wan_audio_runner.py

import os
import gc
import numpy as np
import torch
import torchvision.transforms.functional as TF
from PIL import Image
from contextlib import contextmanager
from typing import Optional, Tuple, Union, List, Dict, Any
from dataclasses import dataclass

from lightx2v.utils.registry_factory import RUNNER_REGISTER
from lightx2v.models.runners.wan.wan_runner import WanRunner
from lightx2v.utils.profiler import ProfilingContext4Debug, ProfilingContext
from lightx2v.models.networks.wan.audio_model import WanAudioModel
from lightx2v.models.networks.wan.lora_adapter import WanLoraWrapper
from lightx2v.models.networks.wan.audio_adapter import AudioAdapter, AudioAdapterPipe, rank0_load_state_dict_from_path
from lightx2v.utils.utils import save_to_video, vae_to_comfyui_image
from lightx2v.models.schedulers.wan.audio.scheduler import ConsistencyModelScheduler

from loguru import logger
from einops import rearrange
import torchaudio as ta
from transformers import AutoFeatureExtractor

from torchvision.transforms import InterpolationMode
from torchvision.transforms.functional import resize

import subprocess
import warnings


@contextmanager
def memory_efficient_inference():
    """Context manager for memory-efficient inference"""
    try:
        yield
    finally:
        if torch.cuda.is_available():
            torch.cuda.empty_cache()
        gc.collect()


def get_optimal_patched_size_with_sp(patched_h, patched_w, sp_size):
    assert sp_size > 0 and (sp_size & (sp_size - 1)) == 0, "sp_size must be a power of 2"

    h_ratio, w_ratio = 1, 1
    while sp_size != 1:
        sp_size //= 2
        if patched_h % 2 == 0:
            patched_h //= 2
            h_ratio *= 2
        elif patched_w % 2 == 0:
            patched_w //= 2
            w_ratio *= 2
        else:
            if patched_h > patched_w:
                patched_h //= 2
                h_ratio *= 2
            else:
                patched_w //= 2
                w_ratio *= 2
    return patched_h * h_ratio, patched_w * w_ratio


def get_crop_bbox(ori_h, ori_w, tgt_h, tgt_w):
    tgt_ar = tgt_h / tgt_w
    ori_ar = ori_h / ori_w
    if abs(ori_ar - tgt_ar) < 0.01:
        return 0, ori_h, 0, ori_w
    if ori_ar > tgt_ar:
        crop_h = int(tgt_ar * ori_w)
        y0 = (ori_h - crop_h) // 2
        y1 = y0 + crop_h
        return y0, y1, 0, ori_w
    else:
        crop_w = int(ori_h / tgt_ar)
        x0 = (ori_w - crop_w) // 2
        x1 = x0 + crop_w
        return 0, ori_h, x0, x1


def isotropic_crop_resize(frames: torch.Tensor, size: tuple):
    """
    frames: (T, C, H, W)
    size: (H, W)
    """
    ori_h, ori_w = frames.shape[2:]
    h, w = size
    y0, y1, x0, x1 = get_crop_bbox(ori_h, ori_w, h, w)
    cropped_frames = frames[:, :, y0:y1, x0:x1]
    resized_frames = resize(cropped_frames, [h, w], InterpolationMode.BICUBIC, antialias=True)
    return resized_frames


def adaptive_resize(img):
    bucket_config = {
        0.667: (np.array([[480, 832], [544, 960], [720, 1280]], dtype=np.int64), np.array([0.2, 0.5, 0.3])),
        1.0: (np.array([[480, 480], [576, 576], [704, 704], [960, 960]], dtype=np.int64), np.array([0.1, 0.1, 0.5, 0.3])),
        1.5: (np.array([[480, 832], [544, 960], [720, 1280]], dtype=np.int64)[:, ::-1], np.array([0.2, 0.5, 0.3])),
    }
    ori_height = img.shape[-2]
    ori_weight = img.shape[-1]
    ori_ratio = ori_height / ori_weight
    aspect_ratios = np.array(np.array(list(bucket_config.keys())))
    closet_aspect_idx = np.argmin(np.abs(aspect_ratios - ori_ratio))
    closet_ratio = aspect_ratios[closet_aspect_idx]
    if ori_ratio < 1.0:
        target_h, target_w = 480, 832
    elif ori_ratio == 1.0:
        target_h, target_w = 480, 480
    else:
        target_h, target_w = 832, 480
    for resolution in bucket_config[closet_ratio][0]:
        if ori_height * ori_weight >= resolution[0] * resolution[1]:
            target_h, target_w = resolution
    cropped_img = isotropic_crop_resize(img, (target_h, target_w))
    return cropped_img, target_h, target_w


