[Models] Qwen3-ASR (#33312)

Signed-off-by: Roger Wang <hey@rogerw.io>

docs/models/supported_models.md

@@ -720,6 +720,7 @@ These models primarily accept the [`LLM.generate`](./generative_models.md#llmgen
 | `Qwen3VLForConditionalGeneration` | Qwen3-VL | T + I<sup>E+</sup> + V<sup>E+</sup> | `Qwen/Qwen3-VL-4B-Instruct`, etc. | ✅︎ | ✅︎ |
 | `Qwen3VLMoeForConditionalGeneration` | Qwen3-VL-MOE | T + I<sup>E+</sup> + V<sup>E+</sup> | `Qwen/Qwen3-VL-30B-A3B-Instruct`, etc. | ✅︎ | ✅︎ |
 | `Qwen3OmniMoeThinkerForConditionalGeneration` | Qwen3-Omni | T + I<sup>E+</sup> + V<sup>E+</sup> + A<sup>+</sup> | `Qwen/Qwen3-Omni-30B-A3B-Instruct`, `Qwen/Qwen3-Omni-30B-A3B-Thinking` | ✅︎ | ✅︎ |
+| `Qwen3ASRForConditionalGeneration` | Qwen3-ASR | T + A<sup>+</sup> | `Qwen/Qwen3-ASR-1.7B` | ✅︎ | ✅︎ |
 | `RForConditionalGeneration` | R-VL-4B | T + I<sup>E+</sup> | `YannQi/R-4B` | | ✅︎ |
 | `SkyworkR1VChatModel` | Skywork-R1V-38B | T + I | `Skywork/Skywork-R1V-38B` | | ✅︎ |
 | `SmolVLMForConditionalGeneration` | SmolVLM2 | T + I | `SmolVLM2-2.2B-Instruct` | ✅︎ | |
@@ -769,6 +770,7 @@ Speech2Text models trained specifically for Automatic Speech Recognition.
 | `Gemma3nForConditionalGeneration` | Gemma3n | `google/gemma-3n-E2B-it`, `google/gemma-3n-E4B-it`, etc. | | |
 | `GlmAsrForConditionalGeneration` | GLM-ASR | `zai-org/GLM-ASR-Nano-2512` | ✅︎ | ✅︎ |
 | `GraniteSpeechForConditionalGeneration` | Granite Speech | `ibm-granite/granite-speech-3.3-2b`, `ibm-granite/granite-speech-3.3-8b`, etc. | ✅︎ | ✅︎ |
+| `Qwen3ASRForConditionalGeneration` | Qwen3-ASR | `Qwen/Qwen3-ASR-1.7B`, etc. | | ✅︎ |
 | `VoxtralForConditionalGeneration` | Voxtral (Mistral format) | `mistralai/Voxtral-Mini-3B-2507`, `mistralai/Voxtral-Small-24B-2507`, etc. | ✅︎ | ✅︎ |
 | `WhisperForConditionalGeneration` | Whisper | `openai/whisper-small`, `openai/whisper-large-v3-turbo`, etc. | | |
 

examples/offline_inference/audio_language.py

@@ -330,6 +330,25 @@ def run_qwen2_5_omni(question: str, audio_count: int):
     )
 
 
+def run_qwen3_asr(question: str, audio_count: int) -> ModelRequestData:
+    model_name = "Qwen/Qwen3-Asr-1.7B"
+
+    audio_in_prompt = "<|audio_start|><|audio_pad|><|audio_end|>\n" * audio_count
+    prompt = f"<|im_start|>user\n{audio_in_prompt}<|im_end|>\n<|im_start|>assistant\n"
+
+    engine_args = EngineArgs(
+        model=model_name,
+        max_model_len=4096,
+        max_num_seqs=5,
+        limit_mm_per_prompt={"audio": audio_count},
+    )
+
+    return ModelRequestData(
+        engine_args=engine_args,
+        prompt=prompt,
+    )
+
+
 # Ultravox 0.5-1B
 def run_ultravox(question: str, audio_count: int) -> ModelRequestData:
     model_name = "fixie-ai/ultravox-v0_5-llama-3_2-1b"
@@ -442,6 +461,7 @@ model_example_map = {
     "phi4_mm": run_phi4mm,
     "qwen2_audio": run_qwen2_audio,
     "qwen2_5_omni": run_qwen2_5_omni,
+    "qwen3_asr": run_qwen3_asr,
     "ultravox": run_ultravox,
     "voxtral": run_voxtral,
     "whisper": run_whisper,

tests/models/registry.py

@@ -944,6 +944,12 @@ _MULTIMODAL_EXAMPLE_MODELS = {
         max_model_len=4096,
         min_transformers_version="4.57",
     ),
+    "Qwen3ASRForConditionalGeneration": _HfExamplesInfo(
+        "Qwen/Qwen3-ASR-1.7B",
+        max_model_len=4096,
+        min_transformers_version="4.57",
+        is_available_online=False,
+    ),
     "RForConditionalGeneration": _HfExamplesInfo("YannQi/R-4B", trust_remote_code=True),
     "SkyworkR1VChatModel": _HfExamplesInfo(
         "Skywork/Skywork-R1V-38B", trust_remote_code=True

