[Misc] Clean up renderers (#36770)

Signed-off-by: DarkLight1337 <tlleungac@connect.ust.hk>

tests/models/multimodal/processing/test_common.py

@@ -6,9 +6,6 @@ from functools import partial
 
 import numpy as np
 import pytest
-from mistral_common.protocol.instruct.chunk import ImageChunk, TextChunk
-from mistral_common.protocol.instruct.messages import UserMessage
-from mistral_common.protocol.instruct.request import ChatCompletionRequest
 from PIL import Image
 
 from vllm.config import ModelConfig
@@ -21,7 +18,10 @@ from vllm.config.multimodal import (
 from vllm.multimodal import MULTIMODAL_REGISTRY, MultiModalDataDict
 from vllm.multimodal.cache import MultiModalProcessorOnlyCache
 from vllm.multimodal.inputs import MultiModalInputs, batched_tensors_equal
-from vllm.multimodal.processing import BaseMultiModalProcessor, InputProcessingContext
+from vllm.multimodal.processing import (
+    BaseMultiModalProcessor,
+    InputProcessingContext,
+)
 from vllm.tokenizers import TokenizerLike, cached_tokenizer_from_config
 from vllm.utils.mistral import is_mistral_tokenizer
 
@@ -74,20 +74,6 @@ def glmasr_patch_mm_data(mm_data: MultiModalDataDict) -> MultiModalDataDict:
     return mm_data
 
 
-# For some multimodal models, tokenizer will always add bos_token
-# at the beginning of prompt by default, causing hf_processor outputs
-# incorrect token ids. So we need use `add_special_tokens=False` here
-# to leave bos_token to be added by the processor.
-_ADD_SPECIAL_TOKENS_OVERRIDES = {
-    "lfm2_vl": False,
-    "nemotron_parse": False,
-    "ovis": False,
-    "ovis2_5": False,
-    "paligemma": False,
-    "ultravox": False,
-    "whisper": False,
-}
-
 _IGNORE_MM_KEYS = {
     # In Ultravox, the audio_features can be different depending on padding
     # The slight difference should not be a problem though, since
@@ -152,63 +138,34 @@ def get_text_token_prompts(
     parsed_data = processor.info.parse_mm_data(mm_data)
     mm_counts = {k: len(vs) for k, vs in parsed_data.items()}
 
-    text_prompt: str | None
-    token_prompt: list[int]
     if is_mistral_tokenizer(tokenizer):
-        # ChatCompletionRequest only supports ImageChunk natively;
-        # for other modalities (e.g. audio), fall back to the model's
-        # own dummy inputs builder which knows the right placeholders.
-        has_non_image = any(
-            k != "image" and count > 0 for k, count in mm_counts.items()
-        )
-
-        if has_non_image:
         inputs = dummy_inputs.get_dummy_processor_inputs(
             model_config.max_model_len,
             mm_counts,
             mm_options={},
-            )
-            text_prompt = None
-            token_prompt = (
-                inputs.prompt
-                if isinstance(inputs.prompt, list)
-                else tokenizer.encode(inputs.prompt, add_special_tokens=False)
+            # Assume all Mistral models define this extra argument
+            mm_data=mm_data,  # type: ignore[call-arg]
         )
-        else:
-            images = parsed_data.get("image", [])
-            request = ChatCompletionRequest(
-                messages=[
-                    UserMessage(
-                        content=[
-                            TextChunk(text=""),
-                            *(ImageChunk(image=image) for image in images),
-                        ]
-                    ),
-                ]
-            )
-            res = tokenizer.mistral.encode_chat_completion(request)
-
-            # Mistral does not support decode_tokens with
-            # skip_special_tokens=False
-            text_prompt = None
-            token_prompt = res.tokens
     else:
         inputs = dummy_inputs.get_dummy_processor_inputs(
             model_config.max_model_len,
             mm_counts,
             mm_options={},
         )
-        # Some models (e.g., Kimi-Audio) return token IDs directly instead of str
+
+    text_prompt: str | None
+    token_prompt: list[int]
     if isinstance(inputs.prompt, list):
         text_prompt = None
         token_prompt = inputs.prompt
-        else:
-            assert isinstance(inputs.prompt, str)
+    elif isinstance(inputs.prompt, str):
         text_prompt = inputs.prompt
         token_prompt = tokenizer.encode(
             text_prompt,
-                add_special_tokens=_ADD_SPECIAL_TOKENS_OVERRIDES.get(model_type, True),
+            **processor.info.get_default_tok_params().get_encode_kwargs(),
         )
+    else:
+        raise TypeError(type(inputs.prompt))
 
