Merge pull request #86 from ModelTC/dev/wan_audio

  feature: audio driven video gen

configs/wan_i2v_audio.json

+{
+    "infer_steps": 5,
+    "target_fps": 16,
+    "video_duration": 12,
+    "audio_sr": 16000,
+    "target_video_length": 81,
+    "target_height": 480,
+    "target_width": 832,
+    "self_attn_1_type": "radial_attn",
+    "cross_attn_1_type": "flash_attn3",
+    "cross_attn_2_type": "flash_attn3",
+    "seed": 42,
+    "sample_guide_scale":1,
+    "sample_shift": 5,
+    "enable_cfg": false,
+    "cpu_offload": false
+}

lightx2v/attentions/__init__.py

@@ -2,6 +2,7 @@ from lightx2v.attentions.common.torch_sdpa import torch_sdpa
 from lightx2v.attentions.common.flash_attn2 import flash_attn2
 from lightx2v.attentions.common.flash_attn3 import flash_attn3
 from lightx2v.attentions.common.sage_attn2 import sage_attn2
+from lightx2v.attentions.common.radial_attn import radial_attn
 
 
 def attention(attention_type="flash_attn2", *args, **kwargs):
@@ -13,5 +14,7 @@ def attention(attention_type="flash_attn2", *args, **kwargs):
         return flash_attn3(*args, **kwargs)
     elif attention_type == "sage_attn2":
         return sage_attn2(*args, **kwargs)
+    elif attention_type == "radial_attn":
+        return radial_attn(*args, **kwargs)
     else:
         raise NotImplementedError(f"Unsupported attention mode: {attention_type}")

lightx2v/attentions/common/radial_attn.py

+import torch
+import flashinfer
+
+###
+###  Code from radial-attention
+###  https://github.com/mit-han-lab/ç/blob/main/radial_attn/attn_mask.py#L150
+###
+
+
+def radial_attn(
+    query, key, value, cu_seqlens_q=None, cu_seqlens_kv=None, max_seqlen_q=None, max_seqlen_kv=None, mask_map=None, sparsity_type="radial", block_size=128, decay_factor=1, model_cls="wan"
+):
+    orig_seqlen, num_head, hidden_dim = query.shape
+    query = pad_qkv(query, block_size=block_size)
+    key = pad_qkv(key, block_size=block_size)
+    value = pad_qkv(value, block_size=block_size)
+
+    mask = mask_map.queryLogMask(query, sparsity_type, block_size=block_size, decay_factor=decay_factor, model_type=model_cls) if mask_map else None
+    seqlen = query.shape[0]
+    workspace_buffer = torch.empty(128 * 1024 * 1024, device=query.device, dtype=torch.uint8)
+    bsr_wrapper = flashinfer.BlockSparseAttentionWrapper(
+        workspace_buffer,
+        backend="fa2",
+    )
+
+    indptr = get_indptr_from_mask(mask, query)
+    indices = get_indices_from_mask(mask, query)
+    bsr_wrapper.plan(
+        indptr=indptr,
+        indices=indices,
+        M=seqlen,
+        N=seqlen,
+        R=block_size,
+        C=block_size,
+        num_qo_heads=num_head,
+        num_kv_heads=num_head,
+        head_dim=hidden_dim,
+        q_data_type=query.dtype,
+        kv_data_type=key.dtype,
+        o_data_type=query.dtype,
+        use_fp16_qk_reduction=True,
+    )
+
+    o = bsr_wrapper.run(query, key, value)
+
+    return o[:orig_seqlen, :, :]
+
+
+def get_indptr_from_mask(mask, query):
+    # query shows the device of the indptr
+    # indptr (torch.Tensor) - the block index pointer of the block-sparse matrix on row dimension,
+    # shape `(MB + 1,)`, where `MB` is the number of blocks in the row dimension.
+    # The first element is always 0, and the last element is the number of blocks in the row dimension.
+    # The rest of the elements are the number of blocks in each row.
+    # the mask is already a block sparse mask
+    indptr = torch.zeros(mask.shape[0] + 1, device=query.device, dtype=torch.int32)
+    indptr[0] = 0
+    row_counts = mask.sum(dim=1).flatten()  # Ensure 1D output [num_blocks_row]
+    indptr[1:] = torch.cumsum(row_counts, dim=0)
+    return indptr
+
+
+def get_indices_from_mask(mask, query):
+    # indices (torch.Tensor) - the block indices of the block-sparse matrix on column dimension,
+    # shape `(nnz,),` where `nnz` is the number of non-zero blocks.
+    # The elements in `indices` array should be less than `NB`: the number of blocks in the column dimension.
+    nonzero_indices = torch.nonzero(mask)
+    indices = nonzero_indices[:, 1].to(dtype=torch.int32, device=query.device)
+    return indices
+
+
+def shrinkMaskStrict(mask, block_size=128):
+    seqlen = mask.shape[0]
+    block_num = seqlen // block_size
+    mask = mask[: block_num * block_size, : block_num * block_size].view(block_num, block_size, block_num, block_size)
+    col_densities = mask.sum(dim=1) / block_size
+    # we want the minimum non-zero column density in the block
+    non_zero_densities = col_densities > 0
+    high_density_cols = col_densities > 1 / 3
+    frac_high_density_cols = high_density_cols.sum(dim=-1) / (non_zero_densities.sum(dim=-1) + 1e-9)
+    block_mask = frac_high_density_cols > 0.6
+    block_mask[0:0] = True
+    block_mask[-1:-1] = True
+    return block_mask
+
+
+def pad_qkv(input_tensor, block_size=128):
+    """
+    Pad the input tensor to be a multiple of the block size.
+    input shape: (seqlen, num_heads, hidden_dim)
+    """
+    seqlen, num_heads, hidden_dim = input_tensor.shape
+    # Calculate the necessary padding
+    padding_length = (block_size - (seqlen % block_size)) % block_size
+    # Create a padded tensor with zeros
+    padded_tensor = torch.zeros((seqlen + padding_length, num_heads, hidden_dim), device=input_tensor.device, dtype=input_tensor.dtype)
+    # Copy the original tensor into the padded tensor
+    padded_tensor[:seqlen, :, :] = input_tensor
+
+    return padded_tensor
+
+
+def get_diagonal_split_mask(i, j, token_per_frame, sparse_type, query):
+    assert sparse_type in ["radial"]
+    dist = abs(i - j)
+    group = dist.bit_length()
+    threshold = 128  # hardcoded threshold for now, which is equal to block-size
+    decay_length = 2 ** token_per_frame.bit_length() / 2**group
+    if decay_length >= threshold:
+        return torch.ones((token_per_frame, token_per_frame), device=query.device, dtype=torch.bool)
+
+    split_factor = int(threshold / decay_length)
+    modular = dist % split_factor
+    if modular == 0:
+        return torch.ones((token_per_frame, token_per_frame), device=query.device, dtype=torch.bool)
+    else:
+        return torch.zeros((token_per_frame, token_per_frame), device=query.device, dtype=torch.bool)
+
+
+def get_window_width(i, j, token_per_frame, sparse_type, num_frame, decay_factor=1, block_size=128, model_type=None):
+    assert sparse_type in ["radial"]
+    dist = abs(i - j)
+    if model_type == "wan":
+        if dist < 1:
+            return token_per_frame
+        if dist == 1:
+            return token_per_frame // 2
+    elif model_type == "hunyuan":
+        if dist <= 1:
+            return token_per_frame
+    else:
+        raise ValueError(f"Unknown model type: {model_type}")
+    group = dist.bit_length()
+    decay_length = 2 ** token_per_frame.bit_length() / 2**group * decay_factor
+    threshold = block_size
+    if decay_length >= threshold:
+        return decay_length
+    else:
+        return threshold
+
+
+def gen_log_mask_shrinked(query, s, video_token_num, num_frame, block_size=128, sparse_type="log", decay_factor=0.5, model_type=None):
+    """
+    A more memory friendly version, we generate the attention mask of each frame pair at a time,
+    shrinks it, and stores it into the final result
+    """
+    final_log_mask = torch.zeros((s // block_size, s // block_size), device=query.device, dtype=torch.bool)
+    token_per_frame = video_token_num // num_frame
+    video_text_border = video_token_num // block_size
+
+    col_indices = torch.arange(0, token_per_frame, device=query.device).view(1, -1)
+    row_indices = torch.arange(0, token_per_frame, device=query.device).view(-1, 1)
+    final_log_mask[video_text_border:] = True
+    final_log_mask[:, video_text_border:] = True
+    for i in range(num_frame):
+        for j in range(num_frame):
+            local_mask = torch.zeros((token_per_frame, token_per_frame), device=query.device, dtype=torch.bool)
+            if j == 0:  # this is attention sink
+                local_mask = torch.ones((token_per_frame, token_per_frame), device=query.device, dtype=torch.bool)
+            else:
+                window_width = get_window_width(i, j, token_per_frame, sparse_type, num_frame, decay_factor=decay_factor, block_size=block_size, model_type=model_type)
+                local_mask = torch.abs(col_indices - row_indices) <= window_width
+                split_mask = get_diagonal_split_mask(i, j, token_per_frame, sparse_type, query)
+                local_mask = torch.logical_and(local_mask, split_mask)
+
+            remainder_row = (i * token_per_frame) % block_size
+            remainder_col = (j * token_per_frame) % block_size
+            # get the padded size
+            all_length_row = remainder_row + ((token_per_frame - 1) // block_size + 1) * block_size
+            all_length_col = remainder_col + ((token_per_frame - 1) // block_size + 1) * block_size
+            padded_local_mask = torch.zeros((all_length_row, all_length_col), device=query.device, dtype=torch.bool)
+            padded_local_mask[remainder_row : remainder_row + token_per_frame, remainder_col : remainder_col + token_per_frame] = local_mask
+            # shrink the mask
+            block_mask = shrinkMaskStrict(padded_local_mask, block_size=block_size)
+            # set the block mask to the final log mask
+            block_row_start = (i * token_per_frame) // block_size
+            block_col_start = (j * token_per_frame) // block_size
+            block_row_end = block_row_start + block_mask.shape[0]
+            block_col_end = block_col_start + block_mask.shape[1]
+            final_log_mask[block_row_start:block_row_end, block_col_start:block_col_end] = torch.logical_or(final_log_mask[block_row_start:block_row_end, block_col_start:block_col_end], block_mask)
+    print(f"mask sparsity: {1 - final_log_mask.sum() / final_log_mask.numel()}")
+    return final_log_mask
+
+
+class MaskMap:
+    def __init__(self, video_token_num=79200, num_frame=22):
+        self.video_token_num = video_token_num
+        self.num_frame = num_frame
+        self.log_mask = None
+
+    def queryLogMask(self, query, sparse_type, block_size=128, decay_factor=0.5, model_type=None):
+        log_mask = torch.ones((query.shape[0] // block_size, query.shape[0] // block_size), device=query.device, dtype=torch.bool)
+        if self.log_mask is None:
+            self.log_mask = gen_log_mask_shrinked(
+                query, query.shape[0], self.video_token_num, self.num_frame, sparse_type=sparse_type, decay_factor=decay_factor, model_type=model_type, block_size=block_size
+            )
+        block_bound = self.video_token_num // block_size
+        log_mask[:block_bound, :block_bound] = self.log_mask[:block_bound, :block_bound]
+        return log_mask

lightx2v/common/ops/attn/attn_weight.py

@@ -37,6 +37,9 @@ else:
         sageattn = None
 
 
+from lightx2v.attentions.common.radial_attn import radial_attn
+
+
 class AttnWeightTemplate(metaclass=ABCMeta):
     def __init__(self, weight_name):
         self.weight_name = weight_name
@@ -70,7 +73,7 @@ class FlashAttn2Weight(AttnWeightTemplate):
     def __init__(self):
         self.config = {}
 
