multitalk and more

This commit is contained in:
DeepBeepMeep 2025-07-08 18:41:19 +02:00
parent eb811e0c52
commit 4908c3c243
25 changed files with 2670 additions and 1266 deletions

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@ -20,6 +20,24 @@ WanGP supports the Wan (and derived models), Hunyuan Video and LTV Video models
**Follow DeepBeepMeep on Twitter/X to get the Latest News**: https://x.com/deepbeepmeep
## 🔥 Latest Updates
### July 6 2025: WanGP v6.6, WanGP offers you **Vace Multitalk Dual Voices Fusionix Infinite** :
**Vace** our beloved super Control Net has been combined with **Multitalk** the new king in town that can animate two people speaking (**Dual Voices**). It is accelerated by the **Fusionix** model and thanks to *Sliding Windows* support and *Adaptive Projected Guidance* (much slower but should reduce the reddish effect with long videos) your two people will be able to talk an for very a long time (which **Infinite** amount of time in the field of video generation).
Of course you will get as well plain *Multitalk* vanilla and also *Multitalk 720p* as a bonus.
And since I am mister nice guy I had enclosed as an exclusivity an *Audio Separator* that will save you time to isolate each voice when using Multitalk with two people.
As I feel like a resting a bit I haven't produced a nice sample Video to illustrate all these new capabilities. But here is the thing, I ams sure you will publish in the *Share Your Best Video* channel your Master Pieces. The best one will be added to the *Announcements Channel* and will bring eternal fame to its author.
But wait, there is more:
- Sliding Windows support has been added anywhere with Wan models, so imagine now with text2video upgraded in 6.5 into a video2video, you can upsample very long videos regardless of your VRAM. The good old image2video model can now reuse the last image to produce new videos (as requested by many of you)
- I have added also the capability to transfer the audio of the original control video and an option to preserve the fps into the generated video, so from now on you will be to upsample / restore your old families video and keep the audio and the original pace. Be aware that the duration will be limited 1000 frames as I still need to add streaming support for unlimited video sizes.
Also, of interest too:
- Extract video info from Videos that have not been generated by WanGP, even better you can also apply post processing (Upsampling / MMAudio) on non WanGP videos
- Force the generated video fps to your liking, works wery well with Vace when using a Control Video
- Ability to chain URLs of Finetune models (for instance put the URLs of a model in your main finetune and reference this finetune in other finetune models to save time)
### July 2 2025: WanGP v6.5.1, WanGP takes care of you: lots of quality of life features:
- View directly inside WanGP the properties (seed, resolutions, length, most settings...) of the past generations
- In one click use the newly generated video as a Control Video or Source Video to be continued

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@ -10,5 +10,6 @@
"num_heads": 40,
"num_layers": 40,
"out_dim": 16,
"text_len": 512
"text_len": 512,
"flf": true
}

15
configs/multitalk.json Normal file
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@ -0,0 +1,15 @@
{
"_class_name": "WanModel",
"_diffusers_version": "0.30.0",
"dim": 5120,
"eps": 1e-06,
"ffn_dim": 13824,
"freq_dim": 256,
"in_dim": 36,
"model_type": "i2v",
"num_heads": 40,
"num_layers": 40,
"out_dim": 16,
"text_len": 512,
"multitalk_output_dim": 768
}

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@ -0,0 +1,17 @@
{
"_class_name": "VaceWanModel",
"_diffusers_version": "0.30.0",
"dim": 5120,
"eps": 1e-06,
"ffn_dim": 13824,
"freq_dim": 256,
"in_dim": 16,
"model_type": "t2v",
"num_heads": 40,
"num_layers": 40,
"out_dim": 16,
"text_len": 512,
"vace_layers": [0, 5, 10, 15, 20, 25, 30, 35],
"vace_in_dim": 96,
"multitalk_output_dim": 768
}

12
finetunes/fantasy.json Normal file
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@ -0,0 +1,12 @@
{
"model":
{
"name": "Fantasy Talking 720p",
"architecture" : "fantasy",
"modules": ["fantasy"],
"description": "The Fantasy Talking model corresponds to the original Wan image 2 video model combined with the Fantasy Speaking module to process an audio Input.",
"URLs": "i2v_720p",
"teacache_coefficients" : [-114.36346466, 65.26524496, -18.82220707, 4.91518089, -0.23412683]
},
"resolution": "1280x720"
}