     return text_prompt, token_prompt
 
@@ -448,7 +405,7 @@ def test_processing_correctness(
         )
     if model_id == "mistralai/Voxtral-Mini-4B-Realtime-2602":
         pytest.skip(
-            "Voxtral Realtime doesn't make use of any place-holder"
+            "Voxtral Realtime doesn't make use of any place-holder "
             "tokens and hence cannot pass the processing "
             "correctness test as is. Let's revisit adapting this "
             "test once more realtime models exist."

vllm/config/model.py

@@ -532,6 +532,22 @@ class ModelConfig:
         self._architecture = arch
         logger.info("Resolved architecture: %s", arch)
 
+        # Set default tokenizer modes based on model architecture
+        if self.tokenizer_mode == "auto":
+            if arch == "Grok1ForCausalLM":
+                self.tokenizer_mode = "grok2"
+            elif arch == "MoonshotKimiaForCausalLM":
+                self.tokenizer_mode = "kimi_audio"
+            elif arch == "QwenVLForConditionalGeneration":
+                self.tokenizer_mode = "qwen_vl"
+
+            if self.tokenizer_mode != "auto":
+                logger.info(
+                    "Defaulting to tokenizer_mode=%r for %s",
+                    self.tokenizer_mode,
+                    arch,
+                )
+
         # Init pooler config if needed
         if self.runner_type == "pooling":
             if self.pooler_config is None:

vllm/model_executor/models/kimi_audio.py

@@ -10,11 +10,13 @@ from typing import Any, ClassVar, Literal
 import numpy as np
 import torch
 import torch.nn as nn
+from huggingface_hub import snapshot_download
 from safetensors import safe_open
 from transformers import BatchFeature
 from transformers import WhisperConfig as HFWhisperConfig
 
 from vllm.config import ModelConfig, SpeechToTextConfig, VllmConfig
+from vllm.config.multimodal import BaseDummyOptions
 from vllm.inputs.data import PromptType, TokensPrompt
 from vllm.model_executor.layers.logits_processor import LogitsProcessor
 from vllm.model_executor.model_loader.weight_utils import (
@@ -47,7 +49,10 @@ from vllm.multimodal.processing import (
     BaseProcessingInfo,
     PromptReplacement,
 )
-from vllm.multimodal.processing.processor import BaseMultiModalProcessor
+from vllm.multimodal.processing.processor import (
+    BaseMultiModalProcessor,
+    ProcessorInputs,
+)
 from vllm.sequence import IntermediateTensors
 from vllm.tokenizers import cached_get_tokenizer
 from vllm.tokenizers.kimi_audio import KimiAudioTokenizer
@@ -59,6 +64,15 @@ from vllm.v1.sample.metadata import SamplingMetadata
 KIMIA_WHISPER_SUBFOLDER = "whisper-large-v3"
 
 
+def _get_whisper_local_path(repo_id: str):
+    if os.path.exists(repo_id):
+        repo_local_path = repo_id
+    else:
+        repo_local_path = snapshot_download(repo_id, local_files_only=True)
+
+    return os.path.join(repo_local_path, KIMIA_WHISPER_SUBFOLDER)
+
+
 def _get_feat_extract_output_lengths(input_lengths: torch.Tensor) -> torch.Tensor:
     """Compute output lengths after Whisper feature extraction.
 
