diff --git a/python/sglang/srt/configs/model_config.py b/python/sglang/srt/configs/model_config.py
index 15dd56361..2356270c0 100644
--- a/python/sglang/srt/configs/model_config.py
+++ b/python/sglang/srt/configs/model_config.py
@@ -1548,6 +1548,7 @@ multimodal_model_archs = [
     "Lfm2VlForConditionalGeneration",
     "LightOnOCRForConditionalGeneration",
     *MIMO_V2_MULTIMODAL_ARCHS,
+    "MiMoV2ASRForCausalLM",
     "MiniCPMO",
     "MiniCPMV",
     "Mistral3ForConditionalGeneration",
diff --git a/python/sglang/srt/entrypoints/openai/transcription_adapters/__init__.py b/python/sglang/srt/entrypoints/openai/transcription_adapters/__init__.py
index 353196fdd..bd8779133 100644
--- a/python/sglang/srt/entrypoints/openai/transcription_adapters/__init__.py
+++ b/python/sglang/srt/entrypoints/openai/transcription_adapters/__init__.py
@@ -7,6 +7,9 @@ from sglang.srt.entrypoints.openai.transcription_adapters.base import (  # noqa:
 )
 
 # Import built-in adapters so they self-register via @register_transcription_adapter.
+from sglang.srt.entrypoints.openai.transcription_adapters.mimo_v2_asr import (  # noqa: F401
+    MiMoV2ASRAdapter,
+)
 from sglang.srt.entrypoints.openai.transcription_adapters.qwen3_asr import (  # noqa: F401
     Qwen3ASRAdapter,
 )
@@ -20,4 +23,5 @@ __all__ = [
     "resolve_adapter",
     "WhisperAdapter",
     "Qwen3ASRAdapter",
+    "MiMoV2ASRAdapter",
 ]
diff --git a/python/sglang/srt/entrypoints/openai/transcription_adapters/mimo_v2_asr.py b/python/sglang/srt/entrypoints/openai/transcription_adapters/mimo_v2_asr.py
new file mode 100644
index 000000000..d074e3c04
--- /dev/null
+++ b/python/sglang/srt/entrypoints/openai/transcription_adapters/mimo_v2_asr.py
@@ -0,0 +1,46 @@
+from __future__ import annotations
+
+from sglang.srt.entrypoints.openai.protocol import (
+    TranscriptionRequest,
+    TranscriptionUsage,
+    TranscriptionVerboseResponse,
+)
+from sglang.srt.entrypoints.openai.transcription_adapters.base import (
+    TranscriptionAdapter,
+    register_transcription_adapter,
+)
+
+
+@register_transcription_adapter("MiMoV2ASR")
+class MiMoV2ASRAdapter(TranscriptionAdapter):
+    """Adapter for MiMo-V2-ASR.
+
+    The multimodal processor (``MiMoV2ASRProcessor``) prepends the audio
+    placeholder ``<|sosp|><|empty|>...<|eosp|>`` when ``input_text`` lacks
+    one, so the request text can stay empty and the adapter only has to
+    supply sampling params and the verbose-response shape.
+    """
+
+    def build_sampling_params(self, request: TranscriptionRequest) -> dict:
+        return {
+            "temperature": request.temperature,
+            "max_new_tokens": 448,
+        }
+
+    def build_verbose_response(
+        self,
+        request: TranscriptionRequest,
+        text: str,
+        ret: dict,
+        tokenizer,
+        usage: TranscriptionUsage,
+    ) -> TranscriptionVerboseResponse:
+        # MiMo-V2-ASR does not emit timestamp tokens; segments stay empty
+        # until a forced-aligner path is added.
+        return TranscriptionVerboseResponse(
+            language=request.language or "auto",
+            duration=round(request.audio_duration_s, 2),
+            text=text,
+            segments=[],
+            usage=usage,
+        )
diff --git a/python/sglang/srt/models/mimo_audio.py b/python/sglang/srt/models/mimo_audio.py
index a90547920..24b18faac 100644
--- a/python/sglang/srt/models/mimo_audio.py
+++ b/python/sglang/srt/models/mimo_audio.py
@@ -8,7 +8,7 @@ import os
 import typing as tp
 from dataclasses import dataclass
 from functools import wraps
-from typing import List, Optional, Tuple
+from typing import Optional, Tuple
 
 import torch
 import torch.nn as nn
@@ -1141,72 +1141,89 @@ class MiMoV2AudioConfig:
         return config
 
 
-class MiMoAudioEncoder(nn.Module):
-    config: MiMoAudioEncoderConfig
+class AudioEncoderMixin:
+    """LM model mixin that adds MiMo audio encoder components.
 
-    def __init__(self, config):
-        super().__init__()
+    Components are attached as top-level attributes (no ``audio_encoder.``
+    prefix), matching the checkpoint state_dict layout. Inner naming
+    variations are normalized via ``AUDIO_WEIGHT_REMAP``.
+
+    Hot config fields are cached as direct ``self.audio_*`` attributes at
+    build time so helper methods can stay short and uniform (no
+    ``self.audio_config.foo`` indirection inside hot paths).
+
+    Subclasses call ``self.build_audio_encoder(audio_config)`` from their
+    ``__init__`` after the language model is constructed; the mixin's
+    ``get_audio_feature`` then handles audio item batching end-to-end.
+    """
+
+    AUDIO_WEIGHT_REMAP: tuple[tuple[str, str], ...] = (
+        ("audio_projection", "projection"),
+        ("speech_group_downcast", "projection"),
+        ("audio_input_local_transformer", "input_local_transformer"),
+    )
+
+    def build_audio_encoder(self, config) -> None:
         if not isinstance(config, MiMoV2AudioConfig):
-            config_dict = (
-                vars(config) if hasattr(config, "__dict__") else config.__dict__
-            )
-            config = MiMoV2AudioConfig(**config_dict)
-        self.config = config
-        self.server_args = get_global_server_args()
-        self.use_data_parallel = get_global_server_args().mm_enable_dp_encoder
-        self.speech_empty_ids = self.parsed_speech_empty_ids()
+            cfg_dict = vars(config) if hasattr(config, "__dict__") else config.__dict__
+            config = MiMoV2AudioConfig(**cfg_dict)
+
         self.audio_channels = config.audio_channels
         self.audio_group_size = config.group_size
         self.audio_segment_size = config.audio_segment_size
-        speech_vocab_size = self._parse_maybe_list(
-            self.config.speech_vocab_size, self.config.audio_channels
+        self.audio_input_local_dim = config.input_local_dim
+        self.audio_input_full_attention = config.input_full_attention
+        self.audio_out_hidden_size = config.out_hidden_size
+
+        speech_vocab_size = config._parse_maybe_list(
+            config.speech_vocab_size, self.audio_channels
         )
+        speech_empty_ids = config._parse_maybe_list(
+            config.speech_zeroemb_idx, self.audio_channels
+        )
+
         input_local_config = Qwen2Config(
-            hidden_size=self.config.input_local_dim,
-            num_hidden_layers=self.config.input_local_layers,
-            num_attention_heads=self.config.input_local_attn_heads,
-            num_key_value_heads=self.config.input_local_attn_heads,
-            intermediate_size=self.config.input_local_intermediate_size,
-            attention_dropout=self.config.input_local_hidden_dropout,
-            rope_theta=self.config.rope_theta,
-            partial_rotary_factor=self.config.partial_rotary_factor,
+            hidden_size=self.audio_input_local_dim,
+            num_hidden_layers=config.input_local_layers,
+            num_attention_heads=config.input_local_attn_heads,
+            num_key_value_heads=config.input_local_attn_heads,
+            intermediate_size=config.input_local_intermediate_size,
+            attention_dropout=config.input_local_hidden_dropout,
+            rope_theta=config.rope_theta,
+            partial_rotary_factor=config.partial_rotary_factor,
         )
-        input_local_config.head_dim = self.config.input_local_head_dim
-
+        input_local_config.head_dim = config.input_local_head_dim
         self.input_local_transformer = Qwen2Model(input_local_config)
-
-        if not self.config.add_post_norm:
+        if not config.add_post_norm:
             self.input_local_transformer.norm = nn.Identity()
 
         self.speech_embeddings = nn.ModuleList(
             [
                 nn.Embedding(
                     speech_vocab_size[i],
-                    self.config.input_local_dim,
-                    padding_idx=self.speech_empty_ids[i],
+                    self.audio_input_local_dim,
+                    padding_idx=speech_empty_ids[i],
                 )
-                for i in range(self.config.audio_channels)
+                for i in range(self.audio_channels)
             ]
         )
 