@dataclass
class AudioSegment:
    """Data class for audio segment information"""

    audio_array: np.ndarray
    start_frame: int
    end_frame: int
    is_last: bool = False
    useful_length: Optional[int] = None


class FramePreprocessor:
    """Handles frame preprocessing including noise and masking"""

    def __init__(self, noise_mean: float = -3.0, noise_std: float = 0.5, mask_rate: float = 0.1):
        self.noise_mean = noise_mean
        self.noise_std = noise_std
        self.mask_rate = mask_rate

    def add_noise(self, frames: np.ndarray, rnd_state: Optional[np.random.RandomState] = None) -> np.ndarray:
        """Add noise to frames"""
        if self.noise_mean is None or self.noise_std is None:
            return frames

        if rnd_state is None:
            rnd_state = np.random.RandomState()

        shape = frames.shape
        bs = 1 if len(shape) == 4 else shape[0]
        sigma = rnd_state.normal(loc=self.noise_mean, scale=self.noise_std, size=(bs,))
        sigma = np.exp(sigma)
        sigma = np.expand_dims(sigma, axis=tuple(range(1, len(shape))))
        noise = rnd_state.randn(*shape) * sigma
        return frames + noise

    def add_mask(self, frames: np.ndarray, rnd_state: Optional[np.random.RandomState] = None) -> np.ndarray:
        """Add mask to frames"""
        if self.mask_rate is None:
            return frames

        if rnd_state is None:
            rnd_state = np.random.RandomState()

        h, w = frames.shape[-2:]
        mask = rnd_state.rand(h, w) > self.mask_rate
        return frames * mask

    def process_prev_frames(self, frames: torch.Tensor) -> torch.Tensor:
        """Process previous frames with noise and masking"""
        frames_np = frames.cpu().detach().numpy()
        frames_np = self.add_noise(frames_np)
        frames_np = self.add_mask(frames_np)
        return torch.from_numpy(frames_np).to(dtype=frames.dtype, device=frames.device)


class AudioProcessor:
    """Handles audio loading and segmentation"""

    def __init__(self, audio_sr: int = 16000, target_fps: int = 16):
        self.audio_sr = audio_sr
        self.target_fps = target_fps

    def load_audio(self, audio_path: str) -> np.ndarray:
        """Load and resample audio"""
        audio_array, ori_sr = ta.load(audio_path)
        audio_array = ta.functional.resample(audio_array.mean(0), orig_freq=ori_sr, new_freq=self.audio_sr)
        return audio_array.numpy()

    def get_audio_range(self, start_frame: int, end_frame: int) -> Tuple[int, int]:
        """Calculate audio range for given frame range"""
        audio_frame_rate = self.audio_sr / self.target_fps
        return round(start_frame * audio_frame_rate), round((end_frame + 1) * audio_frame_rate)

    def segment_audio(self, audio_array: np.ndarray, expected_frames: int, max_num_frames: int, prev_frame_length: int = 5) -> List[AudioSegment]:
        """Segment audio based on frame requirements"""
        segments = []

        # Calculate intervals
        interval_num = 1
        res_frame_num = 0

        if expected_frames <= max_num_frames:
            interval_num = 1
        else:
            interval_num = max(int((expected_frames - max_num_frames) / (max_num_frames - prev_frame_length)) + 1, 1)
            res_frame_num = expected_frames - interval_num * (max_num_frames - prev_frame_length)
            if res_frame_num > 5:
                interval_num += 1

