# 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 partial
from typing import 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.chunk import AudioChunk, TextChunk
from mistral_common.protocol.instruct.messages import 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
from transformers import BatchFeature, WhisperConfig

from vllm.config import ModelConfig, SpeechToTextConfig, VllmConfig
from vllm.config.multimodal import BaseDummyOptions
from vllm.config.speech_to_text import SpeechToTextParams
from vllm.inputs import MultiModalDataDict, PromptType, TokensPrompt
from vllm.logger import init_logger
from vllm.model_executor.layers.quantization import QuantizationConfig
from vllm.model_executor.model_loader.weight_utils import default_weight_loader
from vllm.model_executor.models import SupportsPP
from vllm.model_executor.models.module_mapping import MultiModelKeys
from vllm.model_executor.models.whisper import (
    WhisperEncoder,
    _create_fake_bias_for_k_proj,
)
from vllm.model_executor.models.whisper_causal import WhisperCausalEncoder
from vllm.multimodal import MULTIMODAL_REGISTRY
from vllm.multimodal.inputs import (
    MultiModalFieldConfig,
    MultiModalKwargsItems,
    NestedTensors,
)
from vllm.multimodal.parse import (
    AudioProcessorItems,
    MultiModalDataItems,
    MultiModalDataParser,
)
from vllm.multimodal.processing import BaseDummyInputsBuilder
from vllm.multimodal.processing.processor import (
    BaseMultiModalProcessor,
    BaseProcessingInfo,
    MultiModalProcessingInfo,
    PlaceholderFeaturesInfo,
    ProcessorInputs,
    PromptReplacement,
    PromptUpdate,
    TimingContext,
)
from vllm.sequence import IntermediateTensors
from vllm.tokenizers import cached_tokenizer_from_config
from vllm.tokenizers.mistral import MistralTokenizer
from vllm.transformers_utils.processors.voxtral import (
    MistralCommonFeatureExtractor,
    MistralCommonVoxtralProcessor,
)
from vllm.utils.collection_utils import is_list_of

from .interfaces import SupportsLoRA, SupportsMultiModal, SupportsTranscription
from .utils import init_vllm_registered_model, maybe_prefix

logger = init_logger(__name__)

ISO639_1_SUPPORTED_LANGS = {
    "ar": "Arabic",
    "nl": "Dutch",
    "en": "English",
    "fr": "French",
    "de": "German",
    "hi": "Hindi",
    "it": "Italian",
    "pt": "Portuguese",
    "es": "Spanish",
}


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_feature_extractor(self) -> MistralCommonFeatureExtractor:
        return MistralCommonFeatureExtractor(
            self.get_tokenizer().instruct.audio_encoder
        )

    def get_hf_processor(self, **kwargs) -> MistralCommonVoxtralProcessor:
        return MistralCommonVoxtralProcessor(
            tokenizer=self.get_tokenizer(),
            feature_extractor=self.get_feature_extractor(),
        )

    def get_data_parser(self):
        feature_extractor = self.get_feature_extractor()

        return MultiModalDataParser(
            target_sr=feature_extractor.sampling_rate,
            target_channels=1,
            expected_hidden_size=self._get_expected_hidden_size(),
        )

    def get_supported_mm_limits(self) -> Mapping[str, int | None]:
        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:
        feature_extractor = self.get_feature_extractor()

        return self.get_max_audio_tokens() * int(
            feature_extractor.sampling_rate // feature_extractor.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],
        mm_options: Mapping[str, BaseDummyOptions],
    ) -> MultiModalDataDict:
        num_audios = mm_counts.get("audio", 0)

        target_length = self.info.get_max_audio_array_len()

        audio_overrides = mm_options.get("audio")

        return {
            "audio": self._get_dummy_audios(
                length=target_length,
                num_audios=num_audios,
                overrides=audio_overrides,
            )
        }

