Voxtral (#20970)

Signed-off-by: Patrick von Platen <patrick.v.platen@gmail.com>
Co-authored-by: Cyrus Leung <cyrus.tl.leung@gmail.com>

examples/offline_inference/audio_language.py

@@ -10,7 +10,7 @@ on HuggingFace model repository.
 
 import os
 from dataclasses import asdict
-from typing import NamedTuple, Optional
+from typing import Any, NamedTuple, Optional
 
 from huggingface_hub import snapshot_download
 from transformers import AutoTokenizer
@@ -30,7 +30,9 @@ question_per_audio_count = {
 
 class ModelRequestData(NamedTuple):
     engine_args: EngineArgs
-    prompt: str
+    prompt: Optional[str] = None
+    prompt_token_ids: Optional[dict[str, list[int]]] = None
+    multi_modal_data: Optional[dict[str, Any]] = None
     stop_token_ids: Optional[list[int]] = None
     lora_requests: Optional[list[LoRARequest]] = None
 
@@ -40,6 +42,60 @@ class ModelRequestData(NamedTuple):
 # Unless specified, these settings have been tested to work on a single L4.
 
 
+# Voxtral
+def run_voxtral(question: str, audio_count: int) -> ModelRequestData:
+    from mistral_common.audio import Audio
+    from mistral_common.protocol.instruct.messages import (
+        AudioChunk,
+        RawAudio,
+        TextChunk,
+        UserMessage,
+    )
+    from mistral_common.protocol.instruct.request import ChatCompletionRequest
+    from mistral_common.tokens.tokenizers.mistral import MistralTokenizer
+
+    model_name = "mistralai/Voxtral-Mini-3B-2507"
+    tokenizer = MistralTokenizer.from_hf_hub(model_name)
+
+    engine_args = EngineArgs(
+        model=model_name,
+        max_model_len=8192,
+        max_num_seqs=2,
+        limit_mm_per_prompt={"audio": audio_count},
+        config_format="mistral",
+        load_format="mistral",
+        tokenizer_mode="mistral",
+        enforce_eager=True,
+        enable_chunked_prefill=False,
+    )
+
+    text_chunk = TextChunk(text=question)
+    audios = [
+        Audio.from_file(str(audio_assets[i].get_local_path()), strict=False)
+        for i in range(audio_count)
+    ]
+    audio_chunks = [
+        AudioChunk(input_audio=RawAudio.from_audio(audio)) for audio in audios
+    ]
+
+    messages = [UserMessage(content=[*audio_chunks, text_chunk])]
+
+    req = ChatCompletionRequest(messages=messages, model=model_name)
+
+    tokens = tokenizer.encode_chat_completion(req)
+    prompt_ids, audios = tokens.tokens, tokens.audios
+
+    audios_and_sr = [(au.audio_array, au.sampling_rate) for au in audios]
+
+    multi_modal_data = {"audio": audios_and_sr}
+
+    return ModelRequestData(
+        engine_args=engine_args,
+        prompt_token_ids=prompt_ids,
+        multi_modal_data=multi_modal_data,
+    )
+
+
 # Granite Speech
 def run_granite_speech(question: str, audio_count: int) -> ModelRequestData:
     # NOTE - the setting in this example are somehat different than what is
@@ -243,6 +299,7 @@ def run_whisper(question: str, audio_count: int) -> ModelRequestData:
 
 
 model_example_map = {
+    "voxtral": run_voxtral,
     "granite_speech": run_granite_speech,
     "minicpmo": run_minicpmo,
     "phi4_mm": run_phi4mm,
@@ -311,16 +368,24 @@ def main(args):
         temperature=0.2, max_tokens=64, stop_token_ids=req_data.stop_token_ids
     )
 
-    mm_data = {}
-    if audio_count > 0:
-        mm_data = {
-            "audio": [
-                asset.audio_and_sample_rate for asset in audio_assets[:audio_count]
-            ]
-        }
+    mm_data = req_data.multi_modal_data
+    if not mm_data:
+        mm_data = {}
+        if audio_count > 0:
+            mm_data = {
+                "audio": [
+                    asset.audio_and_sample_rate for asset in audio_assets[:audio_count]
+                ]
+            }
 
     assert args.num_prompts > 0
-    inputs = {"prompt": req_data.prompt, "multi_modal_data": mm_data}
+    inputs = {"multi_modal_data": mm_data}
+
+    if req_data.prompt:
+        inputs["prompt"] = req_data.prompt
+    else:
+        inputs["prompt_token_ids"] = req_data.prompt_token_ids
+
     if args.num_prompts > 1:
         # Batch inference
         inputs = [inputs] * args.num_prompts

requirements/common.txt

@@ -33,7 +33,7 @@ pyzmq >= 25.0.0
 msgspec
 gguf >= 0.13.0
 importlib_metadata; python_version < '3.10'
-mistral_common[opencv] >= 1.6.2
+mistral_common[opencv] >= 1.8.0
 opencv-python-headless >= 4.11.0    # required for video IO
 pyyaml
 six>=1.16.0; python_version > '3.11' # transitive dependency of pandas that needs to be the latest version for python 3.12

requirements/nightly_torch_test.txt

@@ -23,7 +23,7 @@ jiwer # required for audio tests
 timm # required for internvl test
 transformers_stream_generator # required for qwen-vl test
 matplotlib # required for qwen-vl test
-mistral_common[opencv] >= 1.6.2 # required for pixtral test
+mistral_common[opencv] >= 1.8.0 # required for voxtral test
 num2words # required for smolvlm test
 opencv-python-headless >= 4.11.0 # required for video test
 datamodel_code_generator # required for minicpm3 test

requirements/test.in

@@ -28,7 +28,7 @@ torchvision==0.22.0
 transformers_stream_generator # required for qwen-vl test
 mamba_ssm # required for plamo2 test
 matplotlib # required for qwen-vl test
-mistral_common[opencv] >= 1.7.0 # required for pixtral test
+mistral_common[opencv] >= 1.8.0 # required for voxtral test
 num2words # required for smolvlm test
 opencv-python-headless >= 4.11.0 # required for video test
 datamodel_code_generator # required for minicpm3 test