vllm/model_executor/models/qwen3_asr.py

@@ -0,0 +1,567 @@
+# SPDX-License-Identifier: Apache-2.0
+# SPDX-FileCopyrightText: Copyright contributors to the vLLM project
+# Copyright 2026 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 Qwen3-ASR model."""
+
+from collections.abc import Iterable, Mapping, Sequence
+from typing import Any, Literal, cast
+
+import numpy as np
+import torch
+import torch.nn as nn
+from transformers.feature_extraction_utils import BatchFeature
+from transformers.models.whisper import WhisperFeatureExtractor
+
+from vllm.config import ModelConfig, SpeechToTextConfig, VllmConfig
+from vllm.config.multimodal import BaseDummyOptions
+from vllm.inputs.data import PromptType
+from vllm.logger import init_logger
+from vllm.model_executor.models.interfaces import (
+    MultiModalEmbeddings,
+    SupportsMRoPE,
+    SupportsMultiModal,
+    SupportsPP,
+    SupportsTranscription,
+)
+from vllm.model_executor.models.module_mapping import MultiModelKeys
+from vllm.model_executor.models.qwen3 import Qwen3ForCausalLM
+from vllm.model_executor.models.qwen3_omni_moe_thinker import (
+    Qwen2_5OmniAudioFeatureInputs,
+    Qwen3OmniMoeAudioEncoder,
+    Qwen3OmniMoeThinkerMultiModalProcessor,
+)
+from vllm.model_executor.models.utils import (
+    AutoWeightsLoader,
+    WeightsMapper,
+    _merge_multimodal_embeddings,
+    maybe_prefix,
+)
+from vllm.model_executor.models.whisper import ISO639_1_SUPPORTED_LANGS
+from vllm.multimodal import MULTIMODAL_REGISTRY
+from vllm.multimodal.inputs import (
+    AudioItem,
+    ModalityData,
+    MultiModalDataDict,
+    MultiModalFeatureSpec,
+    MultiModalFieldConfig,
+    MultiModalKwargsItems,
+)
+from vllm.multimodal.parse import (
+    AudioProcessorItems,
+    DictEmbeddingItems,
+    ModalityDataItems,
+    MultiModalDataItems,
+    MultiModalDataParser,
+)
+from vllm.multimodal.processing import (
+    BaseDummyInputsBuilder,
+    BaseProcessingInfo,
+    PromptReplacement,
+    PromptUpdate,
+)
+from vllm.sequence import IntermediateTensors
+from vllm.tokenizers import cached_tokenizer_from_config
+from vllm.transformers_utils.configs.qwen3_asr import (
+    Qwen3ASRConfig,
+    Qwen3ASRThinkerConfig,
+)
+from vllm.transformers_utils.processor import cached_processor_from_config
+from vllm.transformers_utils.processors.qwen3_asr import (
+    Qwen3ASRProcessor,
+)
+
+logger = init_logger(__name__)
+
+
+def _get_feat_extract_output_lengths(input_lengths: torch.Tensor):
+    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
+
+
+class Qwen3ASRProcessingInfo(BaseProcessingInfo):
+    def get_hf_config(self):
+        return self.ctx.get_hf_config(Qwen3ASRConfig).thinker_config
+
+    def get_hf_processor(self, **kwargs: object) -> Qwen3ASRProcessor:
+        processor = self.ctx.get_hf_processor(
+            Qwen3ASRProcessor,
+            use_fast=kwargs.pop("use_fast", True),
+            **kwargs,
+        )
+        if not hasattr(processor, "audio_token"):
+            processor.audio_token = "<|audio_pad|>"
+        return processor
+
+    def get_feature_extractor(self, **kwargs: object) -> WhisperFeatureExtractor:
+        hf_processor = self.get_hf_processor(**kwargs)
+        feature_extractor = hf_processor.feature_extractor
+        assert isinstance(feature_extractor, WhisperFeatureExtractor)
+        return feature_extractor
+
+    def get_supported_mm_limits(self) -> Mapping[str, int | None]:
+        return {"audio": None}
+
+
+class Qwen3ASRDummyInputsBuilder(BaseDummyInputsBuilder[Qwen3ASRProcessingInfo]):
+    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],
+        mm_options: Mapping[str, BaseDummyOptions] | None = None,
+    ) -> MultiModalDataDict:
+        num_audios = mm_counts.get("audio", 0)
+
+        feature_extractor = self.info.get_feature_extractor()
+
+        target_audio_length = (
+            min(
+                feature_extractor.chunk_length,
+                30,
+            )
+            * feature_extractor.sampling_rate
+        )
+
+        audio_overrides = mm_options.get("audio") if mm_options else None
+
+        return {
+            "audio": self._get_dummy_audios(
+                length=target_audio_length,
+                num_audios=num_audios,
+                overrides=audio_overrides,
+            ),
+        }
+
+
+def _qwen3asr_field_config(hf_inputs: Mapping[str, torch.Tensor]):
+    audio_feature_lengths = hf_inputs.get("audio_feature_lengths", torch.empty((0,)))
+    return dict(
+        input_audio_features=MultiModalFieldConfig.flat_from_sizes(
+            "audio", audio_feature_lengths, dim=1
+        ),
+        feature_attention_mask=MultiModalFieldConfig.batched("audio"),
+        audio_feature_lengths=MultiModalFieldConfig.batched("audio"),
+    )
+
+
+class Qwen3ASRMultiModalDataParser(MultiModalDataParser):
+    def _parse_audio_data(
+        self,
+        data: dict[str, torch.Tensor] | ModalityData[AudioItem],
+    ) -> ModalityDataItems[Any, Any] | None:
+        if isinstance(data, dict):
+            return DictEmbeddingItems(
+                data,
+                modality="audio",
+                required_fields={"input_audio_features", "audio_feature_lengths"},
+                fields_factory=_qwen3asr_field_config,
+            )
+
+        return super()._parse_audio_data(data)
+
+
+class Qwen3ASRMultiModalProcessor(
+    Qwen3OmniMoeThinkerMultiModalProcessor,
+):
+    def _get_data_parser(self) -> MultiModalDataParser:
+        feature_extractor = self.info.get_feature_extractor()
+        return Qwen3ASRMultiModalDataParser(
+            target_sr=feature_extractor.sampling_rate,
+        )
+
+    def _get_mm_fields_config(
+        self,
+        hf_inputs: BatchFeature,
+        hf_processor_mm_kwargs: Mapping[str, object],
+    ) -> Mapping[str, MultiModalFieldConfig]:
+        return _qwen3asr_field_config(hf_inputs)
+
+    def _get_prompt_updates(
+        self,
+        mm_items: MultiModalDataItems,
+        hf_processor_mm_kwargs: Mapping[str, Any],
+        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 = processor.audio_token
+        audio_token_id = vocab[audio_token]
+
+        out_mm_data = out_mm_kwargs.get_data()
+        audio_feature_lengths = out_mm_data.get("audio_feature_lengths")
+        feature_attention_mask = out_mm_data.get("feature_attention_mask")
+        if audio_feature_lengths is None and feature_attention_mask is None:
+            audio_output_lengths = []
+        elif audio_feature_lengths is not None:
+            audio_output_lens = _get_feat_extract_output_lengths(audio_feature_lengths)
+            audio_output_lengths = audio_output_lens.tolist()
+        elif feature_attention_mask is not None:
+            assert isinstance(feature_attention_mask, torch.Tensor)
+            audio_output_lens = _get_feat_extract_output_lengths(
+                feature_attention_mask.sum(-1)
+            )
+            audio_output_lengths = audio_output_lens.tolist()
+
+        def get_replacement_qwen2_audio(item_idx: int):
+            num_features = audio_output_lengths[item_idx]
+            if num_features == 0:
+                audios = mm_items.get_items("audio", AudioProcessorItems)
+                audio = audios.get(item_idx)
+                raise ValueError(
+                    f"The audio {audio} (len={len(audio)}) is too short "
+                    "to be represented inside the model"
+                )
+
+            return [audio_token_id] * num_features
+
+        return [
+            PromptReplacement(
+                modality="audio",
+                target=audio_token,
+                replacement=get_replacement_qwen2_audio,
+            ),
+        ]
+
+
+@MULTIMODAL_REGISTRY.register_processor(
+    Qwen3ASRMultiModalProcessor,
+    info=Qwen3ASRProcessingInfo,
+    dummy_inputs=Qwen3ASRDummyInputsBuilder,
+)
+class Qwen3ASRForConditionalGeneration(
+    nn.Module,
+    SupportsMultiModal,
+    SupportsPP,
+    SupportsMRoPE,
+    SupportsTranscription,
+):
+    supported_languages = ISO639_1_SUPPORTED_LANGS
+
+    hf_to_vllm_mapper = WeightsMapper(
+        orig_to_new_prefix={
+            "thinker.lm_head.": "language_model.lm_head.",
+            "thinker.model.": "language_model.model.",
+            "thinker.": "",
+        }
+    )
+
+    @classmethod
+    def get_placeholder_str(cls, modality: str, i: int) -> str | None:
+        if modality.startswith("audio"):
+            return "<|audio_start|><|audio_pad|><|audio_end|>"
+
+        raise ValueError("Only audio modality is supported")
+
+    def __init__(self, *, vllm_config: VllmConfig, prefix: str = ""):
+        super().__init__()
+        self.vllm_config = vllm_config  # needed for torch compile forward context
+        thinker_config: Qwen3ASRThinkerConfig = (
+            vllm_config.model_config.hf_config.thinker_config
+        )
+        quant_config = vllm_config.quant_config
+        multimodal_config = vllm_config.model_config.multimodal_config
+        self.config = thinker_config
+        self.multimodal_config = multimodal_config
+
+        self.audio_tower = Qwen3OmniMoeAudioEncoder(
+            thinker_config.audio_config,
+            prefix=maybe_prefix(prefix, "audio_tower"),
+        )
+        self.quant_config = quant_config
+
+        self.language_model = Qwen3ForCausalLM(
+            vllm_config=vllm_config.with_hf_config(
+                thinker_config.text_config, architectures=["Qwen3ForCausalLM"]
+            ),
+            prefix=maybe_prefix(prefix, "language_model"),
+        )
+
+        self.make_empty_intermediate_tensors = (
+            self.language_model.make_empty_intermediate_tensors
+        )
+
+    def _parse_and_validate_audio_input(
+        self, **kwargs: object
+    ) -> Qwen2_5OmniAudioFeatureInputs | None:
+        input_audio_features = kwargs.pop("input_audio_features", None)
+        audio_feature_lengths = kwargs.pop("audio_feature_lengths", None)
+        feature_attention_mask = kwargs.pop("feature_attention_mask", None)
+        if input_audio_features is None:
+            return None
+
+        return Qwen2_5OmniAudioFeatureInputs(
+            type="audio_features",
+            input_features=input_audio_features,
+            audio_feature_lengths=audio_feature_lengths,
+            feature_attention_mask=feature_attention_mask,
+        )
+
+    def _parse_and_validate_multimodal_inputs(self, **kwargs: object) -> dict:
+        mm_input_by_modality = {}
+
+        # Preserve the order of modalities if there are multiple of them
+        # from the order of kwargs.
+        for input_key in kwargs:
+            if (
+                input_key in ("input_audio_features")
+                and "audio" not in mm_input_by_modality
+            ):
+                mm_input_by_modality["audio"] = self._parse_and_validate_audio_input(
+                    **kwargs
+                )
+        return mm_input_by_modality
+
+    def _process_audio_input(
+        self,
+        audio_input: Qwen2_5OmniAudioFeatureInputs,
+        audio_hashes: list[str] | None = None,
+        cached_audio_features: torch.Tensor | None = None,
+    ) -> torch.Tensor:
+        input_features = audio_input["input_features"]
+        audio_feature_lengths = audio_input["audio_feature_lengths"]
+
+        audio_output_lengths = _get_feat_extract_output_lengths(audio_feature_lengths)
+
+        audio_features = self.audio_tower(
+            input_features.to(self.audio_tower.dtype),
+            feature_lens=audio_feature_lengths,
+            aftercnn_lens=audio_output_lengths,
+        )
+        return audio_features.split(audio_output_lengths.tolist())
+
+    def get_language_model(self) -> torch.nn.Module:
+        return self.language_model
+
+    def embed_multimodal(self, **kwargs: object) -> MultiModalEmbeddings | None:
+        mm_input_by_modality = self._parse_and_validate_multimodal_inputs(**kwargs)
+        if not mm_input_by_modality:
+            return []
+
+        # The result multimodal_embeddings is tuple of tensors, with each
+        # tensor correspoending to a multimodal data item (image or video).
+        multimodal_embeddings: tuple[torch.Tensor, ...] = ()
+
+        # NOTE: It is important to iterate over the keys in this dictionary
+        # to preserve the order of the modalities.
+        for modality in mm_input_by_modality:
+            multimodal_input = mm_input_by_modality[modality]
+            if modality == "audio":
+                audio_embeddings = self._process_audio_input(multimodal_input)
+                multimodal_embeddings += tuple(audio_embeddings)
+        return multimodal_embeddings
+
+    def embed_input_ids(
+        self,
+        input_ids: torch.Tensor,
+        multimodal_embeddings: MultiModalEmbeddings | None = None,
