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# ---------------------------------------------------------
# sonic.py  (2025-05 rev – fix AudioProjModel tensor shape)
# ---------------------------------------------------------
import os, math, torch, cv2
import torch.utils.checkpoint
from PIL import Image
from omegaconf import OmegaConf
from tqdm import tqdm
from diffusers import AutoencoderKLTemporalDecoder
from diffusers.schedulers import EulerDiscreteScheduler
from transformers import WhisperModel, CLIPVisionModelWithProjection, AutoFeatureExtractor

from src.utils.util import save_videos_grid, seed_everything
from src.dataset.test_preprocess import image_audio_to_tensor, process_bbox
from src.models.base.unet_spatio_temporal_condition import (
    UNetSpatioTemporalConditionModel, add_ip_adapters,
)
from src.models.audio_adapter.audio_proj import AudioProjModel
from src.models.audio_adapter.audio_to_bucket import Audio2bucketModel
from src.pipelines.pipeline_sonic import SonicPipeline
from src.utils.RIFE.RIFE_HDv3 import RIFEModel
from src.dataset.face_align.align import AlignImage

BASE_DIR = os.path.dirname(os.path.abspath(__file__))


# ------------------------------------------------------------------
#                single image + speech  →  video tensor
# ------------------------------------------------------------------
def test(pipe, cfg, wav_enc, audio_pe, audio2bucket, img_enc,
         width, height, batch):

    # --- batch 차원 맞추기 ------------------------------------------
    for k, v in batch.items():
        if isinstance(v, torch.Tensor):
            batch[k] = v.unsqueeze(0).float().to(pipe.device)

    ref_img   = batch['ref_img']
    clip_img  = batch['clip_images']
    face_mask = batch['face_mask']
    img_emb   = img_enc(clip_img).image_embeds           # (1,1024)

    audio_feat = batch['audio_feature']   # (1,80,T)
    audio_len  = int(batch['audio_len'])
    step       = max(1, int(cfg.step))    # 안전 보정

    window = 16_000                       # 1-초 chunk
    prompt_list, last_list = [], []

    for i in range(0, audio_feat.shape[-1], window):
        chunk = audio_feat[:, :, i:i+window]
        hs    = wav_enc.encoder(chunk, output_hidden_states=True).hidden_states
        prompt_list.append(torch.stack(hs, 2))                     # (1,80,L,384)
        last  = wav_enc.encoder(chunk).last_hidden_state.unsqueeze(-2)
        last_list.append(last)                                     # (1,80,1,384)

    if not prompt_list:
        raise ValueError("❌ No speech recognised in audio.")

    audio_prompts = torch.cat(prompt_list, 1)   # (1,80,*L,384)
    last_prompts  = torch.cat(last_list, 1)     # (1,80,*1,384)

    # pad 규칙 (모델 원 논문과 동일)
    audio_prompts = torch.cat([ torch.zeros_like(audio_prompts[:,:4]),
                                audio_prompts,
                                torch.zeros_like(audio_prompts[:,:6]) ], 1)
    last_prompts  = torch.cat([ torch.zeros_like(last_prompts[:,:24]),
                                last_prompts,
                                torch.zeros_like(last_prompts[:,:26]) ], 1)

    # --------------------------------------------------------------
    total_tok = audio_prompts.shape[1]
    n_chunks  = max(1, math.ceil(total_tok / (2*step)))

    ref_L, aud_L, uncond_L, buckets = [], [], [], []

    for i in tqdm(range(n_chunks), ncols=0):
        st = i * 2 * step

        # ① 조건 오디오 토큰(pad → 10×5×384)
        cond = audio_prompts[:, st:st+10]            # (1,80,10,384) → (1,10,8,384)?
        cond = cond[:, :10]                          # f = 10
        cond = cond.permute(0,2,1,3)                 # (1,10,80,384)
        cond = cond.reshape(1, 10, 10, 5, 384)       # ★ w=10, b=5  (zero-pad auto)
        # ② bucket 추정용 토큰
        buck = last_prompts[:, st:st+50]             # (1,80,50,384)
        if buck.shape[1] < 50:
            pad = torch.zeros(1, 50-buck.shape[1], *buck.shape[2:], device=buck.device)
            buck = torch.cat([buck, pad], 1)
        buck = buck[:, :50].permute(0,2,1,3).reshape(1, 50, 10, 5, 384)

        motion = audio2bucket(buck, img_emb) * 16 + 16

        ref_L.append(ref_img[0])
        aud_L.append(audio_pe(cond).squeeze(0))          # (10,1024)
        uncond_L.append(audio_pe(torch.zeros_like(cond)).squeeze(0))
        buckets.append(motion[0])

    # -------------- diffusion -------------------------------------------------
    vid = pipe(
        ref_img, clip_img, face_mask,
        aud_L, uncond_L, buckets,
        height=height, width=width,
        num_frames=len(aud_L),
        decode_chunk_size=cfg.decode_chunk_size,
        motion_bucket_scale=cfg.motion_bucket_scale,
        fps=cfg.fps,
        noise_aug_strength=cfg.noise_aug_strength,
        min_guidance_scale1=cfg.min_appearance_guidance_scale,
        max_guidance_scale1=cfg.max_appearance_guidance_scale,
        min_guidance_scale2=cfg.audio_guidance_scale,
        max_guidance_scale2=cfg.audio_guidance_scale,
        overlap=cfg.overlap,
        shift_offset=cfg.shift_offset,
        frames_per_batch=cfg.n_sample_frames,
        num_inference_steps=cfg.num_inference_steps,
        i2i_noise_strength=cfg.i2i_noise_strength,
    ).frames

    return (vid*0.5+0.5).clamp(0,1).to(pipe.device).unsqueeze(0).cpu()