-    def apply(self, q, k, v, cu_seqlens_q=None, cu_seqlens_kv=None, max_seqlen_q=None, max_seqlen_kv=None, model_cls=None):
+    def apply(self, q, k, v, cu_seqlens_q=None, cu_seqlens_kv=None, max_seqlen_q=None, max_seqlen_kv=None, model_cls=None, mask_map=None):
         x = flash_attn_varlen_func(
             q,
             k,
@@ -88,7 +91,7 @@ class FlashAttn3Weight(AttnWeightTemplate):
     def __init__(self):
         self.config = {}
 
-    def apply(self, q, k, v, cu_seqlens_q=None, cu_seqlens_kv=None, max_seqlen_q=None, max_seqlen_kv=None, model_cls=None):
+    def apply(self, q, k, v, cu_seqlens_q=None, cu_seqlens_kv=None, max_seqlen_q=None, max_seqlen_kv=None, model_cls=None, mask_map=None):
         x = flash_attn_varlen_func_v3(
             q,
             k,
@@ -101,6 +104,28 @@ class FlashAttn3Weight(AttnWeightTemplate):
         return x
 
 
+@ATTN_WEIGHT_REGISTER("radial_attn")
+class RadialAttnWeight(AttnWeightTemplate):
+    def __init__(self):
+        self.config = {}
+
+    def apply(self, q, k, v, cu_seqlens_q=None, cu_seqlens_kv=None, max_seqlen_q=None, max_seqlen_kv=None, mask_map=None, sparsity_type="radial", block_size=128, decay_factor=1, model_cls="wan"):
+        assert len(q.shape) == 3
+
+        x = radial_attn(
+            q,
+            k,
+            v,
+            mask_map=mask_map,
+            sparsity_type=sparsity_type,
+            block_size=block_size,
+            model_cls=model_cls[:3],  # Use first 3 characters to match "wan", "wan2", etc.
+            decay_factor=decay_factor,
+        )
+        x = x.view(max_seqlen_q, -1)
+        return x
+
+
 @ATTN_WEIGHT_REGISTER("sage_attn2")
 class SageAttn2Weight(AttnWeightTemplate):
     def __init__(self):

lightx2v/infer.py

@@ -14,6 +14,7 @@ from lightx2v.models.runners.hunyuan.hunyuan_runner import HunyuanRunner
 from lightx2v.models.runners.wan.wan_runner import WanRunner
 from lightx2v.models.runners.wan.wan_distill_runner import WanDistillRunner
 from lightx2v.models.runners.wan.wan_causvid_runner import WanCausVidRunner
+from lightx2v.models.runners.wan.wan_audio_runner import WanAudioRunner
 from lightx2v.models.runners.wan.wan_skyreels_v2_df_runner import WanSkyreelsV2DFRunner
 from lightx2v.models.runners.graph_runner import GraphRunner
 from lightx2v.models.runners.cogvideox.cogvidex_runner import CogvideoxRunner
@@ -41,14 +42,19 @@ def init_runner(config):
 
 async def main():
     parser = argparse.ArgumentParser()
-    parser.add_argument("--model_cls", type=str, required=True, choices=["wan2.1", "hunyuan", "wan2.1_distill", "wan2.1_causvid", "wan2.1_skyreels_v2_df", "cogvideox"], default="hunyuan")
+    parser.add_argument(
+        "--model_cls", type=str, required=True, choices=["wan2.1", "hunyuan", "wan2.1_distill", "wan2.1_causvid", "wan2.1_skyreels_v2_df", "cogvideox", "wan2.1_audio"], default="hunyuan"
+    )
     parser.add_argument("--task", type=str, choices=["t2v", "i2v"], default="t2v")
     parser.add_argument("--model_path", type=str, required=True)
     parser.add_argument("--config_json", type=str, required=True)
     parser.add_argument("--use_prompt_enhancer", action="store_true")
 
-    parser.add_argument("--prompt", type=str, required=True)
+    parser.add_argument("--prompt", type=str, default="", help="The input prompt for text-to-video generation")
     parser.add_argument("--negative_prompt", type=str, default="")
+    parser.add_argument("--lora_path", type=str, default="", help="The lora file path")
+    parser.add_argument("--prompt_path", type=str, default="", help="The path to input prompt file")
+    parser.add_argument("--audio_path", type=str, default="", help="The path to input audio file")
     parser.add_argument("--image_path", type=str, default="", help="The path to input image file or path for image-to-video (i2v) task")
     parser.add_argument("--save_video_path", type=str, default="./output_lightx2v.mp4", help="The path to save video path/file")
     args = parser.parse_args()

lightx2v/models/input_encoders/hf/xlm_roberta/model.py

@@ -11,6 +11,9 @@ import torchvision.transforms as T
 from lightx2v.attentions import attention
 from loguru import logger
 from lightx2v.models.input_encoders.hf.q_linear import QuantLinearInt8, QuantLinearFp8
+from einops import rearrange
+from torch import Tensor
+from transformers import CLIPVisionModel
 
 
 __all__ = [
@@ -428,3 +431,51 @@ class CLIPModel:
 
     def to_cpu(self):
         self.model = self.model.cpu()
+
+
+class WanVideoIPHandler:
+    def __init__(self, model_name, repo_or_path, require_grad=False, mode="eval", device="cuda", dtype=torch.float16):
+        # image_processor = CLIPImageProcessor.from_pretrained(
+        #     repo_or_path, subfolder='image_processor')
+        """720P-I2V-diffusers config is
+            "size": {
+                "shortest_edge": 224
+            }
+        and 480P-I2V-diffusers config is
+          "size": {
+            "height": 224,
+            "width": 224
+        }
+        but Wan2.1 official use no_crop resize by default
+        so I don't use CLIPImageProcessor
+        """
+        image_encoder = CLIPVisionModel.from_pretrained(repo_or_path, torch_dtype=dtype)
+        logger.info(f"Using image encoder {model_name} from {repo_or_path}")
+        image_encoder.requires_grad_(require_grad)
+        if mode == "eval":
+            image_encoder.eval()
+        else:
+            image_encoder.train()
+        self.dtype = dtype
+        self.device = device
+        self.image_encoder = image_encoder.to(device=device, dtype=dtype)
+        self.size = (224, 224)
+        mean = [0.48145466, 0.4578275, 0.40821073]
+        std = [0.26862954, 0.26130258, 0.27577711]
+        self.normalize = T.Normalize(mean=mean, std=std)
+        # self.image_processor = image_processor
+
+    def encode(
+        self,
+        img_tensor: Tensor,
+    ):
+        if img_tensor.ndim == 5:  # B C T H W
+            # img_tensor = img_tensor[:, :, 0]
+            img_tensor = rearrange(img_tensor, "B C 1 H W -> B C H W")
+        img_tensor = torch.clamp(img_tensor.float() * 0.5 + 0.5, min=0.0, max=1.0).to(self.device)
+        img_tensor = F.interpolate(img_tensor, size=self.size, mode="bicubic", align_corners=False)
+        img_tensor = self.normalize(img_tensor).to(self.dtype)
+
+        image_embeds = self.image_encoder(pixel_values=img_tensor, output_hidden_states=True)
+
+        return image_embeds.hidden_states[-1]