requirements/test.txt

@@ -305,7 +305,7 @@ mbstrdecoder==1.1.3
     #   typepy
 mdurl==0.1.2
     # via markdown-it-py
-mistral-common==1.7.0
+mistral-common==1.8.0
     # via -r requirements/test.in
 more-itertools==10.5.0
     # via lm-eval
@@ -518,6 +518,8 @@ pyasn1-modules==0.4.2
     # via google-auth
 pybind11==2.13.6
     # via lm-eval
+pycountry==24.6.1
+    # via pydantic-extra-types
 pycparser==2.22
     # via cffi
 pycryptodomex==3.22.0
@@ -528,9 +530,12 @@ pydantic==2.11.5
     #   datamodel-code-generator
     #   mistral-common
     #   mteb
+    #   pydantic-extra-types
     #   ray
 pydantic-core==2.33.2
     # via pydantic
+pydantic-extra-types==2.10.5
+    # via mistral-common
 pygments==2.18.0
     # via rich
 pyparsing==3.2.0
@@ -835,6 +840,7 @@ typing-extensions==4.12.2
     #   pqdm
     #   pydantic
     #   pydantic-core
+    #   pydantic-extra-types
     #   torch
     #   typer
     #   typing-inspection

setup.py

@@ -692,7 +692,8 @@ setup(
         "tensorizer": ["tensorizer==2.10.1"],
         "fastsafetensors": ["fastsafetensors >= 0.1.10"],
         "runai": ["runai-model-streamer", "runai-model-streamer-s3", "boto3"],
-        "audio": ["librosa", "soundfile"],  # Required for audio processing
+        "audio": ["librosa", "soundfile",
+                  "mistral_common[audio]"],  # Required for audio processing
         "video": []  # Kept for backwards compatibility
     },
     cmdclass=cmdclass,

tests/entrypoints/openai/test_transcription_validation.py

@@ -17,6 +17,11 @@ from vllm.assets.audio import AudioAsset
 
 from ...utils import RemoteOpenAIServer
 
+MISTRAL_FORMAT_ARGS = [
+    "--tokenizer_mode", "mistral", "--config_format", "mistral",
+    "--load_format", "mistral"
+]
+
 
 @pytest.fixture
 def mary_had_lamb():
@@ -33,9 +38,18 @@ def winning_call():
 
 
 @pytest.mark.asyncio
-async def test_basic_audio(mary_had_lamb):
-    model_name = "openai/whisper-large-v3-turbo"
+@pytest.mark.parametrize(
+    "model_name",
+    ["openai/whisper-large-v3-turbo", "mistralai/Voxtral-Mini-3B-2507"])
+async def test_basic_audio(mary_had_lamb, model_name):
     server_args = ["--enforce-eager"]
+
+    if model_name.startswith("mistralai"):
+        server_args += MISTRAL_FORMAT_ARGS
+
+        # TODO(PATRICK) - REMOVE AFTER RELEASE
+        return  # skip for now
+
     # Based on https://github.com/openai/openai-cookbook/blob/main/examples/Whisper_prompting_guide.ipynb.
     with RemoteOpenAIServer(model_name, server_args) as remote_server:
         client = remote_server.get_async_client()
@@ -65,10 +79,13 @@ async def test_bad_requests(mary_had_lamb):
 
 
 @pytest.mark.asyncio
-async def test_long_audio_request(mary_had_lamb):
-    model_name = "openai/whisper-large-v3-turbo"
+@pytest.mark.parametrize("model_name", ["openai/whisper-large-v3-turbo"])
+async def test_long_audio_request(mary_had_lamb, model_name):
     server_args = ["--enforce-eager"]
 
+    if model_name.startswith("openai"):
+        return
+
     mary_had_lamb.seek(0)
     audio, sr = librosa.load(mary_had_lamb)
     # Add small silence after each audio for repeatability in the split process
@@ -87,7 +104,8 @@ async def test_long_audio_request(mary_had_lamb):
             response_format="text",
             temperature=0.0)
         out = json.loads(transcription)['text']
-        assert out.count("Mary had a little lamb") == 10
+        counts = out.count("Mary had a little lamb")
+        assert counts == 10, counts
 
 
 @pytest.mark.asyncio

tests/models/registry.py

@@ -440,6 +440,7 @@ _MULTIMODAL_EXAMPLE_MODELS = {
                                                          tokenizer="Isotr0py/Florence-2-tokenizer",  # noqa: E501
                                                          trust_remote_code=True),  # noqa: E501
     "MllamaForConditionalGeneration": _HfExamplesInfo("meta-llama/Llama-3.2-11B-Vision-Instruct"),  # noqa: E501
+    "VoxtralForConditionalGeneration": _HfExamplesInfo("mistralai/Voxtral-Mini-3B-2507", is_available_online=False, tokenizer_mode="mistral"),  # noqa: E501
     "WhisperForConditionalGeneration": _HfExamplesInfo("openai/whisper-large-v3"),  # noqa: E501
 
     # [Cross-encoder]
@@ -513,4 +514,4 @@ class HfExampleModels:
         raise ValueError(f"No example model defined for {model_id}")
 
 
-HF_EXAMPLE_MODELS = HfExampleModels(_EXAMPLE_MODELS)
+HF_EXAMPLE_MODELS = HfExampleModels(_EXAMPLE_MODELS)

vllm/entrypoints/openai/speech_to_text.py

@@ -112,6 +112,7 @@ class OpenAISpeechToText(OpenAIServing):
             prompt = self.model_cls.get_generation_prompt(
                 audio=chunk,
                 stt_config=self.asr_config,
+                model_config=self.model_config,
                 language=lang,
                 task_type=self.task_type,
                 request_prompt=request.prompt)

vllm/model_executor/models/interfaces.py

@@ -722,7 +722,8 @@ class SupportsTranscription(Protocol):
 
     @classmethod
     def get_generation_prompt(cls, audio: np.ndarray,
-                              stt_config: SpeechToTextConfig, language: str,
+                              stt_config: SpeechToTextConfig,
+                              model_config: ModelConfig, language: str,
                               task_type: str,
                               request_prompt: str) -> PromptType:
         """Get the prompt for the ASR model.