-        if self.config.projection_layers == 1:
+        if config.projection_layers == 1:
             self.projection = nn.Linear(
-                self.config.input_local_dim * self.config.group_size,
-                self.config.out_hidden_size,
+                self.audio_input_local_dim * self.audio_group_size,
+                self.audio_out_hidden_size,
                 bias=False,
             )
-        elif self.config.projection_layers == 2:
+        elif config.projection_layers == 2:
             self.projection = AudioProjection(
-                self.config.input_local_dim * self.config.group_size,
-                self.config.input_local_dim * self.config.group_size * 4,
-                self.config.out_hidden_size,
+                self.audio_input_local_dim * self.audio_group_size,
+                self.audio_input_local_dim * self.audio_group_size * 4,
+                self.audio_out_hidden_size,
             )
         else:
-            raise ValueError(
-                f"Invalid projection layers: {self.config.projection_layers}"
-            )
+            raise ValueError(f"Invalid projection layers: {config.projection_layers}")
 
-        model_path = self.server_args.model_path
+        model_path = get_global_server_args().model_path
         if not os.path.isdir(model_path):
             from huggingface_hub import snapshot_download
 
@@ -1216,13 +1233,16 @@ class MiMoAudioEncoder(nn.Module):
             )
         audio_tokenizer_path = os.path.join(model_path, "audio_tokenizer")
         dev = torch.device(f"cuda:{torch.cuda.current_device()}")
-        self.audio_tokenizer = self._load_audio_tokenizer(audio_tokenizer_path, dev)
+        self.audio_tokenizer = self._load_mimo_audio_tokenizer(
+            audio_tokenizer_path, dev
+        )
 
     @staticmethod
-    def _load_audio_tokenizer(path: str, device: torch.device) -> MiMoAudioTokenizer:
+    def _load_mimo_audio_tokenizer(
+        path: str, device: torch.device
+    ) -> MiMoAudioTokenizer:
         """Load MiMoAudioTokenizer manually to avoid new-transformers compat issues."""
         import json
-        import os
 
         from safetensors.torch import load_file
 
@@ -1231,7 +1251,6 @@ class MiMoAudioEncoder(nn.Module):
             config_dict = json.load(f)
         config = MiMoAudioTokenizer.config_class(**config_dict)
         model = MiMoAudioTokenizer(config)
-        # Load weights from safetensors or pytorch bin
         safetensors_path = os.path.join(path, "model.safetensors")
         bin_path = os.path.join(path, "pytorch_model.bin")
         if os.path.exists(safetensors_path):
@@ -1243,43 +1262,34 @@ class MiMoAudioEncoder(nn.Module):
                 f"No model weights found in {path} "
                 "(expected model.safetensors or pytorch_model.bin)"
             )
+        # strict=False: upstream ckpt also carries decoder/vocoder weights
+        # that this encoder-only MiMoAudioTokenizer doesn't materialize.
         model.load_state_dict(state_dict, strict=False)
         model = model.to(device=device, dtype=torch.bfloat16)
         model.eval()
         model.requires_grad_(False)
         return model
 
-    def parsed_speech_empty_ids(self):
-        return self._parse_maybe_list(
-            self.config.speech_zeroemb_idx, self.config.audio_channels
-        )
-
-    def _parse_maybe_list(self, value: str | int, length: int) -> List[int]:
-        if isinstance(value, str) and "-" in value:
-            return [int(s) for s in value.split("-")]
-        return [int(value)] * length
-
-    # adapted from mimo-audio
-    def apply_input_local_transformer(self, speech_embeddings: torch.Tensor):
-        output = self.input_local_transformer(
+    def apply_input_local_transformer(
+        self, speech_embeddings: torch.Tensor
+    ) -> torch.Tensor:
+        return self.input_local_transformer(
             inputs_embeds=speech_embeddings,
             return_dict=True,
-            is_causal=not self.config.input_full_attention,  # for SDPA
-        )
-        return output.last_hidden_state  # [T//group_size, group_size, input_local_dim]
+            is_causal=not self.audio_input_full_attention,  # for SDPA
+        ).last_hidden_state  # [T//group_size, group_size, input_local_dim]
 
     def apply_speech_embeddings(self, audio_codes: torch.Tensor) -> torch.Tensor:
-        num_segments = audio_codes.shape[0]
-        _audio_embeddings = torch.zeros(
-            (num_segments, self.config.group_size, self.config.input_local_dim),
+        embeds = torch.zeros(
+            (audio_codes.shape[0], self.audio_group_size, self.audio_input_local_dim),
             dtype=next(self.speech_embeddings[0].parameters()).dtype,
             device=audio_codes.device,
         )
-        for i in range(self.config.audio_channels):
-            _audio_embeddings.add_(self.speech_embeddings[i](audio_codes[:, :, i]))
-        return _audio_embeddings
+        for i in range(self.audio_channels):
+            embeds.add_(self.speech_embeddings[i](audio_codes[:, :, i]))
+        return embeds
 
-    def process_audio(self, audio):
+    def pad_audio_codes(self, audio: torch.Tensor) -> torch.Tensor:
         T = audio.shape[0]
         audio = audio[:, : self.audio_channels]
         padded_T = (
@@ -1299,17 +1309,19 @@ class MiMoAudioEncoder(nn.Module):
                 + audio[-1, :],
             ],
             dim=0,
-        )  # pad using the last embedding
-        padded_audio = padded_audio.reshape(
+        )
+        return padded_audio.reshape(
             padded_T // self.audio_group_size,
             self.audio_group_size,
             self.audio_channels,
         )
-        return padded_audio
 
     def get_audio_feature(self, items) -> torch.Tensor:
-        # items: already audio-only MultimodalDataItem list from caller.
-        # Each item.feature is either one mel tensor or a list of mel tensors (e.g. long audio split into chunks).
+        """Compute audio features for a list of audio MultimodalDataItem.
+
+        Each item.feature is either a mel tensor or a list of mel tensors
+        (long audio split into chunks).
+        """
         all_mels = []
         for item in items:
             f = item.feature
@@ -1321,9 +1333,8 @@ class MiMoAudioEncoder(nn.Module):
             device = next(self.projection.parameters()).device
             dtype = next(self.projection.parameters()).dtype
             return torch.empty(
-                0, self.config.out_hidden_size, device=device, dtype=dtype
+                0, self.audio_out_hidden_size, device=device, dtype=dtype
             )
-        # Batch tokenize: one encode_batch call for all mels
         device = next(self.audio_tokenizer.encoder.parameters()).device
         code_list = tokenize_audio_batch(
             all_mels,
@@ -1331,20 +1342,16 @@ class MiMoAudioEncoder(nn.Module):
             segment_size=self.audio_segment_size,
             device=device,
         )
-        codecs_to_concat = []
-        for codecs in code_list:
-            padded_codes = self.process_audio(
-                codecs
-            )  # [T//group_size, group_size, audio_channels]
-            codecs_to_concat.append(padded_codes)
-        audio_codes = torch.cat(
-            codecs_to_concat, dim=0
-        )  # [T//group_size, group_size, audio_channels]
+        audio_codes = torch.cat([self.pad_audio_codes(c) for c in code_list], dim=0)
+        embeds = self.apply_input_local_transformer(
+            self.apply_speech_embeddings(audio_codes)
+        )
+        return self.projection(embeds.reshape(embeds.shape[0], -1))
 
-        _audio_embeddings = self.apply_speech_embeddings(audio_codes)
-        audio_embeds = self.apply_input_local_transformer(
-            _audio_embeddings
-        )  #  [T//group_size,  group_size, input_local_dim]
-        B = audio_embeds.shape[0]
-        audio_embeds = self.projection(audio_embeds.reshape(B, -1))
-        return audio_embeds
+    @classmethod
+    def remap_audio_weight_name(cls, name: str) -> str:
+        """Normalize inner audio weight name variations to canonical form."""
+        for src, dst in cls.AUDIO_WEIGHT_REMAP:
+            if src in name:
+                return name.replace(src, dst)
+        return name
diff --git a/python/sglang/srt/models/mimo_v2.py b/python/sglang/srt/models/mimo_v2.py
index 4952148d9..33e35fe17 100644
--- a/python/sglang/srt/models/mimo_v2.py
+++ b/python/sglang/srt/models/mimo_v2.py
@@ -78,7 +78,7 @@ from sglang.srt.model_loader.weight_utils import (
     default_weight_loader,
     kv_cache_scales_loader,
 )
-from sglang.srt.models.mimo_audio import MiMoAudioEncoder, MiMoAudioEncoderConfig
+from sglang.srt.models.mimo_audio import AudioEncoderMixin, MiMoAudioEncoderConfig
 from sglang.srt.models.mimo_vl import MiMoVisionTransformer, MiMoVLVisionConfig
 from sglang.srt.server_args import get_global_server_args
 from sglang.srt.utils import (
@@ -993,7 +993,7 @@ class MiMoV2Model(nn.Module):
                 )
 
 
-class MiMoV2ForCausalLM(nn.Module):
+class MiMoV2ForCausalLM(nn.Module, AudioEncoderMixin):
     # BitandBytes specific attributes
     default_bitsandbytes_target_modules = [
         ".gate_proj.",
@@ -1015,8 +1015,10 @@ class MiMoV2ForCausalLM(nn.Module):
 