lightx2v/models/networks/wan/audio_adapter.py

+import flash_attn
+import math
+import torch
+import torch.nn as nn
+import torch.nn.functional as F
+from diffusers.models.embeddings import TimestepEmbedding, Timesteps
+from einops import rearrange
+from transformers import AutoModel
+from loguru import logger
+
+import os
+import safetensors
+from typing import List, Optional, Tuple, Union
+
+
+def load_safetensors(in_path: str):
+    if os.path.isdir(in_path):
+        return load_safetensors_from_dir(in_path)
+    elif os.path.isfile(in_path):
+        return load_safetensors_from_path(in_path)
+    else:
+        raise ValueError(f"{in_path} does not exist")
+
+
+def load_safetensors_from_path(in_path: str):
+    tensors = {}
+    with safetensors.safe_open(in_path, framework="pt", device="cpu") as f:
+        for key in f.keys():
+            tensors[key] = f.get_tensor(key)
+    return tensors
+
+
+def load_safetensors_from_dir(in_dir: str):
+    tensors = {}
+    safetensors = os.listdir(in_dir)
+    safetensors = [f for f in safetensors if f.endswith(".safetensors")]
+    for f in safetensors:
+        tensors.update(load_safetensors_from_path(os.path.join(in_dir, f)))
+    return tensors
+
+
+def load_pt_safetensors(in_path: str):
+    ext = os.path.splitext(in_path)[-1]
+    if ext in (".pt", ".pth", ".tar"):
+        state_dict = torch.load(in_path, map_location="cpu", weights_only=True)
+    else:
+        state_dict = load_safetensors(in_path)
+    return state_dict
+
+
+def rank0_load_state_dict_from_path(model, in_path: str, strict: bool = True):
+    import torch.distributed as dist
+
+    if (dist.is_initialized() and dist.get_rank() == 0) or (not dist.is_initialized()):
+        state_dict = load_pt_safetensors(in_path)
+        model.load_state_dict(state_dict, strict=strict)
+    if dist.is_initialized():
+        dist.barrier()
+    return model.to(dtype=torch.bfloat16, device="cuda")
+
+
+def linear_interpolation(features, output_len: int):
+    features = features.transpose(1, 2)
+    output_features = F.interpolate(features, size=output_len, align_corners=False, mode="linear")
+    return output_features.transpose(1, 2)
+
+
+def get_q_lens_audio_range(
+    batchsize,
+    n_tokens_per_rank,
+    n_query_tokens,
+    n_tokens_per_frame,
+    sp_rank,
+):
+    if n_query_tokens == 0:
+        q_lens = [1] * batchsize
+        return q_lens, 0, 1
+    idx0 = n_tokens_per_rank * sp_rank
+    first_length = idx0 - idx0 // n_tokens_per_frame * n_tokens_per_frame
+    n_frames = (n_query_tokens - first_length) // n_tokens_per_frame
+    last_length = n_query_tokens - n_frames * n_tokens_per_frame - first_length
+    q_lens = []
+    if first_length > 0:
+        q_lens.append(first_length)
+    q_lens += [n_tokens_per_frame] * n_frames
+    if last_length > 0:
+        q_lens.append(last_length)
+    t0 = idx0 // n_tokens_per_frame
+    idx1 = idx0 + n_query_tokens
+    t1 = math.ceil(idx1 / n_tokens_per_frame)
+    return q_lens * batchsize, t0, t1
+
+
+class PerceiverAttentionCA(nn.Module):
+    def __init__(self, dim_head=128, heads=16, kv_dim=2048, adaLN: bool = False):
+        super().__init__()
+        self.dim_head = dim_head
+        self.heads = heads
+        inner_dim = dim_head * heads
+        kv_dim = inner_dim if kv_dim is None else kv_dim
+        self.norm_kv = nn.LayerNorm(kv_dim)
+        self.norm_q = nn.LayerNorm(inner_dim, elementwise_affine=not adaLN)
+
+        self.to_q = nn.Linear(inner_dim, inner_dim)
+        self.to_kv = nn.Linear(kv_dim, inner_dim * 2)
+        self.to_out = nn.Linear(inner_dim, inner_dim)
+        if adaLN:
+            self.shift_scale_gate = nn.Parameter(torch.randn(1, 3, inner_dim) / inner_dim**0.5)
+        else:
+            shift_scale_gate = torch.zeros((1, 3, inner_dim))
+            shift_scale_gate[:, 2] = 1
+            self.register_buffer("shift_scale_gate", shift_scale_gate, persistent=False)
+
+    def forward(self, x, latents, t_emb, q_lens, k_lens):
+        """x shape (batchsize, latent_frame, audio_tokens_per_latent,
+        model_dim) latents (batchsize, length, model_dim)"""
+        batchsize = len(x)
+        x = self.norm_kv(x)
+        shift, scale, gate = (t_emb + self.shift_scale_gate).chunk(3, dim=1)
+        latents = self.norm_q(latents) * (1 + scale) + shift
+        q = self.to_q(latents)
+        k, v = self.to_kv(x).chunk(2, dim=-1)
+        q = rearrange(q, "B L (H C) -> (B L) H C", H=self.heads)
+        k = rearrange(k, "B T L (H C) -> (B T L) H C", H=self.heads)
+        v = rearrange(v, "B T L (H C) -> (B T L) H C", H=self.heads)
+        out = flash_attn.flash_attn_varlen_func(
+            q=q,
+            k=k,
+            v=v,
+            cu_seqlens_q=torch.cat([q_lens.new_zeros([1]), q_lens]).cumsum(0, dtype=torch.int32).to(q.device, non_blocking=True),
+            cu_seqlens_k=torch.cat([k_lens.new_zeros([1]), k_lens]).cumsum(0, dtype=torch.int32).to(q.device, non_blocking=True),
+            max_seqlen_q=q_lens.max(),
+            max_seqlen_k=k_lens.max(),
+            dropout_p=0.0,
+            softmax_scale=None,
+            causal=False,
+            window_size=(-1, -1),
+            deterministic=False,
+        )
+        out = rearrange(out, "(B L) H C -> B L (H C)", B=batchsize)
+        return self.to_out(out) * gate
+
+
+class AudioProjection(nn.Module):
+    def __init__(
+        self,
+        audio_feature_dim: int = 768,
+        n_neighbors: tuple = (2, 2),
+        num_tokens: int = 32,
+        mlp_dims: tuple = (1024, 1024, 32 * 768),
+        transformer_layers: int = 4,
+    ):
+        super().__init__()
+        mlp = []
+        self.left, self.right = n_neighbors
+        self.audio_frames = sum(n_neighbors) + 1
+        in_dim = audio_feature_dim * self.audio_frames
+        for i, out_dim in enumerate(mlp_dims):
+            mlp.append(nn.Linear(in_dim, out_dim))
+            if i != len(mlp_dims) - 1:
+                mlp.append(nn.ReLU())
+            in_dim = out_dim
+        self.mlp = nn.Sequential(*mlp)
+        self.norm = nn.LayerNorm(mlp_dims[-1] // num_tokens)
+        self.num_tokens = num_tokens
+        if transformer_layers > 0:
+            decoder_layer = nn.TransformerDecoderLayer(d_model=audio_feature_dim, nhead=audio_feature_dim // 64, dim_feedforward=4 * audio_feature_dim, dropout=0.0, batch_first=True)
+            self.transformer_decoder = nn.TransformerDecoder(
+                decoder_layer,
+                num_layers=transformer_layers,
+            )
+        else:
+            self.transformer_decoder = None
+
+    def forward(self, audio_feature, latent_frame):
+        video_frame = (latent_frame - 1) * 4 + 1
+        audio_feature_ori = audio_feature
+        audio_feature = linear_interpolation(audio_feature_ori, video_frame)
+        if self.transformer_decoder is not None:
+            audio_feature = self.transformer_decoder(audio_feature, audio_feature_ori)
+        audio_feature = F.pad(audio_feature, pad=(0, 0, self.left, self.right), mode="replicate")
+        audio_feature = audio_feature.unfold(dimension=1, size=self.audio_frames, step=1)
+        audio_feature = rearrange(audio_feature, "B T C W -> B T (W C)")
+        audio_feature = self.mlp(audio_feature)  # (B, video_frame, C)
+        audio_feature = rearrange(audio_feature, "B T (N C) -> B T N C", N=self.num_tokens)  # (B, video_frame, num_tokens, C)
+        return self.norm(audio_feature)
+
+
+class TimeEmbedding(nn.Module):
+    def __init__(self, dim, time_freq_dim, time_proj_dim):
+        super().__init__()
+        self.timesteps_proj = Timesteps(num_channels=time_freq_dim, flip_sin_to_cos=True, downscale_freq_shift=0)
+        self.time_embedder = TimestepEmbedding(in_channels=time_freq_dim, time_embed_dim=dim)
+        self.act_fn = nn.SiLU()
+        self.time_proj = nn.Linear(dim, time_proj_dim)
+
+    def forward(
+        self,
+        timestep: torch.Tensor,
+    ):
+        timestep = self.timesteps_proj(timestep)
+        time_embedder_dtype = next(iter(self.time_embedder.parameters())).dtype
+        timestep = timestep.to(time_embedder_dtype)
+        temb = self.time_embedder(timestep)
+        timestep_proj = self.time_proj(self.act_fn(temb))
+        return timestep_proj
+
+
+class AudioAdapter(nn.Module):
+    def __init__(
+        self,
+        attention_head_dim=64,
+        num_attention_heads=40,
+        base_num_layers=30,
+        interval=1,
+        audio_feature_dim: int = 768,
+        num_tokens: int = 32,
+        mlp_dims: tuple = (1024, 1024, 32 * 768),
+        time_freq_dim: int = 256,
+        projection_transformer_layers: int = 4,
+    ):
+        super().__init__()
+        self.audio_proj = AudioProjection(
+            audio_feature_dim=audio_feature_dim,
+            n_neighbors=(2, 2),
+            num_tokens=num_tokens,
+            mlp_dims=mlp_dims,
+            transformer_layers=projection_transformer_layers,
+        )
+        # self.num_tokens = num_tokens * 4
+        self.num_tokens_x4 = num_tokens * 4
+        self.audio_pe = nn.Parameter(torch.randn(self.num_tokens_x4, mlp_dims[-1] // num_tokens) * 0.02)
+        ca_num = math.ceil(base_num_layers / interval)
+        self.base_num_layers = base_num_layers
+        self.interval = interval
+        self.ca = nn.ModuleList(
+            [
+                PerceiverAttentionCA(
+                    dim_head=attention_head_dim,
+                    heads=num_attention_heads,
+                    kv_dim=mlp_dims[-1] // num_tokens,
+                    adaLN=time_freq_dim > 0,
+                )
+                for _ in range(ca_num)
+            ]
+        )
+        self.dim = attention_head_dim * num_attention_heads
+        if time_freq_dim > 0:
+            self.time_embedding = TimeEmbedding(self.dim, time_freq_dim, self.dim * 3)
+        else:
+            self.time_embedding = None
+
+    def rearange_audio_features(self, audio_feature: torch.Tensor):
+        # audio_feature (B, video_frame, num_tokens, C)
+        audio_feature_0 = audio_feature[:, :1]
+        audio_feature_0 = torch.repeat_interleave(audio_feature_0, repeats=4, dim=1)
+        audio_feature = torch.cat([audio_feature_0, audio_feature[:, 1:]], dim=1)  # (B, 4 * latent_frame, num_tokens, C)
+        audio_feature = rearrange(audio_feature, "B (T S) N C -> B T (S N) C", S=4)
+        return audio_feature
+
+    def forward(self, audio_feat: torch.Tensor, timestep: torch.Tensor, latent_frame: int, weight: float = 1.0):
+        def modify_hidden_states(hidden_states, grid_sizes, ca_block: PerceiverAttentionCA, x, t_emb, dtype, weight):
+            """thw specify the latent_frame, latent_height, latenf_width after
+            hidden_states is patchified.
+
+            latent_frame does not include the reference images so that the
+            audios and hidden_states are strictly aligned
+            """
+            if len(hidden_states.shape) == 2:  # 扩展batchsize dim
+                hidden_states = hidden_states.unsqueeze(0)  # bs = 1
+            # print(weight)
+            t, h, w = grid_sizes[0].tolist()
+            n_tokens = t * h * w
+            ori_dtype = hidden_states.dtype
+            device = hidden_states.device
+            bs, n_tokens_per_rank = hidden_states.shape[:2]
+            tail_length = n_tokens_per_rank - n_tokens
+
+            n_query_tokens = n_tokens_per_rank - tail_length % n_tokens_per_rank
+
+            if n_query_tokens > 0:
+                hidden_states_aligned = hidden_states[:, :n_query_tokens]
+                hidden_states_tail = hidden_states[:, n_query_tokens:]
+            else:
+                # for ranks that should be excluded from cross-attn, fake cross-attn will be applied so that FSDP works.
+                hidden_states_aligned = hidden_states[:, :1]
+                hidden_states_tail = hidden_states[:, 1:]
+
+            q_lens, t0, t1 = get_q_lens_audio_range(batchsize=bs, n_tokens_per_rank=n_tokens_per_rank, n_query_tokens=n_query_tokens, n_tokens_per_frame=h * w, sp_rank=0)
+            q_lens = torch.tensor(q_lens, device=device, dtype=torch.int32)
+            """
+            processing audio features in sp_state can be moved outside.
+            """
+            x = x[:, t0:t1]
+            x = x.to(dtype)
+            k_lens = torch.tensor([self.num_tokens_x4] * (t1 - t0) * bs, device=device, dtype=torch.int32)
+            assert q_lens.shape == k_lens.shape
+            # ca_block:CrossAttention函数
+            residual = ca_block(x, hidden_states_aligned, t_emb, q_lens, k_lens) * weight
+            residual = residual.to(ori_dtype)  # audio做了CrossAttention之后以Residual的方式注入
+            if n_query_tokens == 0:
+                residual = residual * 0.0
+            hidden_states = torch.cat([hidden_states_aligned + residual, hidden_states_tail], dim=1)
+
+            if len(hidden_states.shape) == 3:  #
+                hidden_states = hidden_states.squeeze(0)  # bs = 1
+            return hidden_states
+
+        x = self.audio_proj(audio_feat, latent_frame)
+        x = self.rearange_audio_features(x)
+        x = x + self.audio_pe
+        if self.time_embedding is not None:
+            t_emb = self.time_embedding(timestep).unflatten(1, (3, -1))
+        else:
+            t_emb = torch.zeros((len(x), 3, self.dim), device=x.device, dtype=x.dtype)
+        ret_dict = {}
+        for block_idx, base_idx in enumerate(range(0, self.base_num_layers, self.interval)):
+            block_dict = {
+                "kwargs": {
+                    "ca_block": self.ca[block_idx],
+                    "x": x,
+                    "weight": weight,
+                    "t_emb": t_emb,
+                    "dtype": x.dtype,
+                },
+                "modify_func": modify_hidden_states,
+            }
+            ret_dict[base_idx] = block_dict
+        return ret_dict
+
+    @classmethod
+    def from_transformer(
+        cls,
+        transformer,
+        audio_feature_dim: int = 1024,
+        interval: int = 1,
+        time_freq_dim: int = 256,
+        projection_transformer_layers: int = 4,
+    ):
+        num_attention_heads = transformer.config["num_heads"]
+        base_num_layers = transformer.config["num_layers"]
+        attention_head_dim = transformer.config["dim"] // num_attention_heads
+
+        audio_adapter = AudioAdapter(
+            attention_head_dim,
+            num_attention_heads,
+            base_num_layers,
+            interval=interval,
+            audio_feature_dim=audio_feature_dim,
+            time_freq_dim=time_freq_dim,
+            projection_transformer_layers=projection_transformer_layers,
+            mlp_dims=(1024, 1024, 32 * audio_feature_dim),
+        )
+        return audio_adapter
+
+    def get_fsdp_wrap_module_list(
+        self,
+    ):
+        ret_list = list(self.ca)
+        return ret_list
+
+    def enable_gradient_checkpointing(
+        self,
+    ):
+        pass
+
+
+class AudioAdapterPipe:
+    def __init__(
+        self, audio_adapter: AudioAdapter, audio_encoder_repo: str = "microsoft/wavlm-base-plus", dtype=torch.float32, device="cuda", generator=None, tgt_fps: int = 15, weight: float = 1.0
+    ) -> None:
+        self.audio_adapter = audio_adapter
+        self.dtype = dtype
+        self.device = device
+        self.generator = generator
+        self.audio_encoder_dtype = torch.float16
+        ##音频编码器
+        self.audio_encoder = AutoModel.from_pretrained(audio_encoder_repo)
+
+        self.audio_encoder.eval()
+        self.audio_encoder.to(device, self.audio_encoder_dtype)
+        self.tgt_fps = tgt_fps
+        self.weight = weight
+        if "base" in audio_encoder_repo:
+            self.audio_feature_dim = 768
+        else:
+            self.audio_feature_dim = 1024
+
+    def update_model(self, audio_adapter):
+        self.audio_adapter = audio_adapter
+
+    def __call__(self, audio_input_feat, timestep, latent_shape: tuple, dropout_cond: callable = None):
+        # audio_input_feat is from AudioPreprocessor
+        latent_frame = latent_shape[2]
+        if len(audio_input_feat.shape) == 1:  # 扩展batchsize = 1
+            audio_input_feat = audio_input_feat.unsqueeze(0)
+            latent_frame = latent_shape[1]
+
+        video_frame = (latent_frame - 1) * 4 + 1
+        audio_length = int(50 / self.tgt_fps * video_frame)
+
+        with torch.no_grad():
+            audio_input_feat = audio_input_feat.to(self.device, self.audio_encoder_dtype)
+            try:
+                audio_feat = self.audio_encoder(audio_input_feat, return_dict=True).last_hidden_state
+            except Exception as err:
+                audio_feat = torch.rand(1, audio_length, self.audio_feature_dim).to(self.device)
+                print(err)
+            audio_feat = audio_feat.to(self.dtype)
+            if dropout_cond is not None:
+                audio_feat = dropout_cond(audio_feat)
+
+        return self.audio_adapter(audio_feat=audio_feat, timestep=timestep, latent_frame=latent_frame, weight=self.weight)