+        *,
+        is_multimodal: torch.Tensor | None = None,
+        handle_oov_mm_token: bool = False,
+    ) -> torch.Tensor:
+        inputs_embeds = self._embed_text_input_ids(
+            input_ids,
+            self.language_model.embed_input_ids,
+            is_multimodal=is_multimodal,
+            handle_oov_mm_token=handle_oov_mm_token,
+        )
+
+        if multimodal_embeddings is None or len(multimodal_embeddings) == 0:
+            return inputs_embeds
+
+        inputs_embeds = _merge_multimodal_embeddings(
+            inputs_embeds=inputs_embeds,
+            multimodal_embeddings=multimodal_embeddings,
+            is_multimodal=is_multimodal,
+        )
+
+        return inputs_embeds
+
+    def forward(
+        self,
+        input_ids: torch.Tensor,
+        positions: torch.Tensor,
+        intermediate_tensors: IntermediateTensors | None = None,
+        inputs_embeds: torch.Tensor | None = None,
+        **kwargs: object,
+    ) -> torch.Tensor | IntermediateTensors:
+        if intermediate_tensors is not None:
+            inputs_embeds = None
+
+        hidden_states = self.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 | None:
+        return self.language_model.compute_logits(hidden_states)
+
+    def load_weights(self, weights: Iterable[tuple[str, torch.Tensor]]) -> set[str]:
+        loader = AutoWeightsLoader(
+            self,
+            skip_prefixes=["talker.", "code2wav."],
+        )
+        loaded_weights = loader.load_weights(weights, mapper=self.hf_to_vllm_mapper)
+
+        return loaded_weights
+
+    def get_mrope_input_positions(
+        self,
+        input_tokens: list[int],
+        mm_features: list[MultiModalFeatureSpec],
+    ) -> tuple[torch.Tensor, int]:
+        seq_len = len(input_tokens)
+
+        if not mm_features:
+            # No audio features, just return linear positions
+            llm_positions = (
+                torch.arange(seq_len, dtype=torch.long).view(1, -1).expand(3, -1)
+            )
+            return llm_positions.clone(), 0
+
+        llm_pos_ids_list: list[torch.Tensor] = []
+        st = 0
+
+        for mm_feature in sorted(mm_features, key=lambda f: f.mm_position.offset):
+            offset = mm_feature.mm_position.offset
+
+            # Get audio feature length from mm_feature data
+            audio_feature_length = mm_feature.data["audio_feature_lengths"].data
+            if isinstance(audio_feature_length, torch.Tensor):
+                audio_feature_length = audio_feature_length.item()
+            audio_len = _get_feat_extract_output_lengths(
+                torch.tensor(audio_feature_length)
+            ).item()
+
+            # Text segment before audio (includes audio_start token)
+            text_len = offset - st
+            st_idx = llm_pos_ids_list[-1].max() + 1 if llm_pos_ids_list else 0
+            text_positions = (
+                torch.arange(text_len, dtype=torch.long).view(1, -1).expand(3, -1)
+                + st_idx
+            )
+            llm_pos_ids_list.append(text_positions)
+            st_idx = st_idx + text_len
+
+            # Audio token segment
+            audio_positions = (
+                torch.arange(audio_len, dtype=torch.long).view(1, -1).expand(3, -1)
+                + st_idx
+            )
+            llm_pos_ids_list.append(audio_positions)
+
+            st = offset + audio_len
+
+        # Handle remaining text (includes audio_end and any trailing text)
+        if st < seq_len:
+            st_idx = llm_pos_ids_list[-1].max() + 1 if llm_pos_ids_list else 0
+            text_len = seq_len - st
+            final_text_positions = (
+                torch.arange(text_len, dtype=torch.long).view(1, -1).expand(3, -1)
+                + st_idx
+            )
+            llm_pos_ids_list.append(final_text_positions)
+
+        llm_positions = torch.cat(llm_pos_ids_list, dim=1).reshape(3, -1)
+        if llm_positions.shape[1] != seq_len:
+            raise RuntimeError("Position ids length mismatch with input ids length")
+
+        mrope_position_delta = (llm_positions.max() + 1 - seq_len).item()
+        return llm_positions, mrope_position_delta
+
+    def get_mm_mapping(self) -> MultiModelKeys:
+        """
+        Get the module prefix in multimodal models
+        """
+        return MultiModelKeys.from_string_field(
+            language_model="language_model",
+            tower_model=["audio_tower."],
+        )
+
+    @classmethod
+    def get_speech_to_text_config(
+        cls, model_config: ModelConfig, task_type: str
+    ) -> SpeechToTextConfig:
+        processor = cached_processor_from_config(model_config)
+        feature_extractor: WhisperFeatureExtractor = processor.feature_extractor
+        return SpeechToTextConfig(
+            max_audio_clip_s=feature_extractor.chunk_length,
+            sample_rate=feature_extractor.sampling_rate,
+        )
+
+    @classmethod
+    def get_generation_prompt(
+        cls,
+        audio: np.ndarray,
+        model_config: ModelConfig,
+        stt_config: SpeechToTextConfig,
+        language: str | None,
+        task_type: Literal["transcribe", "translate"],
+        request_prompt: str,
+        to_language: str | None,
+    ) -> PromptType:
+        """Get the generation prompt to be used for transcription requests."""
+        tokenizer = cached_tokenizer_from_config(model_config)
+        audio_placeholder = cls.get_placeholder_str("audio", 0)
+
+        if task_type not in ("transcribe", "translate"):
+            raise ValueError(
+                f"Unsupported task_type '{task_type}'. "
+                "Supported task types are 'transcribe' and 'translate'."
+            )
+        full_lang_name_to = cls.supported_languages.get(to_language, to_language)
+        if to_language is None:
+            prompt = (
+                f"<|im_start|>user\n{audio_placeholder}<|im_end|>\n"
+                f"<|im_start|>assistant\n"
+            )
+        else:
+            prompt = (
+                f"<|im_start|>user\n{audio_placeholder}<|im_end|>\n"
+                f"<|im_start|>assistant\nlanguage {full_lang_name_to}<asr_text>"
+            )
+
+        prompt_token_ids = tokenizer.encode(prompt)
+        prompt_dict = {
+            "prompt_token_ids": prompt_token_ids,
+            "multi_modal_data": {"audio": audio},
+        }
+        return cast(PromptType, prompt_dict)