     # Prefixes for weight routing in encoder_only/language_only modes
     _LANGUAGE_WEIGHT_PREFIXES = ("model.", "lm_head.")
-    _VISION_AUDIO_WEIGHT_PREFIXES = ("visual.", "vision_model.", "audio_")
-    _VISION_AUDIO_WEIGHT_SUBSTRING = "speech_embeddings"
+    _VISION_WEIGHT_PREFIXES = ("visual.", "vision_model.")
+    # ``audio_`` already covers ``audio_encoder.`` so a single prefix is enough.
+    _AUDIO_WEIGHT_PREFIXES = ("audio_",)
+    _AUDIO_WEIGHT_SUBSTRING = "speech_embeddings"
 
     def __init__(
         self,
@@ -1070,8 +1072,7 @@ class MiMoV2ForCausalLM(nn.Module):
                 quant_config=None,
                 prefix=add_prefix("visual", prefix),
             )
-            self.audio_config = MiMoAudioEncoderConfig(**audio_config)
-            self.audio_encoder = MiMoAudioEncoder(self.audio_config)
+            self.build_audio_encoder(MiMoAudioEncoderConfig(**audio_config))
 
         self._routed_experts_weights_of_layer = LazyValue(
             lambda: (
@@ -1126,9 +1127,6 @@ class MiMoV2ForCausalLM(nn.Module):
         assert video_grid_thw.dim() == 2, video_grid_thw.dim()
         return self.visual(pixel_values, grid_thw=video_grid_thw)
 
-    def get_audio_feature(self, items: List[MultimodalDataItem]) -> torch.Tensor:
-        return self.audio_encoder.get_audio_feature(items)
-
     @torch.inference_mode()
     def encode_video_audio(self, mm_inputs: Dict) -> Optional[torch.Tensor]:
         # EPD-side hook: encode audio tracks pulled from videos and trim to the
@@ -1276,14 +1274,14 @@ class MiMoV2ForCausalLM(nn.Module):
         params_dict = dict(self.named_parameters())
         skipped_mtp_weights = False
 
-        def _is_vision_audio_weight(name):
-            return (
-                name.startswith(self._VISION_AUDIO_WEIGHT_PREFIXES)
-                or self._VISION_AUDIO_WEIGHT_SUBSTRING in name
+        for name, loaded_weight in weights:
+            is_vision_weight = name.startswith(self._VISION_WEIGHT_PREFIXES)
+            is_audio_weight = (
+                name.startswith(self._AUDIO_WEIGHT_PREFIXES)
+                or self._AUDIO_WEIGHT_SUBSTRING in name
             )
 
-        for name, loaded_weight in weights:
-            if not self._is_multimodal and _is_vision_audio_weight(name):
+            if not self._is_multimodal and (is_vision_weight or is_audio_weight):
                 continue
 
             if self.config.encoder_only and name.startswith(
@@ -1291,61 +1289,21 @@ class MiMoV2ForCausalLM(nn.Module):
             ):
                 continue
 
-            if self._is_multimodal and "audio" in name:
-                if "projection" in name:
-                    if (
-                        "audio_encoder.audio_projection" in name
-                        and "audio_encoder.projection" not in name
-                    ):
-                        name = name.replace(
-                            "audio_encoder.audio_projection", "audio_encoder.projection"
-                        )
-                    elif (
-                        "audio_projection" in name
-                        and "audio_encoder.projection" not in name
-                    ):
-                        name = name.replace(
-                            "audio_projection", "audio_encoder.projection"
-                        )
-                    param = params_dict[name]
-                    weight_loader = getattr(
-                        param, "weight_loader", default_weight_loader
+            if self._is_multimodal and is_audio_weight:
+                if name.startswith("audio_encoder."):
+                    name = name[len("audio_encoder.") :]
+                name = self.remap_audio_weight_name(name)
+                if name not in params_dict:
+                    logger.warning(
+                        f"Audio param {name} not found in params_dict, skipping"
                     )
-                    weight_loader(param, loaded_weight)
                     continue
-
-                if "input_local_transformer" in name:
-                    if (
-                        "audio_input_local_transformer" in name
-                        and "audio_encoder.input_local_transformer" not in name
-                    ):
-                        name = name.replace(
-                            "audio_input_local_transformer",
-                            "audio_encoder.input_local_transformer",
-                        )
-                    if name not in params_dict:
-                        logger.warning(
-                            f"Parameter {name} not found in params_dict, skipping"
-                        )
-                        continue
-                    param = params_dict[name]
-                    weight_loader = getattr(
-                        param, "weight_loader", default_weight_loader
-                    )
-                    weight_loader(param, loaded_weight)
-                    continue
-
-            if self._is_multimodal and "speech_embeddings" in name:
-                if (
-                    "speech_embeddings" in name
-                    and "audio_encoder.speech_embeddings" not in name
-                ):
-                    name = name.replace(
-                        "speech_embeddings", "audio_encoder.speech_embeddings"
-                    )
                 param = params_dict[name]
                 weight_loader = getattr(param, "weight_loader", default_weight_loader)
-                weight_loader(param, loaded_weight[: param.shape[0], :])
+                if self._AUDIO_WEIGHT_SUBSTRING in name:
+                    weight_loader(param, loaded_weight[: param.shape[0], :])
+                else:
+                    weight_loader(param, loaded_weight)
                 continue
 