lightx2v/models/networks/wan/audio_model.py

+import os
+import torch
+import time
+import glob
+from lightx2v.models.networks.wan.model import WanModel
+from lightx2v.models.networks.wan.weights.pre_weights import WanPreWeights
+from lightx2v.models.networks.wan.weights.post_weights import WanPostWeights
+from lightx2v.models.networks.wan.weights.transformer_weights import (
+    WanTransformerWeights,
+)
+from lightx2v.models.networks.wan.infer.pre_infer import WanPreInfer
+from lightx2v.models.networks.wan.infer.post_infer import WanPostInfer
+
+from lightx2v.models.networks.wan.infer.pre_infer import WanPreInfer
+from lightx2v.models.networks.wan.infer.pre_wan_audio_infer import WanAudioPreInfer
+from lightx2v.models.networks.wan.infer.post_wan_audio_infer import WanAudioPostInfer
+from lightx2v.models.networks.wan.infer.feature_caching.transformer_infer import WanTransformerInferTeaCaching
+from safetensors import safe_open
+import lightx2v.attentions.distributed.ulysses.wrap as ulysses_dist_wrap
+import lightx2v.attentions.distributed.ring.wrap as ring_dist_wrap
+
+from lightx2v.attentions.common.radial_attn import MaskMap
+
+from lightx2v.models.networks.wan.infer.transformer_infer import (
+    WanTransformerInfer,
+)
+from lightx2v.models.networks.wan.infer.feature_caching.transformer_infer import (
+    WanTransformerInferTeaCaching,
+)
+
+
+class WanAudioModel(WanModel):
+    pre_weight_class = WanPreWeights
+    post_weight_class = WanPostWeights
+    transformer_weight_class = WanTransformerWeights
+
+    def __init__(self, model_path, config, device):
+        super().__init__(model_path, config, device)
+
+    def _init_infer_class(self):
+        self.pre_infer_class = WanAudioPreInfer
+        self.post_infer_class = WanAudioPostInfer
+        if self.config["feature_caching"] == "NoCaching":
+            self.transformer_infer_class = WanTransformerInfer
+        elif self.config["feature_caching"] == "Tea":
+            self.transformer_infer_class = WanTransformerInferTeaCaching
+        else:
+            raise NotImplementedError(f"Unsupported feature_caching type: {self.config['feature_caching']}")
+
+    @torch.no_grad()
+    def infer(self, inputs):
+        if self.config["cpu_offload"]:
+            self.pre_weight.to_cuda()
+            self.post_weight.to_cuda()
+
+        if self.transformer_infer.mask_map is None:
+            _, c, h, w = self.scheduler.latents.shape
+            num_frame = c + 1  # for r2v
+            video_token_num = num_frame * (h // 2) * (w // 2)
+            from loguru import logger
+
+            logger.info(f"video_token_num: {video_token_num}, num_frame: {num_frame}")
+            self.transformer_infer.mask_map = MaskMap(video_token_num, num_frame)
+
+        embed, grid_sizes, pre_infer_out, valid_patch_length = self.pre_infer.infer(self.pre_weight, inputs, positive=True)
+        x = self.transformer_infer.infer(self.transformer_weights, grid_sizes, embed, *pre_infer_out)
+        noise_pred_cond = self.post_infer.infer(self.post_weight, x, embed, grid_sizes, valid_patch_length)[0]
+
+        if self.config["feature_caching"] == "Tea":
+            self.scheduler.cnt += 1
+            if self.scheduler.cnt >= self.scheduler.num_steps:
+                self.scheduler.cnt = 0
+        self.scheduler.noise_pred = noise_pred_cond
+
+        if self.config["enable_cfg"]:
+            embed, grid_sizes, pre_infer_out, valid_patch_length = self.pre_infer.infer(self.pre_weight, inputs, positive=False)
+            x = self.transformer_infer.infer(self.transformer_weights, grid_sizes, embed, *pre_infer_out)
+            noise_pred_uncond = self.post_infer.infer(self.post_weight, x, embed, grid_sizes, valid_patch_length)[0]
+
+            if self.config["feature_caching"] == "Tea":
+                self.scheduler.cnt += 1
+                if self.scheduler.cnt >= self.scheduler.num_steps:
+                    self.scheduler.cnt = 0
+
+            self.scheduler.noise_pred = noise_pred_uncond + self.config.sample_guide_scale * (noise_pred_cond - noise_pred_uncond)
+
+            if self.config["cpu_offload"]:
+                self.pre_weight.to_cpu()
+                self.post_weight.to_cpu()