    def get_dummy_processor_inputs(
        self,
        seq_len: int,
        mm_counts: Mapping[str, int],
        mm_options: Mapping[str, BaseDummyOptions],
        mm_data: MultiModalDataDict | None = None,
    ) -> ProcessorInputs:
        tokenizer = self.info.get_tokenizer()
        feature_extractor = self.info.get_feature_extractor()

        dummy_text = self.get_dummy_text(mm_counts)
        dummy_mm_data = (
            self.get_dummy_mm_data(seq_len, mm_counts, mm_options)
            if mm_data is None
            else mm_data
        )
        dummy_mm_items = self.info.parse_mm_data(dummy_mm_data)
        dummy_audios = (
            [] if "audio" not in dummy_mm_data else dummy_mm_items["audio"].get_all()
        )

        audio_chunks: list[AudioChunk] = []
        format = "wav"
        for audio in dummy_audios:
            audio_item = Audio(
                audio_array=audio,
                sampling_rate=feature_extractor.sampling_rate,
                format=format,
            )
            chunk = AudioChunk.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

        dummy_mm_items = self.info.parse_mm_data(
            # 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_items=dummy_mm_items)


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 _validate_mm_placeholders(
        self,
        mm_placeholders: Mapping[str, list[PlaceholderFeaturesInfo]],
        mm_item_counts: Mapping[str, int],
    ) -> None:
        # mistral_common's tokenizer's does not follow HF's placeholder norms
        # skip validation here
        pass

    def _call_hf_processor(
        self,
        prompt: str,
        mm_data: Mapping[str, object],
        mm_kwargs: Mapping[str, object],
        tok_kwargs: Mapping[str, object],
    ) -> BatchFeature:
        mm_data = dict(mm_data)
        audios = mm_data.pop("audios", [])

        if audios:
            # MistralCommonVoxtralProcessor accepts "audio"
            mm_data["audio"] = audios

        outputs = super()._call_hf_processor(
            prompt=prompt,
            mm_data=mm_data,
            mm_kwargs=mm_kwargs,
            # Avoid padding issue
            tok_kwargs={**tok_kwargs, "return_tensors": None},
        )

        # Missing batch dimension
        if is_list_of(outputs["input_ids"], int):
            outputs["input_ids"] = [outputs["input_ids"]]

        return outputs

    def _get_prompt_updates(
        self,
        mm_items: MultiModalDataItems,
        hf_processor_mm_kwargs: Mapping[str, object],
        out_mm_kwargs: MultiModalKwargsItems,
    ) -> Sequence[PromptUpdate]:
        processor = self.info.get_hf_processor(**hf_processor_mm_kwargs)
        feature_extractor = processor.feature_extractor

        audio_id = processor.audio_token_id
        out_mm_data = out_mm_kwargs.require_data()
        out_audio_items = out_mm_data.get("audio", [])

        def get_replacement(item_idx: int):
            if item_idx < len(out_audio_items):
                out_audio_data = out_audio_items[item_idx].get_data()
                audio_arr = out_audio_data["audio_arrays"]
                if isinstance(audio_arr, (torch.Tensor, np.ndarray)):
                    audio_len = len(audio_arr)
                else:
                    raise TypeError(
                        "Unexpected type for audio_arrays in out_mm_kwargs: "
                        f"{type(audio_arr)}"
                    )
            else:
                # Fallback for unexpected processor outputs.
                audios = mm_items.get_items("audio", AudioProcessorItems)
                audio_len = audios.get_audio_length(item_idx)

            nb_audio_tokens = feature_extractor.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,
        inputs: ProcessorInputs,
        timing_ctx: TimingContext,
    ) -> tuple[list[int], MultiModalProcessingInfo, bool]:
        prompt_ids, mm_info, _ = super()._cached_apply_hf_processor(inputs, timing_ctx)