16
finetunes/flf2v_720p.json Normal file
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@ -0,0 +1,16 @@
{
"model":
{
"name": "First Last Frame to Video 720p (FLF2V)14B",
"architecture" : "flf2v_720p",
"visible" : false,
"description": "The First Last Frame 2 Video model is the official model Image 2 Video model that supports Start and End frames.",
"URLs": [
"https://huggingface.co/DeepBeepMeep/Wan2.1/resolve/main/wan2.1_FLF2V_720p_14B_bf16.safetensors",
"https://huggingface.co/DeepBeepMeep/Wan2.1/resolve/main/wan2.1_FLF2V_720p_14B_quanto_int8.safetensors",
"https://huggingface.co/DeepBeepMeep/Wan2.1/resolve/main/wan2.1_FLF2V_720p_14B_quanto_fp16_int8.safetensors"
],
"auto_quantize": true
},
"resolution": "1280x720"
}

16
finetunes/moviigen.json Normal file
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@ -0,0 +1,16 @@
{
"model":
{
"name": "MoviiGen 1080p 14B",
"architecture" : "t2v",
"description": "MoviiGen 1.1, a cutting-edge video generation model that excels in cinematic aesthetics and visual quality. Use it to generate videos in 720p or 1080p in the 21:9 ratio.",
"URLs": [
"https://huggingface.co/DeepBeepMeep/Wan2.1/resolve/main/wan2.1_moviigen1.1_14B_mbf16.safetensors",
"https://huggingface.co/DeepBeepMeep/Wan2.1/resolve/main/wan2.1_moviigen1.1_14B_quanto_mbf16_int8.safetensors",
"https://huggingface.co/DeepBeepMeep/Wan2.1/resolve/main/wan2.1_moviigen1.1_14B_quanto_mfp16_int8.safetensors"
],
"auto_quantize": true
},
"resolution": "1280x720",
"video_length": 81
}

11
finetunes/multitalk.json Normal file
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@ -0,0 +1,11 @@
{
"model":
{
"name": "Multitalk 480p",
"architecture" : "multitalk",
"modules": ["multitalk"],
"description": "The Multitalk model corresponds to the original Wan image 2 video model combined with the Multitalk module. It lets you have up to two people have a conversation.",
"URLs": "i2v",
"teacache_coefficients" : [-3.02331670e+02, 2.23948934e+02, -5.25463970e+01, 5.87348440e+00, -2.01973289e-01]
}
}

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@ -0,0 +1,13 @@
{
"model":
{
"name": "Multitalk 720p",
"architecture" : "multitalk",
"modules": ["multitalk"],
"description": "The Multitalk model corresponds to the original Wan image 2 video 720p model combined with the Multitalk module. It lets you have up to two people have a conversation.",
"URLs": "i2v_720p",
"teacache_coefficients" : [-114.36346466, 65.26524496, -18.82220707, 4.91518089, -0.23412683],
"auto_quantize": true
},
"resolution": "1280x720"
}

11
finetunes/vace_14B.json Normal file
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@ -0,0 +1,11 @@
{
"model": {
"name": "Vace ControlNet 14B",
"architecture": "vace_14B",
"modules": [
"vace_14B"
],
"description": "The Vace ControlNet model is a powerful model that allows you to control the content of the generated video based of additional custom data : pose or depth video, images or objects you want to see in the video.",
"URLs": "t2v"
}
}

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@ -6,12 +6,7 @@
"vace_14B"
],
"description": "Vace control model enhanced using multiple open-source components and LoRAs to boost motion realism, temporal consistency, and expressive detail.",
"URLs": [
"https://huggingface.co/DeepBeepMeep/Wan2.1/resolve/main/Wan14BT2VFusioniX_fp16.safetensors",
"https://huggingface.co/DeepBeepMeep/Wan2.1/resolve/main/Wan14BT2VFusioniX_quanto_bf16_int8.safetensors",
"https://huggingface.co/DeepBeepMeep/Wan2.1/resolve/main/Wan14BT2VFusioniX_quanto_fp16_int8.safetensors"
],
"auto_quantize": true
"URLs": "t2v_fusionix"
},
"negative_prompt": "",
"prompt": "",