        # Create segments
        for idx in range(interval_num):
            if idx == 0:
                # First segment
                audio_start, audio_end = self.get_audio_range(0, max_num_frames)
                segment_audio = audio_array[audio_start:audio_end]
                useful_length = None

                if expected_frames < max_num_frames:
                    useful_length = segment_audio.shape[0]
                    max_num_audio_length = int((max_num_frames + 1) / self.target_fps * self.audio_sr)
                    segment_audio = np.concatenate((segment_audio, np.zeros(max_num_audio_length - useful_length)), axis=0)

                segments.append(AudioSegment(segment_audio, 0, max_num_frames, False, useful_length))

            elif res_frame_num > 5 and idx == interval_num - 1:
                # Last segment (might be shorter)
                start_frame = idx * max_num_frames - idx * prev_frame_length
                audio_start, audio_end = self.get_audio_range(start_frame, expected_frames)
                segment_audio = audio_array[audio_start:audio_end]
                useful_length = segment_audio.shape[0]

                max_num_audio_length = int((max_num_frames + 1) / self.target_fps * self.audio_sr)
                segment_audio = np.concatenate((segment_audio, np.zeros(max_num_audio_length - useful_length)), axis=0)

                segments.append(AudioSegment(segment_audio, start_frame, expected_frames, True, useful_length))

            else:
                # Middle segments
                start_frame = idx * max_num_frames - idx * prev_frame_length
                end_frame = (idx + 1) * max_num_frames - idx * prev_frame_length
                audio_start, audio_end = self.get_audio_range(start_frame, end_frame)
                segment_audio = audio_array[audio_start:audio_end]

                segments.append(AudioSegment(segment_audio, start_frame, end_frame, False))

        return segments


class VideoGenerator:
    """Handles video generation for each segment"""

    def __init__(self, model, vae_encoder, vae_decoder, config, progress_callback=None):
        self.model = model
        self.vae_encoder = vae_encoder
        self.vae_decoder = vae_decoder
        self.config = config
        self.frame_preprocessor = FramePreprocessor()
        self.progress_callback = progress_callback
        self.total_segments = 1

    def prepare_prev_latents(self, prev_video: Optional[torch.Tensor], prev_frame_length: int) -> Optional[Dict[str, torch.Tensor]]:
        """Prepare previous latents for conditioning"""
        if prev_video is None:
            return None

        device = self.model.device
        dtype = torch.bfloat16
        vae_dtype = torch.float

        tgt_h, tgt_w = self.config.tgt_h, self.config.tgt_w
        prev_frames = torch.zeros((1, 3, self.config.target_video_length, tgt_h, tgt_w), device=device)

        # Extract and process last frames
        last_frames = prev_video[:, :, -prev_frame_length:].clone().to(device)
        last_frames = self.frame_preprocessor.process_prev_frames(last_frames)

        prev_frames[:, :, :prev_frame_length] = last_frames
        prev_latents = self.vae_encoder.encode(prev_frames.to(vae_dtype), self.config)[0].to(dtype)

        # Create mask
        prev_token_length = (prev_frame_length - 1) // 4 + 1
        _, nframe, height, width = self.model.scheduler.latents.shape
        frames_n = (nframe - 1) * 4 + 1
        prev_frame_len = max((prev_token_length - 1) * 4 + 1, 0)

        prev_mask = torch.ones((1, frames_n, height, width), device=device, dtype=dtype)
        prev_mask[:, prev_frame_len:] = 0
        prev_mask = self._wan_mask_rearrange(prev_mask).unsqueeze(0)

        if prev_latents.shape[-2:] != (height, width):
            logger.warning(f"Size mismatch: prev_latents {prev_latents.shape} vs scheduler latents (H={height}, W={width}). Config tgt_h={self.config.tgt_h}, tgt_w={self.config.tgt_w}")
            prev_latents = torch.nn.functional.interpolate(prev_latents, size=(height, width), mode="bilinear", align_corners=False)

        return {"prev_latents": prev_latents, "prev_mask": prev_mask}

    def _wan_mask_rearrange(self, mask: torch.Tensor) -> torch.Tensor:
        """Rearrange mask for WAN model"""
        if mask.ndim == 3:
            mask = mask[None]
        assert mask.ndim == 4
        _, t, h, w = mask.shape
        assert t == ((t - 1) // 4 * 4 + 1)
        mask_first_frame = torch.repeat_interleave(mask[:, 0:1], repeats=4, dim=1)
        mask = torch.concat([mask_first_frame, mask[:, 1:]], dim=1)
        mask = mask.view(mask.shape[1] // 4, 4, h, w)
        return mask.transpose(0, 1)