vllm/model_executor/models/registry.py

@@ -231,6 +231,7 @@ _MULTIMODAL_MODELS = {
     "Phi4MMForCausalLM": ("phi4mm", "Phi4MMForCausalLM"),
     "TarsierForConditionalGeneration": ("tarsier", "TarsierForConditionalGeneration"),  # noqa: E501
     "Tarsier2ForConditionalGeneration": ("qwen2_vl", "Tarsier2ForConditionalGeneration"),  # noqa: E501
+    "VoxtralForConditionalGeneration": ("voxtral", "VoxtralForConditionalGeneration"),  # noqa: E501
     # [Encoder-decoder]
     "Florence2ForConditionalGeneration": ("florence2", "Florence2ForConditionalGeneration"),  # noqa: E501
     "MllamaForConditionalGeneration": ("mllama", "MllamaForConditionalGeneration"),  # noqa: E501

vllm/model_executor/models/voxtral.py

@@ -0,0 +1,691 @@
+# SPDX-License-Identifier: Apache-2.0
+# SPDX-FileCopyrightText: Copyright contributors to the vLLM project
+
+import math
+from collections.abc import Iterable, Mapping, Sequence
+from functools import cached_property
+from math import ceil
+from typing import Optional, Union, cast
+
+import numpy as np
+import regex as re
+import torch
+import torch.nn as nn
+from mistral_common.audio import mel_filter_bank
+from mistral_common.protocol.instruct.messages import (AudioChunk, RawAudio,
+                                                       TextChunk, UserMessage)
+from mistral_common.protocol.instruct.request import ChatCompletionRequest
+from mistral_common.protocol.transcription.request import TranscriptionRequest
+from mistral_common.tokens.tokenizers.audio import Audio, AudioEncoder
+from transformers import TensorType, WhisperConfig
+from transformers.tokenization_utils_base import TextInput
+
+from vllm.config import ModelConfig, SpeechToTextConfig, VllmConfig
+from vllm.inputs.data import PromptType
+from vllm.logger import init_logger
+from vllm.model_executor.model_loader.weight_utils import default_weight_loader
+from vllm.model_executor.models import SupportsPP
+# yapf: disable
+from vllm.model_executor.models.whisper import (
+    WhisperEncoder, WhisperForConditionalGeneration)
+# yapf: enable
+from vllm.model_executor.sampling_metadata import SamplingMetadata
+from vllm.multimodal import MULTIMODAL_REGISTRY
+from vllm.multimodal.inputs import (MultiModalDataDict, MultiModalFieldConfig,
+                                    MultiModalKwargs, NestedTensors)
+from vllm.multimodal.parse import (AudioProcessorItems, MultiModalDataItems,
+                                   MultiModalDataParser)
+from vllm.multimodal.processing import (BaseMultiModalProcessor,
+                                        BaseProcessingInfo, MultiModalHashes,
+                                        PromptReplacement, PromptUpdate)
+from vllm.multimodal.profiling import BaseDummyInputsBuilder, ProcessorInputs
+from vllm.sequence import IntermediateTensors
+from vllm.transformers_utils.tokenizer import (MistralTokenizer,
+                                               cached_tokenizer_from_config)
+
+from .interfaces import (MultiModalEmbeddings, SupportsMultiModal,
+                         SupportsTranscription)
+from .utils import (flatten_bn, init_vllm_registered_model, maybe_prefix,
+                    merge_multimodal_embeddings)
+
+logger = init_logger(__name__)
+
+
+class VoxtralProcessorAdapter:
+    """
+    Provide a HF-compatible interface for
+    :class:`mistral_common.tokens.tokenizers.multimodal.AudioEncoder`.
+    """
+
+    def __init__(self, tokenizer: MistralTokenizer) -> None:
+        super().__init__()
+        self.tokenizer = tokenizer
+
+    @cached_property
+    def _audio_processor(self) -> AudioEncoder:
+        audio_encoder = self.tokenizer.instruct.audio_encoder
+        assert isinstance(audio_encoder, AudioEncoder)
+        return audio_encoder
+
+    @cached_property
+    def audio_token_id(self) -> int:
+        return self._audio_processor.special_ids.audio
+
+    @cached_property
+    def begin_audio_token_id(self) -> int:
+        return self._audio_processor.special_ids.begin_audio
+
+    # @cached_property
+    # def begin_transcript_token_id(self) -> int:
+    #     return self._audio_processor.special_ids.begin_transcript
+
+    # @cached_property
+    # def end_transcript_token_id(self) -> int:
+    #     return self._audio_processor.special_ids.end_transcript
+
+    @cached_property
+    def sampling_rate(self) -> int:
+        return self._audio_processor.audio_config.sampling_rate
+
+    @cached_property
+    def frame_rate(self) -> float:
+        return self._audio_processor.audio_config.frame_rate
+
+    def get_num_audio_tokens(
+        self,
+        audio_length: int,
+    ) -> int:
+        pad_audio_length = self._audio_processor.next_multiple_of_chunk_frames(
+            audio_length, self.sampling_rate)
+        return ceil(pad_audio_length / (self.sampling_rate // self.frame_rate))
+
+    def __call__(
+        self,
+        text: Optional[Union[TextInput, list[TextInput]]] = None,
+        audios: Optional[Union[np.ndarray, list[np.ndarray]]] = None,
+        return_tensors: Optional[Union[str, TensorType]] = None,
+        **kwargs,
+    ) -> Mapping[str, NestedTensors]:
+        if text is None:
+            text = []
+        if not isinstance(text, list):
+            text = [text]
+        if audios is None:
+            audios = []
+        if not isinstance(audios, list):
+            audios = [audios]
+
+        if not audios:
+            input_ids = self.tokenizer(text).input_ids
+            return {"input_ids": torch.tensor(input_ids)}
+
+        # Allow dummy text, which is used for profiling as well as token inputs
+        if any(len(t) > 0 for t in text):
+            raise ValueError(
+                "You've passed text inputs instead of token inputs. "
+                "Make sure to process your input via `mistral_common`'s "
+                "tokenizer or pass a chat completion request. "
+                "For more info, see: "
+                "https://github.com/vllm-project/vllm/issues/8411.")
+
+        audios_tokens = list[torch.Tensor]()
+        audios_processed = list[torch.Tensor]()
+        for audio in audios:
+            assert isinstance(audio, np.ndarray)
+            assert audio.ndim == 1
+
+            # pad if necessary
+            audio = self._audio_processor.pad(audio, self.sampling_rate)
+
+            audio_tokens = [
+                self.begin_audio_token_id
+            ] + [self.audio_token_id] * self.get_num_audio_tokens(len(audio))
+
+            audios_tokens.append(torch.tensor(audio_tokens))
+            audios_processed.append(torch.tensor(audio))
+
+        return {
+            "input_ids": torch.cat(audios_tokens)[None].expand(len(text), -1),
+            "audio_arrays": audios_processed,
+        }
+
+
+class VoxtralProcessingInfo(BaseProcessingInfo):
+
+    def get_tokenizer(self) -> MistralTokenizer:
+        tokenizer = cached_tokenizer_from_config(self.ctx.model_config)
+        if not isinstance(tokenizer, MistralTokenizer):
+            raise ValueError("This model requires `--tokenizer-mode mistral`")
+
+        return tokenizer
+
+    def get_hf_processor(self) -> VoxtralProcessorAdapter:
+        return VoxtralProcessorAdapter(self.get_tokenizer())
