[Model] Add MiDashengLM model support (#23652)

Signed-off-by: chenbing8 <chenbing8@xiaomi.com>
Signed-off-by: bingchen-mi <chenbing8@xiaomi.com>
Co-authored-by: Jee Jee Li <pandaleefree@gmail.com>
Co-authored-by: Isotr0py <mozf@mail2.sysu.edu.cn>

docs/models/supported_models.md

@@ -643,6 +643,7 @@ These models primarily accept the [`LLM.generate`](./generative_models.md#llmgen
 | `LlavaNextForConditionalGeneration` | LLaVA-NeXT | T + I<sup>E+</sup> | `llava-hf/llava-v1.6-mistral-7b-hf`, `llava-hf/llava-v1.6-vicuna-7b-hf`, etc. | | ✅︎ | ✅︎ |
 | `LlavaNextVideoForConditionalGeneration` | LLaVA-NeXT-Video | T + V | `llava-hf/LLaVA-NeXT-Video-7B-hf`, etc. | | ✅︎ | ✅︎ |
 | `LlavaOnevisionForConditionalGeneration` | LLaVA-Onevision | T + I<sup>+</sup> + V<sup>+</sup> | `llava-hf/llava-onevision-qwen2-7b-ov-hf`, `llava-hf/llava-onevision-qwen2-0.5b-ov-hf`, etc. | | ✅︎ | ✅︎ |
+| `MiDashengLMModel` | MiDashengLM | T + A<sup>+</sup> | `mispeech/midashenglm-7b` | | ✅︎ | ✅︎ |
 | `MiniCPMO` | MiniCPM-O | T + I<sup>E+</sup> + V<sup>E+</sup> + A<sup>E+</sup> | `openbmb/MiniCPM-o-2_6`, etc. | ✅︎ | ✅︎ | ✅︎ |
 | `MiniCPMV` | MiniCPM-V | T + I<sup>E+</sup> + V<sup>E+</sup> | `openbmb/MiniCPM-V-2` (see note), `openbmb/MiniCPM-Llama3-V-2_5`, `openbmb/MiniCPM-V-2_6`, `openbmb/MiniCPM-V-4`, `openbmb/MiniCPM-V-4_5`, etc. | ✅︎ | | ✅︎ |
 | `MiniMaxVL01ForConditionalGeneration` | MiniMax-VL | T + I<sup>E+</sup> | `MiniMaxAI/MiniMax-VL-01`, etc. | | ✅︎ | ✅︎ |

examples/offline_inference/audio_language.py

@@ -146,6 +146,36 @@ def run_granite_speech(question: str, audio_count: int) -> ModelRequestData:
     )
 
 
+# MiDashengLM
+def run_midashenglm(question: str, audio_count: int):
+    model_name = "mispeech/midashenglm-7b"
+
+    engine_args = EngineArgs(
+        model=model_name,
+        trust_remote_code=True,
+        max_model_len=4096,
+        max_num_seqs=5,
+        limit_mm_per_prompt={"audio": audio_count},
+    )
+
+    audio_in_prompt = "".join(
+        ["<|audio_bos|><|AUDIO|><|audio_eos|>" for idx in range(audio_count)]
+    )
+
+    default_system = "You are a helpful language and speech assistant."
+
+    prompt = (
+        f"<|im_start|>system\n{default_system}<|im_end|>\n"
+        "<|im_start|>user\n"
+        f"{audio_in_prompt}{question}<|im_end|>\n"
+        "<|im_start|>assistant\n"
+    )
+    return ModelRequestData(
+        engine_args=engine_args,
+        prompt=prompt,
+    )
+
+
 # MiniCPM-O
 def run_minicpmo(question: str, audio_count: int) -> ModelRequestData:
     model_name = "openbmb/MiniCPM-o-2_6"
@@ -352,6 +382,7 @@ model_example_map = {
     "voxtral": run_voxtral,
     "gemma3n": run_gemma3n,
     "granite_speech": run_granite_speech,
+    "midashenglm": run_midashenglm,
     "minicpmo": run_minicpmo,
     "phi4_mm": run_phi4mm,
     "phi4_multimodal": run_phi4_multimodal,

tests/models/multimodal/processing/test_common.py

@@ -302,6 +302,7 @@ def _test_processing_correctness_one(
     "llava-hf/llava-onevision-qwen2-0.5b-ov-hf",
     "meta-llama/Llama-3.2-11B-Vision-Instruct",
     "TIGER-Lab/Mantis-8B-siglip-llama3",
+    "mispeech/midashenglm-7b",
     "openbmb/MiniCPM-Llama3-V-2_5",
     "openbmb/MiniCPM-o-2_6",
     "openbmb/MiniCPM-V-2_6",