        # NOTE: The tokens are already inserted by the chat template
        return prompt_ids, mm_info, True


@MULTIMODAL_REGISTRY.register_processor(
    VoxtralMultiModalProcessor,
    info=VoxtralProcessingInfo,
    dummy_inputs=VoxtralDummyInputsBuilder,
)
class VoxtralForConditionalGeneration(
    nn.Module, SupportsMultiModal, SupportsPP, SupportsLoRA, SupportsTranscription
):
    supported_languages = ISO639_1_SUPPORTED_LANGS
    # transformers' currently has limited support for MistralCommon backend
    # and cached_get_processor. Let's skip until fixed
    skip_warmup_audio_preprocessing = True

    packed_modules_mapping = {
        "qkv_proj": ["q_proj", "k_proj", "v_proj"],
        "gate_up_proj": ["gate_proj", "up_proj"],
    }

    def __init__(self, *, vllm_config: VllmConfig, prefix: str = ""):
        super().__init__()
        self.tokenizer = cached_tokenizer_from_config(vllm_config.model_config)

        # update quant config to so that ignored module and target module names
        # match the vLLM model names
        if hasattr(vllm_config, "quant_config"):
            vllm_config.quant_config = self.maybe_update_quant_config(
                vllm_config.quant_config
            )

        config = vllm_config.model_config.hf_config
        self.config = config
        self.downsample_factor = self.config.audio_config.downsample_factor

        with self._mark_language_model(vllm_config):
            self.language_model = init_vllm_registered_model(
                vllm_config=vllm_config,
                hf_config=config.text_config,
                prefix=maybe_prefix(prefix, "language_model"),
            )

        with self._mark_tower_model(vllm_config, "audio"):
            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,
            )

        self.make_empty_intermediate_tensors = (
            self.language_model.make_empty_intermediate_tensors
        )

    def get_mm_mapping(self) -> MultiModelKeys:
        """Get module prefix for multimodal models to filter LoRA modules."""
        return MultiModelKeys.from_string_field(
            language_model="language_model",
            connector="audio_language_adapter",
            tower_model=["whisper_encoder"],
        )

    def forward(
        self,
        input_ids: torch.Tensor | None,
        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 embed_multimodal(
        self, **kwargs
    ) -> 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 _parse_and_validate_audio_arrays(
        self, **kwargs: object
    ) -> 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(
                f"Incorrect type of audio_arrays. Got type: {type(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,
    ) -> torch.Tensor | None:
        return self.language_model.compute_logits(hidden_states)

    @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,
        stt_params: SpeechToTextParams,
    ) -> PromptType:
        audio = stt_params.audio
        model_config = stt_params.model_config
        stt_config = stt_params.stt_config
        language = stt_params.language

        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=audio.to_base64(audio.format),
            language=language,
        )

        tokenized = tokenizer.instruct.encode_transcription(req)

        return TokensPrompt(
            prompt_token_ids=tokenized.tokens,
            multi_modal_data={
                "audio": [
                    (audio.audio_array, stt_config.sample_rate)
                    for audio in tokenized.audios
                ],
            },
        )

    @classmethod
    def get_num_audio_tokens(
        cls,
        audio_duration_s: float,
        stt_config: SpeechToTextConfig,
        model_config: ModelConfig,
    ) -> int | None:
        """
        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)
        feature_extractor = MistralCommonFeatureExtractor(
            tokenizer.instruct.audio_encoder
        )
        return feature_extractor.get_num_audio_tokens(
            int(audio_duration_s * stt_config.sample_rate)
        )

    def load_weights(self, weights: Iterable[tuple[str, torch.Tensor]]) -> set[str]:
        remapping_rules = [
            (r"mm_streams_embeddings.embedding_module\.(.*)", r"\1"),
            (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",
            ),
            (
                r"audio_language_adapter\.2\.weight",
                r"audio_language_adapter.w_out.weight",
            ),
        ]

        audio_params = dict(
            nn.ModuleDict(
                {
                    "audio_language_adapter": self.audio_language_adapter,
                }
            ).named_parameters()
        )
        weights = _create_fake_bias_for_k_proj(weights, ".wk.weight")

