vllm-cpu-amxbf16 0.11.2.post2__cp310-cp310-manylinux_2_17_x86_64.whl

vllm/model_executor/models/ultravox.py

@@ -0,0 +1,681 @@
+# SPDX-License-Identifier: Apache-2.0
+# SPDX-FileCopyrightText: Copyright contributors to the vLLM project
+
+# Adapted from https://github.com/fixie-ai/ultravox/blob/ecd58c4041030bae2ad15aa6bcf04ab43199ea02/ultravox/model/ultravox_model.py
+"""PyTorch Ultravox model."""
+
+from collections.abc import Iterable, Mapping, Sequence
+from typing import Annotated, Any, Literal, TypeAlias
+
+import torch
+from torch import nn
+from torch.nn import functional as F
+from transformers import BatchFeature, ProcessorMixin
+from transformers.models.whisper import WhisperFeatureExtractor
+from transformers.models.whisper.modeling_whisper import WhisperEncoder
+
+from vllm.config import VllmConfig
+from vllm.config.multimodal import BaseDummyOptions
+from vllm.model_executor.layers.activation import MulAndSilu, get_act_fn
+from vllm.model_executor.layers.layernorm import RMSNorm
+from vllm.model_executor.model_loader import DefaultModelLoader
+from vllm.model_executor.models.module_mapping import MultiModelKeys
+from vllm.multimodal import MULTIMODAL_REGISTRY
+from vllm.multimodal.inputs import (
+    MultiModalDataDict,
+    MultiModalFieldConfig,
+    MultiModalKwargsItems,
+    NestedTensors,
+)
+from vllm.multimodal.parse import MultiModalDataItems, MultiModalDataParser
+from vllm.multimodal.processing import (
+    BaseMultiModalProcessor,
+    BaseProcessingInfo,
+    PromptReplacement,
+    PromptUpdate,
+)
+from vllm.multimodal.profiling import BaseDummyInputsBuilder
+from vllm.sequence import IntermediateTensors
+from vllm.transformers_utils.configs.ultravox import UltravoxConfig
+from vllm.utils.tensor_schema import TensorSchema, TensorShape
+
+from .interfaces import (
+    MultiModalEmbeddings,
+    SupportsLoRA,
+    SupportsMultiModal,
+    SupportsPP,
+)
+from .utils import (
+    AutoWeightsLoader,
+    WeightsMapper,
+    flatten_bn,
+    init_vllm_registered_model,
+    maybe_prefix,
+)
+
+_AUDIO_PLACEHOLDER_OVERRIDE = "<|audio|>"
+_MAX_ENCODER_BATCH_SIZE = 16
+
+
+class UltravoxAudioFeatureInputs(TensorSchema):
+    """
+    Dimensions:
+    - b: batch size
+    - n: number of chunks
+    - t: Time frames (M)
+    - nmb: Number of mel bins
+    """
+
+    type: Literal["audio_features"]
+    data: Annotated[
+        torch.Tensor | list[torch.Tensor] | list[list[torch.Tensor]],
+        TensorShape("bn", "nmb", "t"),
+    ]
+    lens: Annotated[torch.Tensor, TensorShape("bn")]
+    """
+    Length of the audio frames per chunk. Used for attention mask in WhisperEncoder.
+    """
+    token_len: Annotated[torch.Tensor, TensorShape("bn")]
+    """Length of the audio tokens per chunk. Used for flattening the audio features."""
+    num_chunks: Annotated[torch.Tensor, TensorShape("n")]
+    """Number of chunks per audio. Used for flattening the audio features."""