@@ -88,10 +102,10 @@ class KimiAudioWhisperEncoder(WhisperEncoder):
         # Load Whisper config from subfolder (authoritative source)
         # Kimi-Audio stores Whisper config in whisper-large-v3/config.json
         model_path = vllm_config.model_config.model
-        whisper_config_path = os.path.join(model_path, KIMIA_WHISPER_SUBFOLDER)
 
         # Load WhisperConfig from the subfolder
-        whisper_config = HFWhisperConfig.from_pretrained(whisper_config_path)
+        whisper_dir = _get_whisper_local_path(model_path)
+        whisper_config = HFWhisperConfig.from_pretrained(whisper_dir)
 
         # Temporarily replace hf_config for WhisperEncoder.__init__()
         original_config = vllm_config.model_config.hf_config
@@ -114,28 +128,18 @@ class KimiAudioWhisperEncoder(WhisperEncoder):
 class KimiAudioProcessingInfo(BaseProcessingInfo):
     """Processing info for vLLM registry."""
 
-    def get_hf_config(self):
-        return self.ctx.model_config.hf_config
-
     def get_hf_processor(self, **kwargs: object) -> KimiAudioProcessor:
-        """Get KimiAudioProcessor with feature extractor and tokenizer."""
-        # Use vLLM's cached loader for feature extractor
         feature_extractor = cached_feature_extractor_from_config(
             self.ctx.model_config,
             subfolder=KIMIA_WHISPER_SUBFOLDER,
         )
 
-        # Use vLLM's standard tokenizer loading (respects tokenizer_mode)
-        tokenizer = self.get_tokenizer()
-
-        # Construct processor directly
         return KimiAudioProcessor(
             feature_extractor=feature_extractor,
-            tokenizer=tokenizer,
+            tokenizer=self.get_tokenizer(),
         )
 
     def get_feature_extractor(self, **kwargs: object):
-        """Get feature extractor using vLLM's cached loader."""
         return cached_feature_extractor_from_config(
             self.ctx.model_config, subfolder=KIMIA_WHISPER_SUBFOLDER
         )
@@ -144,26 +148,16 @@ class KimiAudioProcessingInfo(BaseProcessingInfo):
         return {"audio": 1}
 
     def get_data_parser(self) -> "KimiAudioMultiModalDataParser":
-        """Get data parser for audio inputs."""
+        feature_extractor = self.get_feature_extractor()
         return KimiAudioMultiModalDataParser(
+            target_sr=feature_extractor.sampling_rate,
             expected_hidden_size=self._get_expected_hidden_size(),
         )
 
 
 class KimiAudioDummyInputsBuilder(BaseDummyInputsBuilder[KimiAudioProcessingInfo]):
-    """Dummy inputs builder for vLLM registry."""
-
-    def get_dummy_text(self, mm_counts: Mapping[str, int]) -> list[int]:
-        """Return dummy text as token IDs directly."""
-        num_audios = mm_counts.get("audio", 0)
-        if num_audios == 0:
-            return [198]  # "Transcribe" tokenized
-        # Return as token IDs directly to avoid tokenizer issues
-        return [
-            KimiAudioProcessor.KIMIA_MEDIA_BEGIN,
-            KimiAudioProcessor.KIMIA_TEXT_BLANK,
-            KimiAudioProcessor.KIMIA_MEDIA_END,
-        ] * num_audios
+    def get_dummy_text(self, mm_counts: Mapping[str, int]) -> str:
+        return ""
 
     def get_dummy_mm_data(
         self,
@@ -186,6 +180,29 @@ class KimiAudioDummyInputsBuilder(BaseDummyInputsBuilder[KimiAudioProcessingInfo
             ),
         }
 