        loaded_weights = set()

        def llm_weights_generator():
            nonlocal loaded_weights
            for name, w in weights:
                is_encoder = False
                for k in [
                    "mm_whisper_embeddings",
                    "mm_streams_embeddings.embedding_module",
                ]:
                    is_encoder |= (
                        name.startswith(k)
                        and not name.startswith(f"{k}.tok_embeddings")
                        and not name.startswith(f"{k}.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

    def maybe_update_quant_config(
        self, quant_config: QuantizationConfig
    ) -> QuantizationConfig:
        """
        Update quant config to so that ignored module and target module names
        match the vLLM model names.
        Right now this is specific for compressed-tensors format and
        load_format mistral.
        """
        remapping_rules = [
            (r"output", r"language_model.lm_head"),
            (
                r"layers\.(\d+)\.attention\.wo",
                r"language_model.model.layers.\1.self_attn.out_proj",
            ),
            (
                r"layers\.(\d+)\.attention\.w(.*)",
                r"language_model.model.layers.\1.self_attn.\2_proj",
            ),
            (
                r"layers\.(\d+)\.feed_forward\.w1",
                r"language_model.model.layers.\1.mlp.gate_proj",
            ),
            (
                r"layers\.(\d+)\.feed_forward\.w2",
                r"language_model.model.layers.\1.mlp.down_proj",
            ),
            (
                r"layers\.(\d+)\.feed_forward\.w3",
                r"language_model.model.layers.\1.mlp.up_proj",
            ),
            (
                r"mm_whisper_embeddings\.whisper_encoder\.transformer\.layers\.(\d+)\.attention.w(.*)",
                r"whisper_encoder.whisper_encoder.layers.\1.layers.self_attn.\2_proj",
            ),
            (
                r"mm_whisper_embeddings\.whisper_encoder\.transformer\.layers\.(\d+)\.attention.wo",
                r"whisper_encoder.whisper_encoder.layers.\1.layers.self_attn.out_proj",
            ),
            (
                r"mm_whisper_embeddings\.whisper_encoder\.transformer\.layers\.(\d+)\.feed_forward.w(\d+)",
                r"whisper_encoder.whisper_encoder.layers.\1.layers.mlp.fc\2",
            ),
            (
                r"mm_whisper_embeddings\.whisper_encoder\.conv_layers\.0",
                r"whisper_encoder.whisper_encoder.conv1",
            ),
            (
                r"mm_whisper_embeddings\.whisper_encoder\.conv_layers\.1",
                r"whisper_encoder.whisper_encoder.conv2",
            ),
            (
                r"mm_whisper_embeddings\.audio_language_projection\.0",
                r"audio_language_adapter.w_in",
            ),
            (
                r"mm_whisper_embeddings\.audio_language_projection\.2",
                r"audio_language_adapter.w_out",
            ),
        ]

        # Update ignore list
        if hasattr(quant_config, "ignore"):
            mistral_ignore = []
            for name in quant_config.ignore:
                mistral_name = name
                for pattern, repl in remapping_rules:
                    if re.fullmatch(pattern, name):
                        mistral_name = re.sub(pattern, repl, name)
                mistral_ignore.append(mistral_name)
            quant_config.ignore = mistral_ignore

        # Update target list
        if hasattr(quant_config, "config_groups"):
            config_groups = quant_config.config_groups
            for group_name in config_groups:
                if "targets" in config_groups[group_name]:
                    targets = []
                    for name in config_groups[group_name]["targets"]:
                        mistral_name = name
                        for pattern, repl in remapping_rules:
                            if re.fullmatch(pattern, name):
                                mistral_name = re.sub(pattern, repl, name)
                        targets.append(mistral_name)
                config_groups[group_name]["targets"] = targets
            quant_config.config_groups = config_groups