tests/models/registry.py

@@ -460,6 +460,8 @@ _MULTIMODAL_EXAMPLE_MODELS = {
                                                       max_transformers_version="4.48",  # noqa: E501
                                                       transformers_version_reason="HF model is not compatible.",  # noqa: E501
                                                       hf_overrides={"architectures": ["MantisForConditionalGeneration"]}),  # noqa: E501
+    "MiDashengLMModel": _HfExamplesInfo("mispeech/midashenglm-7b",
+                            trust_remote_code=True),
     "MiniCPMO": _HfExamplesInfo("openbmb/MiniCPM-o-2_6",
                                 trust_remote_code=True),
     "MiniCPMV": _HfExamplesInfo("openbmb/MiniCPM-Llama3-V-2_5",

vllm/model_executor/models/midashenglm.py

@@ -0,0 +1,788 @@
+# SPDX-License-Identifier: Apache-2.0
+# SPDX-FileCopyrightText: Copyright contributors to the vLLM project
+# Copyright 2025 Horizon team, Xiaomi MiLM Plus.
+# Copyright 2024 The Qwen team.
+# Copyright 2023 The vLLM team.
+# Copyright 2022 EleutherAI and the HuggingFace Inc. team. All rights reserved.
+#
+# This code is based on EleutherAI's GPT-NeoX library and the GPT-NeoX
+# and OPT implementations in this library. It has been modified from its
+# original forms to accommodate minor architectural differences compared
+# to GPT-NeoX and OPT used by the Meta AI team that trained the model.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+"""Inference-only MiDashengLM model compatible with HuggingFace weights."""
+import collections
+import collections.abc
+from collections.abc import Iterable, Mapping, Sequence
+from typing import Any, Callable, Optional, TypedDict, Union, cast
+
+import numpy as np
+import torch
+import torch.nn as nn
+import torchaudio.transforms as audio_transforms
+from transformers import BatchFeature
+
+from vllm.attention.layer import MultiHeadAttention
+from vllm.config import VllmConfig
+from vllm.distributed import get_tensor_model_parallel_world_size
+from vllm.model_executor.layers.activation import get_act_fn
+from vllm.model_executor.layers.linear import (ColumnParallelLinear,
+                                               QKVParallelLinear,
+                                               RowParallelLinear)
+from vllm.model_executor.layers.quantization import QuantizationConfig
+from vllm.model_executor.model_loader.utils import set_default_torch_dtype
+from vllm.model_executor.sampling_metadata import SamplingMetadata
+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 (BaseMultiModalProcessor,
+                                        BaseProcessingInfo, PromptReplacement,
+                                        PromptUpdate, PromptUpdateDetails)
+from vllm.multimodal.profiling import BaseDummyInputsBuilder
+from vllm.sequence import IntermediateTensors
+from vllm.transformers_utils.configs.midashenglm import DashengConfig
+
+from .interfaces import MultiModalEmbeddings, SupportsMultiModal, SupportsPP
+from .utils import (AutoWeightsLoader, init_vllm_registered_model,
+                    maybe_prefix, merge_multimodal_embeddings)
+
+_Tuple2 = Union[int, tuple[int, int], Sequence[int]]
+
+
+def _resolve_tuple2(x: _Tuple2) -> tuple[int, int]:
+    if isinstance(x, collections.abc.Sequence):
+        assert len(x) == 2, (
+            f"Expected a sequence of length 2, got {x} with length {len(x)}")
+        return cast(tuple[int, int], tuple(x))
+    return (x, x)
+
+
+def calculate_mel_frames_dasheng(
+    audio_length_samples: int,
+    n_fft: int = 512,
+    hop_size: int = 160,
+    dasheng_subsampling: int = 4,
+    center=True,
+    model_subsampling: int = 5,
+) -> int:
+    """Calculate the number of Mel-spectrogram frames."""
+    if center:
+        audio_length_samples = audio_length_samples + n_fft
+
+    return (int(1 + ((audio_length_samples - n_fft) / hop_size)) //
+            dasheng_subsampling // model_subsampling)
+
+
+class AudioPatchEmbed(nn.Module):
+
+    def __init__(
+        self,
+        input_size: _Tuple2 = 64,
+        patch_size: _Tuple2 = 16,
+        patch_stride: _Tuple2 = 16,
+        in_chans: int = 1,
+        embed_dim: int = 768,
+        norm_layer: Optional[Callable] = None,
+        flatten: bool = False,
+    ):
+        super().__init__()
+        self.input_size = _resolve_tuple2(input_size)
+        self.patch_size = _resolve_tuple2(patch_size)
+        self.patch_stride = _resolve_tuple2(patch_stride)
+        self.grid_size = (
+            self.input_size[0] // self.patch_stride[0],
+            self.input_size[1] // self.patch_stride[1],
+        )
+        self.num_patches = self.grid_size[0] * self.grid_size[1]
+        self.flatten = flatten
+
+        self.proj = nn.Conv2d(
+            in_chans,
+            embed_dim,
+            kernel_size=self.patch_size,