    @torch.no_grad()
    def generate_segment(self, inputs: Dict[str, Any], audio_features: torch.Tensor, prev_video: Optional[torch.Tensor] = None, prev_frame_length: int = 5, segment_idx: int = 0) -> torch.Tensor:
        """Generate video segment"""
        # Update inputs with audio features
        inputs["audio_encoder_output"] = audio_features

        # Reset scheduler for non-first segments
        if segment_idx > 0:
            self.model.scheduler.reset()

        # Prepare previous latents - ALWAYS needed, even for first segment
        device = self.model.device
        dtype = torch.bfloat16
        vae_dtype = torch.float
        tgt_h, tgt_w = self.config.tgt_h, self.config.tgt_w
        max_num_frames = self.config.target_video_length

        if segment_idx == 0:
            # First segment - create zero frames
            prev_frames = torch.zeros((1, 3, max_num_frames, tgt_h, tgt_w), device=device)
            prev_latents = self.vae_encoder.encode(prev_frames.to(vae_dtype), self.config)[0].to(dtype)
            prev_len = 0
        else:
            # Subsequent segments - use previous video
            previmg_encoder_output = self.prepare_prev_latents(prev_video, prev_frame_length)
            if previmg_encoder_output:
                prev_latents = previmg_encoder_output["prev_latents"]
                prev_len = (prev_frame_length - 1) // 4 + 1
            else:
                # Fallback to zeros if prepare_prev_latents fails
                prev_frames = torch.zeros((1, 3, max_num_frames, tgt_h, tgt_w), device=device)
                prev_latents = self.vae_encoder.encode(prev_frames.to(vae_dtype), self.config)[0].to(dtype)
                prev_len = 0

        # Create mask for prev_latents
        _, nframe, height, width = self.model.scheduler.latents.shape
        frames_n = (nframe - 1) * 4 + 1
        prev_frame_len = max((prev_len - 1) * 4 + 1, 0)

        prev_mask = torch.ones((1, frames_n, height, width), device=device, dtype=dtype)
        prev_mask[:, prev_frame_len:] = 0
        prev_mask = self._wan_mask_rearrange(prev_mask).unsqueeze(0)

        if prev_latents.shape[-2:] != (height, width):
            logger.warning(f"Size mismatch: prev_latents {prev_latents.shape} vs scheduler latents (H={height}, W={width}). Config tgt_h={self.config.tgt_h}, tgt_w={self.config.tgt_w}")
            prev_latents = torch.nn.functional.interpolate(prev_latents, size=(height, width), mode="bilinear", align_corners=False)

        # Always set previmg_encoder_output
        inputs["previmg_encoder_output"] = {"prev_latents": prev_latents, "prev_mask": prev_mask}

        # Run inference loop
        total_steps = self.model.scheduler.infer_steps
        for step_index in range(total_steps):
            logger.info(f"==> Segment {segment_idx}, Step {step_index}/{total_steps}")

            with ProfilingContext4Debug("step_pre"):
                self.model.scheduler.step_pre(step_index=step_index)

            with ProfilingContext4Debug("infer"):
                self.model.infer(inputs)

            with ProfilingContext4Debug("step_post"):
                self.model.scheduler.step_post()

            if self.progress_callback:
                segment_progress = (segment_idx * total_steps + step_index + 1) / (self.total_segments * total_steps)
                self.progress_callback(int(segment_progress * 100), 100)