+
+    def get_supported_mm_limits(self) -> Mapping[str, Optional[int]]:
+        return {"audio": 5}  # Performance tends to degrade after 5
+
+    def get_mm_max_tokens_per_item(
+        self,
+        seq_len: int,
+        mm_counts: Mapping[str, int],
+    ) -> Mapping[str, int]:
+        return {"audio": self.get_max_audio_tokens()}
+
+    def get_max_audio_tokens(self) -> int:
+        return self.ctx.model_config.max_model_len
+
+    def get_max_audio_array_len(self) -> int:
+        processor = self.get_hf_processor()
+        return self.get_max_audio_tokens() * int(
+            processor.sampling_rate // processor.frame_rate)
+
+
+class VoxtralDummyInputsBuilder(BaseDummyInputsBuilder[VoxtralProcessingInfo]):
+
+    def get_dummy_text(self, mm_counts: Mapping[str, int]) -> str:
+        return ""
+
+    def get_dummy_mm_data(
+        self,
+        seq_len: int,
+        mm_counts: Mapping[str, int],
+    ) -> MultiModalDataDict:
+        num_audios = mm_counts.get("audio", 0)
+
+        target_length = self.info.get_max_audio_array_len()
+
+        return {
+            "audio":
+            self._get_dummy_audios(length=target_length, num_audios=num_audios)
+        }
+
+    def get_dummy_processor_inputs(
+        self,
+        seq_len: int,
+        mm_counts: Mapping[str, int],
+    ) -> 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)
+        dummy_audios = dummy_mm_data.get("audio", [])
+
+        audio_chunks: list[AudioChunk] = []
+        format = "wav"
+        for audio in dummy_audios:
+            audio_item = Audio(
+                audio_array=audio,
+                sampling_rate=self.info.get_hf_processor().sampling_rate,
+                format=format,
+            )
+            chunk = AudioChunk(input_audio=RawAudio.from_audio(audio_item))
+            audio_chunks.append(chunk)
+
+        request = ChatCompletionRequest(messages=[
+            UserMessage(content=[TextChunk(text=dummy_text), *audio_chunks]),
+        ])
+        res = tokenizer.mistral.encode_chat_completion(request)
+        dummy_tokens = res.tokens
+        # whixtral tokenizer adds padding to the audio
+        # so we need to update the audio arrays
+        dummy_mm_data["audio"] = [a.audio_array for a in res.audios]
+
+        return ProcessorInputs(prompt=dummy_tokens, mm_data=dummy_mm_data)
+
+
+class VoxtralMultiModalProcessor(BaseMultiModalProcessor[VoxtralProcessingInfo]
+                                 ):
+
+    def _get_mm_fields_config(
+        self,
+        hf_inputs: Mapping[str, NestedTensors],
+        hf_processor_mm_kwargs: Mapping[str, object],
+    ) -> Mapping[str, MultiModalFieldConfig]:
+        return dict(audio_arrays=MultiModalFieldConfig.batched("audio"))
+
+    def _get_prompt_updates(
+        self,
+        mm_items: MultiModalDataItems,
+        hf_processor_mm_kwargs: Mapping[str, object],
+        out_mm_kwargs: MultiModalKwargs,
+    ) -> Sequence[PromptUpdate]:
+        processor = self.info.get_hf_processor(**hf_processor_mm_kwargs)
+
+        audio_id = processor.audio_token_id
+
+        def get_replacement(item_idx: int):
+            audios = mm_items.get_items("audio", AudioProcessorItems)
+            audio_len = audios.get_audio_length(item_idx)
+
+            nb_audio_tokens = processor.get_num_audio_tokens(audio_len)
+
+            return [audio_id] * nb_audio_tokens
+
+        return [
+            PromptReplacement(
+                modality="audio",
+                target="",  # Never match the prompt (see below note)
+                replacement=get_replacement,
+            ),
+        ]
+
+    def _cached_apply_hf_processor(
+        self,
+        prompt: Union[str, list[int]],
+        mm_data_items: MultiModalDataItems,
+        hf_processor_mm_kwargs: Mapping[str, object],
+        tokenization_kwargs: Mapping[str, object],
+        *,
+        return_mm_hashes: bool,
+    ) -> tuple[list[int], MultiModalKwargs, Optional[MultiModalHashes], bool]:
+        prompt_ids, mm_kwargs, mm_hashes, _ = super(
+        )._cached_apply_hf_processor(
+            prompt=prompt,
+            mm_data_items=mm_data_items,
+            hf_processor_mm_kwargs=hf_processor_mm_kwargs,
+            tokenization_kwargs=tokenization_kwargs,
+            return_mm_hashes=return_mm_hashes,
+        )
+
+        # NOTE: The tokens are already inserted by the chat template
+        return prompt_ids, mm_kwargs, mm_hashes, True
+
+    def _get_data_parser(self) -> MultiModalDataParser:
+        sampling_rate = self.info.get_hf_processor().sampling_rate
+        return MultiModalDataParser(target_sr=sampling_rate)
+
+
+@MULTIMODAL_REGISTRY.register_processor(VoxtralMultiModalProcessor,
+                                        info=VoxtralProcessingInfo,
+                                        dummy_inputs=VoxtralDummyInputsBuilder)
+class VoxtralForConditionalGeneration(nn.Module, SupportsMultiModal,
+                                      SupportsPP, SupportsTranscription):
+
+    def __init__(self, *, vllm_config: VllmConfig, prefix: str = ""):
+        super().__init__()
+        self.tokenizer = cached_tokenizer_from_config(vllm_config.model_config)
+
+        config = vllm_config.model_config.hf_config
+        self.config = config
+        self.downsample_factor = self.config.audio_config.downsample_factor
+
+        self.language_model = init_vllm_registered_model(
+            vllm_config=vllm_config,
+            hf_config=config.text_config,
+            prefix=maybe_prefix(prefix, "language_model"),
+        )
+        self.whisper_encoder = VoxtralEncoderModel(
+            vllm_config.with_hf_config(config.audio_config),
+            prefix=maybe_prefix(prefix, "whisper_encoder"),
+        )
+        self.audio_language_adapter = AudioLanguageAdapter(
+            hidden_size=config.audio_config.d_model * self.downsample_factor,
+            dim=config.text_config.hidden_size,
+        )
+
+    def get_language_model(self) -> torch.nn.Module:
+        return self.language_model
+
+    def forward(
+        self,
+        input_ids: torch.Tensor,
+        positions: torch.Tensor,
+        intermediate_tensors: Optional[IntermediateTensors] = None,
+        inputs_embeds: Optional[torch.Tensor] = None,
+        **kwargs: object,
+    ) -> Union[torch.Tensor, IntermediateTensors]:
+        if intermediate_tensors is not None:
+            inputs_embeds = None
+
+        # NOTE: In v1, inputs_embeds is always generated at model runner, this
+        # condition is for v0 compatibility.
+        elif inputs_embeds is None:
+            audio_embeddings = self.get_multimodal_embeddings(**kwargs)
+            inputs_embeds = self.get_input_embeddings(input_ids,
+                                                      audio_embeddings)
+            input_ids = None
+
+        hidden_states = self.language_model.model(input_ids,
+                                                  positions,
+                                                  intermediate_tensors,
+                                                  inputs_embeds=inputs_embeds)
+
+        return hidden_states
+
+    def get_multimodal_embeddings(