+
+
+class UltravoxAudioEmbeddingInputs(TensorSchema):
+    """
+    Dimensions:
+    - b: batch size
+    - na: number of audios
+    - afs: audio feature size
+    - hs: hidden size
+    """
+
+    type: Literal["audio_embeds"]
+    data: Annotated[
+        torch.Tensor | list[torch.Tensor], TensorShape("b", "na", "afs", "hs")
+    ]
+
+
+UltravoxAudioInputs: TypeAlias = (
+    UltravoxAudioFeatureInputs | UltravoxAudioEmbeddingInputs
+)
+
+
+class UltravoxProcessingInfo(BaseProcessingInfo):
+    def get_hf_processor(self, **kwargs: object) -> ProcessorMixin:
+        config = self.ctx.model_config.hf_config
+        hf_processor = self.ctx.get_hf_processor(**kwargs)
+
+        # NOTE: Ultravox processing definition uses '<|eot_id|>' as the
+        # placeholder that will cause confusion with the actual end of turn
+        # token, thus we override placeholder with a reserved token.
+        hf_processor.audio_token_replacement = _AUDIO_PLACEHOLDER_OVERRIDE
+        hf_processor.audio_replacement_token_id = config.audio_token_index
+
+        return hf_processor
+
+    def get_feature_extractor(self, **kwargs: object) -> WhisperFeatureExtractor:
+        hf_processor = self.get_hf_processor(**kwargs)
+        audio_processor = hf_processor.audio_processor  # type: ignore
+        feature_extractor = audio_processor.feature_extractor  # type: ignore
+        assert isinstance(feature_extractor, WhisperFeatureExtractor)
+        return feature_extractor
+
+    def get_supported_mm_limits(self) -> Mapping[str, int | None]:
+        return {"audio": None}
+
+
+class UltravoxDummyInputsBuilder(BaseDummyInputsBuilder[UltravoxProcessingInfo]):
+    def get_dummy_text(self, mm_counts: Mapping[str, int]) -> str:
+        num_audios = mm_counts.get("audio", 0)
+
+        return "<|audio|>" * num_audios
+
+    def get_dummy_mm_data(
+        self,
+        seq_len: int,
+        mm_counts: Mapping[str, int],
+        mm_options: Mapping[str, BaseDummyOptions] | None = None,
+    ) -> MultiModalDataDict:
+        feature_extractor = self.info.get_feature_extractor()
+
+        sampling_rate = feature_extractor.sampling_rate
+        audio_len = (
+            feature_extractor.chunk_length * sampling_rate * _MAX_ENCODER_BATCH_SIZE
+        )
+        num_audios = mm_counts.get("audio", 0)
+
+        audio_overrides = mm_options.get("audio") if mm_options else None
+
+        return {
+            "audio": self._get_dummy_audios(
+                length=audio_len, num_audios=num_audios, overrides=audio_overrides
+            )
+        }
+
+
+class UltravoxMultiModalProcessor(BaseMultiModalProcessor[UltravoxProcessingInfo]):
+    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, object],
+        tok_kwargs: Mapping[str, object],
+    ) -> BatchFeature:
+        # Text-only input not supported in composite processor
+        if not mm_data.get("audios", []):
+            prompt_ids = self.info.get_tokenizer().encode(
+                prompt, add_special_tokens=False
+            )
+            prompt_ids = self._apply_hf_processor_tokens_only(prompt_ids)
+            return BatchFeature(dict(input_ids=[prompt_ids]), tensor_type="pt")
+
+        mm_data = dict(mm_data)
+        audios = mm_data.pop("audios", [])
+        assert isinstance(audios, list)
+
+        feature_extractor = self.info.get_feature_extractor(**mm_kwargs)
+        mm_kwargs = dict(
+            **mm_kwargs,
+            sampling_rate=feature_extractor.sampling_rate,
+            include_audio_num_chunks=True,
+        )
+
+        item_processor_data = dict(**mm_data, audios=audios)
+
+        # some tokenizer kwargs are incompatible with UltravoxProcessor
+        tok_kwargs.pop("padding", None)
+        tok_kwargs.pop("truncation", None)
+
+        output = super()._call_hf_processor(
+            prompt=prompt,
+            mm_data=item_processor_data,
+            mm_kwargs=mm_kwargs,
+            tok_kwargs=tok_kwargs,
+        )
+        output["audio_features"] = output.pop("audio_values")
+