# ------------------------------------------------------------------
#                       Sonic  wrapper
# ------------------------------------------------------------------
class Sonic:
    config_file = os.path.join(BASE_DIR, "config/inference/sonic.yaml")
    config      = OmegaConf.load(config_file)

    def __init__(self, device_id=0, enable_interpolate_frame=True):
        cfg                = self.config
        cfg.use_interframe = enable_interpolate_frame
        self.device        = f"cuda:{device_id}" if torch.cuda.is_available() and device_id>=0 else "cpu"
        cfg.pretrained_model_name_or_path = os.path.join(BASE_DIR, cfg.pretrained_model_name_or_path)

        self._load_models(cfg)
        print("Sonic init done")

    # model-loader (unchanged, but with tiny clean-ups) ------------------------
    def _load_models(self, cfg):
        dtype = {"fp16": torch.float16, "fp32": torch.float32, "bf16": torch.bfloat16}[cfg.weight_dtype]

        vae = AutoencoderKLTemporalDecoder.from_pretrained(cfg.pretrained_model_name_or_path, subfolder="vae", variant="fp16")
        sched = EulerDiscreteScheduler.from_pretrained(cfg.pretrained_model_name_or_path, subfolder="scheduler")
        img_enc = CLIPVisionModelWithProjection.from_pretrained(cfg.pretrained_model_name_or_path, subfolder="image_encoder", variant="fp16")
        unet = UNetSpatioTemporalConditionModel.from_pretrained(cfg.pretrained_model_name_or_path, subfolder="unet", variant="fp16")
        add_ip_adapters(unet, [32], [cfg.ip_audio_scale])

        self.audio2token  = AudioProjModel(10, 5, 384, 1024, 1024, 32).to(self.device)
        self.audio2bucket = Audio2bucketModel(50, 1, 384, 1024, 1024, 1, 2).to(self.device)

        unet.load_state_dict (torch.load(os.path.join(BASE_DIR, cfg.unet_checkpoint_path),          map_location="cpu"))
        self.audio2token.load_state_dict (torch.load(os.path.join(BASE_DIR, cfg.audio2token_checkpoint_path),  map_location="cpu"))
        self.audio2bucket.load_state_dict(torch.load(os.path.join(BASE_DIR, cfg.audio2bucket_checkpoint_path), map_location="cpu"))

        self.whisper = WhisperModel.from_pretrained(os.path.join(BASE_DIR, "checkpoints/whisper-tiny")).to(self.device).eval()
        self.whisper.requires_grad_(False)

        self.feature_extractor = AutoFeatureExtractor.from_pretrained(os.path.join(BASE_DIR, "checkpoints/whisper-tiny"))
        self.face_det = AlignImage(self.device, det_path=os.path.join(BASE_DIR, "checkpoints/yoloface_v5m.pt"))
        if cfg.use_interframe:
            self.rife = RIFEModel(device=self.device); self.rife.load_model(os.path.join(BASE_DIR, "checkpoints/RIFE/"))

        for m in (img_enc, vae, unet): m.to(dtype)
        self.pipe = SonicPipeline(unet=unet, image_encoder=img_enc, vae=vae, scheduler=sched).to(device=self.device, dtype=dtype)
        self.image_encoder = img_enc

    # ------------------------------------------------------------------
    def preprocess(self, img_path, expand_ratio=1.0):
        img = cv2.imread(img_path)
        _, _, boxes = self.face_det(img, maxface=True)
        if boxes:
            x,y,w,h = boxes[0]; return {"face_num":1,"crop_bbox":process_bbox((x,y,x+w,y+h),expand_ratio,*img.shape[:2])}
        return {"face_num":0,"crop_bbox":None}

    # ------------------------------------------------------------------
    @torch.no_grad()
    def process(self, img_path, wav_path, out_path,
                min_resolution=512, inference_steps=25,
                dynamic_scale=1.0, keep_resolution=False, seed=None):

        cfg = self.config
        if seed is not None: cfg.seed = seed
        cfg.num_inference_steps = inference_steps
        cfg.motion_bucket_scale = dynamic_scale
        seed_everything(cfg.seed)

        sample = image_audio_to_tensor(
            self.face_det, self.feature_extractor,
            img_path, wav_path, limit=-1,
            image_size=min_resolution, area=cfg.area,
        )
        if sample is None: return -1

        h,w = sample['ref_img'].shape[-2:]
        resolution = (f"{Image.open(img_path).width//2*2}x{Image.open(img_path).height//2*2}"
                      if keep_resolution else f"{w}x{h}")

        video = test(self.pipe, cfg, self.whisper, self.audio2token,
                     self.audio2bucket, self.image_encoder, w, h, sample)

        if cfg.use_interframe:                       # RIFE interpolation
            out = video.to(self.device); frames=[]
            for i in tqdm(range(out.shape[2]-1), ncols=0):
                mid = self.rife.inference(out[:,:,i], out[:,:,i+1]).clamp(0,1)
                frames += [out[:,:,i], mid]
            frames.append(out[:,:,-1]); video = torch.stack(frames,2).cpu()

        tmp = out_path.replace(".mp4","_noaudio.mp4")
        save_videos_grid(video, tmp, n_rows=video.shape[0], fps=cfg.fps*(2 if cfg.use_interframe else 1))
        os.system(f"ffmpeg -i '{tmp}' -i '{wav_path}' -s {resolution} "
                  f"-vcodec libx264 -acodec aac -crf 18 -shortest '{out_path}' -y -loglevel error")
        os.remove(tmp); return 0