             if self._is_multimodal and "visual" in name:
diff --git a/python/sglang/srt/models/mimo_v2_asr.py b/python/sglang/srt/models/mimo_v2_asr.py
new file mode 100644
index 000000000..4d63f3a9e
--- /dev/null
+++ b/python/sglang/srt/models/mimo_v2_asr.py
@@ -0,0 +1,162 @@
+"""MiMo-V2-ASR model.
+
+Reuses the LM scaffold of ``MiMoForCausalLM`` and adds audio encoder
+components via ``AudioEncoderMixin``. The encoder modules are attached as
+top-level attributes (no ``audio_encoder.`` prefix) so the checkpoint
+state_dict aligns 1:1 with ``self.named_parameters()``.
+"""
+
+import logging
+from typing import Any, Iterable, List, Optional, Tuple
+
+import torch
+
+from sglang.srt.managers.mm_utils import (
+    MultiModalityDataPaddingPatternMultimodalTokens,
+    general_mm_embed_routine,
+)
+from sglang.srt.managers.schedule_batch import MultimodalInputs
+from sglang.srt.model_executor.forward_batch_info import ForwardBatch, PPProxyTensors
+from sglang.srt.model_loader.weight_utils import default_weight_loader
+from sglang.srt.models import mimo_audio as _mimo_audio_module
+from sglang.srt.models.mimo import MiMoForCausalLM
+from sglang.srt.models.mimo_audio import AudioEncoderMixin, MiMoAudioEncoderConfig
+
+logger = logging.getLogger(__name__)
+
+
+def _maybe_override_audio_attn_for_blackwell() -> None:
+    """Swap mimo_audio.flash_attn_varlen_func to upstream FA2 on GPUs that
+    sgl-kernel's FA3 doesn't support.
+
+    sgl-kernel FA3 only covers sm80/86/89/90 — on Blackwell consumer cards
+    (sm_120 / RTX 50xx) its varlen kernel raises NotImplementedError. ASR is
+    small enough to be deployed on those GPUs, so when FA3 isn't supported
+    we replace the module-level reference with upstream flash-attn (FA2),
+    which works on sm_120. No-op on supported GPUs (FA3 stays).
+
+    MiMo-V2 (the heavy multimodal model) is only deployed on H100/A100, so
+    this override never triggers in its hot path.
+    """
+    try:
+        from sgl_kernel.flash_attn import is_fa3_supported
+    except ImportError:
+        return
+    if is_fa3_supported():
+        return
+    try:
+        from flash_attn import flash_attn_varlen_func
+    except ImportError as e:
+        raise RuntimeError(
+            "MiMo-V2-ASR audio encoder needs upstream flash-attn on this GPU "
+            "(sgl-kernel FA3 doesn't support sm_120). Install with "
+            "`pip install flash-attn --no-build-isolation`."
+        ) from e
+    _mimo_audio_module.flash_attn_varlen_func = flash_attn_varlen_func
+
+
+MiMoV2ASRConfig = Any
+
+# Top-level audio sub-module name prefixes (after AUDIO_WEIGHT_REMAP). Loaded
+# directly by default_weight_loader because the LM branch's qkv/gate-up fused
+# stacked-params mapping doesn't apply to the vanilla HF Qwen2Model used
+# inside the audio encoder.
+_AUDIO_NAME_PREFIXES: Tuple[str, ...] = (
+    "projection.",
+    "input_local_transformer.",
+    "speech_embeddings.",
+)
+
+# Training-only weights present in checkpoint but not used at inference.
+# Checked AFTER the audio-prefix load path so substring matching here is
+# safe: legitimate audio weights (``input_local_transformer.*``) are
+# already consumed by ``_AUDIO_NAME_PREFIXES`` above.
+_SKIP_NAME_SUBSTRINGS: Tuple[str, ...] = (
+    "hidden_states_downcast",
+    "local_transformer",
+)
+
+
+class MiMoV2ASRForCausalLM(MiMoForCausalLM, AudioEncoderMixin):
+    def __init__(
+        self,
+        config: MiMoV2ASRConfig,
+        quant_config=None,
+        prefix: str = "",
+    ) -> None:
+        _maybe_override_audio_attn_for_blackwell()
+        super().__init__(config, quant_config=quant_config, prefix=prefix)
+        self.build_audio_encoder(MiMoAudioEncoderConfig(**config.audio_config))
+
+    def pad_input_ids(self, input_ids: List[int], mm_inputs: MultimodalInputs):
+        pattern = MultiModalityDataPaddingPatternMultimodalTokens()
+        return pattern.pad_input_tokens(input_ids, mm_inputs)
+
+    def get_input_embeddings(self):
+        if getattr(self.config, "encoder_only", False):
+            return None
+        return self.model.embed_tokens
+
+    @torch.no_grad()
+    def forward(
+        self,
+        input_ids: torch.Tensor,
+        positions: torch.Tensor,
+        forward_batch: ForwardBatch,
+        input_embeds: torch.Tensor = None,
+        get_embedding: bool = False,
+        pp_proxy_tensors: Optional[PPProxyTensors] = None,
+    ) -> torch.Tensor:
+        if getattr(self.config, "encoder_only", False):
+            raise NotImplementedError(
+                "forward() is not supported in encoder_only mode. "
+                "Use get_audio_feature() instead."
+            )
+
+        hidden_states = general_mm_embed_routine(
+            input_ids=input_ids,
+            forward_batch=forward_batch,
+            language_model=self.model,
+            multimodal_model=self,
+            positions=positions,
+            pp_proxy_tensors=pp_proxy_tensors,
+        )
+
+        if not get_embedding:
+            return self.logits_processor(
+                input_ids, hidden_states, self.lm_head, forward_batch
+            )
+        return self.pooler(hidden_states, forward_batch)
+
+    def load_weights(self, weights: Iterable[Tuple[str, torch.Tensor]]):
+        params_dict = dict(self.named_parameters())
+        deferred: List[Tuple[str, torch.Tensor]] = []
+
+        for name, loaded_weight in weights:
+            if name.startswith("audio_encoder."):
+                name = name[len("audio_encoder.") :]
+            name = self.remap_audio_weight_name(name)
+
+            if name.startswith(_AUDIO_NAME_PREFIXES):
+                if name not in params_dict:
+                    logger.warning(
+                        f"Audio param {name} not found in params_dict, skipping"
+                    )
+                    continue
+                param = params_dict[name]
+                weight_loader = getattr(param, "weight_loader", default_weight_loader)
+                if name.startswith("speech_embeddings."):
+                    weight_loader(param, loaded_weight[: param.shape[0], :])
+                else:
+                    weight_loader(param, loaded_weight)
+                continue
+
+            if any(s in name for s in _SKIP_NAME_SUBSTRINGS):
+                continue
+
+            deferred.append((name, loaded_weight))
+
+        super().load_weights(iter(deferred))
+
+
+EntryClass = MiMoV2ASRForCausalLM
diff --git a/python/sglang/srt/multimodal/processors/mimo_audio.py b/python/sglang/srt/multimodal/processors/mimo_audio.py
new file mode 100644
index 000000000..3bae8ea16
--- /dev/null
+++ b/python/sglang/srt/multimodal/processors/mimo_audio.py
@@ -0,0 +1,318 @@
+"""Stateful audio preprocessing pipeline shared by MiMo multimodal and ASR processors."""
+
+import io
+import math
+import os
+import time
+from collections import OrderedDict
+from dataclasses import dataclass
+from typing import Optional
+
+import numpy as np
+import pybase64
+import requests
+import torch
+
+from sglang.utils import logger
+
+try:
+    from torchcodec.decoders import AudioDecoder
+except ImportError:
+    logger.warning(
+        "torchcodec is not installed; audio inputs will fail at request time"
+    )
+    AudioDecoder = None
+
+try:
+    import torchaudio
+    from torchaudio.transforms import MelSpectrogram
+except ImportError:
+    logger.warning(
+        "torchaudio is not installed; audio inputs will fail at request time"
+    )
+    torchaudio = None
+    MelSpectrogram = None
+
+
+@dataclass
+class AudioInput:
+    """
+    if audio is str or bytes, only load it as mel spectrogram.
+    if audio is tuple, it is (waveform, original_sr)
+    if audio is torch.Tensor, it is tokenized input ids with shape (T, n_vq+).
+    if audio is np.ndarray, it is a pre-loaded waveform (1D, already resampled).
+    """
+
+    audio: str | bytes | tuple | torch.Tensor | np.ndarray
+
+    def __post_init__(self):
+        if not isinstance(self.audio, (str, bytes, tuple, torch.Tensor, np.ndarray)):
+            raise ValueError(
+                f"audio must be a str, bytes, tuple, torch.Tensor, or np.ndarray, but got {type(self.audio)}"
+            )
+        if isinstance(self.audio, tuple):
+            if (
+                len(self.audio) != 2
+                or not isinstance(self.audio[0], torch.Tensor)
+                or not isinstance(self.audio[1], (int, float))
+            ):
+                raise ValueError(
+                    f"audio must be a tuple of (waveform-T, original_sr-int/float), but got {len(self.audio)} elements and {type(self.audio[0])} and {type(self.audio[1])}"
+                )
+            if self.audio[0].ndim != 1:
+                raise ValueError(
+                    f"waveform must be a 1D tensor, but got {self.audio[0].ndim}D tensor"
+                )
+            if self.audio[1] <= 0:
+                raise ValueError(
+                    f"original_sr must be a positive number, but got {self.audio[1]}"
+                )
+        if isinstance(self.audio, torch.Tensor) and self.audio.ndim != 2:
+            raise ValueError(
+                f"audio must be a 2D tensor, but got {self.audio.ndim}D tensor"
+            )
+
+
+class MiMoAudioPipeline:
+    """Stateful audio preprocessing pipeline.
+
+    Composable: held by both MiMoProcessor (multimodal) and MiMoV2ASRProcessor.
+    Owns the mel spectrogram, resampler cache, http session, and the special
+    token ids for ``<|sosp|> <|empty|>* <|eosp|>`` placeholders.
+    """
+
+    def __init__(
+        self,
+        *,
+        audio_token_id: int,
+        audio_start_token_id: int,
+        audio_end_token_id: int,
+        audio_kernel_size: int = 3,
+        audio_stride_size: int = 2,
+        audio_avg_pooler: int = 2,
+        audio_group_size: int = 4,
+        audio_channels: int = 8,
+        audio_sampling_rate: int = 24000,
+        audio_nfft: int = 960,