+        return output
+
+    def _get_mm_fields_config(
+        self,
+        hf_inputs: BatchFeature,
+        hf_processor_mm_kwargs: Mapping[str, object],
+    ) -> Mapping[str, MultiModalFieldConfig]:
+        num_chunks = hf_inputs.get("audio_num_chunks", torch.zeros(0))
+        return dict(
+            # to handle longer than 30s audio, each audio might be split
+            # into multiple chunks as such, their batch dimension can be
+            # higher than the number of audio samples
+            audio_features=MultiModalFieldConfig.flat_from_sizes("audio", num_chunks),
+            audio_token_len=MultiModalFieldConfig.flat_from_sizes("audio", num_chunks),
+            audio_lens=MultiModalFieldConfig.flat_from_sizes("audio", num_chunks),
+            # num_chunks can convert audio_chunked to audio batch dimension
+            audio_num_chunks=MultiModalFieldConfig.batched("audio"),
+            audio_embeds=MultiModalFieldConfig.batched("audio"),
+        )
+
+    def _get_prompt_updates(
+        self,
+        mm_items: MultiModalDataItems,
+        hf_processor_mm_kwargs: Mapping[str, Any],
+        out_mm_kwargs: MultiModalKwargsItems,
+    ) -> Sequence[PromptUpdate]:
+        hf_processor = self.info.get_hf_processor(**hf_processor_mm_kwargs)
+
+        replacement_id = hf_processor.audio_replacement_token_id  # type: ignore
+
+        # Each audio can be split into multiple chunks.
+        # chunks_start_idx[i] indicates the start index of the chunks
+        # belonging to the i-th audio.
+        out_mm_data = out_mm_kwargs.get_data()
+        num_chunks = out_mm_data.get("audio_num_chunks", torch.zeros(0))
+        chunks_start_idx: torch.Tensor = torch.cumsum(
+            num_chunks, dim=0, dtype=torch.int32
+        )
+        chunks_start_idx = torch.cat(
+            [torch.tensor([0], dtype=torch.int32), chunks_start_idx]
+        )
+
+        def get_replacement_ultravox(item_idx: int):
+            start = chunks_start_idx[item_idx]
+            end = chunks_start_idx[item_idx + 1]
+            audio_token_len = out_mm_data["audio_token_len"][start:end].sum()
+            return [replacement_id] * int(audio_token_len)  # type: ignore
+
+        return [
+            PromptReplacement(
+                modality="audio",
+                target="<|audio|>",
+                replacement=get_replacement_ultravox,
+            )
+        ]
+
+
+class StackAudioFrames(nn.Module):
+    """
+    Stack the audio embedding frames to reduce the sequence length by a factor
+    of `stack_factor`.
+    """
+
+    def __init__(self, stack_factor: int = 8):
+        super().__init__()
+        self.stack_factor = stack_factor
+
+    def forward(self, audio_embeds: torch.Tensor) -> torch.Tensor:
+        B, T, C = audio_embeds.shape
+        T_pad = (T + self.stack_factor - 1) // self.stack_factor * self.stack_factor
+        audio_embeds = F.pad(audio_embeds, (0, 0, 0, T_pad - T))
+        B, T, C = audio_embeds.shape
+        audio_embeds = audio_embeds.view(
+            B, T // self.stack_factor, C * self.stack_factor
+        )
+        return audio_embeds
+
+
+class UltravoxProjector(nn.Module):
+    def __init__(self, config: UltravoxConfig):
+        super().__init__()
+        self.hidden_dim = config.hidden_size
+        self._pad_and_stack = StackAudioFrames(config.stack_factor)
+        dim_in = config.audio_config.hidden_size * config.stack_factor
+        self.ln_pre = RMSNorm(dim_in)
+        self.linear_1 = nn.Linear(dim_in, self.hidden_dim, bias=False)
+        dim_mid = self.hidden_dim
+
+        if config.projector_act == "swiglu":
+            self.act = MulAndSilu()
+            dim_mid = dim_mid // 2
+        else:
+            self.act = get_act_fn(config.projector_act)
+
+        dim_out = config.text_config.hidden_size
+        self.linear_2 = nn.Linear(dim_mid, dim_out, bias=False)
+
+        # Ultravox v0.4.1 and below use layer_norm after the second linear layer
+        # while v0.5.0 and above uses layer_norm after the first linear layer.