lightx2v/models/networks/wan/infer/post_wan_audio_infer.py

+import math
+import torch
+import torch.cuda.amp as amp
+from loguru import logger
+from lightx2v.models.networks.wan.infer.post_infer import WanPostInfer
+
+
+class WanAudioPostInfer(WanPostInfer):
+    def __init__(self, config):
+        self.out_dim = config["out_dim"]
+        self.patch_size = (1, 2, 2)
+
+    def set_scheduler(self, scheduler):
+        self.scheduler = scheduler
+
+    def infer(self, weights, x, e, grid_sizes, valid_patch_length):
+        if e.dim() == 2:
+            modulation = weights.head_modulation.tensor  # 1, 2, dim
+            e = (modulation + e.unsqueeze(1)).chunk(2, dim=1)
+        elif e.dim() == 3:  # For Diffustion forcing
+            modulation = weights.head_modulation.tensor.unsqueeze(2)  # 1, 2, seq, dim
+            e = (modulation + e.unsqueeze(1)).chunk(2, dim=1)
+            e = [ei.squeeze(1) for ei in e]
+
+        norm_out = torch.nn.functional.layer_norm(x, (x.shape[1],), None, None, 1e-6).type_as(x)
+        out = norm_out * (1 + e[1].squeeze(0)) + e[0].squeeze(0)
+        x = weights.head.apply(out)
+
+        x = x[:, :valid_patch_length]
+
+        x = self.unpatchify(x, grid_sizes)
+        return [u.float() for u in x]
+
+    def unpatchify(self, x, grid_sizes):
+        x = x.unsqueeze(0)
+        c = self.out_dim
+        out = []
+        for u, v in zip(x, grid_sizes.tolist()):
+            u = u[: math.prod(v)].view(*v, *self.patch_size, c)
+            u = torch.einsum("fhwpqrc->cfphqwr", u)
+            u = u.reshape(c, *[i * j for i, j in zip(v, self.patch_size)])
+            out.append(u)
+        return out

lightx2v/models/networks/wan/infer/pre_wan_audio_infer.py

+import torch
+import math
+from .utils import rope_params, sinusoidal_embedding_1d
+from lightx2v.models.networks.wan.infer.pre_infer import WanPreInfer
+from loguru import logger
+
+
+class WanAudioPreInfer(WanPreInfer):
+    def __init__(self, config):
+        assert (config["dim"] % config["num_heads"]) == 0 and (config["dim"] // config["num_heads"]) % 2 == 0
+        d = config["dim"] // config["num_heads"]
+
+        self.task = config["task"]
+        self.freqs = torch.cat(
+            [
+                rope_params(1024, d - 4 * (d // 6)),
+                rope_params(1024, 2 * (d // 6)),
+                rope_params(1024, 2 * (d // 6)),
+            ],
+            dim=1,
+        ).cuda()
+        self.freq_dim = config["freq_dim"]
+        self.dim = config["dim"]
+        self.text_len = config["text_len"]
+
+    def infer(self, weights, inputs, positive):
+        ltnt_channel = self.scheduler.latents.size(0)
+
+        prev_latents = inputs["previmg_encoder_output"]["prev_latents"].unsqueeze(0)
+        prev_mask = inputs["previmg_encoder_output"]["prev_mask"]
+
+        hidden_states = self.scheduler.latents.unsqueeze(0)
+        hidden_states = torch.cat([hidden_states[:, :ltnt_channel], prev_latents, prev_mask], dim=1)
+        hidden_states = hidden_states.squeeze(0)
+
+        x = [hidden_states]
+        t = torch.stack([self.scheduler.timesteps[self.scheduler.step_index]])
+
+        audio_dit_blocks = []
+        audio_encoder_output = inputs["audio_encoder_output"]
+        audio_model_input = {
+            "audio_input_feat": audio_encoder_output.to(hidden_states.device),
+            "latent_shape": hidden_states.shape,
+            "timestep": t,
+        }
+        audio_dit_blocks.append(inputs["audio_adapter_pipe"](**audio_model_input))
+
+        if positive:
+            context = inputs["text_encoder_output"]["context"]
+        else:
+            context = inputs["text_encoder_output"]["context_null"]
+        seq_len = self.scheduler.seq_len
+
+        clip_fea = inputs["image_encoder_output"]["clip_encoder_out"]
+        ref_image_encoder = inputs["image_encoder_output"]["vae_encode_out"]
+        batch_size = len(x)
+        num_channels, num_frames, height, width = x[0].shape
+        _, ref_num_channels, ref_num_frames, _, _ = ref_image_encoder.shape
+
+        if ref_num_channels != num_channels:
+            zero_padding = torch.zeros(
+                (batch_size, num_channels - ref_num_channels, ref_num_frames, height, width),
+                dtype=self.scheduler.latents.dtype,
+                device=self.scheduler.latents.device,
+            )
+            ref_image_encoder = torch.concat([ref_image_encoder, zero_padding], dim=1)
+        y = list(torch.unbind(ref_image_encoder, dim=0))  # 第一个batch维度变成list
+        # embeddings
+        x = [weights.patch_embedding.apply(u.unsqueeze(0)) for u in x]
+        x_grid_sizes = torch.stack([torch.tensor(u.shape[2:], dtype=torch.long) for u in x])
+        x = [u.flatten(2).transpose(1, 2) for u in x]
+        seq_lens = torch.tensor([u.size(1) for u in x], dtype=torch.long).cuda()
+        assert seq_lens.max() <= seq_len
+        x = torch.cat([torch.cat([u, u.new_zeros(1, seq_len - u.size(1), u.size(2))], dim=1) for u in x])
+        valid_patch_length = x[0].size(0)
+        y = [weights.patch_embedding.apply(u.unsqueeze(0)) for u in y]
+        y_grid_sizes = torch.stack([torch.tensor(u.shape[2:], dtype=torch.long) for u in y])
+        y = [u.flatten(2).transpose(1, 2).squeeze(0) for u in y]
+
+        x = [torch.cat([a, b], dim=0) for a, b in zip(x, y)]
+        x = torch.stack(x, dim=0)
+
+        embed = sinusoidal_embedding_1d(self.freq_dim, t.flatten())
+        embed = weights.time_embedding_0.apply(embed)
+        embed = torch.nn.functional.silu(embed)
+        embed = weights.time_embedding_2.apply(embed)
+        embed0 = torch.nn.functional.silu(embed)
+
+        embed0 = weights.time_projection_1.apply(embed0).unflatten(1, (6, self.dim))
+
+        # text embeddings
+        stacked = torch.stack([torch.cat([u, u.new_zeros(self.text_len - u.size(0), u.size(1))]) for u in context])
+        out = weights.text_embedding_0.apply(stacked.squeeze(0))
+        out = torch.nn.functional.gelu(out, approximate="tanh")
+        context = weights.text_embedding_2.apply(out)
+
+        if self.task == "i2v":
+            context_clip = weights.proj_0.apply(clip_fea)
+            context_clip = weights.proj_1.apply(context_clip)
+            context_clip = torch.nn.functional.gelu(context_clip, approximate="none")
+            context_clip = weights.proj_3.apply(context_clip)
+            context_clip = weights.proj_4.apply(context_clip)
+
+            context = torch.concat([context_clip, context], dim=0)
+
+        return (embed, x_grid_sizes, (x.squeeze(0), embed0.squeeze(0), seq_lens, self.freqs, context, audio_dit_blocks), valid_patch_length)

lightx2v/models/networks/wan/infer/transformer_infer.py

 import torch
-from .utils import compute_freqs, compute_freqs_dist, apply_rotary_emb, apply_rotary_emb_chunk
+from .utils import compute_freqs, compute_freqs_dist, compute_freqs_audio, compute_freqs_audio_dist, apply_rotary_emb, apply_rotary_emb_chunk
 from lightx2v.common.offload.manager import (
     WeightAsyncStreamManager,
     LazyWeightAsyncStreamManager,
 )
 from lightx2v.common.transformer_infer.transformer_infer import BaseTransformerInfer
 from lightx2v.utils.envs import *
+from loguru import logger
+import os
 
 
 class WanTransformerInfer(BaseTransformerInfer):
@@ -21,6 +23,8 @@ class WanTransformerInfer(BaseTransformerInfer):
         self.parallel_attention = None
         self.apply_rotary_emb_func = apply_rotary_emb_chunk if config.get("rotary_chunk", False) else apply_rotary_emb
         self.clean_cuda_cache = self.config.get("clean_cuda_cache", False)
+        self.mask_map = None
+
         if self.config["cpu_offload"]:
             if "offload_ratio" in self.config:
                 offload_ratio = self.config["offload_ratio"]
@@ -64,10 +68,10 @@ class WanTransformerInfer(BaseTransformerInfer):
         return cu_seqlens_q, cu_seqlens_k
 
     @torch.compile(disable=not CHECK_ENABLE_GRAPH_MODE())
-    def infer(self, weights, grid_sizes, embed, x, embed0, seq_lens, freqs, context):
-        return self.infer_func(weights, grid_sizes, embed, x, embed0, seq_lens, freqs, context)
+    def infer(self, weights, grid_sizes, embed, x, embed0, seq_lens, freqs, context, audio_dit_blocks=None):
+        return self.infer_func(weights, grid_sizes, embed, x, embed0, seq_lens, freqs, context, audio_dit_blocks)
 