+        audio_hop_length: int = 240,
+        audio_window_size: int = 960,
+        audio_fmin: int = 0,
+        audio_fmax: Optional[int] = None,
+        audio_n_mels: int = 128,
+        audio_input_id_per_second: int = 25,
+        max_resamplers: int = 16,
+    ) -> None:
+        self.audio_token_id = audio_token_id
+        self.audio_start_token_id = audio_start_token_id
+        self.audio_end_token_id = audio_end_token_id
+
+        self.audio_kernel_size = audio_kernel_size
+        self.audio_stride_size = audio_stride_size
+        self.audio_avg_pooler = audio_avg_pooler
+        self.audio_group_size = audio_group_size
+        self.audio_channels = audio_channels
+
+        self.audio_sampling_rate = audio_sampling_rate
+        self.audio_nfft = audio_nfft
+        self.audio_hop_length = audio_hop_length
+        self.audio_window_size = audio_window_size
+        self.audio_fmin = audio_fmin
+        self.audio_fmax = audio_fmax
+        self.audio_n_mels = audio_n_mels
+        self.audio_input_id_per_second = audio_input_id_per_second
+
+        self.mel_spectrogram_kwargs = dict(
+            sample_rate=audio_sampling_rate,
+            n_fft=audio_nfft,
+            hop_length=audio_hop_length,
+            win_length=audio_window_size,
+            f_min=audio_fmin,
+            f_max=audio_fmax,
+            n_mels=audio_n_mels,
+            power=1.0,
+            center=True,
+        )
+        self._mel_spectrogram = None
+        self._resamplers: "OrderedDict[int, torchaudio.transforms.Resample]" = (
+            OrderedDict()
+        )
+        self._resamplers_max = max_resamplers
+
+        self.http_session = requests.Session()
+
+    @property
+    def audio_token_per_second(self) -> float:
+        return self.audio_input_id_per_second / self.audio_group_size
+
+    @staticmethod
+    def _ensure_audio_dependencies() -> None:
+        if torchaudio is None or MelSpectrogram is None:
+            raise RuntimeError(
+                "torchaudio is required for audio inputs; install torchaudio"
+            )
+
+    @property
+    def mel_spectrogram(self):
+        self._ensure_audio_dependencies()
+        if self._mel_spectrogram is None:
+            self._mel_spectrogram = MelSpectrogram(**self.mel_spectrogram_kwargs)
+        return self._mel_spectrogram
+
+    def compute_audio_token_len(self, mel_len: int) -> int:
+        n = mel_len + 3 - self.audio_kernel_size
+        n = (n + 2 - self.audio_kernel_size) // self.audio_stride_size + 1
+        n = n // self.audio_avg_pooler + int(n % self.audio_avg_pooler != 0)
+        return math.ceil(n / self.audio_group_size)
+
+    def preprocess_audio(self, audio):
+        """Load audio source → log-mel spectrogram + token length.
+
+        Input: filename string, bytes, or tuple of (waveform, original_sr).
+        Output: (mel-spectrogram tensor [T, n_mels], audio_token_len int).
+        """
+        self._ensure_audio_dependencies()
+        assert isinstance(
+            audio, (str, bytes, tuple)
+        ), f"audio must be a str, bytes or tuple, but got {type(audio)}"
+        if isinstance(audio, tuple):
+            waveform, original_sr = audio
+        else:
+            if isinstance(audio, bytes):
+                file = io.BytesIO(audio)
+            elif isinstance(audio, str):
+                if audio.startswith("data:"):
+                    file = io.BytesIO(
+                        pybase64.b64decode(audio.split(",")[1], validate=True)
+                    )
+                elif audio.startswith("http://") or audio.startswith("https://"):
+                    dl_start = time.perf_counter()
+                    timeout = int(os.getenv("REQUEST_TIMEOUT", "5"))
+                    try:
+                        response = self.http_session.get(
+                            audio, stream=True, timeout=timeout
+                        )
+                        dl_elapsed_ms = (time.perf_counter() - dl_start) * 1000
+                        if dl_elapsed_ms > 1000.0:
+                            content_len = len(response.content)
+                            logger.warning(
+                                f"Slow audio download: {dl_elapsed_ms:.2f}ms, "
+                                f"size={content_len / 1024:.1f}KB, url={audio}"
+                            )
+                        file = io.BytesIO(response.content)
+                        response.close()
+                    except Exception as e:
+                        dl_elapsed_ms = (time.perf_counter() - dl_start) * 1000
+                        logger.error(
+                            f"Failed to download audio: {dl_elapsed_ms:.2f}ms, "
+                            f"error={type(e).__name__}: {e}, url={audio}"
+                        )
+                        raise
+                else:
+                    file = audio
+            if AudioDecoder is None:
+                raise RuntimeError(
+                    "torchcodec is required for audio decoding; install with `pip install torchcodec`."
+                )
+            try:
+                samples = AudioDecoder(file).get_all_samples()
+            except RuntimeError as e:
+                audio_source = (
+                    audio
+                    if isinstance(audio, str)
+                    and (audio.startswith("http://") or audio.startswith("https://"))
+                    else "<bytes or base64>"
+                )
+                logger.error(f"Failed to decode audio: {e}, source={audio_source}")
+                raise ValueError(
+                    f"Invalid audio format: source={audio_source}, detail={e}"
+                ) from e
+            waveform = samples.data
+            original_sr = samples.sample_rate
+
+        if original_sr != self.audio_sampling_rate:
+            if original_sr in self._resamplers:
+                self._resamplers.move_to_end(original_sr)
+            else:
+                if len(self._resamplers) >= self._resamplers_max:
+                    self._resamplers.popitem(last=False)
+                self._resamplers[original_sr] = torchaudio.transforms.Resample(
+                    orig_freq=original_sr, new_freq=self.audio_sampling_rate
+                )
+            waveform = self._resamplers[original_sr](waveform)
+        if waveform.ndim == 2:
+            waveform = waveform.mean(dim=0)
+        spec = self.mel_spectrogram(waveform[None, :])
+        spec = torch.log(torch.clip(spec, min=1e-7)).squeeze()
+        spec = spec.transpose(0, 1)
+
+        audio_token_len = self.compute_audio_token_len(spec.shape[0])
+        return spec, audio_token_len
+
+    def process_audio(self, audio_input: AudioInput):
+        """Dispatch on the underlying audio payload.
+
+        - str/bytes/tuple/np.ndarray waveform → returns (mel-spec, token_len) tuple
+        - 2D tensor of pre-tokenized audio codes → returns padded codes tensor
+          shaped [T//group, group, channels]
+        """
+        audio = audio_input.audio
+        if isinstance(audio, np.ndarray):
+            waveform = torch.from_numpy(audio).float()
+            audio = (waveform, self.audio_sampling_rate)
+        if isinstance(audio, (str, bytes, tuple)):
+            return self.preprocess_audio(audio)
+
+        assert (
+            audio.shape[1] >= self.audio_channels
+        ), f"audio must have at least {self.audio_channels} channels, but got {audio.shape[1]}"
+        T = audio.shape[0]
+        audio = audio[:, : self.audio_channels].to(torch.long)
+        padded_T = (
+            (T + self.audio_group_size - 1)
+            // self.audio_group_size
+            * self.audio_group_size
+        )
+        padded_audio = torch.cat(
+            [
+                audio,
+                torch.zeros(padded_T - T, self.audio_channels, dtype=torch.long)
+                + audio[-1, :],
+            ],
+            dim=0,
+        )
+        padded_audio = padded_audio.reshape(
+            padded_T // self.audio_group_size,
+            self.audio_group_size,
+            self.audio_channels,
+        )
+        return padded_audio
+
+    def build_audio_placeholder_ids(self, audio_token_len: int) -> list[int]:
+        return (
+            [self.audio_start_token_id]
+            + [self.audio_token_id] * audio_token_len
+            + [self.audio_end_token_id]
+        )
+
+    def process_audio_input(self, audio_input: AudioInput) -> dict:
+        """Run process_audio and produce the placeholder input_ids.
+
+        Replaces the duplicated _process_audio_content bodies in both processors.
+        Returns dict with input_ids, audio_input (mel or codes), and is_tokenized.
+        """
+        processed = self.process_audio(audio_input)
+        if isinstance(processed, tuple):
+            is_tokenized = False
+            audio_spec, audio_token_len = processed
+            payload = audio_spec
+        else:
+            is_tokenized = True
+            audio_token_len = processed.shape[0]
+            payload = processed
+
+        return {
+            "input_ids": self.build_audio_placeholder_ids(audio_token_len),
+            "audio_input": payload,
+            "audio_token_len": audio_token_len,
+            "is_tokenized": is_tokenized,
+        }
diff --git a/python/sglang/srt/multimodal/processors/mimo_v2.py b/python/sglang/srt/multimodal/processors/mimo_v2.py
index 0c6c82d71..91397692f 100644
--- a/python/sglang/srt/multimodal/processors/mimo_v2.py
+++ b/python/sglang/srt/multimodal/processors/mimo_v2.py
@@ -3,21 +3,16 @@
 import asyncio
 import base64
 import copy
-import io
 import json
 import math
-import os
 import re
 import subprocess
-import time
-from collections import OrderedDict
 from concurrent.futures import ThreadPoolExecutor, as_completed
 from dataclasses import dataclass, field
 from io import BytesIO
 from typing import List, Literal, Optional, Union
 