+            stride=self.patch_stride,
+        )
+        self.norm = norm_layer(embed_dim) if norm_layer else nn.Identity()
+
+    def forward(self, x: torch.Tensor) -> torch.Tensor:
+        x = self.proj(x)
+        if self.flatten:
+            x = torch.permute(torch.flatten(
+                x, 2, 3), (0, 2, 1))  # rearrange(x, "b c f t -> b (f t) c")
+        x = self.norm(x)
+        return x
+
+
+class LayerScale(nn.Module):
+
+    def __init__(self, dim, init_values=1e-5, inplace=False):
+        super().__init__()
+        self.inplace = inplace
+        self.gamma = nn.Parameter(init_values * torch.ones(dim))
+
+    def forward(self, x: torch.Tensor) -> torch.Tensor:
+        return x.mul_(self.gamma) if self.inplace else x * self.gamma
+
+
+class DashengMlp(nn.Module):
+
+    def __init__(
+        self,
+        in_features: int,
+        hidden_features: Optional[int] = None,
+        out_features: Optional[int] = None,
+        quant_config: Optional[QuantizationConfig] = None,
+        prefix: str = "",
+    ):
+        super().__init__()
+        out_features = out_features or in_features
+        hidden_features = hidden_features or in_features
+        self.fc1 = ColumnParallelLinear(input_size=in_features,
+                                        output_size=hidden_features,
+                                        quant_config=quant_config,
+                                        prefix=f"{prefix}.fc1")
+        self.act = get_act_fn("gelu")
+        self.fc2 = RowParallelLinear(input_size=hidden_features,
+                                     output_size=out_features,
+                                     quant_config=quant_config,
+                                     prefix=f"{prefix}.fc2")
+
+    def forward(self, x: torch.Tensor) -> torch.Tensor:
+        x, _ = self.fc1(x)
+        x = self.act(x)
+        x, _ = self.fc2(x)
+        return x
+
+
+class DashengAttention(nn.Module):
+
+    def __init__(
+        self,
+        dim: int,
+        num_heads: int = 8,
+        qkv_bias: bool = False,
+        causal: bool = False,
+        quant_config: Optional[QuantizationConfig] = None,
+        prefix: str = "",
+    ):
+        super().__init__()
+        assert dim % num_heads == 0, "dim should be divisible by num_heads"
+        self.embed_dim = dim
+        tp_size = get_tensor_model_parallel_world_size()
+        self.total_num_heads = num_heads
+        assert self.total_num_heads % tp_size == 0
+        self.num_heads = self.total_num_heads // tp_size
+        if self.total_num_heads >= tp_size:
+            # Number of heads is greater than TP size, so we partition
+            # the KV heads across multiple tensor parallel GPUs.
+            assert self.total_num_heads % tp_size == 0
+        else:
+            # Number of heads is less than TP size, so we replicate
+            # the KV heads across multiple tensor parallel GPUs.
+            assert tp_size % self.total_num_heads == 0
+        self.num_kv_heads = max(1, self.total_num_heads // tp_size)
+        self.head_dim = self.embed_dim // self.total_num_heads
+        self.q_size = self.num_heads * self.head_dim
+        self.kv_size = self.num_kv_heads * self.head_dim
+        self.scale = self.head_dim**-0.5
+
+        self.qkv = QKVParallelLinear(
+            hidden_size=self.embed_dim,
+            head_size=self.head_dim,
+            total_num_heads=self.total_num_heads,
+            total_num_kv_heads=self.total_num_heads,
+            bias=qkv_bias,
+            quant_config=quant_config,
+            prefix=f"{prefix}.qkv",
+        )
+        self.attn = MultiHeadAttention(
+            self.num_heads,
+            self.head_dim,
+            self.scale,
+            num_kv_heads=self.num_kv_heads,
+        )
+        self.proj = RowParallelLinear(
+            input_size=dim,
+            output_size=dim,
+            quant_config=quant_config,
+            prefix=f"{prefix}.proj",
+        )
+        self.causal = causal
+
+    def forward(self, x: torch.Tensor, mask: Optional[torch.Tensor] = None):
+        B, N, C = x.shape
+
+        qkv_out, _ = self.qkv(x)
+        q, k, v = qkv_out.split([self.q_size, self.kv_size, self.kv_size],
+                                dim=-1)
+
+        attn_out = self.attn(q, k, v)
+        C_local = attn_out.numel() // (B * N)  # C_local for parallel
+        attn_out = attn_out.view(B, N, C_local)
+
+        x, _ = self.proj(attn_out)
+
+        return x
+
+
+class DashengBlock(nn.Module):
+
+    def __init__(
+        self,
+        dim: int,
+        num_heads: int,
+        mlp_ratio: float = 4.0,
+        qkv_bias: bool = False,
+        init_values: Optional[float] = None,
+        quant_config: Optional[QuantizationConfig] = None,
+        prefix: str = "",
+    ):
+        super().__init__()