-    def _infer_with_offload(self, weights, grid_sizes, embed, x, embed0, seq_lens, freqs, context):
+    def _infer_with_offload(self, weights, grid_sizes, embed, x, embed0, seq_lens, freqs, context, audio_dit_blocks=None):
         for block_idx in range(self.blocks_num):
             if block_idx == 0:
                 self.weights_stream_mgr.active_weights[0] = weights.blocks[0]
@@ -92,7 +96,7 @@ class WanTransformerInfer(BaseTransformerInfer):
 
         return x
 
-    def _infer_with_phases_offload(self, weights, grid_sizes, embed, x, embed0, seq_lens, freqs, context):
+    def _infer_with_phases_offload(self, weights, grid_sizes, embed, x, embed0, seq_lens, freqs, context, audio_dit_blocks=None):
         for block_idx in range(weights.blocks_num):
             for phase_idx in range(self.phases_num):
                 if block_idx == 0 and phase_idx == 0:
@@ -133,7 +137,7 @@ class WanTransformerInfer(BaseTransformerInfer):
 
         return x
 
-    def _infer_with_phases_lazy_offload(self, weights, grid_sizes, embed, x, embed0, seq_lens, freqs, context):
+    def _infer_with_phases_lazy_offload(self, weights, grid_sizes, embed, x, embed0, seq_lens, freqs, context, audio_dit_blocks=None):
         self.weights_stream_mgr.prefetch_weights_from_disk(weights)
 
         for block_idx in range(weights.blocks_num):
@@ -194,7 +198,22 @@ class WanTransformerInfer(BaseTransformerInfer):
 
         return x
 
-    def _infer_without_offload(self, weights, grid_sizes, embed, x, embed0, seq_lens, freqs, context):
+    def zero_temporal_component_in_3DRoPE(self, valid_token_length, rotary_emb=None):
+        if rotary_emb is None:
+            return None
+        self.use_real = False
+        rope_t_dim = 44
+        if self.use_real:
+            freqs_cos, freqs_sin = rotary_emb
+            freqs_cos[valid_token_length:, :, :rope_t_dim] = 0
+            freqs_sin[valid_token_length:, :, :rope_t_dim] = 0
+            return freqs_cos, freqs_sin
+        else:
+            freqs_cis = rotary_emb
+            freqs_cis[valid_token_length:, :, : rope_t_dim // 2] = 0
+            return freqs_cis
+
+    def _infer_without_offload(self, weights, grid_sizes, embed, x, embed0, seq_lens, freqs, context, audio_dit_blocks=None):
         for block_idx in range(self.blocks_num):
             x = self.infer_block(
                 weights.blocks[block_idx],
@@ -206,6 +225,12 @@ class WanTransformerInfer(BaseTransformerInfer):
                 freqs,
                 context,
             )
+
+            if audio_dit_blocks is not None and len(audio_dit_blocks) > 0:
+                for ipa_out in audio_dit_blocks:
+                    if block_idx in ipa_out:
+                        cur_modify = ipa_out[block_idx]
+                        x = cur_modify["modify_func"](x, grid_sizes, **cur_modify["kwargs"])
         return x
 
     def infer_block(self, weights, grid_sizes, embed, x, embed0, seq_lens, freqs, context):
@@ -265,14 +290,23 @@ class WanTransformerInfer(BaseTransformerInfer):
         v = weights.self_attn_v.apply(norm1_out).view(s, n, d)
 
         if not self.parallel_attention:
-            freqs_i = compute_freqs(q.size(2) // 2, grid_sizes, freqs)
+            if self.config.get("audio_sr", False):
+                freqs_i = compute_freqs_audio(q.size(2) // 2, grid_sizes, freqs)
+            else:
+                freqs_i = compute_freqs(q.size(2) // 2, grid_sizes, freqs)
         else:
-            freqs_i = compute_freqs_dist(q.size(0), q.size(2) // 2, grid_sizes, freqs)
+            if self.config.get("audio_sr", False):
+                freqs_i = compute_freqs_audio_dist(q.size(0), q.size(2) // 2, grid_sizes, freqs)
+            else:
+                freqs_i = compute_freqs_dist(q.size(2) // 2, grid_sizes, freqs)
+
+        freqs_i = self.zero_temporal_component_in_3DRoPE(seq_lens, freqs_i)
 
         q = self.apply_rotary_emb_func(q, freqs_i)
         k = self.apply_rotary_emb_func(k, freqs_i)
 
-        cu_seqlens_q, cu_seqlens_k = self._calculate_q_k_len(q, k_lens=seq_lens)
+        k_lens = torch.empty_like(seq_lens).fill_(freqs_i.size(0))
+        cu_seqlens_q, cu_seqlens_k = self._calculate_q_k_len(q, k_lens=k_lens)
 
         if self.clean_cuda_cache:
             del freqs_i, norm1_out, norm1_weight, norm1_bias
@@ -288,6 +322,7 @@ class WanTransformerInfer(BaseTransformerInfer):
                 max_seqlen_q=q.size(0),
                 max_seqlen_kv=k.size(0),
                 model_cls=self.config["model_cls"],
+                mask_map=self.mask_map,
             )
         else:
             attn_out = self.parallel_attention(
@@ -353,7 +388,6 @@ class WanTransformerInfer(BaseTransformerInfer):
                 q,
                 k_lens=torch.tensor([k_img.size(0)], dtype=torch.int32, device=k.device),
             )
-
             img_attn_out = weights.cross_attn_2.apply(
                 q=q,
                 k=k_img,

lightx2v/models/networks/wan/infer/utils.py

 import torch
 import torch.distributed as dist
+from loguru import logger
 from lightx2v.utils.envs import *
 
 
@@ -19,6 +20,45 @@ def compute_freqs(c, grid_sizes, freqs):
     return freqs_i
 
 
+def compute_freqs_audio(c, grid_sizes, freqs):
+    freqs = freqs.split([c - 2 * (c // 3), c // 3, c // 3], dim=1)
+    f, h, w = grid_sizes[0].tolist()
+    f = f + 1  ##for r2v add 1 channel
+    seq_len = f * h * w
+    freqs_i = torch.cat(
+        [
+            freqs[0][:f].view(f, 1, 1, -1).expand(f, h, w, -1),
+            freqs[1][:h].view(1, h, 1, -1).expand(f, h, w, -1),
+            freqs[2][:w].view(1, 1, w, -1).expand(f, h, w, -1),
+        ],
+        dim=-1,
+    ).reshape(seq_len, 1, -1)
+
+    return freqs_i
+
+
+def compute_freqs_audio_dist(s, c, grid_sizes, freqs):
+    world_size = dist.get_world_size()
+    cur_rank = dist.get_rank()
+    freqs = freqs.split([c - 2 * (c // 3), c // 3, c // 3], dim=1)
+    f, h, w = grid_sizes[0].tolist()
+    f = f + 1
+    seq_len = f * h * w
+    freqs_i = torch.cat(
+        [
+            freqs[0][:f].view(f, 1, 1, -1).expand(f, h, w, -1),
+            freqs[1][:h].view(1, h, 1, -1).expand(f, h, w, -1),
+            freqs[2][:w].view(1, 1, w, -1).expand(f, h, w, -1),
+        ],
+        dim=-1,
+    ).reshape(seq_len, 1, -1)
+
+    freqs_i = pad_freqs(freqs_i, s * world_size)
+    s_per_rank = s
+    freqs_i_rank = freqs_i[(cur_rank * s_per_rank) : ((cur_rank + 1) * s_per_rank), :, :]
+    return freqs_i_rank
+
+
 def compute_freqs_causvid(c, grid_sizes, freqs, start_frame=0):
     freqs = freqs.split([c - 2 * (c // 3), c // 3, c // 3], dim=1)
     f, h, w = grid_sizes[0].tolist()

lightx2v/models/networks/wan/lora_adapter.py

@@ -50,6 +50,7 @@ class WanLoraWrapper:
         self.model._init_weights(weight_dict)
 
         logger.info(f"Applied LoRA: {lora_name} with alpha={alpha}")
+        del lora_weights  # 删除节约显存
         return True
 
     @torch.no_grad()
@@ -84,7 +85,8 @@ class WanLoraWrapper:
             if name in lora_pairs:
                 if name not in self.override_dict:
                     self.override_dict[name] = param.clone().cpu()
-
+                # import pdb
+                # pdb.set_trace()
                 name_lora_A, name_lora_B = lora_pairs[name]
                 lora_A = lora_weights[name_lora_A].to(param.device, param.dtype)
                 lora_B = lora_weights[name_lora_B].to(param.device, param.dtype)

lightx2v/models/networks/wan/weights/transformer_weights.py

@@ -184,7 +184,7 @@ class WanSelfAttention(WeightModule):
             sparge_ckpt = torch.load(self.config["sparge_ckpt"])
             self.self_attn_1.load(sparge_ckpt)
         else:
-            self.add_module("self_attn_1", ATTN_WEIGHT_REGISTER[self.config["attention_type"]]())
+            self.add_module("self_attn_1", ATTN_WEIGHT_REGISTER[self.config["self_attn_1_type"]]())
         if self.quant_method in ["smoothquant", "awq"]:
             self.add_module(
                 "smooth_norm1_weight",
@@ -275,7 +275,7 @@ class WanCrossAttention(WeightModule):
                 self.lazy_load_file,
             ),
         )
-        self.add_module("cross_attn_1", ATTN_WEIGHT_REGISTER[self.config["attention_type"]]())
+        self.add_module("cross_attn_1", ATTN_WEIGHT_REGISTER[self.config["cross_attn_1_type"]]())
 
         if self.config.task == "i2v":
             self.add_module(
@@ -304,7 +304,7 @@ class WanCrossAttention(WeightModule):
                     self.lazy_load_file,
                 ),
             )
-            self.add_module("cross_attn_2", ATTN_WEIGHT_REGISTER[self.config["attention_type"]]())
+            self.add_module("cross_attn_2", ATTN_WEIGHT_REGISTER[self.config["cross_attn_2_type"]]())
 
 
 class WanFFN(WeightModule):