+    def get_dummy_processor_inputs(
+        self,
+        seq_len: int,
+        mm_counts: Mapping[str, int],
+        mm_options: Mapping[str, BaseDummyOptions],
+    ) -> ProcessorInputs:
+        dummy_mm_data = self.get_dummy_mm_data(seq_len, mm_counts, mm_options)
+        dummy_mm_items = self.info.parse_mm_data(dummy_mm_data)
+
+        num_audios = mm_counts.get("audio", 0)
+        dummy_tokens = (
+            [198]
+            if num_audios == 0
+            else [
+                KimiAudioProcessor.KIMIA_MEDIA_BEGIN,
+                KimiAudioProcessor.KIMIA_TEXT_BLANK,
+                KimiAudioProcessor.KIMIA_MEDIA_END,
+            ]
+            * num_audios
+        )
+
+        return ProcessorInputs(prompt=dummy_tokens, mm_data_items=dummy_mm_items)
+
 
 # Field config for Kimi-Audio multimodal data
 _KIMIAUDIO_FIELD_CONFIG = {
@@ -197,10 +214,6 @@ _KIMIAUDIO_FIELD_CONFIG = {
 class KimiAudioMultiModalDataParser(MultiModalDataParser):
     """Custom data parser for Kimi-Audio multimodal data."""
 
-    def __init__(self, **kwargs):
-        # Whisper expects 16kHz audio
-        super().__init__(target_sr=16000, **kwargs)
-
     def _parse_audio_data(
         self,
         data: dict[str, torch.Tensor] | ModalityData[AudioItem],
@@ -589,9 +602,8 @@ class KimiAudioForConditionalGeneration(
         loaded = loader.load_weights(main_weights, mapper=self.hf_to_vllm_mapper)
 
         # Load Whisper encoder weights from subfolder
-        whisper_path = os.path.join(
-            self.model_path, f"{KIMIA_WHISPER_SUBFOLDER}/model.safetensors"
-        )
+        whisper_dir = _get_whisper_local_path(self.model_path)
+        whisper_path = os.path.join(whisper_dir, "model.safetensors")
         if os.path.exists(whisper_path):
             whisper_loaded = self._load_whisper_weights_from_file(whisper_path)
             loaded.update(whisper_loaded)

vllm/model_executor/models/mllama4.py

@@ -63,12 +63,10 @@ from vllm.multimodal.processing import (
     BaseDummyInputsBuilder,
     BaseMultiModalProcessor,
     BaseProcessingInfo,
-    InputProcessingContext,
     PromptReplacement,
     PromptUpdate,
     PromptUpdateDetails,
 )
-from vllm.renderers import TokenizeParams
 from vllm.sequence import IntermediateTensors
 from vllm.utils.tensor_schema import TensorSchema, TensorShape
 
@@ -546,9 +544,6 @@ class Llama4VisionModel(nn.Module):
 
 
 class Mllama4ProcessingInfo(BaseProcessingInfo):
-    def __init__(self, ctx: InputProcessingContext) -> None:
-        super().__init__(ctx)
-
     def get_hf_config(self) -> Llama4Config:
         return self.ctx.get_hf_config(Llama4Config)
 
@@ -557,9 +552,6 @@ class Mllama4ProcessingInfo(BaseProcessingInfo):
             Llama4Processor, use_fast=kwargs.pop("use_fast", True), **kwargs
         )
 
