    def __call__(
+        self,
+        raw_speech: np.ndarray | list[float] | list[np.ndarray] | list[list[float]],
+        truncation: bool = True,
+        pad_to_multiple_of: int | None = None,
+        return_tensors: str | TensorType | None = None,
+        return_attention_mask: bool | None = None,
+        padding: str | None = "max_length",
+        max_length: int | None = None,
+        sampling_rate: int | None = None,
+        do_normalize: bool | None = None,
+        device: str | None = "cpu",
+        return_token_timestamps: bool | None = None,
+        **kwargs,
+    ) -> BatchFeature:
+        is_batched = isinstance(raw_speech, (list, tuple)) and (
+            isinstance(raw_speech[0], (np.ndarray, tuple, list))
+        )
+
+        if is_batched:
+            raw_speech = [
+                np.asarray([speech], dtype=np.float32).T for speech in raw_speech
+            ]
+        elif not is_batched and not isinstance(raw_speech, np.ndarray):
+            raw_speech = np.asarray(raw_speech, dtype=np.float32)
+        elif isinstance(raw_speech, np.ndarray) and raw_speech.dtype is np.dtype(
+            np.float64
+        ):
+            raw_speech = raw_speech.astype(np.float32)
+
+        if not is_batched:
+            raw_speech = [np.asarray([raw_speech]).T]
+
+        batched_speech = BatchFeature({"input_features": raw_speech})
+
+        padded_inputs = self.pad(
+            batched_speech,
+            padding=padding,
+            max_length=max_length if max_length else self.n_samples,
+            truncation=truncation,
+            pad_to_multiple_of=pad_to_multiple_of,
+            return_attention_mask=return_attention_mask or do_normalize,
+        )
+
+        input_features = padded_inputs.get("input_features").transpose(2, 0, 1)
+
+        self.frontend = WavFrontend(**self.frontend_conf)
+        input_features, speech_lengths = self.extract_fbank(
+            input_features[0],
+            data_type=kwargs.get("data_type", "sound"),
+            frontend=self.frontend,
+            is_final=True,
+        )
+        olens = 1 + (speech_lengths - 3 + 2 * 1) // 2
+        olens = 1 + (olens - 3 + 2 * 1) // 2
+        fake_token_len = (olens - 1) // 2 + 1
+        if isinstance(input_features[0], list):
+            padded_inputs["input_features"] = [
+                np.asarray(feature, dtype=np.float32) for feature in input_features
+            ]
+
+        else:
+            padded_inputs["input_features"] = input_features
+
+        if return_tensors is not None:
+            padded_inputs = padded_inputs.convert_to_tensors(return_tensors)
+
+        padded_inputs["speech_lengths"] = speech_lengths
+        padded_inputs["fake_token_len"] = fake_token_len
+
+        return padded_inputs
+
+
+class FunASRProcessor(ProcessorMixin):
+    r"""
+    Constructs a FunASR processor which wraps a FunASR feature extractor and
+    a FunASR tokenizer into a single processor.
+
+    [`FunASRProcessor`] offers all the functionalities of
+    [`FunASRFeatureExtractor`] and [`Qwen2Tokenizer`]. See the
+    [`~FunASRProcessor.__call__`] and [`~FunASRProcessor.decode`] for more
+    information.
+
+    Args:
+        feature_extractor (`FunASRFeatureExtractor`): An instance of
+            [`FunASRFeatureExtractor`].
+            The feature extractor is a required input.
+        tokenizer (`Qwen2Tokenizer`):
+            An instance of [`Qwen2Tokenizer`]. The tokenizer is a required
+            input.
+    """
+
+    feature_extractor_class = "FunASRFeatureExtractor"
+    tokenizer_class = ("Qwen2Tokenizer", "Qwen2TokenizerFast")
+
+    def __init__(
+        self,
+        feature_extractor,
+        tokenizer,
+        audio_token="<|AUDIO|>",
+    ):
+        super().__init__(feature_extractor, tokenizer)
+        self.current_processor = self.feature_extractor
+        self._in_target_context_manager = False
+        self.audio_token = (
+            tokenizer.audio_token if hasattr(tokenizer, "audio_token") else audio_token
+        )
+        self.audio_token_id = tokenizer.convert_tokens_to_ids(self.audio_token)
+
+    def get_decoder_prompt_ids(self, task=None, language=None, no_timestamps=True):
+        return self.tokenizer.get_decoder_prompt_ids(
+            task=task, language=language, no_timestamps=no_timestamps
+        )
+
+    def __call__(self, *args, **kwargs):
+        """
+        Forwards the `audio` argument to FunASRFeatureExtractor's
+        [`~FunASRFeatureExtractor.__call__`] and the `text` argument to
+        [`~Qwen2Tokenizer.__call__`]. Please refer to the docstring of the
+        above two methods for more information.
+        """
+        if self._in_target_context_manager:
+            return self.current_processor(*args, **kwargs)
+
+        audio = kwargs.pop("audio", None)
+        sampling_rate = kwargs.pop("sampling_rate", None)
+        text = kwargs.pop("text", None)
+        if len(args) > 0:
+            audio = args[0]
+            args = args[1:]
+
+        if text is None:
+            raise ValueError("You need to specify `text` input to process.")
+        elif isinstance(text, str):
+            text = [text]
+        elif not isinstance(text, list) and not isinstance(text[0], str):
+            raise ValueError(
+                "Invalid input text. Please provide a string, or a list of strings"
+            )
+
+        if audio is not None:
+            # ensure we have as much audios as audio tokens
+            num_audio_tokens = sum(sample.count(self.audio_token) for sample in text)
+            num_audios = 1 if type(audio) is np.ndarray else len(audio)
+            if num_audio_tokens != num_audios:
+                raise ValueError(
+                    f"Found {num_audio_tokens} {self.audio_token} token{'s' if num_audio_tokens > 1 else ''} in provided text but received {num_audios} audio{'s' if num_audios > 1 else ''}"  # noqa: E501
+                )
+            inputs = self.feature_extractor(
+                audio, *args, sampling_rate=sampling_rate, **kwargs
+            )
+
+            expanded_text = []
+            for sample in text:
+                replace_str = []
+                while self.audio_token in sample:
+                    num_audio_tokens = inputs["fake_token_len"].item()
+
+                    expanded_audio_token = self.audio_token * num_audio_tokens
+
+                    replace_str.append(expanded_audio_token)
+                    sample = sample.replace(self.audio_token, "<placeholder>", 1)
+
+                while "<placeholder>" in sample:
+                    sample = sample.replace("<placeholder>", replace_str.pop(0), 1)
+                expanded_text.append(sample)
+            text = expanded_text
+
+        if text is not None:
+            encodings = self.tokenizer(text, **kwargs)
+
+        if text is None:
+            return inputs
+
+        elif audio is None:
+            return encodings
+        else:
+            inputs["labels"] = encodings["input_ids"]
+
+            return inputs
+
+    def get_prompt_ids(self, text: str, return_tensors="np"):
+        return self.tokenizer.get_prompt_ids(text, return_tensors=return_tensors)
+
+
+AutoFeatureExtractor.register("FunASRFeatureExtractor", FunASRFeatureExtractor)
+AutoProcessor.register("FunASRProcessor", FunASRProcessor)