+        self, **kwargs
+    ) -> Union[list[torch.Tensor], torch.Tensor, tuple[torch.Tensor, ...],
+               None]:
+        audio_inputs = self._parse_and_validate_audio_arrays(**kwargs)
+        if audio_inputs is None:
+            return None
+
+        audio_embeddings = self.whisper_encoder(audio_inputs)
+
+        for i, audio_embedding in enumerate(audio_embeddings):
+            seq_len, dim = audio_embedding.shape
+            # Pad such that seq_len is divisible by downsample_factor
+            target_seq_len = self.downsample_factor * math.ceil(
+                seq_len / self.downsample_factor)
+            audio_embedding = torch.nn.functional.pad(
+                audio_embedding,
+                (0, 0, 0, target_seq_len - seq_len),
+            )
+            audio_embeddings[i] = audio_embedding.reshape(
+                target_seq_len // self.downsample_factor,
+                dim * self.downsample_factor)
+
+        # Concat, project and resplit
+        audio_embeddings_packed = torch.cat(audio_embeddings, dim=0)
+        audio_embeddings_packed = self.audio_language_adapter(
+            audio_embeddings_packed)
+        audio_embeddings = torch.split(audio_embeddings_packed,
+                                       [a.shape[0] for a in audio_embeddings],
+                                       dim=0)
+
+        return audio_embeddings
+
+    def get_input_embeddings(
+        self,
+        input_ids: torch.Tensor,
+        multimodal_embeddings: Optional[MultiModalEmbeddings] = None,
+    ) -> torch.Tensor:
+        audio_encoder = self.tokenizer.instruct.audio_encoder
+        audio_tok_id = audio_encoder.audio_token
+
+        inputs_embeds = self.language_model.get_input_embeddings(input_ids)
+        if multimodal_embeddings is not None:
+            inputs_embeds = merge_multimodal_embeddings(
+                input_ids, inputs_embeds, multimodal_embeddings, audio_tok_id)
+        return inputs_embeds
+
+    def _parse_and_validate_audio_arrays(
+            self, **kwargs: object) -> Union[list[torch.Tensor], None]:
+        audio_arrays = kwargs.pop("audio_arrays", None)
+        if audio_arrays is None:
+            return None
+
+        if not isinstance(audio_arrays, (torch.Tensor, list)):
+            raise ValueError("Incorrect type of audio_arrays. "
+                             f"Got type: {type(audio_arrays)}")
+
+        audio_arrays = flatten_bn(audio_arrays)
+        if isinstance(audio_arrays, torch.Tensor):
+            audio_arrays = list(audio_arrays.unbind(0))
+        return audio_arrays
+
+    def compute_logits(
+        self,
+        hidden_states: torch.Tensor,
+        sampling_metadata: SamplingMetadata,
+    ) -> Optional[torch.Tensor]:
+        return self.language_model.compute_logits(hidden_states,
+                                                  sampling_metadata)
+
+    @classmethod
+    def get_speech_to_text_config(cls, model_config: ModelConfig,
+                                  task_type: str) -> SpeechToTextConfig:
+        tokenizer = cached_tokenizer_from_config(model_config)
+        audio_config = tokenizer.instruct.audio_encoder.audio_config
+        max_audio_clip_s = audio_config.chunk_length_s
+        sample_rate = audio_config.sampling_rate
+        return SpeechToTextConfig(
+            max_audio_clip_s=max_audio_clip_s,
+            sample_rate=sample_rate,
+            # mistral_common and whisper encoder take care of chunking
+            min_energy_split_window_size=None,
+        )
+
+    @classmethod
+    # for speech-to-text transcription
+    def get_generation_prompt(cls, audio: np.ndarray,
+                              model_config: ModelConfig,
+                              stt_config: SpeechToTextConfig, language: str,
+                              task_type: str,
+                              request_prompt: str) -> PromptType:
+        tokenizer = cached_tokenizer_from_config(model_config)
+        audio = Audio(audio, int(stt_config.sample_rate),
+                      format="wav")  # lossless
+        req = TranscriptionRequest(model=model_config.model,
+                                   audio=RawAudio.from_audio(audio),
+                                   language=language)
+
+        tokenized = tokenizer.instruct.encode_transcription(req)
+        audio = (tokenized.audios[0].audio_array, stt_config.sample_rate)
+        prompts_dict = {"multi_modal_data": {"audio": audio}}
+        prompts_dict["prompt_token_ids"] = tokenized.tokens
+        return cast(PromptType, prompts_dict)
+
+    @classmethod
+    def validate_language(cls, language: str) -> bool:
+        # same as whisper
+        return WhisperForConditionalGeneration.validate_language(language)
+
+    @classmethod
+    def get_num_audio_tokens(cls, audio_duration_s: float,
+                             stt_config: SpeechToTextConfig,
+                             model_config: ModelConfig) -> Optional[int]:
+        """
+        Map from audio duration to number of audio tokens produced by the ASR 
+        model, without running a forward pass.
+        This is used for estimating the amount of processing for this audio.
+        """
+        tokenizer = cached_tokenizer_from_config(model_config)
+        adapter = VoxtralProcessorAdapter(tokenizer)
+        return adapter.get_num_audio_tokens(
+            int(audio_duration_s * stt_config.sample_rate))
+
+    def load_weights(self, weights: Iterable[tuple[str,
+                                                   torch.Tensor]]) -> set[str]:
+        # fmt: off
+        remapping_rules = [
+            (r"mm_whisper_embeddings\.(.*)", r"\1"),
+            (r"audio_language_projection\.(.*)", r"audio_language_adapter.\1"),
+            (r"audio_language_adapter\.0\.weight", r"audio_language_adapter.w_in.weight"),  # noqa: E501
+            (r"audio_language_adapter\.2\.weight", r"audio_language_adapter.w_out.weight"),  # noqa: E501
+        ]
+        # fmt: on
+
+        audio_params = dict(
+            nn.ModuleDict({
+                "audio_language_adapter":
+                self.audio_language_adapter,
+            }).named_parameters())
+
+        loaded_weights = set()
+
+        def llm_weights_generator():
+            nonlocal loaded_weights
+            for name, w in weights:
+                is_encoder = (
+                    name.startswith("mm_whisper_embeddings") and
+                    not name.startswith("mm_whisper_embeddings.tok_embeddings")
+                    and not name.startswith(
+                        "mm_whisper_embeddings.audio_language_projection"))
+
+                for pattern, repl in remapping_rules:
+                    if re.fullmatch(pattern, name):
+                        name = re.sub(pattern, repl, name)
+
+                if is_encoder:
+                    name = self.whisper_encoder.load_weight((name, w))
+                    loaded_weights.add(f"whisper_encoder.{name}")
+                    continue
+
+                if name in audio_params:
+                    param = audio_params[name]
+                    with torch.no_grad():
+                        default_weight_loader(param, w)