        # Decode latents
        latents = self.model.scheduler.latents
        generator = self.model.scheduler.generator
        gen_video = self.vae_decoder.decode(latents, generator=generator, config=self.config)
        gen_video = torch.clamp(gen_video, -1, 1).to(torch.float)

        return gen_video


@RUNNER_REGISTER("wan2.1_audio")
class WanAudioRunner(WanRunner):  # type:ignore
    def __init__(self, config):
        super().__init__(config)
        self._audio_adapter_pipe = None
        self._audio_processor = None
        self._video_generator = None
        self._audio_preprocess = None

    def initialize(self):
        """Initialize all models once for multiple runs"""

        # Initialize audio processor
        audio_sr = self.config.get("audio_sr", 16000)
        target_fps = self.config.get("target_fps", 16)
        self._audio_processor = AudioProcessor(audio_sr, target_fps)

        # Initialize scheduler
        self.init_scheduler()

    def init_scheduler(self):
        """Initialize consistency model scheduler"""
        scheduler = ConsistencyModelScheduler(self.config)
        self.model.set_scheduler(scheduler)

    def load_audio_adapter_lazy(self):
        """Lazy load audio adapter when needed"""
        if self._audio_adapter_pipe is not None:
            return self._audio_adapter_pipe

        # Audio adapter
        audio_adapter_path = self.config["model_path"] + "/audio_adapter.safetensors"
        audio_adapter = AudioAdapter.from_transformer(
            self.model,
            audio_feature_dim=1024,
            interval=1,
            time_freq_dim=256,
            projection_transformer_layers=4,
        )
        audio_adapter = rank0_load_state_dict_from_path(audio_adapter, audio_adapter_path, strict=False)

        # Audio encoder
        device = self.model.device
        audio_encoder_repo = self.config["model_path"] + "/audio_encoder"
        self._audio_adapter_pipe = AudioAdapterPipe(audio_adapter, audio_encoder_repo=audio_encoder_repo, dtype=torch.bfloat16, device=device, generator=torch.Generator(device), weight=1.0)

        return self._audio_adapter_pipe

    def prepare_inputs(self):
        """Prepare inputs for the model"""
        image_encoder_output = None

        if os.path.isfile(self.config.image_path):
            with ProfilingContext("Run Img Encoder"):
                vae_encode_out, clip_encoder_out = self.run_image_encoder(self.config, self.vae_encoder)
                image_encoder_output = {
                    "clip_encoder_out": clip_encoder_out,
                    "vae_encode_out": vae_encode_out,
                }

        with ProfilingContext("Run Text Encoder"):
            img = Image.open(self.config["image_path"]).convert("RGB")
            text_encoder_output = self.run_text_encoder(self.config["prompt"], img)

        self.set_target_shape()

        return {"text_encoder_output": text_encoder_output, "image_encoder_output": image_encoder_output, "audio_adapter_pipe": self.load_audio_adapter_lazy()}

    def run_pipeline(self, save_video=True):
        """Optimized pipeline with modular components"""
        # Ensure models are initialized
        self.initialize()

        assert self._audio_processor is not None
        assert self._audio_preprocess is not None

        # Initialize video generator if needed
        if self._video_generator is None:
            self._video_generator = VideoGenerator(self.model, self.vae_encoder, self.vae_decoder, self.config, self.progress_callback)

        # Prepare inputs
        with memory_efficient_inference():
            if self.config["use_prompt_enhancer"]:
                self.config["prompt_enhanced"] = self.post_prompt_enhancer()

            self.inputs = self.prepare_inputs()
            # Re-initialize scheduler after image encoding sets correct dimensions
            self.init_scheduler()
            self.model.scheduler.prepare(self.inputs["image_encoder_output"])

        # Re-create video generator with updated model/scheduler
        self._video_generator = VideoGenerator(self.model, self.vae_encoder, self.vae_decoder, self.config, self.progress_callback)

        # Process audio
        audio_array = self._audio_processor.load_audio(self.config["audio_path"])
        video_duration = self.config.get("video_duration", 5)
        target_fps = self.config.get("target_fps", 16)
        max_num_frames = self.config.get("target_video_length", 81)

        audio_len = int(audio_array.shape[0] / self._audio_processor.audio_sr * target_fps)
        expected_frames = min(max(1, int(video_duration * target_fps)), audio_len)

        # Segment audio
        audio_segments = self._audio_processor.segment_audio(audio_array, expected_frames, max_num_frames)

        self._video_generator.total_segments = len(audio_segments)