+        if config.projector_ln_mid:
+            self.ln_mid: nn.Module = RMSNorm(dim_mid)
+            self.ln_post = nn.Identity()
+        else:
+            self.ln_mid = nn.Identity()
+            self.ln_post = RMSNorm(dim_out)
+
+    def forward(self, audio_features: torch.Tensor) -> torch.Tensor:
+        audio_features = self._pad_and_stack(audio_features)
+        audio_features = self.ln_pre(audio_features)
+        hidden_states = self.linear_1(audio_features)
+        hidden_states = self.act(hidden_states)
+        hidden_states = self.ln_mid(hidden_states)
+        hidden_states = self.linear_2(hidden_states)
+        hidden_states = self.ln_post(hidden_states)
+        return hidden_states
+
+
+class ModifiedWhisperEncoder(WhisperEncoder):
+    """
+    Encoder portion of OpenAI's Whisper model.
+
+    This implementation is a slightly modified version of HF Transformers'
+    Whisper Encoder, with only a few fixes:
+    1. base_model_prefix updated to allow for doing `.from_pretrained`
+       directly on the encoder
+    2. allow less than 30 second of audio padding to be passed in:
+        - relaxed ValueError check for `input_features` length to be less
+           than or equal to `expected_seq_length` instead of strictly equal
+        - embed_pos is now sliced to match the length of `inputs_embeds`
+
+    Original: https://github.com/huggingface/transformers/blob/main/src/transformers/models/whisper/modeling_whisper.py
+    See commentary: https://github.com/huggingface/transformers/issues/25744
+    """
+
+    base_model_prefix = "model.encoder"
+
+    def __init__(self, *args, **kwargs):
+        super().__init__(*args, **kwargs)
+        self.config.is_decoder = False
+
+    @property
+    def max_context_length(self):
+        return (
+            self.config.max_source_positions
+            * self.conv1.stride[0]
+            * self.conv2.stride[0]
+        )
+
+    def get_attention_mask_by_audio_len(
+        self, audio_lens: torch.Tensor | None, hidden_states: torch.Tensor
+    ):
+        """
+        Create attention mask based on audio lengths to mask out padding tokens
+        For each sample in batch:
+        - Convert raw audio length to feature length after convolutions
+        - Create bool mask: True for valid positions and False for padding
+        - Convert to attention mask format expected by transformer layers
+        (1.0 for positions to attend to, large negative for positions to ignore)
+        This masking ensures consistent behavior between training and inference
+        by preventing the model from attending to padding tokens in both cases
+        """
+        if audio_lens is None:
+            return None
+
+        audio_feature_len = self._get_feat_extract_output_lengths(audio_lens)
+        max_seq_len = hidden_states.shape[1]
+        attention_mask = torch.arange(max_seq_len, device=hidden_states.device)[
+            None, :
+        ].lt(audio_feature_len.view(-1, 1))
+        attention_mask = self.get_extended_attention_mask(
+            attention_mask,
+            None,
+            dtype=hidden_states.dtype,
+        )
+        return attention_mask
+
+    def forward(
+        self,
+        input_features: torch.Tensor,
+        audio_lens: torch.Tensor | None = None,
+    ):
+        expected_seq_length = self.max_context_length
+        if input_features.shape[-1] > expected_seq_length:
+            raise ValueError(
+                f"Whisper expects the mel input features to be of length "
+                f"{expected_seq_length} or less, but found "
+                f"{input_features.shape[-1]}. Make sure to pad the input mel "
+                f"features to {expected_seq_length}."