-    def get_default_tok_params(self) -> TokenizeParams:
-        return super().get_default_tok_params().with_kwargs(add_special_tokens=False)
-
     def get_supported_mm_limits(self) -> Mapping[str, int | None]:
         # Although vLLM can support more images from an infra capability
         # perspective, we do not recommend using >10 images in practice.
@@ -597,10 +589,6 @@ class Mllama4MultiModalProcessor(BaseMultiModalProcessor[Mllama4ProcessingInfo])
         mm_kwargs: Mapping[str, object],
         tok_kwargs: Mapping[str, object],
     ) -> BatchFeature:
-        tokenizer = self.info.get_tokenizer()
-
-        if mm_data is None:
-            return tokenizer(prompt, add_special_tokens=False)  # exclude bos
         processed_outputs = super()._call_hf_processor(
             prompt=prompt,
             mm_data=mm_data,

vllm/model_executor/models/pixtral.py

@@ -172,12 +172,20 @@ class PixtralDummyInputsBuilder(BaseDummyInputsBuilder[PixtralProcessingInfo]):
         seq_len: int,
         mm_counts: Mapping[str, int],
         mm_options: Mapping[str, BaseDummyOptions],
+        mm_data: MultiModalDataDict | None = None,
     ) -> ProcessorInputs:
         tokenizer = self.info.get_tokenizer()
 
         dummy_text = self.get_dummy_text(mm_counts)
-        dummy_mm_data = self.get_dummy_mm_data(seq_len, mm_counts, mm_options)
-        dummy_images = dummy_mm_data.get("image", [])
+        dummy_mm_data = (
+            self.get_dummy_mm_data(seq_len, mm_counts, mm_options)
+            if mm_data is None
+            else mm_data
+        )
+        dummy_mm_items = self.info.parse_mm_data(dummy_mm_data)
+        dummy_images = (
+            [] if "image" not in dummy_mm_data else dummy_mm_items["image"].get_all()
+        )
 
         request = ChatCompletionRequest(
             messages=[
@@ -192,8 +200,6 @@ class PixtralDummyInputsBuilder(BaseDummyInputsBuilder[PixtralProcessingInfo]):
         res = tokenizer.mistral.encode_chat_completion(request)
         dummy_tokens = res.tokens
 
-        dummy_mm_items = self.info.parse_mm_data(dummy_mm_data)
-
         return ProcessorInputs(prompt=dummy_tokens, mm_data_items=dummy_mm_items)
 
 

vllm/model_executor/models/voxtral.py

@@ -150,13 +150,21 @@ class VoxtralDummyInputsBuilder(BaseDummyInputsBuilder[VoxtralProcessingInfo]):
         seq_len: int,
         mm_counts: Mapping[str, int],
         mm_options: Mapping[str, BaseDummyOptions],
+        mm_data: MultiModalDataDict | None = None,
     ) -> ProcessorInputs:
         tokenizer = self.info.get_tokenizer()
         feature_extractor = self.info.get_hf_processor().feature_extractor
 
         dummy_text = self.get_dummy_text(mm_counts)
-        dummy_mm_data = self.get_dummy_mm_data(seq_len, mm_counts, mm_options)
-        dummy_audios = dummy_mm_data.get("audio", [])
+        dummy_mm_data = (
+            self.get_dummy_mm_data(seq_len, mm_counts, mm_options)
+            if mm_data is None
+            else mm_data
+        )
+        dummy_mm_items = self.info.parse_mm_data(dummy_mm_data)
+        dummy_audios = (
+            [] if "audio" not in dummy_mm_data else dummy_mm_items["audio"].get_all()
+        )
 
         audio_chunks: list[AudioChunk] = []
         format = "wav"

vllm/renderers/qwen_vl.py

@@ -6,11 +6,10 @@ from vllm.config import VllmConfig
 from vllm.tokenizers import cached_get_tokenizer
 from vllm.tokenizers.qwen_vl import QwenVLTokenizer
 
-from .base import BaseRenderer
 from .hf import HfRenderer
 
 
-class QwenVLRenderer(BaseRenderer[QwenVLTokenizer]):
+class QwenVLRenderer(HfRenderer):
     @classmethod
     def from_config(  # type: ignore[override]
         cls,

vllm/renderers/registry.py

@@ -80,13 +80,6 @@ def renderer_from_config(config: "VllmConfig", **kwargs):
         model_config, **kwargs
     )
 