+        self.norm1 = nn.LayerNorm(dim, eps=1e-6)
+        self.attn = DashengAttention(
+            dim,
+            num_heads=num_heads,
+            qkv_bias=qkv_bias,
+            quant_config=quant_config,
+            prefix=f"{prefix}.attn",
+        )
+        self.ls1 = (LayerScale(dim, init_values=init_values)
+                    if init_values else nn.Identity())
+
+        self.norm2 = nn.LayerNorm(dim, eps=1e-6)
+        self.mlp = DashengMlp(
+            in_features=dim,
+            hidden_features=int(dim * mlp_ratio),
+            quant_config=quant_config,
+            prefix=f"{prefix}.mlp",
+        )
+        self.ls2 = (LayerScale(dim, init_values=init_values)
+                    if init_values else nn.Identity())
+
+    # Kwargs usually has a mask parameter that is passed to Attention
+    def forward(
+        self,
+        x: torch.Tensor,
+        mask: Optional[torch.Tensor] = None,
+    ) -> torch.Tensor:
+        x = x + self.ls1(self.attn(self.norm1(x), mask))
+        x = x + self.ls2(self.mlp(self.norm2(x)))
+        return x
+
+
+class DashengAudioTransformer(nn.Module):
+
+    def __init__(
+        self,
+        config: DashengConfig,
+        quant_config: Optional[QuantizationConfig] = None,
+        prefix: str = "",
+    ):
+        super().__init__()
+
+        self.target_length = config.target_length
+        self.hop_length = config.hop_length
+
+        self._init_front_end(config)
+
+        self.init_bn = nn.BatchNorm2d(config.n_mels, momentum=0.01)
+
+        self.patch_embed = AudioPatchEmbed(
+            input_size=(config.n_mels, config.target_length),
+            embed_dim=config.embed_dim,
+            in_chans=config.input_channels,
+            patch_size=config.patch_size,
+            flatten=False,
+            patch_stride=config.patch_stride,
+        )
+
+        self.time_pos_embed = nn.Parameter(
+            torch.empty(1, config.embed_dim, 1, self.patch_embed.grid_size[1]))
+        self.freq_pos_embed = nn.Parameter(
+            torch.empty(1, config.embed_dim, self.patch_embed.grid_size[0], 1))
+        self.blocks = nn.ModuleList(
+            DashengBlock(
+                dim=config.embed_dim,
+                num_heads=config.num_heads,
+                mlp_ratio=config.mlp_ratio,
+                qkv_bias=config.qkv_bias,
+                init_values=config.init_values,
+                quant_config=quant_config,
+                prefix=f"{prefix}.block{i}",
+            ) for i in range(config.depth))
+        self.norm = nn.LayerNorm(config.embed_dim, eps=1e-6)
+
+    def _init_front_end(self, config):
+        with set_default_torch_dtype(torch.float32):
+            self.front_end = nn.Sequential(
+                audio_transforms.MelSpectrogram(
+                    f_min=config.f_min,
+                    f_max=config.f_max,
+                    center=config.center,
+                    win_length=config.win_length,
+                    hop_length=config.hop_length,
+                    sample_rate=config.sample_rate,
+                    n_fft=config.n_fft,
+                    n_mels=config.n_mels,
+                ),
+                audio_transforms.AmplitudeToDB(top_db=120),
+            )
+
+            mel_spectrogram = self.front_end[0]
+            fb = mel_spectrogram.mel_scale.fb
+            win = mel_spectrogram.spectrogram.window
+            mel_spectrogram.mel_scale.fb = fb.to(torch.bfloat16).to(
+                torch.float32)
+            mel_spectrogram.spectrogram.window = win.to(torch.bfloat16).to(
+                torch.float32)
+
+    def forward_features(
+        self,
+        x: torch.Tensor,
+        mask: Optional[torch.Tensor] = None,
+    ) -> torch.Tensor:
+        t = x.shape[-1]
+        x = x + self.time_pos_embed[:, :, :, :t]
+        x = (x + self.freq_pos_embed[:, :, :, :]
+             )  # Just to support __getitem__ in posembed
+        x = torch.permute(torch.flatten(x, 2, 3),
+                          (0, 2, 1))  # rearrange(x, "b c f t -> b (f t) c")
+        for block in self.blocks:
+            x = block(x, mask)
+        x = self.norm(x)
+        return x
+
+    def _to_mask(self, lengths: torch.Tensor, max_length: int) -> torch.Tensor:
+        batch_size = len(lengths)
+        idx = torch.arange(max_length, device=lengths.device)
+        idx = idx.repeat(batch_size).view(batch_size, max_length)
+        mask = (idx < lengths.unsqueeze(-1)).bool()
+        return mask
+
+    def forward(
+        self,
+        x: torch.Tensor,
+        x_length: Optional[torch.Tensor] = None,
+    ) -> tuple[torch.Tensor, Optional[torch.Tensor]]:
+        x = self.front_end(x)
+        x = x.to(self.time_pos_embed.dtype)
+        target_length_in_patches = self.target_length // 4
+        x = x.unsqueeze(1)
+        x = torch.permute(x, (0, 2, 1, 3))