        # Generate video segments
        gen_video_list = []
        cut_audio_list = []
        prev_video = None

        for idx, segment in enumerate(audio_segments):
            # Update seed for each segment
            self.config.seed = self.config.seed + idx
            torch.manual_seed(self.config.seed)
            logger.info(f"Processing segment {idx + 1}/{len(audio_segments)}, seed: {self.config.seed}")

            # Process audio features
            audio_features = self._audio_preprocess(segment.audio_array, sampling_rate=self._audio_processor.audio_sr, return_tensors="pt").input_values.squeeze(0).to(self.model.device)

            # Generate video segment
            with memory_efficient_inference():
                gen_video = self._video_generator.generate_segment(
                    self.inputs.copy(),  # Copy to avoid modifying original
                    audio_features,
                    prev_video=prev_video,
                    prev_frame_length=5,
                    segment_idx=idx,
                )

            # Extract relevant frames
            start_frame = 0 if idx == 0 else 5
            start_audio_frame = 0 if idx == 0 else int(6 * self._audio_processor.audio_sr / target_fps)

            if segment.is_last and segment.useful_length:
                end_frame = segment.end_frame - segment.start_frame
                gen_video_list.append(gen_video[:, :, start_frame:end_frame].cpu())
                cut_audio_list.append(segment.audio_array[start_audio_frame : segment.useful_length])
            elif segment.useful_length and expected_frames < max_num_frames:
                gen_video_list.append(gen_video[:, :, start_frame:expected_frames].cpu())
                cut_audio_list.append(segment.audio_array[start_audio_frame : segment.useful_length])
            else:
                gen_video_list.append(gen_video[:, :, start_frame:].cpu())
                cut_audio_list.append(segment.audio_array[start_audio_frame:])

            # Update prev_video for next iteration
            prev_video = gen_video

            # Clean up GPU memory after each segment
            del gen_video
            torch.cuda.empty_cache()

        # Merge results
        with memory_efficient_inference():
            gen_lvideo = torch.cat(gen_video_list, dim=2).float()
            merge_audio = np.concatenate(cut_audio_list, axis=0).astype(np.float32)
            comfyui_images = vae_to_comfyui_image(gen_lvideo)

        # Apply frame interpolation if configured
        if "video_frame_interpolation" in self.config and self.vfi_model is not None:
            interpolation_target_fps = self.config["video_frame_interpolation"]["target_fps"]
            logger.info(f"Interpolating frames from {target_fps} to {interpolation_target_fps}")
            comfyui_images = self.vfi_model.interpolate_frames(
                comfyui_images,
                source_fps=target_fps,
                target_fps=interpolation_target_fps,
            )
            target_fps = interpolation_target_fps

        # Convert audio to ComfyUI format
        audio_waveform = torch.from_numpy(merge_audio).unsqueeze(0).unsqueeze(0)
        comfyui_audio = {"waveform": audio_waveform, "sample_rate": self._audio_processor.audio_sr}

        # Save video if requested
        if save_video and self.config.get("save_video_path", None):
            self._save_video_with_audio(comfyui_images, merge_audio, target_fps)

        # Final cleanup
        self.end_run()

        return comfyui_images, comfyui_audio

    def _save_video_with_audio(self, images, audio_array, fps):
        """Save video with audio"""
        import tempfile

        with tempfile.NamedTemporaryFile(suffix=".mp4", delete=False) as video_tmp:
            video_path = video_tmp.name

        with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as audio_tmp:
            audio_path = audio_tmp.name

        try:
            save_to_video(images, video_path, fps)
            ta.save(audio_path, torch.tensor(audio_array[None]), sample_rate=self._audio_processor.audio_sr)  # type: ignore

            output_path = self.config.get("save_video_path")
            parent_dir = os.path.dirname(output_path)
            if parent_dir and not os.path.exists(parent_dir):
                os.makedirs(parent_dir, exist_ok=True)

            subprocess.call(["/usr/bin/ffmpeg", "-y", "-i", video_path, "-i", audio_path, output_path])