-    # Override tokenizer_mode for Kimi-Audio models
-    if model_config.architecture == "MoonshotKimiaForCausalLM":
-        tokenizer_mode = "kimi_audio"
-        # Update model_config so other components (e.g., multimodal registry)
-        # also use the correct tokenizer mode
-        model_config.tokenizer_mode = "kimi_audio"
-
     if (
         model_config.tokenizer_mode == "auto"
         and model_config.model_impl == "terratorch"

vllm/tokenizers/registry.py

@@ -159,18 +159,6 @@ def resolve_tokenizer_args(
     ):
         tokenizer_mode = "mistral"
 
-    # Try to use Grok2 tiktoken tokenizer if possible
-    if tokenizer_mode == "auto" and any_pattern_in_repo_files(
-        model_name_or_path=str(tokenizer_name),
-        allow_patterns=["tokenizer.tok.json"],
-        revision=revision,
-    ):
-        tokenizer_mode = "grok2"
-
-    # Model-specific tokenizers
-    if tokenizer_mode == "auto" and "/Qwen-VL" in str(tokenizer_name):
-        tokenizer_mode = "qwen_vl"
-
     # Fallback to HF tokenizer
     if tokenizer_mode == "auto":
         tokenizer_mode = "hf"

vllm/transformers_utils/processors/glm4v.py

 # SPDX-License-Identifier: Apache-2.0
 # SPDX-FileCopyrightText: Copyright contributors to the vLLM project
+
+# Adapted from
+# https://github.com/zai-org/CogAgent
 from transformers import PreTrainedTokenizer
 from transformers.image_processing_utils_fast import BaseImageProcessorFast
 from transformers.image_utils import PILImageResampling

vllm/transformers_utils/processors/kimi_audio.py

 # SPDX-License-Identifier: Apache-2.0
 # SPDX-FileCopyrightText: Copyright contributors to the vLLM project
 
-# ruff: noqa
-# mypy: ignore-errors
-# coding=utf-8
-# Copyright 2026 The Moonshot AI team and the HuggingFace Inc. team. All rights reserved.
+# Copyright 2026 The Moonshot AI team and the HuggingFace Inc. team.
+# All rights reserved.
 #
 # Licensed under the Apache License, Version 2.0 (the "License");
 # you may not use this file except in compliance with the License.
@@ -19,42 +17,13 @@
 # limitations under the License.
 """Processor for Kimi-Audio ASR model."""
 
-from collections.abc import Mapping
-from typing import Any
-
 import numpy as np
-import torch
-from transformers import AutoFeatureExtractor, BatchFeature, ProcessorMixin
+from transformers import BatchFeature, ProcessorMixin
 from transformers.audio_utils import AudioInput
-from transformers.tokenization_utils_base import TextInput
-
-from vllm.tokenizers.kimi_audio import KimiAudioTokenizer
-
-
-def _get_feat_extract_output_lengths(input_lengths: torch.Tensor) -> torch.Tensor:
-    """Compute output lengths after Whisper feature extraction."""
-    input_lengths_leave = input_lengths % 100
-    feat_lengths = (input_lengths_leave - 1) // 2 + 1
-    output_lengths = (
-        ((feat_lengths - 1) // 2 + 1 - 1) // 2 + 1 + (input_lengths // 100) * 13
-    )
-    return output_lengths
+from transformers.tokenization_utils_base import PreTokenizedInput, TextInput
 