vllm/model_executor/models/registry.py

@@ -436,6 +436,10 @@ _MULTIMODAL_MODELS = {
         "qwen3_omni_moe_thinker",
         "Qwen3OmniMoeThinkerForConditionalGeneration",
     ),
+    "Qwen3ASRForConditionalGeneration": (
+        "qwen3_asr",
+        "Qwen3ASRForConditionalGeneration",
+    ),
     "Qwen3VLForConditionalGeneration": ("qwen3_vl", "Qwen3VLForConditionalGeneration"),  # noqa: E501
     "Qwen3VLMoeForConditionalGeneration": (
         "qwen3_vl_moe",

vllm/transformers_utils/config.py

@@ -97,6 +97,7 @@ _CONFIG_REGISTRY: dict[str, type[PretrainedConfig]] = LazyConfigDict(
     ultravox="UltravoxConfig",
     step3_vl="Step3VLConfig",
     step3_text="Step3TextConfig",
+    qwen3_asr="Qwen3ASRConfig",
     qwen3_next="Qwen3NextConfig",
     lfm2_moe="Lfm2MoeConfig",
     tarsier2="Tarsier2Config",

vllm/transformers_utils/configs/__init__.py

@@ -52,6 +52,7 @@ _CLASS_TO_MODULE: dict[str, str] = {
     "Step3VLConfig": "vllm.transformers_utils.configs.step3_vl",
     "Step3VisionEncoderConfig": "vllm.transformers_utils.configs.step3_vl",
     "Step3TextConfig": "vllm.transformers_utils.configs.step3_vl",
+    "Qwen3ASRConfig": "vllm.transformers_utils.configs.qwen3_asr",
     "Qwen3NextConfig": "vllm.transformers_utils.configs.qwen3_next",
     "Tarsier2Config": "vllm.transformers_utils.configs.tarsier2",
     # Special case: DeepseekV3Config is from HuggingFace Transformers
@@ -94,6 +95,7 @@ __all__ = [
     "Step3VLConfig",
     "Step3VisionEncoderConfig",
     "Step3TextConfig",
+    "Qwen3ASRConfig",
     "Qwen3NextConfig",
     "Tarsier2Config",
 ]