+        x = self.init_bn(x)
+        x = torch.permute(x, (0, 2, 1, 3))
+
+        x = self.patch_embed(x)
+        t = x.shape[-1]
+
+        input_splits = x.split(target_length_in_patches, dim=-1)
+
+        if x_length is not None:
+            assert len(x_length) == len(x), (
+                "batchsizes of input x and x_length need to be same")
+            assert x_length.ndim == 1, "Lengths are of size (B,)"
+            scaled_lengths = (x_length / (self.hop_length * 4)).long()
+            mask = self._to_mask(max_length=t, lengths=scaled_lengths)
+            split_masks = mask.logical_not().split(target_length_in_patches,
+                                                   dim=-1)
+        else:
+            mask = None
+            split_masks = [None] * len(input_splits)
+
+        outputs = []
+
+        for split_x, split_mask in zip(input_splits, split_masks):
+            forward_kwargs = {}
+            forward_kwargs["mask"] = split_mask
+            split_x = self.forward_features(split_x, **forward_kwargs)
+            outputs.append(split_x)
+        x = torch.cat(outputs, dim=1)
+        return x, mask
+
+
+class AudioProjectorSubsample(nn.Module):
+
+    def __init__(
+        self,
+        in_dim: int,
+        out_dim: int,
+        downsample_rate=5,
+        dtype: Optional[torch.dtype] = None,
+        quant_config: Optional[QuantizationConfig] = None,
+        prefix: str = "",
+    ):
+        super().__init__()
+        self.k = downsample_rate
+        self.net = nn.Sequential(
+            ColumnParallelLinear(
+                input_size=in_dim * self.k,
+                output_size=out_dim,
+                quant_config=quant_config,
+                prefix=f"{prefix}.net.0",
+                return_bias=False,
+            ), get_act_fn("gelu"),
+            RowParallelLinear(
+                input_size=out_dim,
+                output_size=out_dim,
+                quant_config=quant_config,
+                prefix=f"{prefix}.net.2",
+                return_bias=False,
+            ))
+
+    def forward(self, x, mask=None):
+        batch_size, seq_len, dim = x.shape
+        num_frames_to_discard = seq_len % self.k
+        if num_frames_to_discard > 0:
+            x = x[:, :-num_frames_to_discard, :]
+            if mask is not None:
+                mask = mask[:, :-num_frames_to_discard]
+        if mask is None:
+            mask = torch.ones(x.shape[:-1], dtype=torch.long, device=x.device)
+        x = x.reshape(batch_size, -1, self.k *
+                      dim)  # rearrange(x, "b (s k) d -> b s (k d)", k=self.k)
+        for layer in self.net:
+            x = layer(x)
+        mask = mask.reshape(
+            batch_size, -1,
+            self.k)  # rearrange(mask, "b (s k) -> b s k", k=self.k)
+        mask = mask.any(dim=-1).long()
+        return x, mask
+
+
+# === Audio Inputs === #
+class MiDashengLMAudioInputs(TypedDict):
+    input_values: torch.Tensor
+    """Shape: `(num_audios, num_sampling_points)`"""
+    audio_length: torch.Tensor
+    """Shape: `(num_audios, 1)`"""
+
+
+class MiDashengLMProcessingInfo(BaseProcessingInfo):
+
+    def get_hf_config(self):
+        return self.ctx.get_hf_config()
+
+    def get_feature_extractor(self):
+        hf_processor = self.get_hf_processor()
+        feature_extractor = hf_processor.feature_extractor
+        return feature_extractor
+
+    def get_supported_mm_limits(self) -> Mapping[str, Optional[int]]:
+        return {"audio": None}
+
+    def get_min_audio_len(self):
+        return 3200
+
+    def get_max_audio_len(self):
+        return 160000
+
+
+class MiDashengLMDummyInputsBuilder(
+        BaseDummyInputsBuilder[MiDashengLMProcessingInfo]):
+
+    def get_dummy_text(self, mm_counts: Mapping[str, int]) -> str:
+        num_audios = mm_counts.get("audio", 0)
+
+        hf_processor = self.info.get_hf_processor()
+        audio_token = hf_processor.audio_token
+
+        return audio_token * num_audios
+
+    def get_dummy_mm_data(
+        self,
+        seq_len: int,
+        mm_counts: Mapping[str, int],
+    ) -> MultiModalDataDict:
+        num_audios = mm_counts.get("audio", 0)
+
+        return {
+            "audio":
+            self._get_dummy_audios(length=self.info.get_max_audio_len(),
+                                   num_audios=num_audios)
+        }
+
+
+class MiDashengLMMultiModalProcessor(
+        BaseMultiModalProcessor[MiDashengLMProcessingInfo]):
+
+    def _get_data_parser(self) -> MultiModalDataParser:
+        feature_extractor = self.info.get_feature_extractor()
+        return MultiModalDataParser(target_sr=feature_extractor.sampling_rate)
+
+    def _call_hf_processor(
+        self,
+        prompt: str,
+        mm_data: Mapping[str, object],
+        mm_kwargs: Mapping[str, Any],
+        tok_kwargs: Mapping[str, object],