 import numpy as np
-import pybase64
 import requests
 import torch
 import torch.nn.functional as F
@@ -39,20 +34,14 @@ from sglang.srt.multimodal.processors.base_processor import (
     BaseMultimodalProcessor,
     MultimodalSpecialTokens,
 )
+from sglang.srt.multimodal.processors.mimo_audio import (
+    AudioInput,
+    MiMoAudioPipeline,
+)
 from sglang.srt.multimodal.processors.qwen_vl import smart_nframes
 from sglang.srt.utils import ImageData, VideoData
 from sglang.utils import logger
 
-try:
-    import torchaudio
-    from torchaudio.transforms import MelSpectrogram
-except ImportError:
-    logger.warning(
-        "torchaudio is not installed; audio inputs will fail at request time"
-    )
-    torchaudio = None
-    MelSpectrogram = None
-
 
 @dataclass
 class ImageInput:
@@ -112,45 +101,6 @@ class VideoInput:
         )
 
 
-@dataclass
-class AudioInput:
-    """
-    if audio is str or bytes, only load it as mel spectrogram.
-    if audio is tuple, it is (waveform, original_sr)
-    if audio is torch.Tensor, it is tokenized input ids with shape (T, n_vq+).
-    if audio is np.ndarray, it is a pre-loaded waveform (1D, already resampled).
-    """
-
-    audio: str | bytes | tuple | torch.Tensor | np.ndarray
-
-    def __post_init__(self):
-        if not isinstance(self.audio, (str, bytes, tuple, torch.Tensor, np.ndarray)):
-            raise ValueError(
-                f"audio must be a str, bytes, tuple, torch.Tensor, or np.ndarray, but got {type(self.audio)}"
-            )
-        if isinstance(self.audio, tuple):
-            if (
-                len(self.audio) != 2
-                or not isinstance(self.audio[0], torch.Tensor)
-                or not isinstance(self.audio[1], (int, float))
-            ):
-                raise ValueError(
-                    f"audio must be a tuple of (waveform-T, original_sr-int/float), but got {len(self.audio)} elements and {type(self.audio[0])} and {type(self.audio[1])}"
-                )
-            if self.audio[0].ndim != 1:
-                raise ValueError(
-                    f"waveform must be a 1D tensor, but got {self.audio[0].ndim}D tensor"
-                )
-            if self.audio[1] <= 0:
-                raise ValueError(
-                    f"original_sr must be a positive number, but got {self.audio[1]}"
-                )
-        if isinstance(self.audio, torch.Tensor) and self.audio.ndim != 2:
-            raise ValueError(
-                f"audio must be a 2D tensor, but got {self.audio.ndim}D tensor"
-            )
-
-
 @dataclass
 class VideoAudioInput:
     video: str | bytes | tuple[torch.Tensor, torch.Tensor]
@@ -334,11 +284,9 @@ class MiMoProcessor:
         audio_fmin=0,
         audio_fmax=None,
         audio_n_mels=128,
-        audio_segment_size=6000,
         audio_channels=8,
         audio_group_size=4,
         audio_input_id_per_second=25,
-        audio_zeroemb_idx=4096,
         image_min_pixels=None,
         image_max_pixels=None,
         video_min_pixels=None,
@@ -395,11 +343,8 @@ class MiMoProcessor:
 
         self.image_token_id = image_token_id
         self.video_token_id = video_token_id
-        self.audio_token_id = audio_token_id
         self.vision_start_token_id = vision_start_token_id
         self.vision_end_token_id = vision_end_token_id
-        self.audio_start_token_id = audio_start_token_id
-        self.audio_end_token_id = audio_end_token_id
         self.video_start_token_id = video_start_token_id
         self.video_end_token_id = video_end_token_id
         self.pad_token_id = pad_token_id
@@ -411,60 +356,24 @@ class MiMoProcessor:
 
         self.video_tokens_per_second = video_tokens_per_second
 
-        self.audio_sampling_rate = audio_sampling_rate
-        self.audio_nfft = audio_nfft
-        self.audio_hop_length = audio_hop_length
-        self.audio_window_size = audio_window_size
-        self.audio_fmin = audio_fmin
-        self.audio_fmax = audio_fmax
-        self.audio_n_mels = audio_n_mels
-
-        self.audio_segment_size = audio_segment_size
-
-        self.audio_kernel_size = audio_kernel_size
-        self.audio_stride_size = audio_stride_size
-        self.audio_avg_pooler = audio_avg_pooler
-
-        self.mel_spectrogram_kwargs = dict(
-            sample_rate=audio_sampling_rate,
-            n_fft=audio_nfft,
-            hop_length=audio_hop_length,
-            win_length=audio_window_size,
-            f_min=audio_fmin,
-            f_max=audio_fmax,
-            n_mels=audio_n_mels,
-            power=1.0,
-            center=True,
+        self.audio_pipeline = MiMoAudioPipeline(
+            audio_token_id=audio_token_id,
+            audio_start_token_id=audio_start_token_id,
+            audio_end_token_id=audio_end_token_id,
+            audio_kernel_size=audio_kernel_size,
+            audio_stride_size=audio_stride_size,
+            audio_avg_pooler=audio_avg_pooler,
+            audio_group_size=audio_group_size,
+            audio_channels=audio_channels,
+            audio_sampling_rate=audio_sampling_rate,
+            audio_nfft=audio_nfft,
+            audio_hop_length=audio_hop_length,
+            audio_window_size=audio_window_size,
+            audio_fmin=audio_fmin,
+            audio_fmax=audio_fmax,
+            audio_n_mels=audio_n_mels,
+            audio_input_id_per_second=audio_input_id_per_second,
         )
-        self._mel_spectrogram = None
-        self._resamplers = OrderedDict()
-        self._resamplers_max = 16
-
-        self.audio_channels = audio_channels
-        self.audio_group_size = audio_group_size
-        self.audio_input_id_per_second = audio_input_id_per_second
-        if isinstance(audio_zeroemb_idx, int):
-            self.audio_zeroemb_idxs = torch.tensor(
-                [audio_zeroemb_idx] * self.audio_channels, dtype=torch.int32
-            )
-        elif isinstance(audio_zeroemb_idx, list):
-            if len(audio_zeroemb_idx) == 1:
-                self.audio_zeroemb_idxs = torch.tensor(
-                    audio_zeroemb_idx * self.audio_channels, dtype=torch.int32
-                )
-            elif len(audio_zeroemb_idx) == self.audio_channels:
-                self.audio_zeroemb_idxs = torch.tensor(
-                    audio_zeroemb_idx, dtype=torch.int32
-                )
-            else:
-                raise ValueError(
-                    f"audio_zeroemb_idx must be a list of 1 or {self.audio_channels} integers, but got {len(audio_zeroemb_idx)}"
-                )
-        else:
-            raise ValueError(
-                f"audio_zeroemb_idx must be an integer or a list of {self.audio_channels} integers, but got {type(audio_zeroemb_idx)}"
-            )
-
         assert image_min_pixels is not None
         assert image_max_pixels is not None
         assert video_min_pixels is not None
@@ -487,10 +396,18 @@ class MiMoProcessor:
             "min_frames": min_frames,
         }
 
-        self.http_session = requests.Session()
         for k in kwargs:
             logger.info(f"[Warning] Ignored unknown parameter {k} for MiMoProcessor")
 
+    def __getattr__(self, name):
+        # Delegate audio_pipeline fields so callers can use self.audio_token_id
+        # etc. directly. Only triggers when normal attribute lookup fails;
+        # __dict__.get avoids recursion before audio_pipeline is assigned.
+        pipeline = self.__dict__.get("audio_pipeline")
+        if pipeline is not None and hasattr(pipeline, name):
+            return getattr(pipeline, name)
+        raise AttributeError(name)
+
     @classmethod
     def from_hf_config(cls, hf_config, mm_config=None, **overrides):
         # Params must come from hf_config.processor_config so E and D agree;
@@ -576,13 +493,13 @@ class MiMoProcessor:
             return _ffprobe_has_audio(path_or_data, stdin=None, label=path_or_data)
 
         if isinstance(path_or_data, bytes):
-            source = io.BytesIO(path_or_data)
+            source = BytesIO(path_or_data)
         elif (
             isinstance(path_or_data, str)
             and path_or_data.startswith("data:")
             and ";base64," in path_or_data
         ):
-            source = io.BytesIO(base64.b64decode(path_or_data.split(";base64,")[1]))
+            source = BytesIO(base64.b64decode(path_or_data.split(";base64,")[1]))
         else:
             source = path_or_data  # local path or file://
         try:
@@ -667,7 +584,9 @@ class MiMoProcessor:
                 all_timestamps.extend(aligned_ts[::step].tolist())
 
                 if self.has_audio_track(video_blob):
-                    audio_spec, audio_token_len = self.preprocess_audio(video_blob)
+                    audio_spec, audio_token_len = self.audio_pipeline.preprocess_audio(
+                        video_blob
+                    )
                     units = self._build_video_audio_units(
                         grid,
                         aligned_ts,
@@ -704,7 +623,7 @@ class MiMoProcessor:
             for audio in mm_data:
                 if isinstance(audio, np.ndarray):
                     audio = (torch.from_numpy(audio).float(), self.audio_sampling_rate)
-                spec, token_len = self.preprocess_audio(audio)
+                spec, token_len = self.audio_pipeline.preprocess_audio(audio)
                 all_specs.append(spec)
                 all_lens.append(token_len)
             return {
@@ -714,20 +633,6 @@ class MiMoProcessor:
 
         raise ValueError(f"Unsupported modality for EPD preprocessing: {modality}")
 
-    @property
-    def mel_spectrogram(self):
-        self._ensure_audio_dependencies()
-        if self._mel_spectrogram is None:
-            self._mel_spectrogram = MelSpectrogram(**self.mel_spectrogram_kwargs)
-        return self._mel_spectrogram
-
-    @staticmethod
-    def _ensure_audio_dependencies():
-        if torchaudio is None or MelSpectrogram is None:
-            raise RuntimeError(
-                "torchaudio is required for audio inputs; install torchaudio"
-            )
-
     def prepare_image_kwargs(self, image: ImageInput):
         kwargs = {}
         for k in ["min_pixels", "max_pixels"]:
@@ -764,95 +669,6 @@ class MiMoProcessor:
             raise ValueError("Video sampling strategy not specified")
         return kwargs
 