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@ -0,0 +1,41 @@
{
"model": {
"name": "Vace Multitalk FusioniX 14B",
"architecture": "vace_multitalk_14B",
"modules": [
"vace_14B",
"multitalk"
],
"description": "Vace control model enhanced using multiple open-source components and LoRAs to boost motion realism, temporal consistency, and expressive detail. And it that's not sufficient Vace is combined with Multitalk.",
"URLs": [
"https://huggingface.co/DeepBeepMeep/Wan2.1/resolve/main/Wan14BT2VFusioniX_fp16.safetensors",
"https://huggingface.co/DeepBeepMeep/Wan2.1/resolve/main/Wan14BT2VFusioniX_quanto_bf16_int8.safetensors",
"https://huggingface.co/DeepBeepMeep/Wan2.1/resolve/main/Wan14BT2VFusioniX_quanto_fp16_int8.safetensors"
],
"auto_quantize": true
},
"negative_prompt": "",
"prompt": "",
"resolution": "832x480",
"video_length": 81,
"seed": -1,
"num_inference_steps": 10,
"guidance_scale": 1,
"flow_shift": 5,
"embedded_guidance_scale": 6,
"repeat_generation": 1,
"multi_images_gen_type": 0,
"tea_cache_setting": 0,
"tea_cache_start_step_perc": 0,
"loras_multipliers": "",
"temporal_upsampling": "",
"spatial_upsampling": "",
"RIFLEx_setting": 0,
"slg_switch": 0,
"slg_start_perc": 10,
"slg_end_perc": 90,
"cfg_star_switch": 0,
"cfg_zero_step": -1,
"prompt_enhancer": "",
"activated_loras": []
}

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@ -131,23 +131,13 @@ from pathlib import Path
import torch
def remux_with_audio(video_path: Path, output_path: Path, audio: torch.Tensor, sampling_rate: int):
"""Remux video with new audio using FFmpeg."""
from wan.utils.utils import extract_audio_tracks, combine_video_with_audio_tracks, cleanup_temp_audio_files
with tempfile.NamedTemporaryFile(suffix='.wav', delete=False) as f:
temp_path = Path(f.name)
try:
# Write audio as WAV
temp_path_str= str(temp_path)
import torchaudio
torchaudio.save(str(temp_path), audio.unsqueeze(0) if audio.dim() == 1 else audio, sampling_rate)
# Remux with FFmpeg
subprocess.run([
'ffmpeg', '-i', str(video_path), '-i', str(temp_path),
'-c:v', 'copy', '-c:a', 'aac', '-map', '0:v', '-map', '1:a',
'-shortest', '-y', str(output_path)
], check=True, capture_output=True)
finally:
torchaudio.save(temp_path_str, audio.unsqueeze(0) if audio.dim() == 1 else audio, sampling_rate)
combine_video_with_audio_tracks(video_path, [temp_path_str], output_path )
temp_path.unlink(missing_ok=True)
def remux_with_audio_old(video_path: Path, audio: torch.Tensor, output_path: Path, sampling_rate: int):

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@ -9,7 +9,8 @@ import os
from collections import defaultdict
from pathlib import Path
from typing import Optional, Union
import matplotlib
matplotlib.use('TkAgg')
import matplotlib.pyplot as plt
import numpy as np
import torch

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@ -5,6 +5,8 @@ from PIL import Image, ImageDraw, ImageOps
import numpy as np
from typing import Union
from segment_anything import sam_model_registry, SamPredictor, SamAutomaticMaskGenerator
import matplotlib
matplotlib.use('TkAgg')
import matplotlib.pyplot as plt
import PIL
from .mask_painter import mask_painter as mask_painter2