vllm/transformers_utils/configs/qwen3_asr.py

@@ -0,0 +1,436 @@
+# 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 Qwen team, Alibaba Group 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.
+# 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 transformers.configuration_utils import PretrainedConfig
+from transformers.modeling_rope_utils import rope_config_validation
+from transformers.utils import logging
+
+logger = logging.get_logger(__name__)
+
+
+class Qwen3ASRAudioEncoderConfig(PretrainedConfig):
+    r"""
+    This is the configuration class to store the configuration of a [`Qwen3ASRAudioEncoder`]. It is used to instantiate a
+    Qwen3-ASR audio encoder according to the specified arguments, defining the model architecture. Instantiating a
+    configuration with the defaults will yield a similar configuration to that of the audio encoder of the Qwen2-Audio
+    architecture.
+
+    e.g. [Qwen/Qwen3-ASR-1.7B](https://huggingface.co/Qwen/Qwen3-ASR-1.7B)
+
+    Configuration objects inherit from [`PretrainedConfig`] and can be used to control the model outputs. Read the
+    documentation from [`PretrainedConfig`] for more information.
+
+    Args:
+        num_mel_bins (`int`, *optional*, defaults to 128):
+            Number of mel features used per input features. Should correspond to the value used in the
+            `Qwen3ASRProcessor` class.
+        encoder_layers (`int`, *optional*, defaults to 32):
+            Number of encoder layers.
+        encoder_attention_heads (`int`, *optional*, defaults to 20):
+            Number of attention heads for each attention layer in the Transformer encoder.
+        encoder_ffn_dim (`int`, *optional*, defaults to 5120):
+            Dimensionality of the "intermediate" (often named feed-forward) layer in encoder.
+        d_model (`int`, *optional*, defaults to 1280):
+            Dimensionality of the layers.
+        dropout (`float`, *optional*, defaults to 0.0):
+            The dropout probability for all fully connected layers in the embeddings, encoder, and pooler.
+        attention_dropout (`float`, *optional*, defaults to 0.0):
+            The dropout ratio for the attention probabilities.
+        activation_function (`str`, *optional*, defaults to `"gelu"`):
+            The non-linear activation function (function or string) in the encoder and pooler. If string, `"gelu"`,
+            `"relu"`, `"silu"` and `"gelu_new"` are supported.
+        activation_dropout (`float`, *optional*, defaults to 0.0):
+            The dropout ratio for activations inside the fully connected layer.
+        scale_embedding (`bool`, *optional*, defaults to `False`):
+            Scale embeddings by diving by sqrt(d_model).
+        initializer_range (`float`, *optional*, defaults to 0.02):
+            The standard deviation of the truncated_normal_initializer for initializing all weight matrices.
+        max_source_positions (`int`, *optional*, defaults to 1500):
+            The maximum sequence length of log-mel filter-bank features that this model might ever be used with.
+        n_window (`int`, *optional*, defaults to 100):
+            The chunk for conv and flash attn in AudioEncoder.
+        output_dim (`int`, *optional*, defaults to 3584):
+            The output dimension of AudioEncoder.
+
+    Example:
+
+    ```python
+    >>> from transformers import Qwen3ASRAudioEncoderConfig, Qwen3ASRAudioEncoder
+
+    >>> # Initializing a Qwen3ASRAudioEncoderConfig
+    >>> configuration = Qwen3ASRAudioEncoderConfig()
+
+    >>> # Initializing a Qwen3ASRAudioEncoder (with random weights)
+    >>> model = Qwen3ASRAudioEncoder(configuration)
+
+    >>> # Accessing the model configuration
+    >>> configuration = model.config
+    ```"""
+
+    model_type = "qwen3_asr_audio_encoder"
+
+    def __init__(
+        self,
+        num_mel_bins=128,
+        encoder_layers=32,
+        encoder_attention_heads=20,
+        encoder_ffn_dim=5120,
+        d_model=1280,
+        dropout=0,
+        attention_dropout=0,
+        activation_function="gelu",
+        activation_dropout=0,
+        scale_embedding=False,
+        initializer_range=0.02,
+        max_source_positions=1500,
+        n_window=100,
+        output_dim=3584,
+        n_window_infer=400,
+        conv_chunksize=500,
+        downsample_hidden_size=480,
+        **kwargs,
+    ):
+        super().__init__(**kwargs)
+
+        self.num_mel_bins = num_mel_bins
+        self.d_model = d_model
+        self.encoder_layers = encoder_layers
+        self.encoder_attention_heads = encoder_attention_heads
+        self.encoder_ffn_dim = encoder_ffn_dim
+        self.dropout = dropout
+        self.attention_dropout = attention_dropout
+        self.activation_function = activation_function
+        self.activation_dropout = activation_dropout
+        self.num_hidden_layers = encoder_layers
+        self.initializer_range = initializer_range
+        self.scale_embedding = (
+            scale_embedding  # scale factor will be sqrt(d_model) if True
+        )
+        self.max_source_positions = max_source_positions
+        self.n_window = n_window
+        self.output_dim = output_dim
+        self.n_window_infer = n_window_infer
+        self.conv_chunksize = conv_chunksize
+        self.downsample_hidden_size = downsample_hidden_size
+
+
+class Qwen3ASRTextConfig(PretrainedConfig):
+    r"""
+    This is the configuration class to store the configuration of a [`Qwen3ASRTextModel`]. It is used to instantiate a
+    Qwen3-ASR model according to the specified arguments, defining the model architecture. Instantiating a configuration
+    with the defaults will yield a similar configuration to that of
+    Qwen3-ASR-1.7B [Qwen/Qwen3-ASR-1.7B](https://huggingface.co/Qwen/Qwen3-ASR-1.7B)
+
+    Configuration objects inherit from [`PretrainedConfig`] and can be used to control the model outputs. Read the
+    documentation from [`PretrainedConfig`] for more information.
+
+    Args:
+        vocab_size (`int`, *optional*, defaults to 151936):
+            Vocabulary size of the Qwen3ASR model. Defines the number of different tokens that can be represented by the
+            `inputs_ids` passed when calling [`Qwen3ASRModel`]
+        hidden_size (`int`, *optional*, defaults to 4096):
+            Dimension of the hidden representations.
+        intermediate_size (`int`, *optional*, defaults to 22016):
+            Dimension of the MLP representations.
+        num_hidden_layers (`int`, *optional*, defaults to 32):
+            Number of hidden layers in the Transformer encoder.
+        num_attention_heads (`int`, *optional*, defaults to 32):
+            Number of attention heads for each attention layer in the Transformer encoder.
+        num_key_value_heads (`int`, *optional*, defaults to 32):
+            This is the number of key_value heads that should be used to implement Grouped Query Attention. If
+            `num_key_value_heads=num_attention_heads`, the model will use Multi Head Attention (MHA), if
+            `num_key_value_heads=1` the model will use Multi Query Attention (MQA) otherwise GQA is used. When
+            converting a multi-head checkpoint to a GQA checkpoint, each group key and value head should be constructed
+            by meanpooling all the original heads within that group. For more details, check out [this
+            paper](https://huggingface.co/papers/2305.13245). If it is not specified, will default to `32`.
+        head_dim (`int`, *optional*, defaults to 128):
+            The dimension of the head. If not specified, will default to `hidden_size // num_attention_heads`.
+        hidden_act (`str` or `function`, *optional*, defaults to `"silu"`):
+            The non-linear activation function (function or string) in the decoder.
+        max_position_embeddings (`int`, *optional*, defaults to 128000):
+            The maximum sequence length that this model might ever be used with.
+        initializer_range (`float`, *optional*, defaults to 0.02):
+            The standard deviation of the truncated_normal_initializer for initializing all weight matrices.
+        rms_norm_eps (`float`, *optional*, defaults to 1e-06):
+            The epsilon used by the rms normalization layers.
+        use_cache (`bool`, *optional*, defaults to `True`):
+            Whether or not the model should return the last key/values attentions (not used by all models). Only
+            relevant if `config.is_decoder=True`.
+        tie_word_embeddings (`bool`, *optional*, defaults to `False`):
+            Whether the model's input and output word embeddings should be tied.
+        rope_theta (`float`, *optional*, defaults to 5000000.0):
+            The base period of the RoPE embeddings.
+        rope_scaling (`Dict`, *optional*):
+            Dictionary containing the scaling configuration for the RoPE embeddings. NOTE: if you apply new rope type
+            and you expect the model to work on longer `max_position_embeddings`, we recommend you to update this value
+            accordingly.
+            Expected contents:
+                `rope_type` (`str`):
+                    The sub-variant of RoPE to use. Can be one of ['default', 'linear', 'dynamic', 'yarn', 'longrope',
+                    'llama3'], with 'default' being the original RoPE implementation.
+                `factor` (`float`, *optional*):
+                    Used with all rope types except 'default'. The scaling factor to apply to the RoPE embeddings. In
+                    most scaling types, a `factor` of x will enable the model to handle sequences of length x *
+                    original maximum pre-trained length.
+                `original_max_position_embeddings` (`int`, *optional*):
+                    Used with 'dynamic', 'longrope' and 'llama3'. The original max position embeddings used during
+                    pretraining.
+                `attention_factor` (`float`, *optional*):
+                    Used with 'yarn' and 'longrope'. The scaling factor to be applied on the attention
+                    computation. If unspecified, it defaults to value recommended by the implementation, using the