-    def preprocess_audio(self, audio: str | bytes):
-        self._ensure_audio_dependencies()
-        """
-        - Input: audio filename string, bytes, or tuple of (waveform, original_sr)
-        - Output:
-            - mel spectrogram: torch.Tensor (T, n_mels)
-            - number of tokens: int
-        """
-        assert isinstance(
-            audio, (str, bytes, tuple)
-        ), f"audio must be a str, bytes or tuple, but got {type(audio)}"
-        if isinstance(audio, tuple):
-            waveform, original_sr = audio
-        else:
-            if isinstance(audio, bytes):
-                file = io.BytesIO(audio)
-            elif isinstance(audio, str):
-                if audio.startswith("data:"):
-                    file = io.BytesIO(
-                        pybase64.b64decode(audio.split(",")[1], validate=True)
-                    )
-                elif audio.startswith("http://") or audio.startswith("https://"):
-                    dl_start = time.perf_counter()
-                    timeout = int(os.getenv("REQUEST_TIMEOUT", "5"))
-                    try:
-                        response = self.http_session.get(
-                            audio, stream=True, timeout=timeout
-                        )
-                        dl_elapsed_ms = (time.perf_counter() - dl_start) * 1000
-                        if dl_elapsed_ms > 1000.0:
-                            content_len = len(response.content)
-                            logger.warning(
-                                f"Slow audio download: {dl_elapsed_ms:.2f}ms, "
-                                f"size={content_len / 1024:.1f}KB, url={audio}"
-                            )
-                        file = io.BytesIO(response.content)
-                        response.close()
-                    except Exception as e:
-                        dl_elapsed_ms = (time.perf_counter() - dl_start) * 1000
-                        logger.error(
-                            f"Failed to download audio: {dl_elapsed_ms:.2f}ms, "
-                            f"error={type(e).__name__}: {e}, url={audio}"
-                        )
-                        raise
-                else:
-                    file = audio
-            try:
-                samples = AudioDecoder(file).get_all_samples()
-            except RuntimeError as e:
-                audio_source = (
-                    audio
-                    if isinstance(audio, str)
-                    and (audio.startswith("http://") or audio.startswith("https://"))
-                    else "<bytes or base64>"
-                )
-                logger.error(f"Failed to decode audio: {e}, source={audio_source}")
-                raise ValueError(
-                    f"Invalid audio format: source={audio_source}, detail={e}"
-                ) from e
-            waveform = samples.data
-            original_sr = samples.sample_rate
-
-        if original_sr != self.audio_sampling_rate:
-            if original_sr in self._resamplers:
-                self._resamplers.move_to_end(original_sr)
-            else:
-                if len(self._resamplers) >= self._resamplers_max:
-                    self._resamplers.popitem(last=False)
-                self._resamplers[original_sr] = torchaudio.transforms.Resample(
-                    orig_freq=original_sr, new_freq=self.audio_sampling_rate
-                )
-            waveform = self._resamplers[original_sr](waveform)
-        if waveform.ndim == 2:
-            waveform = waveform.mean(dim=0)
-        spec = self.mel_spectrogram(waveform[None, :])
-        spec = torch.log(torch.clip(spec, min=1e-7)).squeeze()
-        spec = spec.transpose(0, 1)
-
-        audio_token_len = spec.shape[0] + 3 - self.audio_kernel_size
-        audio_token_len = (
-            audio_token_len + 2 - self.audio_kernel_size
-        ) // self.audio_stride_size + 1
-        audio_token_len = audio_token_len // self.audio_avg_pooler + int(
-            audio_token_len % self.audio_avg_pooler != 0
-        )
-        audio_token_len = math.ceil(audio_token_len / self.audio_group_size)
-
-        return spec, audio_token_len
-
     def process_image(self, image: ImageInput):
         kwargs = self.prepare_image_kwargs(image)
         image = image.image
@@ -1014,40 +830,6 @@ class MiMoProcessor:
         )
         return visual_patches, thw_grid, aligned_timestamps, video_meta
 
-    def process_audio(self, audio: AudioInput):
-        audio = audio.audio
-        if isinstance(audio, np.ndarray):
-            waveform = torch.from_numpy(audio).float()
-            audio = (waveform, self.audio_sampling_rate)
-        if isinstance(audio, (str, bytes, tuple)):
-            audio_spec, audio_token_len = self.preprocess_audio(audio)
-            return audio_spec, audio_token_len
-
-        assert (
-            audio.shape[1] >= self.audio_channels
-        ), f"audio must have at least {self.audio_channels} channels, but got {audio.shape[1]}"
-        T = audio.shape[0]
-        audio = audio[:, : self.audio_channels].to(torch.long)
-        padded_T = (
-            (T + self.audio_group_size - 1)
-            // self.audio_group_size
-            * self.audio_group_size
-        )
-        padded_audio = torch.cat(
-            [
-                audio,
-                torch.zeros(padded_T - T, self.audio_channels, dtype=torch.long)
-                + audio[-1, :],
-            ],
-            dim=0,
-        )
-        padded_audio = padded_audio.reshape(
-            padded_T // self.audio_group_size,
-            self.audio_group_size,
-            self.audio_channels,
-        )
-        return padded_audio
-
     def _process_videos_parallel(self, contents):
         video_contents_info = []
         for idx, content in enumerate(contents):
@@ -1171,29 +953,17 @@ class MiMoProcessor:
         }
 
     def _process_audio_content(self, content, verbose):
-        processed_audio = self.process_audio(content.content)
-        if isinstance(processed_audio, tuple):
-            is_tokenized = False
-            audio_spec, audio_token_len = processed_audio
-            audio_input = audio_spec
-        else:
-            is_tokenized = True
-            audio_token_len = processed_audio.shape[0]
-            audio_input = processed_audio
-        _input_ids = (
-            [self.audio_start_token_id]
-            + [self.audio_token_id] * audio_token_len
-            + [self.audio_end_token_id]
-        )
-
+        result = self.audio_pipeline.process_audio_input(content.content)
         verbose_str = ""
         if verbose:
-            verbose_str = f"Audio (is_tokenized={is_tokenized}): [<audio_start> {audio_token_len}*<audio> <audio_end>]\n"
-
+            verbose_str = (
+                f"Audio (is_tokenized={result['is_tokenized']}): "
+                f"[<audio_start> {result['audio_token_len']}*<audio> <audio_end>]\n"
+            )
         return {
-            "input_ids": _input_ids,
-            "audio_input": audio_input,
-            "is_tokenized": is_tokenized,
+            "input_ids": result["input_ids"],
+            "audio_input": result["audio_input"],
+            "is_tokenized": result["is_tokenized"],
             "verbose": verbose_str,
         }
 
@@ -1219,7 +989,7 @@ class MiMoProcessor:
         grid_t_timestamps = timestamps[
             :: self.temporal_patch_size * self.temporal_compression_ratio
         ]
-        audio_token_per_second = self.audio_input_id_per_second / self.audio_group_size
+        audio_token_per_second = self.audio_token_per_second
 
         units = []
         for i in range(len(grid_t_timestamps)):
@@ -1363,7 +1133,7 @@ class MiMoProcessor:
         self, content_idx, content, video_results, verbose
     ):
         visual_patches, thw_grid, timestamps, video_meta = video_results[content_idx]
-        processed_audio = self.process_audio(content.content)
+        processed_audio = self.audio_pipeline.process_audio(content.content)
 
         if isinstance(processed_audio, tuple):
             assert (
@@ -2208,6 +1978,11 @@ class MiMoV2Processor(BaseMultimodalProcessor):
                         audio_source = raw_audio_item.get("url", loaded_audio)
                     elif isinstance(raw_audio_item, (str, bytes, torch.Tensor)):
                         audio_source = raw_audio_item
+                    else:
+                        raise ValueError(
+                            f"unsupported audio item: loaded={type(loaded_audio).__name__}, "
+                            f"raw={type(raw_audio_item).__name__}"
+                        )
 