            logger.info(f"Saved video with audio to: {output_path}")

        finally:
            # Clean up temp files
            if os.path.exists(video_path):
                os.remove(video_path)
            if os.path.exists(audio_path):
                os.remove(audio_path)

    def load_transformer(self):
        """Load transformer with LoRA support"""
        base_model = WanAudioModel(self.config.model_path, self.config, self.init_device)

        if self.config.get("lora_configs") and self.config.lora_configs:
            assert not self.config.get("dit_quantized", False) or self.config.mm_config.get("weight_auto_quant", False)
            lora_wrapper = WanLoraWrapper(base_model)
            for lora_config in self.config.lora_configs:
                lora_path = lora_config["path"]
                strength = lora_config.get("strength", 1.0)
                lora_name = lora_wrapper.load_lora(lora_path)
                lora_wrapper.apply_lora(lora_name, strength)
                logger.info(f"Loaded LoRA: {lora_name} with strength: {strength}")

        # XXX: trick
        self._audio_preprocess = AutoFeatureExtractor.from_pretrained(self.config["model_path"], subfolder="audio_encoder")

        return base_model

    def run_image_encoder(self, config, vae_model):
        """Run image encoder"""

        ref_img = Image.open(config.image_path)
        ref_img = (np.array(ref_img).astype(np.float32) - 127.5) / 127.5
        ref_img = torch.from_numpy(ref_img).to(vae_model.device)
        ref_img = rearrange(ref_img, "H W C -> 1 C H W")
        ref_img = ref_img[:, :3]

        adaptive = config.get("adaptive_resize", False)

        if adaptive:
            # Use adaptive_resize to modify aspect ratio
            ref_img, h, w = adaptive_resize(ref_img)

            patched_h = h // self.config.vae_stride[1] // self.config.patch_size[1]
            patched_w = w // self.config.vae_stride[2] // self.config.patch_size[2]

        else:
            h, w = ref_img.shape[2:]
            aspect_ratio = h / w
            max_area = config.target_height * config.target_width

            patched_h = round(np.sqrt(max_area * aspect_ratio) // config.vae_stride[1] // config.patch_size[1])
            patched_w = round(np.sqrt(max_area / aspect_ratio) // config.vae_stride[2] // config.patch_size[2])

        patched_h, patched_w = get_optimal_patched_size_with_sp(patched_h, patched_w, 1)

        config.lat_h = patched_h * self.config.patch_size[1]
        config.lat_w = patched_w * self.config.patch_size[2]

        config.tgt_h = config.lat_h * self.config.vae_stride[1]
        config.tgt_w = config.lat_w * self.config.vae_stride[2]

        logger.info(f"[wan_audio] adaptive_resize: {adaptive}, tgt_h: {config.tgt_h}, tgt_w: {config.tgt_w}, lat_h: {config.lat_h}, lat_w: {config.lat_w}")

        cond_frms = torch.nn.functional.interpolate(ref_img, size=(config.tgt_h, config.tgt_w), mode="bicubic")

        # clip encoder
        clip_encoder_out = self.image_encoder.visual([cond_frms], self.config).squeeze(0).to(torch.bfloat16)

        # vae encode
        cond_frms = rearrange(cond_frms, "1 C H W -> 1 C 1 H W")
        vae_encode_out = vae_model.encode(cond_frms.to(torch.float), config)
        if isinstance(vae_encode_out, list):
            vae_encode_out = torch.stack(vae_encode_out, dim=0).to(torch.bfloat16)

        return vae_encode_out, clip_encoder_out

    def set_target_shape(self):
        """Set target shape for generation"""
        ret = {}
        num_channels_latents = 16
        if self.config.task == "i2v":
            self.config.target_shape = (
                num_channels_latents,
                (self.config.target_video_length - 1) // self.config.vae_stride[0] + 1,
                self.config.lat_h,
                self.config.lat_w,
            )
            ret["lat_h"] = self.config.lat_h
            ret["lat_w"] = self.config.lat_w
        else:
            error_msg = "t2v task is not supported in WanAudioRunner"
            assert False, error_msg

        ret["target_shape"] = self.config.target_shape
        return ret