lightx2v/models/runners/wan/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 lightx2v.utils.registry_factory import RUNNER_REGISTER
+from lightx2v.models.runners.wan.wan_runner import WanRunner
+from lightx2v.models.runners.default_runner import DefaultRunner
+from lightx2v.models.schedulers.wan.scheduler import WanScheduler
+from lightx2v.models.networks.wan.model import WanModel
+from lightx2v.utils.profiler import ProfilingContext4Debug, ProfilingContext
+from lightx2v.models.input_encoders.hf.t5.model import T5EncoderModel
+from lightx2v.models.input_encoders.hf.xlm_roberta.model import CLIPModel, WanVideoIPHandler
+from lightx2v.models.networks.wan.audio_model import WanAudioModel
+from lightx2v.models.networks.wan.lora_adapter import WanLoraWrapper
+from lightx2v.models.video_encoders.hf.wan.vae import WanVAE
+
+from lightx2v.models.networks.wan.audio_adapter import AudioAdapter, AudioAdapterPipe, rank0_load_state_dict_from_path
+
+from loguru import logger
+import torch.distributed as dist
+from einops import rearrange
+import torchaudio as ta
+from transformers import AutoFeatureExtractor
+
+from torchvision.datasets.folder import IMG_EXTENSIONS
+from torchvision.transforms import InterpolationMode
+from torchvision.transforms.functional import resize
+
+import subprocess
+import warnings
+from typing import Optional, Tuple, Union
+
+
+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, size, 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]
+    target_h, target_w = 480, 832
+    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
+
+
+def array_to_video(
+    image_array: np.ndarray,
+    output_path: str,
+    fps: Union[int, float] = 30,
+    resolution: Optional[Union[Tuple[int, int], Tuple[float, float]]] = None,
+    disable_log: bool = False,
+    lossless: bool = True,
+) -> None:
+    if not isinstance(image_array, np.ndarray):
+        raise TypeError("Input should be np.ndarray.")
+    assert image_array.ndim == 4
+    assert image_array.shape[-1] == 3
+    if resolution:
+        height, width = resolution
+        width += width % 2
+        height += height % 2
+    else:
+        image_array = pad_for_libx264(image_array)
+        height, width = image_array.shape[1], image_array.shape[2]
+    if lossless:
+        command = [
+            "/usr/bin/ffmpeg",
+            "-y",  # (optional) overwrite output file if it exists
+            "-f",
+            "rawvideo",
+            "-s",
+            f"{int(width)}x{int(height)}",  # size of one frame
+            "-pix_fmt",
+            "bgr24",
+            "-r",
+            f"{fps}",  # frames per second
+            "-loglevel",
+            "error",
+            "-threads",
+            "4",
+            "-i",
+            "-",  # The input comes from a pipe
+            "-vcodec",
+            "libx264rgb",
+            "-crf",
+            "0",
+            "-an",  # Tells FFMPEG not to expect any audio
+            output_path,
+        ]
+    else:
+        command = [
+            "/usr/bin/ffmpeg",
+            "-y",  # (optional) overwrite output file if it exists
+            "-f",
+            "rawvideo",
+            "-s",
+            f"{int(width)}x{int(height)}",  # size of one frame
+            "-pix_fmt",
+            "bgr24",
+            "-r",
+            f"{fps}",  # frames per second
+            "-loglevel",
+            "error",
+            "-threads",
+            "4",
+            "-i",
+            "-",  # The input comes from a pipe
+            "-vcodec",
+            "libx264",
+            "-an",  # Tells FFMPEG not to expect any audio
+            output_path,
+        ]
+
+    if not disable_log:
+        print(f'Running "{" ".join(command)}"')
+    process = subprocess.Popen(
+        command,
+        stdin=subprocess.PIPE,
+        stderr=subprocess.PIPE,
+    )
+    if process.stdin is None or process.stderr is None:
+        raise BrokenPipeError("No buffer received.")
+    index = 0
+    while True:
+        if index >= image_array.shape[0]:
+            break
+        process.stdin.write(image_array[index].tobytes())
+        index += 1
+    process.stdin.close()
+    process.stderr.close()
+    process.wait()
+
+
+def pad_for_libx264(image_array):
+    if image_array.ndim == 2 or (image_array.ndim == 3 and image_array.shape[2] == 3):
+        hei_index = 0
+        wid_index = 1
+    elif image_array.ndim == 4 or (image_array.ndim == 3 and image_array.shape[2] != 3):
+        hei_index = 1
+        wid_index = 2
+    else:
+        return image_array
+    hei_pad = image_array.shape[hei_index] % 2
+    wid_pad = image_array.shape[wid_index] % 2
+    if hei_pad + wid_pad > 0:
+        pad_width = []
+        for dim_index in range(image_array.ndim):
+            if dim_index == hei_index:
+                pad_width.append((0, hei_pad))
+            elif dim_index == wid_index:
+                pad_width.append((0, wid_pad))
+            else:
+                pad_width.append((0, 0))
+        values = 0
+        image_array = np.pad(image_array, pad_width, mode="constant", constant_values=values)
+    return image_array
+
+
+def generate_unique_path(path):
+    if not os.path.exists(path):
+        return path
+    root, ext = os.path.splitext(path)
+    index = 1
+    new_path = f"{root}-{index}{ext}"
+    while os.path.exists(new_path):
+        index += 1
+        new_path = f"{root}-{index}{ext}"
+    return new_path
+
+
+def save_to_video(gen_lvideo, out_path, target_fps):
+    gen_lvideo = rearrange(gen_lvideo, "B C T H W -> B T H W C")
+    gen_lvideo = (gen_lvideo[0].cpu().numpy() * 127.5 + 127.5).astype(np.uint8)
+    gen_lvideo = gen_lvideo[..., ::-1].copy()
+    generate_unique_path(out_path)
+    array_to_video(gen_lvideo, output_path=out_path, fps=target_fps, lossless=False)
+
+
+def save_audio(
+    audio_array,
+    audio_name: str,
+    video_name: str = None,
+    sr: int = 16000,
+):
+    logger.info(f"Saving audio to {audio_name} type: {type(audio_array)}")
+
+    ta.save(
+        audio_name,
+        torch.tensor(audio_array[None]),
+        sample_rate=sr,
+    )
+
+    out_video = f"{video_name[:-4]}_with_audio.mp4"
+    # 确保父目录存在
+    parent_dir = os.path.dirname(out_video)
+    if parent_dir and not os.path.exists(parent_dir):
+        os.makedirs(parent_dir, exist_ok=True)
+
+    # 如果输出视频已存在，先删除
+    if os.path.exists(out_video):
+        os.remove(out_video)
+
+    cmd = f"/usr/bin/ffmpeg -i {video_name} -i {audio_name} {out_video}"
+    subprocess.call(cmd, shell=True)
+
+
+@RUNNER_REGISTER("wan2.1_audio")
+class WanAudioRunner(WanRunner):
+    def __init__(self, config):
+        super().__init__(config)
+
+    def load_audio_models(self):
+        ##音频特征提取器
+        self.audio_preprocess = AutoFeatureExtractor.from_pretrained(self.config["model_path"], subfolder="audio_encoder")
+
+        ##音频驱动视频生成adapter
+        audio_adapter_path = self.config["model_path"] + "/audio_adapter.safetensors"
+        audio_adaper = 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_adaper, audio_adapter_path, strict=False)
+
+        ##音频特征编码器
+        device = self.model.device
+        audio_encoder_repo = self.config["model_path"] + "/audio_encoder"
+        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 audio_adapter_pipe
+
+    def load_transformer(self):
+        base_model = WanAudioModel(self.config.model_path, self.config, self.init_device)
+
+        if self.config.lora_path:
+            assert not self.config.get("dit_quantized", False) or self.config.mm_config.get("weight_auto_quant", False)
+            lora_wrapper = WanLoraWrapper(base_model)
+            lora_name = lora_wrapper.load_lora(self.config.lora_path)
+            lora_wrapper.apply_lora(lora_name, self.config.strength_model)
+            logger.info(f"Loaded LoRA: {lora_name}")
+
+        return base_model
+
+    def load_image_encoder(self):
+        clip_model_dir = self.config["model_path"] + "/image_encoder"
+        image_encoder = WanVideoIPHandler("CLIPModel", repo_or_path=clip_model_dir, require_grad=False, mode="eval", device=self.init_device, dtype=torch.float16)
+
+        return image_encoder
+
+    def run_image_encoder(self, config, vae_model):
+        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]
+
+        # resize and crop image
+        cond_frms, tgt_h, tgt_w = adaptive_resize(ref_img)
+        config.tgt_h = tgt_h
+        config.tgt_w = tgt_w
+        clip_encoder_out = self.image_encoder.encode(cond_frms).squeeze(0).to(torch.bfloat16)
+
+        cond_frms = rearrange(cond_frms, "1 C H W -> 1 C 1 H W")
+        lat_h, lat_w = tgt_h // 8, tgt_w // 8
+        config.lat_h = lat_h
+        config.lat_w = lat_w
+        vae_encode_out = vae_model.encode(cond_frms.to(torch.float), config)
+        if isinstance(vae_encode_out, list):  #
+            # list转tensor
+            vae_encode_out = torch.stack(vae_encode_out, dim=0).to(torch.bfloat16)
+
+        return vae_encode_out, clip_encoder_out
+
+    def run_input_encoder_internal(self):
+        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,
+                }
+                logger.info(f"clip_encoder_out:{clip_encoder_out.shape} vae_encode_out:{vae_encode_out.shape}")
+        with ProfilingContext("Run Text Encoder"):
+            with open(self.config["prompt_path"], "r", encoding="utf-8") as f:
+                prompt = f.readline().strip()
+                logger.info(f"Prompt: {prompt}")
+                img = Image.open(self.config["image_path"]).convert("RGB")
+                text_encoder_output = self.run_text_encoder(prompt, img)
+
+        self.set_target_shape()
+        self.inputs = {"text_encoder_output": text_encoder_output, "image_encoder_output": image_encoder_output}
+
+        del self.image_encoder  # 删除ref的clip模型，只使用一次
+        gc.collect()
+        torch.cuda.empty_cache()
+
+    def set_target_shape(self):
+        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 1 == 0, error_msg
+
+        ret["target_shape"] = self.config.target_shape
+        return ret
+
+    def run(self):
+        def load_audio(in_path: str, sr: float = 16000):
+            audio_array, ori_sr = ta.load(in_path)
+            audio_array = ta.functional.resample(audio_array.mean(0), orig_freq=ori_sr, new_freq=sr)
+            return audio_array.numpy()
+
+        def get_audio_range(start_frame: int, end_frame: int, fps: float, audio_sr: float = 16000):
+            audio_frame_rate = audio_sr / fps
+            return round(start_frame * audio_frame_rate), round((end_frame + 1) * audio_frame_rate)
+
+        self.inputs["audio_adapter_pipe"] = self.load_audio_models()
+
+        # process audio
+        audio_sr = self.config.get("audio_sr", 16000)
+        max_num_frames = self.config.get("target_video_length", 81)  # wan2.1一段最多81帧，5秒，16fps
+        target_fps = self.config.get("target_fps", 16)  # 音视频同步帧率
+        video_duration = self.config.get("video_duration", 5)  # 期望视频输出时长
+        audio_array = load_audio(self.config["audio_path"], sr=audio_sr)
+        audio_len = int(audio_array.shape[0] / audio_sr * target_fps)
+        prev_frame_length = 5
+        prev_token_length = (prev_frame_length - 1) // 4 + 1
+        max_num_audio_length = int((max_num_frames + 1) / target_fps * audio_sr)
+
+        interval_num = 1
+        # expected_frames
+        expected_frames = min(max(1, int(float(video_duration) * target_fps)), audio_len)
+        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
+
+        audio_start, audio_end = get_audio_range(0, expected_frames, fps=target_fps, audio_sr=audio_sr)
+        audio_array_ori = audio_array[audio_start:audio_end]
+
+        gen_video_list = []
+        cut_audio_list = []
+        # reference latents
+
+        tgt_h = self.config.tgt_h
+        tgt_w = self.config.tgt_w
+        device = self.model.scheduler.latents.device
+        dtype = torch.bfloat16
+        vae_dtype = torch.float
+
+        for idx in range(interval_num):
+            torch.manual_seed(42 + idx)
+            logger.info(f"###  manual_seed: {42 + idx} ####")
+            useful_length = -1
+            if idx == 0:  # 第一段 Condition padding0
+                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
+                audio_start, audio_end = get_audio_range(0, max_num_frames, fps=target_fps, audio_sr=audio_sr)
+                audio_array = audio_array_ori[audio_start:audio_end]
+                if expected_frames < max_num_frames:
+                    useful_length = audio_array.shape[0]
+                    audio_array = np.concatenate((audio_array, np.zeros(max_num_audio_length)[: max_num_audio_length - useful_length]), axis=0)
+                audio_input_feat = self.audio_preprocess(audio_array, sampling_rate=audio_sr, return_tensors="pt").input_values.squeeze(0)
+
+            elif res_frame_num > 5 and idx == interval_num - 1:  # 最后一段可能不够81帧
+                prev_frames = torch.zeros((1, 3, max_num_frames, tgt_h, tgt_w), device=device)
+                prev_frames[:, :, :prev_frame_length] = gen_video_list[-1][:, :, -prev_frame_length:]
+                prev_latents = self.vae_encoder.encode(prev_frames.to(vae_dtype), self.config)[0].to(dtype)
+                prev_len = prev_token_length
+                audio_start, audio_end = get_audio_range(idx * max_num_frames - idx * prev_frame_length, expected_frames, fps=target_fps, audio_sr=audio_sr)
+                audio_array = audio_array_ori[audio_start:audio_end]
+                useful_length = audio_array.shape[0]
+                audio_array = np.concatenate((audio_array, np.zeros(max_num_audio_length)[: max_num_audio_length - useful_length]), axis=0)
+                audio_input_feat = self.audio_preprocess(audio_array, sampling_rate=audio_sr, return_tensors="pt").input_values.squeeze(0)
+
+            else:  # 中间段满81帧带pre_latens
+                prev_frames = torch.zeros((1, 3, max_num_frames, tgt_h, tgt_w), device=device)
+                prev_frames[:, :, :prev_frame_length] = gen_video_list[-1][:, :, -prev_frame_length:]
+                prev_latents = self.vae_encoder.encode(prev_frames.to(vae_dtype), self.config)[0].to(dtype)
+                prev_len = prev_token_length
+                audio_start, audio_end = get_audio_range(idx * max_num_frames - idx * prev_frame_length, (idx + 1) * max_num_frames - idx * prev_frame_length, fps=target_fps, audio_sr=audio_sr)
+                audio_array = audio_array_ori[audio_start:audio_end]
+                audio_input_feat = self.audio_preprocess(audio_array, sampling_rate=audio_sr, return_tensors="pt").input_values.squeeze(0)
+
+            self.inputs["audio_encoder_output"] = audio_input_feat.to(device)
+
+            if idx != 0:
+                self.model.scheduler.reset()
+
+            if prev_latents is not None:
+                ltnt_channel, nframe, height, width = self.model.scheduler.latents.shape
+                bs = 1
+                prev_mask = torch.zeros((bs, 1, nframe, height, width), device=device, dtype=dtype)
+                if prev_len > 0:
+                    prev_mask[:, :, :prev_len] = 1.0
+
+                previmg_encoder_output = {
+                    "prev_latents": prev_latents,
+                    "prev_mask": prev_mask,
+                }
+                self.inputs["previmg_encoder_output"] = previmg_encoder_output
+
+            for step_index in range(self.model.scheduler.infer_steps):
+                logger.info(f"==> step_index: {step_index} / {self.model.scheduler.infer_steps}")
+
+                with ProfilingContext4Debug("step_pre"):
+                    self.model.scheduler.step_pre(step_index=step_index)
+
+                with ProfilingContext4Debug("infer"):
+                    self.model.infer(self.inputs)
+
+                with ProfilingContext4Debug("step_post"):
+                    self.model.scheduler.step_post()
+
+            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)
+            start_frame = 0 if idx == 0 else prev_frame_length
+            start_audio_frame = 0 if idx == 0 else int((prev_frame_length + 1) * audio_sr / target_fps)
+
+            if res_frame_num > 5 and idx == interval_num - 1:
+                gen_video_list.append(gen_video[:, :, start_frame:res_frame_num])
+                cut_audio_list.append(audio_array[start_audio_frame:useful_length])
+            elif expected_frames < max_num_frames and useful_length != -1:
+                gen_video_list.append(gen_video[:, :, start_frame:expected_frames])
+                cut_audio_list.append(audio_array[start_audio_frame:useful_length])
+            else:
+                gen_video_list.append(gen_video[:, :, start_frame:])
+                cut_audio_list.append(audio_array[start_audio_frame:])
+
+        gen_lvideo = torch.cat(gen_video_list, dim=2).float()
+        merge_audio = np.concatenate(cut_audio_list, axis=0).astype(np.float32)
+        out_path = self.config.save_video_path
+        audio_file = os.path.join("./", "audio_merge.wav")
+        save_to_video(gen_lvideo, out_path, target_fps)
+        save_audio(merge_audio, audio_file, out_path)
+        os.remove(out_path)
+        os.remove(audio_file)
+
+    async def run_pipeline(self):
+        if self.config["use_prompt_enhancer"]:
+            self.config["prompt_enhanced"] = self.post_prompt_enhancer()
+
+        self.run_input_encoder_internal()
+        self.set_target_shape()
+
+        self.init_scheduler()
+        self.model.scheduler.prepare(self.inputs["image_encoder_output"])
+        self.run()
+        self.end_run()
+
+        gc.collect()
+        torch.cuda.empty_cache()