 
 class KimiAudioProcessor(ProcessorMixin):
-    r"""
-    Constructs a Kimi-Audio processor.
-
-    [`KimiAudioProcessor`] offers all the functionalities of [`WhisperFeatureExtractor`], and a tokenizer.
-    See the [`~KimiAudioProcessor.__call__`] and [`~KimiAudioProcessor.decode`] for more information.
-
-    Args:
-        feature_extractor ([`WhisperFeatureExtractor`], *optional*):
-            The audio feature extractor.
-        tokenizer ([`PreTrainedTokenizer`], *optional*):
-            The text tokenizer.
-    """
-
     # Required for ProcessorMixin
     attributes = ["feature_extractor", "tokenizer"]
     feature_extractor_class = "AutoFeatureExtractor"
@@ -69,44 +38,30 @@ class KimiAudioProcessor(ProcessorMixin):
     AUDIO_SEQ_LEN: int = 376
 
     def __init__(self, feature_extractor=None, tokenizer=None, **kwargs):
-        # Pass feature_extractor and tokenizer to parent ProcessorMixin
-        super().__init__(
-            feature_extractor=feature_extractor,
-            tokenizer=tokenizer,
-            **kwargs,
-        )
-
-    def check_argument_for_proper_class(self, attribute_name: str, argument: Any):
-        """Override to skip class validation for custom tokenizer."""
-        # Skip validation for tokenizer since KimiAudioTokenizer doesn't inherit
-        # from PreTrainedTokenizerBase but is compatible
-        if attribute_name == "tokenizer" and argument is not None:
-            return
-        # For other attributes, use default validation
-        super().check_argument_for_proper_class(attribute_name, argument)
+        self.feature_extractor = feature_extractor
+        self.tokenizer = tokenizer
 
     def __call__(
         self,
-        text: TextInput = None,
-        audio: AudioInput = None,
+        text: TextInput
+        | PreTokenizedInput
+        | list[TextInput]
+        | list[PreTokenizedInput]
+        | None = None,
+        audio: AudioInput | None = None,
         return_tensors: str = "pt",
         **kwargs,
     ) -> BatchFeature:
-        """
-        Main method to prepare for the model one or several sequences(s) and audio(s).
+        if text is not None:
+            if not isinstance(text, list):
+                text = [text]
 
-        Args:
-            text (`str`, `List[str]`):
-                The sequence or batch of sequences to be encoded.
-            audio (`np.ndarray`, `List[np.ndarray]`):
-                The audio or batch of audio to be prepared. Each audio can be a NumPy array.
-            return_tensors (`str`):
-                The type of tensors to return ("pt", "np", etc.)
-        """
-        if text is None:
-            raise ValueError("You need to specify either a `text` input to process.")
+            text_inputs = self.tokenizer(
+                text, return_tensors=return_tensors, padding=True
+            )
+        else:
+            text_inputs = {}
 
-        # Process audio if provided
         if audio is not None:
             # Ensure audio is a list
             if isinstance(audio, np.ndarray):
@@ -144,19 +99,6 @@ class KimiAudioProcessor(ProcessorMixin):
         else:
             audio_inputs = {}
 
-        # Handle text input - can be string or token IDs from vLLM processor
-        if isinstance(text, list) and len(text) > 0 and isinstance(text[0], int):
-            # Text is already token IDs (from vLLM processor) - just wrap
-            text_inputs = {"input_ids": torch.tensor([text], dtype=torch.long)}
-        else:
-            # Text is string - tokenize
-            if not isinstance(text, list):
-                text = [text]
-
-            text_inputs = self.tokenizer(
-                text, return_tensors=return_tensors, padding=True
-            )
-
         return BatchFeature(
             data={**text_inputs, **audio_inputs},
             tensor_type=return_tensors,

vllm/transformers_utils/processors/qwen_vl.py

 # SPDX-License-Identifier: Apache-2.0
 # SPDX-FileCopyrightText: Copyright contributors to the vLLM project
+
+# Adapted from
+# https://huggingface.co/Qwen/Qwen-VL/blob/main/modeling_qwen.py
+# Copyright (c) Alibaba Cloud.
 from transformers.image_processing_utils_fast import BaseImageProcessorFast
 from transformers.image_utils import PILImageResampling
 from transformers.processing_utils import ProcessorMixin