+                    loaded_weights.add(name)
+                else:
+                    yield (name, w)
+
+        for name in self.language_model.load_weights(llm_weights_generator()):
+            loaded_weights.add(f"language_model.{name}")
+
+        # potentially manually add position embeddings
+        sin_key = "whisper_encoder.whisper_encoder.embed_positions.weight"
+        if sin_key not in loaded_weights:
+            # make sure we don't hit an error here
+            loaded_weights.add(sin_key)
+
+        return loaded_weights
+
+
+class AudioLanguageAdapter(nn.Module):
+
+    def __init__(self, hidden_size: int, dim: int) -> None:
+        super().__init__()
+        self.w_in = nn.Linear(hidden_size, dim, bias=False)
+        self.gelu = nn.GELU()
+        self.w_out = nn.Linear(dim, dim, bias=False)
+
+    def forward(self, x: torch.Tensor) -> torch.Tensor:
+        return self.w_out(self.gelu(self.w_in(x)))
+
+
+class VoxtralEncoderModel(nn.Module):
+    packed_modules_mapping = {"qkv_proj": ["q_proj", "k_proj", "v_proj"]}
+
+    # fmt: off
+    mistral_remapping = [
+        (r"whisper_encoder\.conv_layers\.0\.(weight|bias)", r"whisper_encoder.conv1.\1"), # noqa: E501
+        (r"whisper_encoder\.conv_layers\.1\.(weight|bias)", r"whisper_encoder.conv2.\1"), # noqa: E501
+        (r"whisper_encoder\.transformer\.layers\.(\d+)\.attention\.w([qkv])\.(weight|bias)", r"whisper_encoder.layers.\1.self_attn.\2_proj.\3"), # noqa: E501
+        (r"whisper_encoder\.transformer\.layers\.(\d+)\.attention\.wo\.(weight|bias)", r"whisper_encoder.layers.\1.self_attn.out_proj.\2"), # noqa: E501
+        (r"whisper_encoder\.transformer\.layers\.(\d+)\.attention_norm\.(weight|bias)", r"whisper_encoder.layers.\1.self_attn_layer_norm.\2"), # noqa: E501
+        (r"whisper_encoder\.transformer\.layers\.(\d+)\.feed_forward\.w1\.(weight|bias)", r"whisper_encoder.layers.\1.mlp.fc1.\2"), # noqa: E501
+        (r"whisper_encoder\.transformer\.layers\.(\d+)\.feed_forward\.w2\.(weight|bias)", r"whisper_encoder.layers.\1.mlp.fc2.\2"), # noqa: E501
+        (r"whisper_encoder\.transformer\.layers\.(\d+)\.ffn_norm\.(weight|bias)", r"whisper_encoder.layers.\1.final_layer_norm.\2"), # noqa: E501
+        (r"whisper_encoder\.transformer\.norm\.(weight|bias)", r"whisper_encoder.layer_norm.\1"), # noqa: E501
+    ]
+    # fmt: on
+
+    def __init__(
+        self,
+        vllm_config: VllmConfig,
+        *,
+        prefix: str = "",
+    ) -> None:
+        super().__init__()
+        self.config = cast(WhisperConfig, vllm_config.model_config.hf_config)
+        self.dtype: torch.dtype = vllm_config.model_config.dtype
+        self.whisper_encoder = WhisperEncoder(vllm_config=vllm_config,
+                                              prefix=maybe_prefix(
+                                                  prefix, "whisper_encoder"),
+                                              is_standalone_encoder=True,
+                                              init_in_fp32=True)
+        mel_filters = mel_filter_bank(
+            num_frequency_bins=1 + self.config.window_size // 2,
+            num_mel_bins=self.config.num_mel_bins,
+            min_frequency=0.0,
+            max_frequency=8000.0,
+            sampling_rate=self.config.sampling_rate,
+        )
+        self.mel_filters = torch.tensor(mel_filters, dtype=torch.float32)
+
+    def compute_whisper_melspec(
+        self,
+        audio_waveforms: torch.Tensor,
+    ) -> torch.Tensor:
+        input_dtype = audio_waveforms.dtype
+        window = torch.hann_window(self.config.window_size).to(
+            audio_waveforms.device)
+        stft = torch.stft(
+            audio_waveforms,
+            self.config.window_size,
+            self.config.hop_length,
+            window=window,
+            return_complex=True,
+        )
+        magnitudes = stft[..., :-1].abs()**2
+        mel_spec = self.mel_filters.T @ magnitudes
+        log_spec = torch.clamp(mel_spec, min=1e-10).log10()
+        log_spec = torch.maximum(log_spec, log_spec.max() - 8.0)
+        log_spec = (log_spec + 4.0) / 4.0
+        return log_spec.to(input_dtype)
+
+    @property
+    def downsample_factor(self) -> int:
+        return self.whisper_encoder.conv1.stride[
+            0] * self.whisper_encoder.conv2.stride[0]
+
+    @property
+    def chunk_size(self) -> int:
+        return self.config.max_source_positions * self.downsample_factor
+
+    def prepare_inputs_for_conv(
+        self,
+        audio_waveforms: list[torch.Tensor],
+    ) -> tuple[torch.Tensor, list[int]]:
+        assert isinstance(audio_waveforms, list)
+        # list[num_mel_bins, seq_len]
+        input_features = [
+            self.compute_whisper_melspec(audio).to(self.dtype)
+            for audio in audio_waveforms
+        ]
+
+        chunked_features: list[torch.Tensor] = []
+        chunks_per_example: list[int] = []
+        for feature in input_features:
+            chunks = feature.split(self.chunk_size, dim=-1)
+            chunked_features += chunks
+            chunks_per_example.append(len(chunks))
+
+        # [total_num_chunks, num_mel_bins, chunk_size]
+        return torch.stack(chunked_features), chunks_per_example
+
+    def forward(
+        self, input_features: Union[torch.Tensor, list[torch.Tensor]]
+    ) -> list[torch.Tensor]:
+        if not isinstance(input_features, list):
+            input_features = [input_features]
+
+        # Split long inputs into chunks
+        input_embeds, chunks_per_example = (
+            self.prepare_inputs_for_conv(input_features))
+
+        # [total_num_chunks, ceil(chunk_size / downsample_factor), hidden_size]
+        out = self.whisper_encoder([input_embeds])
+
+        # Re-concatenate the chunks
+        chunk_idx = 0
+        results = []
+        for n_chunks in chunks_per_example:
+            result = out[chunk_idx:chunk_idx + n_chunks].flatten(0, 1)
+            results.append(result)
+            chunk_idx += n_chunks
+
+        return results
+
+    def load_weight(self, weight: tuple[str, torch.Tensor]) -> str:
+        stacked_params_mapping = [
+            # (param_name, shard_name, shard_id)
+            ("qkv_proj", "q_proj", "q"),
+            ("qkv_proj", "k_proj", "k"),
+            ("qkv_proj", "v_proj", "v"),
+        ]
+        params_dict = dict(self.named_parameters())
+
+        name, loaded_weight = weight
+        for pattern, repl in self.mistral_remapping:
+            if re.fullmatch(pattern, name):
+                name = re.sub(pattern, repl, name)
+
+        for (param_name, weight_name, shard_id) in stacked_params_mapping:
+            if weight_name not in name:
+                continue
+            name = name.replace(weight_name, param_name)
+
+            param = params_dict[name]
+            weight_loader = param.weight_loader
+            weight_loader(param, loaded_weight, shard_id)
+            break
+        else:
+            param = params_dict[name]
+            weight_loader = getattr(param, "weight_loader",
+                                    default_weight_loader)
+            weight_loader(param, loaded_weight)
+
+        return name