+    ) -> BatchFeature:
+        audios = mm_data.pop("audios", [])
+
+        # + Padding
+        min_audio_len = self.info.get_min_audio_len()
+        processed_audios = [
+            np.pad(audio, (0, min_audio_len - audio.shape[-1]),
+                   mode='constant',
+                   constant_values=0) if isinstance(audio, np.ndarray)
+            and audio.shape[-1] < min_audio_len else audio for audio in audios
+        ]
+
+        if processed_audios:
+            mm_data["audio"] = processed_audios
+
+        if not mm_data.get("audio", []):
+            prompt_ids = self.info.get_tokenizer().encode(prompt)
+            prompt_ids = self._apply_hf_processor_tokens_only(prompt_ids)
+            return BatchFeature(dict(input_ids=[prompt_ids]), tensor_type="pt")
+
+        mm_kwargs = dict(**mm_kwargs, )
+
+        return super()._call_hf_processor(
+            prompt=prompt,
+            mm_data=mm_data,
+            mm_kwargs=mm_kwargs,
+            tok_kwargs=tok_kwargs,
+        )
+
+    def _get_mm_fields_config(
+        self,
+        hf_inputs: BatchFeature,
+        hf_processor_mm_kwargs: Mapping[str, object],
+    ) -> Mapping[str, MultiModalFieldConfig]:
+        return dict(
+            input_values=MultiModalFieldConfig.batched("audio"),
+            audio_length=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()
+
+        audio_token = getattr(processor, "audio_token", "<|AUDIO|>")
+        audio_bos_token = getattr(processor, "audio_bos_token",
+                                  "<|audio_bos|>")
+        audio_eos_token = getattr(processor, "audio_eos_token",
+                                  "<|audio_eos|>")
+
+        audio_token_id = vocab[audio_token]
+        audio_bos_id = vocab[audio_bos_token]
+        audio_eos_id = vocab[audio_eos_token]
+
+        out_mm_data = out_mm_kwargs.get_data()
+        audio_length = out_mm_data.get("audio_length")
+        if audio_length is None:
+            audio_output_lengths = []
+        else:
+            audio_length_np = audio_length.cpu().numpy() if isinstance(
+                audio_length, torch.Tensor) else audio_length
+            audio_output_lengths = [
+                max(1, calculate_mel_frames_dasheng(
+                    int(length)))  # at least one frame
+                for length in audio_length_np
+            ]
+
+        def get_replacement_midashenglm(item_idx: int):
+            num_features = audio_output_lengths[item_idx]
+            audio_tokens = [audio_token_id] * num_features
+
+            return PromptUpdateDetails.select_token_id(
+                [audio_bos_id] + audio_tokens + [audio_eos_id],
+                embed_token_id=audio_token_id,
+            )
+
+        return [
+            PromptReplacement(
+                modality="audio",
+                target=audio_token,
+                replacement=get_replacement_midashenglm,
+            )
+        ]
+
+
+@MULTIMODAL_REGISTRY.register_processor(
+    MiDashengLMMultiModalProcessor,
+    info=MiDashengLMProcessingInfo,
+    dummy_inputs=MiDashengLMDummyInputsBuilder,
+)
+class MiDashengLMModel(nn.Module, SupportsMultiModal, SupportsPP):
+
+    @classmethod
+    def get_placeholder_str(cls, modality: str, i: int) -> Optional[str]:
+        if modality.startswith("audio"):
+            return "<|audio_bos|><|AUDIO|><|audio_eos|>"
+
+        raise ValueError("Only audio modality is supported")
+
+    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
+
+        # Initialize audio components
+        self.audio_encoder = DashengAudioTransformer(
+            config.audio_encoder_config,
+            quant_config=quant_config,
+            prefix=maybe_prefix(prefix, "audio_encoder"),
+        )
+        self.audio_projector = AudioProjectorSubsample(
+            in_dim=config.audio_encoder_config.embed_dim,
+            out_dim=config.text_config.hidden_size,
+            downsample_rate=config.subsample_factor,
+            quant_config=quant_config,
+            prefix=maybe_prefix(prefix, "audio_projector"),
+        )
+
+        # Initialize language model (decoder)
+        self.decoder = init_vllm_registered_model(
+            vllm_config=vllm_config,
+            hf_config=config.text_config,
+            prefix=maybe_prefix(prefix, "decoder"),
+            architectures=["Qwen2ForCausalLM"],
+        )
+
+        self.quant_config = quant_config
+        self.make_empty_intermediate_tensors = (
+            self.decoder.make_empty_intermediate_tensors)
+
+    def _validate_and_reshape_mm_tensor(self, mm_input: object,
+                                        name: str) -> torch.Tensor:
+        if not isinstance(mm_input, (torch.Tensor, list)):
+            raise ValueError(f"Incorrect type of {name}. "
+                             f"Got type: {type(mm_input)}")