+                    `factor` field to infer the suggested value.
+                `beta_fast` (`float`, *optional*):
+                    Only used with 'yarn'. Parameter to set the boundary for extrapolation (only) in the linear
+                    ramp function. If unspecified, it defaults to 32.
+                `beta_slow` (`float`, *optional*):
+                    Only used with 'yarn'. Parameter to set the boundary for interpolation (only) in the linear
+                    ramp function. If unspecified, it defaults to 1.
+                `short_factor` (`list[float]`, *optional*):
+                    Only used with 'longrope'. The scaling factor to be applied to short contexts (<
+                    `original_max_position_embeddings`). Must be a list of numbers with the same length as the hidden
+                    size divided by the number of attention heads divided by 2
+                `long_factor` (`list[float]`, *optional*):
+                    Only used with 'longrope'. The scaling factor to be applied to long contexts (<
+                    `original_max_position_embeddings`). Must be a list of numbers with the same length as the hidden
+                    size divided by the number of attention heads divided by 2
+                `low_freq_factor` (`float`, *optional*):
+                    Only used with 'llama3'. Scaling factor applied to low frequency components of the RoPE
+                `high_freq_factor` (`float`, *optional*):
+                    Only used with 'llama3'. Scaling factor applied to high frequency components of the RoPE
+        attention_bias (`bool`, defaults to `False`, *optional*, defaults to `False`):
+            Whether to use a bias in the query, key, value and output projection layers during self-attention.
+        attention_dropout (`float`, *optional*, defaults to 0.0):
+            The dropout ratio for the attention probabilities.
+
+    ```python
+    >>> from transformers import Qwen3ASRTextModel, Qwen3ASRTextConfig
+
+    >>> # Initializing a Qwen3ASR style configuration
+    >>> configuration = Qwen3ASRTextConfig()
+
+    >>> # Initializing a model from the Qwen3-VL-7B style configuration
+    >>> model = Qwen3ASRTextModel(configuration)
+
+    >>> # Accessing the model configuration
+    >>> configuration = model.config
+    ```"""
+
+    model_type = "qwen3_asr_text"
+    base_config_key = "text_config"
+
+    def __init__(
+        self,
+        vocab_size=151936,
+        hidden_size=4096,
+        intermediate_size=22016,
+        num_hidden_layers=32,
+        num_attention_heads=32,
+        num_key_value_heads=32,
+        head_dim=128,
+        hidden_act="silu",
+        max_position_embeddings=128000,
+        initializer_range=0.02,
+        rms_norm_eps=1e-6,
+        use_cache=True,
+        tie_word_embeddings=False,
+        rope_theta=5000000.0,
+        rope_scaling=None,
+        attention_bias=False,
+        attention_dropout=0.0,
+        **kwargs,
+    ):
+        self.vocab_size = vocab_size
+        self.max_position_embeddings = max_position_embeddings
+        self.hidden_size = hidden_size
+        self.intermediate_size = intermediate_size
+        self.num_hidden_layers = num_hidden_layers
+        self.num_attention_heads = num_attention_heads
+
+        # for backward compatibility
+        if num_key_value_heads is None:
+            num_key_value_heads = num_attention_heads
+
+        self.num_key_value_heads = num_key_value_heads
+        self.head_dim = head_dim
+        self.hidden_act = hidden_act
+        self.initializer_range = initializer_range
+        self.rms_norm_eps = rms_norm_eps
+        self.use_cache = use_cache
+        self.rope_theta = rope_theta
+        self.rope_scaling = rope_scaling
+        self.attention_bias = attention_bias
+        self.attention_dropout = attention_dropout
+        # Validate the correctness of rotary position embeddings parameters
+        # BC: if there is a 'type' field, move it to 'rope_type'.
+        if self.rope_scaling is not None and "type" in self.rope_scaling:
+            self.rope_scaling["rope_type"] = self.rope_scaling["type"]
+        rope_config_validation(self)
+
+        super().__init__(tie_word_embeddings=tie_word_embeddings, **kwargs)
+
+
+class Qwen3ASRThinkerConfig(PretrainedConfig):
+    r"""
+    This is the configuration class to store the configuration of a [`Qwen3ASRThinker`]. It is used to instantiate a
+    Qwen3-ASR-Thinker model according to the specified arguments, defining the model architecture. Instantiating a
+    configuration with the defaults will yield a similar configuration to that of the thinker component of the Qwen3-Omni
+    architecture.
+
+    e.g. [Qwen/Qwen3-ASR-1.7B](https://huggingface.co/Qwen/Qwen3-ASR-1.7B)
+
+    Configuration objects inherit from [`PretrainedConfig`] and can be used to control the model outputs. Read the
+    documentation from [`PretrainedConfig`] for more information.
+
+    Args:
+        audio_config (`dict`, *optional*):
+            The config dictionary of the audio backbone.
+        text_config (`dict`, *optional*):
+            The config dictionary of the text backbone.
+        audio_token_id (`int`, *optional*, defaults to 151646):
+            The audio token id to encode the audio prompt.
+        audio_start_token_id (`int`, *optional*, defaults to 151647):
+            The audio start token id to encode the audio prompt.
+        user_token_id (`int`, *optional*, defaults to 872):
+            The user token id to encode the user token.
+        initializer_range (`float`, *optional*, defaults to 0.02):
+            The standard deviation of the truncated_normal_initializer for initializing all weight matrices.
+
+    Example:
+
+    ```python
+    >>> from transformers import Qwen3ASRThinkerModel, Qwen3ASRThinkerConfig
+
+    >>> # Initializing a default Qwen3ASRThinkerConfig
+    >>> configuration = Qwen3ASRThinkerConfig()
+
+    >>> # Initializing a model (with random weights) from the default configuration
+    >>> model = Qwen3ASRThinkerModel(configuration)
+
+    >>> # Accessing the model configuration
+    >>> configuration = model.config
+    ```"""
+
+    model_type = "qwen3_asr_thinker"
+
+    attribute_map = {}
+    sub_configs = {
+        "audio_config": Qwen3ASRAudioEncoderConfig,
+        "text_config": Qwen3ASRTextConfig,
+    }
+
+    def __init__(
+        self,
+        audio_config=None,
+        text_config=None,
+        audio_token_id=151646,
+        audio_start_token_id=151647,
+        user_token_id=872,
+        initializer_range=0.02,
+        **kwargs,
+    ):
+        super().__init__(**kwargs)
+        self.user_token_id = user_token_id
+        self.audio_start_token_id = audio_start_token_id
+        self.initializer_range = initializer_range
+
+        if isinstance(audio_config, dict):
+            audio_config = Qwen3ASRAudioEncoderConfig(**audio_config)
+        elif audio_config is None:
+            audio_config = Qwen3ASRAudioEncoderConfig()
+        self.audio_config = audio_config
+
+        if isinstance(text_config, dict):
+            text_config = Qwen3ASRTextConfig(**text_config)
+        elif text_config is None:
+            text_config = Qwen3ASRTextConfig()
+        self.text_config = text_config
+        self.audio_token_id = audio_token_id
+
+
+class Qwen3ASRConfig(PretrainedConfig):
+    """
+    This is the configuration class to store the configuration of a [`Qwen3ASRForConditionalGeneration`]. It is used to instantiate a Qwen3ASR
+    model according to the specified sub-models configurations, defining the model architecture.
+
+    Instantiating a configuration with the defaults will yield a similar configuration to that of the
+    [Qwen/Qwen3-ASR-1.7B](https://huggingface.co/Qwen/Qwen3-ASR-1.7B) architecture.
+
+    Configuration objects inherit from [`PretrainedConfig`] and can be used to control the model outputs. Read the
+    documentation from [`PretrainedConfig`] for more information.
+
+    Args:
+        thinker_config (`dict`, *optional*): Configuration of the underlying thinker sub-model.
+        support_languages (`List[str]`, *optional*): The languages supported by the model.
+
+    Example:
+
+    ```python
+    >>> from transformers import (
+    ...     Qwen3ASRThinkerConfig,
+    ...     Qwen3ASRForConditionalGeneration,
+    ...     Qwen3ASRConfig,
+    ... )
+
+    >>> # Initializing a Qwen3ASR style configuration
+    >>> configuration = Qwen3ASRConfig()
+
+    >>> # Initializing a model from the configuration
+    >>> model = Qwen3ASRForConditionalGeneration(configuration)
+
+    >>> # Accessing the model configuration
+    >>> configuration = model.config
+    ```"""
+
+    model_type = "qwen3_asr"
+    sub_configs = {
+        "thinker_config": Qwen3ASRThinkerConfig,
+    }
+
+    def __init__(
+        self,
+        thinker_config=None,
+        support_languages=None,
+        **kwargs,
+    ):
+        super().__init__(**kwargs)
+        if thinker_config is None:
+            thinker_config = {}
+            logger.info(
+                "thinker_config is None. Initializing thinker model with default values"
+            )
+
+        self.thinker_config = Qwen3ASRThinkerConfig(**thinker_config)
+        self.support_languages = support_languages
+
+    def get_text_config(self, decoder=False) -> "PretrainedConfig":
+        """
+        Returns the config that is meant to be used with text IO. On most models, it is the original config instance
+        itself. On specific composite models, it is under a set of valid names.
+
+        Args:
+            decoder (`Optional[bool]`, *optional*, defaults to `False`):
+                If set to `True`, then only search for decoder config names.
+        """
+        # Overridden for deeply nested config like Qwen2.5-Omni. We don't have any omni model
+        # except for Qwen yet. This has to be generalized if more deeply nested configs are
+        # added. NOTE: currently method used only by vLLM
+        return self.thinker_config.get_text_config()
+
+
+__all__ = ["Qwen3ASRConfig", "Qwen3ASRThinkerConfig", "Qwen3ASRAudioEncoderConfig"]