vllm/model_executor/models/whisper.py

@@ -3,6 +3,7 @@
 
 import math
 from collections.abc import Iterable, Mapping, Sequence
+from contextlib import nullcontext
 from typing import Optional, TypedDict, Union, cast
 
 import numpy as np
@@ -13,6 +14,7 @@ from transformers import (BatchFeature, WhisperConfig, WhisperFeatureExtractor,
 from transformers.models.whisper.modeling_whisper import sinusoids
 
 from vllm.attention import Attention, AttentionType
+from vllm.attention.layer import MultiHeadAttention
 from vllm.config import (CacheConfig, ModelConfig, SpeechToTextConfig,
                          VllmConfig)
 from vllm.distributed import get_tensor_model_parallel_world_size
@@ -26,6 +28,7 @@ from vllm.model_executor.layers.logits_processor import LogitsProcessor
 from vllm.model_executor.layers.quantization.base_config import (
     QuantizationConfig)
 from vllm.model_executor.layers.vocab_parallel_embedding import ParallelLMHead
+from vllm.model_executor.model_loader.utils import set_default_torch_dtype
 from vllm.model_executor.model_loader.weight_utils import default_weight_loader
 from vllm.model_executor.sampling_metadata import SamplingMetadata
 from vllm.multimodal import MULTIMODAL_REGISTRY, NestedTensors
@@ -178,6 +181,7 @@ class WhisperAttention(nn.Module):
         cache_config: Optional[CacheConfig] = None,
         quant_config: Optional[QuantizationConfig] = None,
         prefix: str = "",
+        standalone_encoder: bool = False,
     ):
         super().__init__()
         self.embed_dim = embed_dim
@@ -213,16 +217,24 @@ class WhisperAttention(nn.Module):
             quant_config=quant_config,
             prefix=f"{prefix}.out_proj",
         )
-        self.attn = Attention(
-            self.num_heads,
-            self.head_dim,
-            self.scaling,
-            num_kv_heads=self.num_kv_heads,
-            cache_config=cache_config,
-            quant_config=quant_config,
-            prefix=f"{prefix}.attn",
-            attn_type=self.attn_type,
-        )
+        if standalone_encoder:
+            self.attn = MultiHeadAttention(
+                self.num_heads,
+                self.head_dim,
+                self.scaling,
+                num_kv_heads=self.num_kv_heads,
+            )
+        else:
+            self.attn = Attention(
+                self.num_heads,
+                self.head_dim,
+                self.scaling,
+                num_kv_heads=self.num_kv_heads,
+                cache_config=cache_config,
+                quant_config=quant_config,
+                prefix=f"{prefix}.attn",
+                attn_type=self.attn_type,
+            )
 
     def _init_qkv(
         self,
@@ -357,7 +369,11 @@ class WhisperMLP(nn.Module):
 
 class WhisperEncoderLayer(nn.Module):
 
-    def __init__(self, *, vllm_config: VllmConfig, prefix: str = ""):
+    def __init__(self,
+                 *,
+                 vllm_config: VllmConfig,
+                 prefix: str = "",
+                 is_standalone_encoder: bool = False):
         super().__init__()
         config = vllm_config.model_config.hf_config
         cache_config = vllm_config.cache_config
@@ -371,6 +387,7 @@ class WhisperEncoderLayer(nn.Module):
             cache_config=cache_config,
             quant_config=quant_config,
             prefix=f"{prefix}.self_attn",
+            standalone_encoder=is_standalone_encoder,
         )
         self.self_attn_layer_norm = nn.LayerNorm(self.embed_dim)
         self.mlp = WhisperMLP(
@@ -462,10 +479,16 @@ class WhisperDecoderLayer(nn.Module):
 
 class WhisperEncoder(nn.Module):
 