+        if isinstance(mm_input, torch.Tensor):
+            return torch.concat(list(mm_input))
+        else:
+            return torch.concat(mm_input)
+
+    def _parse_and_validate_audio_input(
+            self, **kwargs: object) -> Optional[MiDashengLMAudioInputs]:
+        input_values = kwargs.pop("input_values", None)
+        audio_length = kwargs.pop("audio_length", None)
+
+        if input_values is None:
+            return None
+        input_values = self._validate_and_reshape_mm_tensor(
+            input_values, "input_values")
+        audio_length = self._validate_and_reshape_mm_tensor(
+            audio_length, "audio_length")
+        if not isinstance(input_values, (torch.Tensor, list)):
+            raise ValueError("Incorrect type of audio input features. "
+                             f"Got type: {type(input_values)}")
+
+        return MiDashengLMAudioInputs(
+            input_values=input_values,
+            audio_length=audio_length,
+        )
+
+    def _process_audio_input(
+            self, audio_input: MiDashengLMAudioInputs) -> torch.Tensor:
+        # Process audio through encoder and projector
+        input_values = audio_input["input_values"]
+        audio_length = audio_input["audio_length"]
+
+        encoder_out, encoder_atts = self.audio_encoder(input_values,
+                                                       audio_length)
+        audio_embeddings, _ = self.audio_projector(encoder_out, encoder_atts)
+        audio_embeddings = audio_embeddings.to(
+            audio_input["input_values"].dtype)
+        batch_size, max_audio_tokens, embed_dim = audio_embeddings.shape
+
+        audio_length_np = audio_length.cpu().numpy() if isinstance(
+            audio_length, torch.Tensor) else audio_length
+        audio_output_lengths = [
+            max(1, calculate_mel_frames_dasheng(
+                int(length)))  # at least one frame
+            for length in audio_length_np
+        ]
+        audio_output_lengths = torch.tensor(audio_output_lengths).to(
+            audio_embeddings.device)
+
+        audio_feature_mask = (torch.arange(
+            max_audio_tokens,
+            device=audio_embeddings.device).unsqueeze(0).expand(
+                batch_size, max_audio_tokens)
+                              < audio_output_lengths.unsqueeze(1))
+
+        masked_audio_features = audio_embeddings[audio_feature_mask].view(
+            -1, embed_dim)
+
+        return torch.split(masked_audio_features,
+                           audio_output_lengths.tolist())
+
+    def get_language_model(self) -> torch.nn.Module:
+        return self.decoder
+
+    def get_multimodal_embeddings(self,
+                                  **kwargs: object) -> MultiModalEmbeddings:
+        audio_input = self._parse_and_validate_audio_input(**kwargs)
+
+        if audio_input is None:
+            return []
+        return self._process_audio_input(audio_input)
+
+    def get_input_embeddings(
+        self,
+        input_ids: torch.Tensor,
+        multimodal_embeddings: Optional[MultiModalEmbeddings] = None,
+    ) -> torch.Tensor:
+        inputs_embeds = self.decoder.get_input_embeddings(input_ids)
+        if multimodal_embeddings and len(multimodal_embeddings) > 0:
+            inputs_embeds = merge_multimodal_embeddings(
+                input_ids,
+                inputs_embeds,
+                multimodal_embeddings,
+                self.config.audio_token_id,
+            )
+        return inputs_embeds
+
+    def forward(
+        self,
+        input_ids: torch.Tensor,
+        positions: torch.Tensor,
+        intermediate_tensors: Optional[IntermediateTensors] = None,
+        inputs_embeds: Optional[torch.Tensor] = None,
+        **kwargs: object,
+    ) -> Union[torch.Tensor, IntermediateTensors]:
+        if intermediate_tensors is not None:
+            inputs_embeds = None
+        elif inputs_embeds is None:
+            multimodal_embeddings = self.get_multimodal_embeddings(**kwargs)
+            inputs_embeds = self.get_input_embeddings(input_ids,
+                                                      multimodal_embeddings)
+            input_ids = None
+
+        return self.decoder.model(input_ids,
+                                  positions,
+                                  intermediate_tensors,
+                                  inputs_embeds=inputs_embeds)
+
+    def compute_logits(
+        self,
+        hidden_states: torch.Tensor,
+        sampling_metadata: SamplingMetadata,
+    ) -> Optional[torch.Tensor]:
+        return self.decoder.compute_logits(hidden_states, sampling_metadata)
+
+    def load_weights(self, weights: Iterable[tuple[str,
+                                                   torch.Tensor]]) -> set[str]:
+        loader = AutoWeightsLoader(self)
+        return loader.load_weights(weights)