                     contents.append(
                         Content(
diff --git a/python/sglang/srt/multimodal/processors/mimo_v2_asr.py b/python/sglang/srt/multimodal/processors/mimo_v2_asr.py
new file mode 100644
index 000000000..3ebba3d69
--- /dev/null
+++ b/python/sglang/srt/multimodal/processors/mimo_v2_asr.py
@@ -0,0 +1,286 @@
+"""MiMo-V2-ASR multimodal processor.
+
+Audio preprocessing is delegated to :class:`MiMoAudioPipeline`; this
+processor only handles the special-token contract and content interleaving.
+"""
+
+import asyncio
+import re
+from dataclasses import dataclass
+from typing import List, Literal, Union
+
+import numpy as np
+import torch
+
+from sglang.srt.managers.schedule_batch import (
+    Modality,
+    MultimodalDataItem,
+    MultimodalProcessorOutput,
+)
+from sglang.srt.models.mimo_v2_asr import MiMoV2ASRForCausalLM
+from sglang.srt.multimodal.processors.base_processor import (
+    BaseMultimodalProcessor,
+    MultimodalSpecialTokens,
+)
+from sglang.srt.multimodal.processors.mimo_audio import (
+    AudioInput,
+    MiMoAudioPipeline,
+)
+from sglang.utils import logger
+
+TextInput = str | list[int]
+
+
+@dataclass
+class _Content:
+    type: Literal["text", "audio"]
+    content: TextInput | AudioInput
+
+
+class MiMoV2ASRProcessor(BaseMultimodalProcessor):
+    """ASR-only MiMo processor.
+
+    Wires three special tokens into the input id stream around each audio
+    span: ``<|sosp|> <|empty|>* <|eosp|>``. The actual mel/codec preparation
+    is owned by :class:`MiMoAudioPipeline`, which is shared with the
+    multimodal MiMo-V2 processor.
+    """
+
+    models = [MiMoV2ASRForCausalLM]
+
+    AUDIO_PAD_TOKEN = "<|empty|>"
+    AUDIO_START_TOKEN = "<|sosp|>"
+    AUDIO_END_TOKEN = "<|eosp|>"
+
+    AUDIO_REGEX = re.compile(r"<\|sosp\|>(?:<\|empty\|>)+<\|eosp\|>")
+
+    def __init__(self, hf_config, server_args, _processor, *args, **kwargs):
+        super().__init__(hf_config, server_args, _processor, *args, **kwargs)
+        self.tokenizer = _processor
+
+        self.audio_pipeline = MiMoAudioPipeline(
+            audio_token_id=self._resolve_special_token_id(self.AUDIO_PAD_TOKEN),
+            audio_start_token_id=self._resolve_special_token_id(self.AUDIO_START_TOKEN),
+            audio_end_token_id=self._resolve_special_token_id(self.AUDIO_END_TOKEN),
+            audio_sampling_rate=24000,
+        )
+
+        self.mm_tokens = MultimodalSpecialTokens(
+            audio_token=f"{self.AUDIO_START_TOKEN}{self.AUDIO_PAD_TOKEN}{self.AUDIO_END_TOKEN}",
+            audio_token_id=self.audio_token_id,
+            audio_token_regex=self.AUDIO_REGEX,
+        ).build(_processor)
+
+    def __getattr__(self, name):
+        # Delegate audio_pipeline fields so callers can use self.audio_token_id
+        # etc. directly. Only triggers when normal attribute lookup fails;
+        # __dict__.get avoids recursion before audio_pipeline is assigned.
+        pipeline = self.__dict__.get("audio_pipeline")
+        if pipeline is not None and hasattr(pipeline, name):
+            return getattr(pipeline, name)
+        raise AttributeError(name)
+
+    def _resolve_special_token_id(self, name: str) -> int:
+        tid = self.tokenizer.convert_tokens_to_ids(name)
+        if tid is None or tid == self.tokenizer.unk_token_id:
+            raise ValueError(
+                f"tokenizer missing required special token {name!r}; "
+                "checkpoint vocab does not match MiMo-V2-ASR"
+            )
+        return int(tid)
+
+    def _process_contents(self, contents: List[_Content]):
+        """Run pipeline + tokenizer over an interleaved content list.
+
+        Returns ``(input_ids: Tensor[L], audio_inputs: list[Tensor],
+        position_ids: Tensor[3,L], rope_deltas: Tensor[1,1])``.
+        """
+        input_ids: List[int] = []
+        audio_inputs: List[torch.Tensor] = []
+
+        for content in contents:
+            if content.type == "text":
+                if isinstance(content.content, str):
+                    input_ids.extend(self.tokenizer.encode(content.content))
+                else:
+                    input_ids.extend(content.content)
+            elif content.type == "audio":
+                result = self.audio_pipeline.process_audio_input(content.content)
+                audio_inputs.append(result["audio_input"])
+                input_ids.extend(result["input_ids"])
+
+        ids = torch.as_tensor(input_ids)
+        position_ids = torch.arange(ids.shape[0]).expand(3, -1)
+        rope_deltas = torch.zeros((1, 1), dtype=torch.int32)
+        return ids, audio_inputs, position_ids, rope_deltas
+
+    def process_mm_data(
+        self, input_text, images=None, videos=None, audios=None, **kwargs
+    ) -> dict:
+        if audios and not self.AUDIO_REGEX.search(input_text or ""):
+            input_text = f"{self.mm_tokens.audio_token}{input_text or ''}"
+
+        processed_audios: List[Union[tuple, torch.Tensor]] = []
+        if audios:
+            for audio in audios:
+                if isinstance(audio, np.ndarray):
+                    audio_tensor = torch.from_numpy(audio).float()
+                elif isinstance(audio, torch.Tensor):
+                    audio_tensor = audio.float()
+                else:
+                    processed_audios.append(audio)
+                    continue
+                if audio_tensor.ndim == 1:
+                    processed_audios.append(
+                        (audio_tensor.cpu().contiguous(), self.audio_sampling_rate)
+                    )
+                else:
+                    processed_audios.append(audio_tensor.cpu().contiguous())
+
+        contents: List[_Content] = []
+        if input_text and processed_audios:
+            multimodal_tokens_pattern = self.mm_tokens.get_combined_regex()
+            text_parts = re.split(multimodal_tokens_pattern, input_text)
+            audio_iter = iter(processed_audios)
+
+            for text_part in text_parts:
+                if multimodal_tokens_pattern.match(text_part):
+                    modality = self.mm_tokens.get_modality_of_token(text_part)
+                    if modality == Modality.AUDIO:
+                        try:
+                            audio = next(audio_iter)
+                            contents.append(
+                                _Content(type="audio", content=AudioInput(audio=audio))
+                            )
+                        except StopIteration:
+                            pass
+                else:
+                    if text_part:
+                        contents.append(_Content(type="text", content=text_part))
+        else:
+            contents.extend(
+                _Content(type="audio", content=AudioInput(audio=audio))
+                for audio in processed_audios
+            )
+
+        if not contents:
+            ids = self.tokenizer(
+                input_text or "",
+                return_tensors="pt",
+                add_special_tokens=True,
+            ).input_ids
+            return {"input_ids": ids}
+
+        input_ids, audio_inputs, position_ids, rope_deltas = self._process_contents(
+            contents
+        )
+
+        ret: dict = {
+            "input_ids": input_ids,
+            "mrope_positions": position_ids,
+            "mrope_position_delta": rope_deltas,
+        }
+        if audio_inputs:
+            ret["audio_features"] = audio_inputs
+        return ret
+
+    async def process_mm_data_async(
+        self,
+        image_data: List[Union[str, bytes]],
+        audio_data: List[Union[str, bytes]],
+        input_text,
+        request_obj,
+        *args,
+        **kwargs,
+    ):
+        if audio_data is None:
+            audio_data = getattr(request_obj, "audio_data", [])
+        if not audio_data:
+            return None
+        if not self.AUDIO_REGEX.search(input_text):
+            input_text = f"{self.mm_tokens.audio_token}{input_text}"
+
+        base_output = await self.load_mm_data(
+            prompt=input_text,
+            image_data=[],
+            video_data=[],
+            audio_data=audio_data,
+            multimodal_tokens=self.mm_tokens,
+            audio_sample_rate=self.audio_sampling_rate,
+        )
+        multimodal_tokens_pattern = self.mm_tokens.get_combined_regex()
+
+        raw_audio_data = audio_data or []
+        loaded_audio_iter = iter(base_output.audios)
+        raw_audio_iter = iter(raw_audio_data)
+
+        text_parts = re.split(multimodal_tokens_pattern, base_output.input_text)
+        contents: List[_Content] = []
+
+        for text_part in text_parts:
+            if multimodal_tokens_pattern.match(text_part):
+                modality = self.mm_tokens.get_modality_of_token(text_part)
+                assert modality is not None
+
+                if modality == Modality.AUDIO:
+                    loaded_audio = next(loaded_audio_iter)
+                    raw_audio_item = next(raw_audio_iter)
+
+                    if isinstance(loaded_audio, np.ndarray):
+                        audio_source = loaded_audio
+                    elif isinstance(raw_audio_item, dict):
+                        audio_source = raw_audio_item.get("url", loaded_audio)
+                    elif isinstance(raw_audio_item, (str, bytes, torch.Tensor)):
+                        audio_source = raw_audio_item
+                    else:
+                        raise ValueError(
+                            f"unsupported audio item: loaded={type(loaded_audio).__name__}, "
+                            f"raw={type(raw_audio_item).__name__}"
+                        )
+
+                    contents.append(
+                        _Content(
+                            type="audio",
+                            content=AudioInput(audio=audio_source),
+                        )
+                    )
+            else:
+                if text_part:
+                    contents.append(_Content(type="text", content=text_part))
+
+        loop = asyncio.get_running_loop()
+        try:
+            input_ids, audio_inputs, position_ids, rope_deltas = (
+                await loop.run_in_executor(
+                    self.io_executor,
+                    lambda: self._process_contents(contents),
+                )
+            )
+        except RuntimeError as e:
+            logger.error(f"MiMo ASR processor failed in process_mm_data_async: {e}")
+            raise ValueError(f"Multimodal data is corrupted or cannot be decoded: {e}")
+
+        input_ids_flat = input_ids.flatten()
+        if audio_inputs:
+            mm_items = [
+                MultimodalDataItem(
+                    modality=Modality.AUDIO,
+                    feature=audio_inputs,
+                    offsets=self.get_mm_items_offset(
+                        input_ids=input_ids_flat,
+                        mm_token_id=self.audio_token_id,
+                    ),
+                )
+            ]
+        else:
+            mm_items = []
+
+        return MultimodalProcessorOutput(
+            mm_items=mm_items,
+            input_ids=input_ids_flat.tolist(),
+            audio_token_id=self.audio_token_id,
+            audio_start_id=self.audio_start_token_id,
+            audio_end_id=self.audio_end_token_id,
+            mrope_positions=position_ids,
+            mrope_position_delta=rope_deltas,
+        )