-    def __init__(self, *, vllm_config: VllmConfig, prefix: str = ""):
+    def __init__(self,
+                 *,
+                 vllm_config: VllmConfig,
+                 prefix: str = "",
+                 is_standalone_encoder: bool = False,
+                 init_in_fp32: bool = False):
         super().__init__()
         config = vllm_config.model_config.hf_config
         embed_dim = config.d_model
+        self.is_standalone_encoder = is_standalone_encoder
         self.num_mel_bins = config.num_mel_bins
         self.max_source_positions = config.max_source_positions
         self.embed_scale = (math.sqrt(embed_dim)
@@ -480,17 +503,25 @@ class WhisperEncoder(nn.Module):
                                kernel_size=3,
                                stride=2,
                                padding=1)
-        self.embed_positions = nn.Embedding(self.max_source_positions,
-                                            embed_dim)
         self.start_layer, self.end_layer, self.layers = make_layers(
             config.encoder_layers,
             lambda prefix: WhisperEncoderLayer(vllm_config=vllm_config,
-                                               prefix=f"{prefix}.layers"),
+                                               prefix=f"{prefix}.layers",
+                                               is_standalone_encoder=
+                                               is_standalone_encoder),
             prefix=f"{prefix}.layers",
         )
         self.layer_norm = nn.LayerNorm(config.d_model)
 
-        with torch.no_grad():
+        maybe_fp32_init_ctx = set_default_torch_dtype(
+            torch.float32) if init_in_fp32 else nullcontext()
+
+        with (
+                torch.no_grad(),
+                maybe_fp32_init_ctx,
+        ):
+            self.embed_positions = nn.Embedding(self.max_source_positions,
+                                                embed_dim)
             self.embed_positions.weight.copy_(
                 sinusoids(*self.embed_positions.weight.shape))
 
@@ -499,8 +530,10 @@ class WhisperEncoder(nn.Module):
         for features in input_features:
             embeds = nn.functional.gelu(self.conv1(features))
             embeds = nn.functional.gelu(self.conv2(embeds))
-            embeds = embeds.permute(1, 0)
-            embeds = embeds + self.embed_positions.weight[:embeds.size(0), :]
+            embeds = embeds.transpose(-1, -2)
+            embeds = (embeds +
+                      self.embed_positions.weight[:embeds.size(-2), :]).to(
+                          embeds.dtype)
             hidden_states.append(embeds)
         hidden_states = torch.cat(hidden_states)
 
@@ -792,10 +825,14 @@ class WhisperForConditionalGeneration(nn.Module, SupportsTranscription,
                              f"or {list(ISO639_1_OTHER_LANGS.values())}")
 
     @classmethod
-    def get_generation_prompt(cls, audio: np.ndarray,
-                              stt_config: SpeechToTextConfig, language: str,
-                              task_type: str,
-                              request_prompt: str) -> PromptType:
+    def get_generation_prompt(
+            cls,
+            audio: np.ndarray,
+            model_config: ModelConfig,  # not needed here
+            stt_config: SpeechToTextConfig,
+            language: str,
+            task_type: str,
+            request_prompt: str) -> PromptType:
         prompt = {
             "encoder_prompt": {
                 # Whisper does not support encoder prompt.

vllm/transformers_utils/configs/mistral.py

@@ -2,7 +2,7 @@
 # SPDX-FileCopyrightText: Copyright contributors to the vLLM project
 from typing import Any
 
-from transformers import PretrainedConfig
+from transformers import PretrainedConfig, WhisperConfig
 
 from vllm.logger import init_logger
 
@@ -24,9 +24,21 @@ def adapt_config_dict(config_dict: dict[str, Any],
 
     if bool(config_dict.get("yarn")):
         config_dict = _remap_mistral_yarn_args(config_dict)
-    if bool((config_dict.get("multimodal") or {}).get("vision_encoder_args")
-            or config_dict.get("vision_encoder")):
+
+    is_vision = ((config_dict.get("multimodal")
+                  or {}).get("vision_encoder_args")
+                 or config_dict.get("vision_encoder"))
+    is_audio = bool(
+        ((config_dict.get("multimodal") or {}).get("whisper_model_args")
+         or {}).get("encoder_args"))
+
+    assert not (is_vision and is_audio), \
+        "Vision and audio are mutually exclusive"
+
+    if is_vision:
         config_dict = _remap_mistral_vision_args(config_dict)
+    if is_audio:
+        config_dict = _remap_mistral_audio_args(config_dict)
 
     config = PretrainedConfig.from_dict(config_dict)
 
@@ -118,3 +130,35 @@ def _remap_mistral_quantization_args(config: dict) -> dict:
     config["quantization_config"] = quantization_config
 
     return config
+
+
+def _remap_mistral_audio_args(config: dict) -> dict:
+    whisper_args = config["multimodal"].pop("whisper_model_args")
+    encoder_args = whisper_args["encoder_args"]
+    downsample_args = whisper_args["downsample_args"]
+
+    quant_config = config.get("quantization_config")
+    config = {
+        "model_type":
+        "whixtral",
+        "architectures": ["VoxtralForConditionalGeneration"],
+        "text_config":
+        PretrainedConfig.from_dict(config),
+        "audio_config":
+        WhisperConfig(
+            num_mel_bins=encoder_args["audio_encoding_args"]["num_mel_bins"],
+            window_size=encoder_args["audio_encoding_args"]["window_size"],
+            sampling_rate=encoder_args["audio_encoding_args"]["sampling_rate"],
+            hop_length=encoder_args["audio_encoding_args"]["hop_length"],
+            downsample_factor=downsample_args["downsample_factor"],
+            d_model=encoder_args["dim"],
+            encoder_layers=encoder_args["n_layers"],
+            encoder_ffn_dim=encoder_args["hidden_dim"],
+            encoder_attention_heads=encoder_args["n_heads"],
+            vocab_size=encoder_args["vocab_size"],
+            max_source_positions=encoder_args["max_source_positions"],
+        )
+    }
+    if quant_config:
+        config["quantization_config"] = quant_config
+    return config