vllm/model_executor/models/registry.py

@@ -236,6 +236,7 @@ _MULTIMODAL_MODELS = {
     "LlavaNextVideoForConditionalGeneration": ("llava_next_video", "LlavaNextVideoForConditionalGeneration"),  # noqa: E501
     "LlavaOnevisionForConditionalGeneration": ("llava_onevision", "LlavaOnevisionForConditionalGeneration"),  # noqa: E501
     "MantisForConditionalGeneration": ("llava", "MantisForConditionalGeneration"),  # noqa: E501
+    "MiDashengLMModel": ("midashenglm", "MiDashengLMModel"),
     "MiniMaxVL01ForConditionalGeneration": ("minimax_vl_01", "MiniMaxVL01ForConditionalGeneration"),  # noqa: E501
     "MiniCPMO": ("minicpmo", "MiniCPMO"),
     "MiniCPMV": ("minicpmv", "MiniCPMV"),

vllm/transformers_utils/config.py

@@ -71,6 +71,7 @@ _CONFIG_REGISTRY: dict[str, type[PretrainedConfig]] = LazyConfigDict(
     jais="JAISConfig",
     mlp_speculator="MLPSpeculatorConfig",
     medusa="MedusaConfig",
+    midashenglm="MiDashengLMConfig",
     eagle="EAGLEConfig",
     speculators="SpeculatorsConfig",
     nemotron="NemotronConfig",

vllm/transformers_utils/configs/__init__.py

@@ -17,6 +17,7 @@ from vllm.transformers_utils.configs.falcon import RWConfig
 from vllm.transformers_utils.configs.jais import JAISConfig
 from vllm.transformers_utils.configs.kimi_vl import KimiVLConfig
 from vllm.transformers_utils.configs.medusa import MedusaConfig
+from vllm.transformers_utils.configs.midashenglm import MiDashengLMConfig
 from vllm.transformers_utils.configs.mlp_speculator import MLPSpeculatorConfig
 from vllm.transformers_utils.configs.moonvit import MoonViTConfig
 from vllm.transformers_utils.configs.nemotron import NemotronConfig
@@ -36,6 +37,7 @@ __all__ = [
     "RWConfig",
     "JAISConfig",
     "MedusaConfig",
+    "MiDashengLMConfig",
     "MLPSpeculatorConfig",
     "MoonViTConfig",
     "KimiVLConfig",

vllm/transformers_utils/configs/midashenglm.py

@@ -0,0 +1,101 @@
+# SPDX-License-Identifier: Apache-2.0
+# SPDX-FileCopyrightText: Copyright contributors to the vLLM project
+# Copyright 2025 Horizon team, Xiaomi MiLM Plus.
+# Copyright 2024 The Qwen team.
+# Copyright 2023 The vLLM team.
+# Copyright 2022 EleutherAI and the HuggingFace Inc. team. All rights reserved.
+#
+# This code is based on EleutherAI's GPT-NeoX library and the GPT-NeoX
+# and OPT implementations in this library. It has been modified from its
+# original forms to accommodate minor architectural differences compared
+# to GPT-NeoX and OPT used by the Meta AI team that trained the model.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+from typing import Optional, Union
+
+from transformers import PretrainedConfig
+from transformers.models.qwen2_5_omni.configuration_qwen2_5_omni import (
+    Qwen2_5OmniTextConfig)
+
+
+class DashengConfig(PretrainedConfig):
+    model_type = "midashenglm_dasheng_encoder"
+
+    def __init__(
+        self,
+        embed_dim: int = 768,
+        outputdim: int = 527,
+        patch_size: Union[int, tuple[int, int]] = 16,
+        patch_stride: Union[int, tuple[int, int]] = 16,
+        input_channels: int = 1,
+        target_length: int = 1012,
+        depth: int = 12,
+        num_heads: int = 12,
+        mlp_ratio: float = 4.0,
+        qkv_bias: bool = True,
+        init_values: Optional[float] = None,
+        drop_rate: float = 0.0,
+        attn_drop_rate: float = 0.0,
+        f_min: float = 0.0,
+        f_max: float = 8000.0,
+        center: bool = True,
+        win_length: int = 512,
+        hop_length: int = 160,
+        sample_rate: int = 16000,
+        n_fft: int = 512,
+        n_mels: int = 64,
+        **kwargs,
+    ):
+        self.embed_dim = embed_dim
+        self.outputdim = outputdim
+        self.patch_size = patch_size
+        self.patch_stride = patch_stride
+        self.input_channels = input_channels
+        self.target_length = target_length
+        self.depth = depth
+        self.num_heads = num_heads
+        self.mlp_ratio = mlp_ratio
+        self.qkv_bias = qkv_bias
+        self.init_values = init_values
+        self.drop_rate = drop_rate
+        self.attn_drop_rate = attn_drop_rate
+        self.f_min = f_min
+        self.f_max = f_max
+        self.center = center
+        self.win_length = win_length
+        self.hop_length = hop_length
+        self.sample_rate = sample_rate
+        self.n_fft = n_fft
+        self.n_mels = n_mels
+        super().__init__(**kwargs)
+
+
+class MiDashengLMConfig(PretrainedConfig):
+    model_type = "midashenglm"
+
+    def __init__(
+        self,
+        audio_encoder_config: Optional[dict] = None,
+        subsample_factor: int = 5,
+        text_config: Optional[dict] = None,
+        audio_token_id: Optional[int] = None,
+        **kwargs,
+    ):
+        self.audio_encoder_config = DashengConfig(
+            **(audio_encoder_config or {}))
+        self.subsample_factor = subsample_factor
+        self.text_config = (Qwen2_5OmniTextConfig(
+            **text_config) if text_config else Qwen2_5OmniTextConfig())
+        self.text_config.rope_scaling = None  # uses_mrope is false
+        self.audio_token_id = audio_token_id
+        super().__init__(**kwargs)