vllm/transformers_utils/processors/qwen3_asr.py

@@ -0,0 +1,231 @@
+# 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 Qwen team, Alibaba Group 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.
+# 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.
+import regex as re
+
+import numpy as np
+
+from transformers import AutoProcessor
+from transformers.audio_utils import AudioInput
+from transformers.feature_extraction_utils import BatchFeature
+from transformers.processing_utils import ProcessingKwargs, ProcessorMixin
+from transformers.tokenization_utils_base import TextInput
+
+
+class Qwen3ASRProcessorKwargs(ProcessingKwargs, total=False):
+    _defaults = {
+        "text_kwargs": {
+            "padding": False,
+            "padding_side": "left",
+        },
+        "audio_kwargs": {
+            "sampling_rate": 16000,
+            "padding": True,
+            "return_attention_mask": True,
+        },
+    }
+
+
+def _get_feat_extract_output_lengths(input_lengths):
+    """
+    Computes the output length of the convolutional layers and the output length of the audio encoder
+    """
+
+    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
+
+
+class Qwen3ASRProcessor(ProcessorMixin):
+    r"""
+    Constructs a Qwen3ASR processor.
+    [`Qwen3ASRProcessor`] offers all the functionalities of [`WhisperFeatureExtractor`], and [`Qwen2TokenizerFast`]. See the
+    [`~Qwen3ASRProcessor.__call__`] and [`~Qwen3ASRProcessor.decode`] for more information.
+
+    Args:
+        feature_extractor ([`WhisperFeatureExtractor`], *optional*):
+            The audio feature extractor.
+        tokenizer ([`Qwen2TokenizerFast`], *optional*):
+            The text tokenizer.
+        chat_template (`Optional[str]`, *optional*):
+            The Jinja template to use for formatting the conversation. If not provided, the default chat template is used.
+    """
+
+    attributes = ["feature_extractor", "tokenizer"]
+    feature_extractor_class = "WhisperFeatureExtractor"
+    tokenizer_class = ("Qwen2Tokenizer", "Qwen2TokenizerFast")
+
+    def __init__(self, feature_extractor=None, tokenizer=None, chat_template=None):
+        super().__init__(feature_extractor, tokenizer, chat_template=chat_template)
+        self.audio_token = self.tokenizer.audio_token
+        self.audio_bos_token = self.tokenizer.audio_bos_token
+        self.audio_eos_token = self.tokenizer.audio_eos_token
+
+    def __call__(
+        self,
+        text: TextInput = None,
+        audio: AudioInput = None,
+        **kwargs,
+    ) -> BatchFeature:
+        """
+        Main method to prepare for the model one or several sequences(s) and audio(s). This method forwards the `text`
+        and `kwargs` arguments to Qwen2TokenizerFast's [`~Qwen2TokenizerFast.__call__`] if `text` is not `None` to encode
+        the text. To prepare the audio(s), this method forwards the `audio` and `kwargs` arguments to
+        WhisperFeatureExtractor's [`~WhisperFeatureExtractor.__call__`] if `audio` is not `None`. Please refer to the doctsring
+        of the above two methods for more information.
+
+        Args:
+            text (`str`, `List[str]`, `List[List[str]]`):
+                The sequence or batch of sequences to be encoded. Each sequence can be a string or a list of strings
+                (pretokenized string). If the sequences are provided as list of strings (pretokenized), you must set
+                `is_split_into_words=True` (to lift the ambiguity with a batch of sequences).
+            audio (`np.ndarray`, `List[np.ndarray]`):
+                The audio or batch of audio to be prepared. Each audio can be a NumPy array.
+        """
+
+        if text is None:
+            raise ValueError("You need to specify either a `text` input to process.")
+
+        output_kwargs = self._merge_kwargs(
+            Qwen3ASRProcessorKwargs,
+            tokenizer_init_kwargs=self.tokenizer.init_kwargs,
+            **kwargs,
+        )
+
+        if audio is not None:
+            output_kwargs["audio_kwargs"]["padding"] = True
+            output_kwargs["audio_kwargs"]["truncation"] = False
+            audio_inputs = self.feature_extractor(
+                audio, **output_kwargs["audio_kwargs"]
+            )
+            audio_inputs["feature_attention_mask"] = audio_inputs.pop(
+                "attention_mask"
+            )  # rename feature_attention_mask to prevent conflicts later on
+            audio_inputs["input_features"] = audio_inputs.pop(
+                "input_features"
+            )  # rename input_features to prevent conflicts later on
+            audio_lengths = iter(
+                _get_feat_extract_output_lengths(
+                    audio_inputs["feature_attention_mask"].sum(-1)
+                )
+            )
+        else:
+            audio_inputs = {}
+            audio_lengths = iter([])
+
+        if not isinstance(text, list):
+            text = [text]
+
+        text = self.replace_multimodal_special_tokens(
+            text,
+            audio_lengths,
+        )
+
+        texts_inputs = self.tokenizer(text, **output_kwargs["text_kwargs"])
+
+        return BatchFeature(
+            data={**texts_inputs, **audio_inputs},
+            tensor_type=kwargs.get("return_tensors"),
+        )
+
+    def replace_multimodal_special_tokens(
+        self,
+        text,
+        audio_lengths,
+    ):
+        processed_text = []
+        for sample in text:
+            positions = []
+            special_tokens = [re.escape(tok) for tok in [self.audio_token]]
+            pattern = "|".join(special_tokens)
+            positions = sorted(
+                [
+                    (match.start(), match.group())
+                    for match in re.finditer(pattern, sample)
+                ]
+            )
+            positions.sort(key=lambda x: x[0])
+
+            for _, special_token in positions:
+                if special_token == self.audio_token:
+                    sample = sample.replace(
+                        self.audio_token,
+                        "<|audio_placeholder|>" * next(audio_lengths),
+                        1,
+                    )
+
+            sample = sample.replace("<|audio_placeholder|>", self.audio_token)
+            processed_text.append(sample)
+        return processed_text
+
+    def get_chunked_index(
+        self, token_indices: np.ndarray, tokens_per_chunk: int
+    ) -> list[tuple[int, int]]:
+        """
+        Splits token index list into chunks based on token value ranges.
+
+        Given a list of token indices, returns a list of (start, end) index tuples representing
+        slices of the list where the token values fall within successive ranges of `tokens_per_chunk`.
+
+        For example, if `tokens_per_chunk` is 1000, the function will create chunks such that:
+        - the first chunk contains token values < 1000,
+        - the second chunk contains values >= 1000 and < 2000, and so on.
+
+        Parameters:
+            token_indices (`np.ndarray`): A monotonically increasing list of token index values.
+            tokens_per_chunk (`int`): Number of tokens per chunk (used as the chunk size threshold).
+
+        Returns:
+            `list[tuple[int, int]]`: A list of tuples, each representing the start (inclusive)
+                                and end (exclusive) indices of a chunk in `token_indices`.
+        """
+
+        def _iter():
+            i, start_idx = 0, 0  # skip bos token
+            current_chunk = 1
+            while i < len(token_indices):  # skip eos token
+                if token_indices[i] >= current_chunk * tokens_per_chunk:
+                    yield (start_idx, i)
+                    start_idx = i
+                    current_chunk += 1
+                i += 1
+            yield (start_idx, len(token_indices))
+
+        return list(_iter())
+
+    def apply_chat_template(self, conversations, chat_template=None, **kwargs):
+        return super().apply_chat_template(conversations, chat_template, **kwargs)
+
+    @property
+    def model_input_names(self):
+        tokenizer_input_names = self.tokenizer.model_input_names
+        feature_extractor_input_names = self.feature_extractor.model_input_names
+        return list(
+            dict.fromkeys(
+                tokenizer_input_names
+                + feature_extractor_input_names
+                + ["feature_attention_mask"]
+            )
+        )
+
+
+AutoProcessor.register("Qwen3ASRProcessor", Qwen3ASRProcessor)