+            )
+
+        inputs_embeds = nn.functional.gelu(self.conv1(input_features))
+        inputs_embeds = nn.functional.gelu(self.conv2(inputs_embeds))
+
+        inputs_embeds = inputs_embeds.permute(0, 2, 1)
+        embed_pos = self.embed_positions.weight[: inputs_embeds.size(-2)]
+
+        hidden_states = inputs_embeds + embed_pos
+        hidden_states = nn.functional.dropout(
+            hidden_states, p=self.dropout, training=self.training
+        )
+
+        attention_mask = self.get_attention_mask_by_audio_len(audio_lens, hidden_states)
+
+        for encoder_layer in self.layers:
+            layer_outputs = encoder_layer(
+                hidden_states,
+                attention_mask,
+                layer_head_mask=None,
+            )
+
+            hidden_states = layer_outputs[0]
+
+        hidden_states = self.layer_norm(hidden_states)
+        return hidden_states
+
+
+@MULTIMODAL_REGISTRY.register_processor(
+    UltravoxMultiModalProcessor,
+    info=UltravoxProcessingInfo,
+    dummy_inputs=UltravoxDummyInputsBuilder,
+)
+class UltravoxModel(nn.Module, SupportsMultiModal, SupportsPP, SupportsLoRA):
+    merge_by_field_config = True
+
+    packed_modules_mapping = {
+        "qkv_proj": ["q_proj", "k_proj", "v_proj"],
+        "gate_up_proj": ["gate_proj", "up_proj"],
+    }
+
+    hf_to_vllm_mapper = WeightsMapper(
+        orig_to_new_prefix={"audio_tower.model.encoder.": "audio_tower."}
+    )
+
+    @classmethod
+    def get_placeholder_str(cls, modality: str, i: int) -> str | None:
+        if modality.startswith("audio"):
+            return "<|audio|>"
+
+        raise ValueError("Only audio modality is supported")
+
+    def __init__(self, *, vllm_config: VllmConfig, prefix: str = ""):
+        super().__init__()
+        config: UltravoxConfig = vllm_config.model_config.hf_config
+        multimodal_config = vllm_config.model_config.multimodal_config
+        self.config = config
+        self.multi_modal_config = multimodal_config
+        assert self.multi_modal_config
+
+        self.secondary_weights = []
+        self.audio_tower = ModifiedWhisperEncoder(config.audio_config)
+        if config.audio_model_id is not None:
+            # this prefix is not for initialization, but for loading weights
+            # note the trailing dot
+            self.secondary_weights.append(
+                DefaultModelLoader.Source(
+                    model_or_path=config.audio_model_id,
+                    revision=None,
+                    prefix="audio_tower.",
+                )
+            )
+        self.multi_modal_projector = UltravoxProjector(config)
+        self.language_model = init_vllm_registered_model(
+            vllm_config=vllm_config,
+            hf_config=config.wrapped_model_config,
+            prefix=maybe_prefix(prefix, "language_model"),
+        )
+        if config.text_model_id is not None:
+            # this prefix is not for initialization, but for loading weights
+            # note the trailing dot
+            self.secondary_weights.append(
+                DefaultModelLoader.Source(
+                    model_or_path=config.text_model_id,
+                    revision=None,
+                    prefix="language_model.",
+                )
+            )
+
+        self.make_empty_intermediate_tensors = (
+            self.language_model.make_empty_intermediate_tensors
+        )
+
+    def get_mm_mapping(self) -> MultiModelKeys:
+        """
+        Get the module prefix in multimodal models
+        """
+        return MultiModelKeys.from_string_field(
+            language_model="language_model.",
+            connector="multi_modal_projector.",
+            tower_model="audio_tower.",
+        )
+