        return quant_config


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"]}

    mistral_remapping = [
        (r"mm_streams_embeddings.embedding_module\.(.*)", r"\1"),
        (
            r"whisper_encoder\.conv_layers\.0\.(weight|bias)",
            r"whisper_encoder.conv1.\1",
        ),
        (
            r"whisper_encoder\.conv_layers\.1\.(weight|bias)",
            r"whisper_encoder.conv2.\1",
        ),
        (
            r"whisper_encoder\.conv_layers\.0\.conv\.(weight|bias)",
            r"whisper_encoder.conv1.\1",
        ),  # noqa: E501
        (
            r"whisper_encoder\.conv_layers\.1\.conv\.(weight|bias)",
            r"whisper_encoder.conv2.\1",
        ),  # noqa: E501
        (
            r"whisper_encoder\.transformer\.layers\.(\d+)\.attention\.w([qkv])\.(weight|bias)",  # noqa: E501
            r"whisper_encoder.layers.\1.self_attn.\2_proj.\3",
        ),
        (
            r"whisper_encoder\.transformer\.layers\.(\d+)\.attention\.wo\.(weight|bias)",  # noqa: E501
            r"whisper_encoder.layers.\1.self_attn.out_proj.\2",
        ),
        (
            r"whisper_encoder\.transformer\.layers\.(\d+)\.attention_norm\.(weight|bias)",  # noqa: E501
            r"whisper_encoder.layers.\1.self_attn_layer_norm.\2",
        ),
        (
            r"whisper_encoder\.transformer\.layers\.(\d+)\.feed_forward\.w1\.(weight|bias)",  # noqa: E501
            r"whisper_encoder.layers.\1.mlp.fc1.\2",
        ),
        (
            r"whisper_encoder\.transformer\.layers\.(\d+)\.feed_forward\.w2\.(weight|bias)",  # noqa: E501
            r"whisper_encoder.layers.\1.mlp.fc2.\2",
        ),
        (
            r"whisper_encoder\.transformer\.layers\.(\d+)\.feed_forward\.w3\.(weight|bias)",
            r"whisper_encoder.layers.\1.mlp.fc3.\2",
        ),  # noqa: E501
        (
            r"whisper_encoder\.transformer\.layers\.(\d+)\.ffn_norm\.(weight|bias)",
            r"whisper_encoder.layers.\1.final_layer_norm.\2",
        ),
        (
            r"whisper_encoder\.transformer\.norm\.(weight|bias)",
            r"whisper_encoder.layer_norm.\1",
        ),
    ]

    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.is_causal = getattr(self.config, "is_causal", False)
        if self.is_causal:
            WhisperEncoderCls = WhisperCausalEncoder
        else:
            WhisperEncoderCls = partial(WhisperEncoder, init_in_fp32=True)

        self.whisper_encoder = WhisperEncoderCls(
            vllm_config=vllm_config,
            prefix=maybe_prefix(prefix, "whisper_encoder"),
        )
        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, device=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()

        if global_log_mel_max := self.config.global_log_mel_max:
            if not isinstance(global_log_mel_max, float):
                raise TypeError(f"{global_log_mel_max=} needs to be of type float.")
            log_spec_max = torch.tensor(
                global_log_mel_max,
                device=log_spec.device,
                dtype=log_spec.dtype,
            )
        else:
            log_spec_max = log_spec.max()

        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: 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_mapping = []

        if self.is_causal:
            # For `WhisperCausalEncoder` we need
            # some more renaming
            stacked_params_mapping.extend(
                [
                    (".mlp.gate_up_proj", ".mlp.fc1", 0),
                    (".mlp.gate_up_proj", ".mlp.fc3", 1),
                ]
            )
            params_mapping.extend(
                [
                    (".mlp.down_proj", ".mlp.fc2"),
                ]
            )
        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:
            for param_name, weight_name in params_mapping:
                if weight_name not in name:
                    continue
                name = name.replace(weight_name, param_name)

            param = params_dict[name]
            weight_loader = getattr(param, "weight_loader", default_weight_loader)
            weight_loader(param, loaded_weight)

        return name