lightx2v/models/schedulers/wan/scheduler.py

 import math
 import numpy as np
 import torch
+import gc
 from typing import List, Optional, Tuple, Union
 from lightx2v.models.schedulers.scheduler import BaseScheduler
 
@@ -115,6 +116,17 @@ class WanScheduler(BaseScheduler):
         x0_pred = sample - sigma_t * model_output
         return x0_pred
 
+    def reset(self):
+        self.model_outputs = [None] * self.solver_order
+        self.timestep_list = [None] * self.solver_order
+        self.last_sample = None
+        self.noise_pred = None
+        self.this_order = None
+        self.lower_order_nums = 0
+        self.prepare_latents(self.config.target_shape, dtype=torch.float32)
+        gc.collect()
+        torch.cuda.empty_cache()
+
     def multistep_uni_p_bh_update(
         self,
         model_output: torch.Tensor,

scripts/run_wan_i2v_audio.sh

+#!/bin/bash
+
+# set path and first
+lightx2v_path="/mnt/Text2Video/wangshankun/lightx2v"
+model_path="/mnt/Text2Video/wangshankun/HF_Cache/Wan2.1-I2V-Audio-14B-720P/"
+lora_path="/mnt/Text2Video/wangshankun/HF_Cache/Wan21_T2V_14B_lightx2v_cfg_step_distill_lora_rank32.safetensors"
+# check section
+if [ -z "${CUDA_VISIBLE_DEVICES}" ]; then
+    cuda_devices=0
+    echo "Warn: CUDA_VISIBLE_DEVICES is not set, using default value: ${cuda_devices}, change at shell script or set env variable."
+    export CUDA_VISIBLE_DEVICES=${cuda_devices}
+fi
+
+if [ -z "${lightx2v_path}" ]; then
+    echo "Error: lightx2v_path is not set. Please set this variable first."
+    exit 1
+fi
+
+if [ -z "${model_path}" ]; then
+    echo "Error: model_path is not set. Please set this variable first."
+    exit 1
+fi
+
+export TOKENIZERS_PARALLELISM=false
+
+export PYTHONPATH=${lightx2v_path}:$PYTHONPATH
+
+export ENABLE_PROFILING_DEBUG=true
+export ENABLE_GRAPH_MODE=false
+export DTYPE=BF16
+
+python -m lightx2v.infer \
+--model_cls wan2.1_audio \
+--task i2v \
+--model_path $model_path \
+--config_json ${lightx2v_path}/configs/wan_i2v_audio.json \
+--prompt_path ${lightx2v_path}/assets/inputs/audio/15.txt \
+--negative_prompt 色调艳丽，过曝，静态，细节模糊不清，字幕，风格，作品，画作，画面，静止，整体发灰，最差质量，低质量，JPEG压缩残留，丑陋的，残缺的，多余的手指，画得不好的手部，画得不好的脸部，畸形的，毁容的，形态畸形的肢体，手指融合，静止不动的画面，杂乱的背景，三条腿，背景人很多，倒着走 \
+--image_path ${lightx2v_path}/assets/inputs/audio/15.png \
+--audio_path ${lightx2v_path}/assets/inputs/audio/15.wav \
+--save_video_path ${lightx2v_path}/save_results/output_lightx2v_wan_i2v_audio.mp4 \
+--lora_path ${lora_path}