+    def _audio_features_to_embeddings(
+        self, input_features: torch.Tensor, audio_lens: torch.Tensor
+    ) -> torch.Tensor:
+        audio_features = input_features.to(self.audio_tower.dtype)
+        batch_size = audio_features.size(0)
+        audio_embeddings = []
+
+        # Process audio features in batches to keep memory usage predictable
+        for start in range(0, batch_size, _MAX_ENCODER_BATCH_SIZE):
+            end = min(start + _MAX_ENCODER_BATCH_SIZE, batch_size)
+            # Process through audio tower
+            batch_features = self.audio_tower(
+                audio_features[start:end], audio_lens[start:end]
+            )
+            batch_features = batch_features.to(self.audio_tower.dtype)
+
+            # Process through projector
+            batch_embeddings = self.multi_modal_projector(batch_features)
+            audio_embeddings.append(batch_embeddings)
+
+        # Concatenate results
+        audio_embeddings = torch.cat(audio_embeddings, dim=0)
+        return audio_embeddings
+
+    def _parse_and_validate_audio_input(
+        self, **kwargs: object
+    ) -> UltravoxAudioInputs | None:
+        audio_features = kwargs.pop("audio_features", None)
+        audio_embeds = kwargs.pop("audio_embeds", None)
+        audio_lens = kwargs.pop("audio_lens", None)
+        audio_token_len = kwargs.pop("audio_token_len", None)
+        audio_num_chunks = kwargs.pop("audio_num_chunks", None)
+
+        if audio_features is None and audio_embeds is None:
+            return None
+
+        if audio_features is not None:
+            return UltravoxAudioFeatureInputs(
+                type="audio_features",
+                data=audio_features,
+                lens=audio_lens,
+                token_len=audio_token_len,
+                num_chunks=audio_num_chunks,
+            )
+
+        if audio_embeds is not None:
+            return UltravoxAudioEmbeddingInputs(type="audio_embeds", data=audio_embeds)
+
+        raise AssertionError("This line should be unreachable.")
+
+    def _process_audio_input(
+        self,
+        audio_input: UltravoxAudioInputs,
+    ) -> NestedTensors | tuple[torch.Tensor, ...]:
+        if audio_input["type"] == "audio_embeds":
+            return audio_input["data"]
+
+        # Pad and concatenate audio features
+        # [[B1, 80, M1], [B2, 80, M2]] -> [B1+B2, 80, max(M1, M2)]
+        audio_features = pad_and_concat_to_dim3(audio_input["data"])
+
+        audio_lens = audio_input["lens"]
+        audio_token_len = audio_input["token_len"]
+
+        embeddings = self._audio_features_to_embeddings(audio_features, audio_lens)
+
+        # We should flatten and concatenate embeddings based on token lengths
+        # For example, with token_len = [4, 2, 3], flattened_embeddings will be
+        # concat(embeddings[0][:4], embeddings[1][:2], embeddings[2][:3])
+
+        # Create a mask of valid indices based on token lengths
+        max_len = embeddings.shape[1]
+        indices = torch.arange(max_len, device=embeddings.device).expand(
+            embeddings.shape[0], -1
+        )
+        mask = indices < audio_token_len[:, None]
+        # Apply mask and flatten
+        flattened_embeddings = embeddings[mask]
+
+        # Return one tensor per input audio
+        embed_lens = [
+            chunk_lens.sum().item()
+            for chunk_lens in audio_token_len.split(audio_input["num_chunks"].tolist())
+        ]
+        return flattened_embeddings.split(embed_lens)
+
+    def get_language_model(self) -> torch.nn.Module:
+        return self.language_model
+
+    def embed_multimodal(self, **kwargs: object) -> MultiModalEmbeddings:
+        audio_input = self._parse_and_validate_audio_input(**kwargs)
+        if audio_input is None:
+            return []
+        audio_embeddings = self._process_audio_input(audio_input)
+        return audio_embeddings
+
+    def embed_input_ids(
+        self,
+        input_ids: torch.Tensor,
+        multimodal_embeddings: MultiModalEmbeddings | None = None,
+        *,
+        is_multimodal: torch.Tensor | None = None,
+        # Multi-modal token ID may exceed vocab size
+        handle_oov_mm_token: bool = True,
+    ) -> torch.Tensor:
+        # This is to satisfy the type checker for each overload
+        if multimodal_embeddings is None or is_multimodal is None:
+            return super().embed_input_ids(input_ids)
+
+        return super().embed_input_ids(
+            input_ids,
+            multimodal_embeddings=multimodal_embeddings,
+            is_multimodal=is_multimodal,
+            handle_oov_mm_token=handle_oov_mm_token,
+        )
+
+    def forward(
+        self,
+        input_ids: torch.Tensor,
+        positions: torch.Tensor,
+        intermediate_tensors: torch.Tensor | None = None,
+        inputs_embeds: torch.Tensor | None = None,
+        **kwargs,
+    ) -> torch.Tensor | IntermediateTensors:
+        """Run forward pass for Ultravox
+
+        One key thing to understand is the `input_ids` already accounts for the
+        positions of the to-be-inserted audio embeddings. The to-be-inserted
+        audio has a size that is essentially 6.25 tokens per second of audio.
+
+        This way, the `positions` and `attn_metadata` are consistent
+        with the `input_ids`.
+
+        Args:
+            input_ids: Flattened (concatenated) input_ids corresponding to a
+                batch.
+            positions: Position indices for the input tokens.
+            intermediate_tensors: Intermediate tensors from prior forward pass.
+            inputs_embeds: Optional tensor of input embeddings.
+
+        """
+
+        if intermediate_tensors is not None:
+            inputs_embeds = None
+
+        language_model = self.language_model
+        if hasattr(language_model, "language_model"):
+            language_model = language_model.language_model
+
+        hidden_states = language_model.model(
+            input_ids, positions, intermediate_tensors, inputs_embeds=inputs_embeds
+        )
+        return hidden_states
+
+    def compute_logits(self, hidden_states: torch.Tensor) -> torch.Tensor:
+        return self.language_model.compute_logits(hidden_states)
+
+    def load_weights(self, weights: Iterable[tuple[str, torch.Tensor]]) -> set[str]:
+        loader = AutoWeightsLoader(self, ignore_unexpected_prefixes=["audio_tower."])
+        return loader.load_weights(weights, mapper=self.hf_to_vllm_mapper)
+
+
+def pad_and_concat_to_dim3(
+    features: torch.Tensor | list[torch.Tensor] | list[list[torch.Tensor]],
+) -> torch.Tensor:
+    """
+    Pad and concatenate a list of tensors.
+
+    output:
+        Tensor of shape [B, C, M] where M is the maximum length of the input
+        tensors, B is the sum of the batch sizes of the input tensors.
+        C must be the same for all input tensors.
+    """
+    if isinstance(features, torch.Tensor):
+        if features.ndim > 3:
+            # Flatten [B, N, 80, M] -> [B * N, 80, M]
+            features = flatten_bn(features)
+
+        return features
+
+    features = [pad_and_concat_to_dim3(f) for f in features]
+
+    max_len = max(f.shape[-1] for f in features)
+    # Ensure all features have dim=3
+    features = [f.view(-1, *f.shape[-2:]) for f in features]
+    # Pad and concatenate:
+    # [[B1, 80, M1], [B2, 80, M2]] -> [B1+B2, 80, max(M1, M2)]
+    features = [F.pad(f, (0, max_len - f.shape[-1])) for f in features]
+    return torch.cat(features)