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@ -0,0 +1,922 @@
import torch
import torchaudio
import numpy as np
import os
import warnings
from pathlib import Path
from typing import Dict, List, Tuple
import argparse
from concurrent.futures import ThreadPoolExecutor
import gc
import logging
verbose_output = True
# Suppress specific warnings before importing pyannote
warnings.filterwarnings("ignore", category=UserWarning, module="pyannote.audio.models.blocks.pooling")
warnings.filterwarnings("ignore", message=".*TensorFloat-32.*", category=UserWarning)
warnings.filterwarnings("ignore", message=".*std\\(\\): degrees of freedom.*", category=UserWarning)
warnings.filterwarnings("ignore", message=".*speechbrain.pretrained.*was deprecated.*", category=UserWarning)
warnings.filterwarnings("ignore", message=".*Module 'speechbrain.pretrained'.*", category=UserWarning)
# logging.getLogger('speechbrain').setLevel(logging.WARNING)
# logging.getLogger('speechbrain.utils.checkpoints').setLevel(logging.WARNING)
os.environ["SB_LOG_LEVEL"] = "WARNING"
import speechbrain
def xprint(t = None):
if verbose_output:
print(t)
# Configure TF32 before any CUDA operations to avoid reproducibility warnings
if torch.cuda.is_available():
torch.backends.cuda.matmul.allow_tf32 = True
torch.backends.cudnn.allow_tf32 = True
try:
from pyannote.audio import Pipeline
PYANNOTE_AVAILABLE = True
except ImportError:
PYANNOTE_AVAILABLE = False
print("Install: pip install pyannote.audio")
class OptimizedPyannote31SpeakerSeparator:
def __init__(self, hf_token: str = None, local_model_path: str = None,
vad_onset: float = 0.2, vad_offset: float = 0.8):
"""
Initialize with Pyannote 3.1 pipeline with tunable VAD sensitivity.
"""
embedding_path = "ckpts/pyannote/pyannote_model_wespeaker-voxceleb-resnet34-LM.bin"
segmentation_path = "ckpts/pyannote/pytorch_model_segmentation-3.0.bin"
xprint(f"Loading segmentation model from: {segmentation_path}")
xprint(f"Loading embedding model from: {embedding_path}")
try:
from pyannote.audio import Model
from pyannote.audio.pipelines import SpeakerDiarization
# Load models directly
segmentation_model = Model.from_pretrained(segmentation_path)
embedding_model = Model.from_pretrained(embedding_path)
xprint("Models loaded successfully!")
# Create pipeline manually
self.pipeline = SpeakerDiarization(
segmentation=segmentation_model,
embedding=embedding_model,
clustering='AgglomerativeClustering'
)
# Instantiate with default parameters
self.pipeline.instantiate({
'clustering': {
'method': 'centroid',
'min_cluster_size': 12,
'threshold': 0.7045654963945799
},
'segmentation': {
'min_duration_off': 0.0
}
})
xprint("Pipeline instantiated successfully!")
# Send to GPU if available
if torch.cuda.is_available():
xprint("CUDA available, moving pipeline to GPU...")
self.pipeline.to(torch.device("cuda"))
else:
xprint("CUDA not available, using CPU...")
except Exception as e:
xprint(f"Error loading pipeline: {e}")
xprint(f"Error type: {type(e)}")
import traceback
traceback.xprint_exc()
raise
self.hf_token = hf_token
self._overlap_pipeline = None
def separate_audio(self, audio_path: str, output1, output2 ) -> Dict[str, str]:
"""Optimized main separation function with memory management."""
xprint("Starting optimized audio separation...")
self._current_audio_path = os.path.abspath(audio_path)
# Suppress warnings during processing
with warnings.catch_warnings():
warnings.filterwarnings("ignore", category=UserWarning)
# Load audio
waveform, sample_rate = self.load_audio(audio_path)
# Perform diarization
diarization = self.perform_optimized_diarization(audio_path)
# Create masks
masks = self.create_optimized_speaker_masks(diarization, waveform.shape[1], sample_rate)
# Apply background preservation
final_masks = self.apply_optimized_background_preservation(masks, waveform.shape[1])
# Clear intermediate results
del masks
if torch.cuda.is_available():
torch.cuda.empty_cache()
gc.collect()
# Save outputs efficiently
output_paths = self._save_outputs_optimized(waveform, final_masks, sample_rate, audio_path, output1, output2)
return output_paths
def _extract_both_speaking_regions(
self,
diarization,
audio_length: int,
sample_rate: int
) -> np.ndarray:
"""
Detect regions where 2 speakers talk simultaneously
using pyannote/overlapped-speech-detection.
Falls back to manual pair-wise detection if the model
is unavailable.
"""
xprint("Extracting overlap with dedicated pipeline…")
both_speaking_mask = np.zeros(audio_length, dtype=bool)
# ── 1) try the proper overlap model ────────────────────────────────
overlap_pipeline = self._get_overlap_pipeline()
# try the path stored by separate_audio otherwise whatever the
# diarization object carries (may be None)
audio_uri = getattr(self, "_current_audio_path", None) \
or getattr(diarization, "uri", None)
if overlap_pipeline and audio_uri:
try:
with warnings.catch_warnings():
warnings.filterwarnings("ignore", category=UserWarning)
overlap_annotation = overlap_pipeline(audio_uri)
for seg in overlap_annotation.get_timeline().support():
s = max(0, int(seg.start * sample_rate))
e = min(audio_length, int(seg.end * sample_rate))
if s < e:
both_speaking_mask[s:e] = True
t = np.sum(both_speaking_mask) / sample_rate
xprint(f" Found {t:.1f}s of overlapped speech (model) ")
return both_speaking_mask
except Exception as e:
xprint(f" ⚠ Overlap model failed: {e}")
# ── 2) fallback = brute-force pairwise intersection ────────────────
xprint(" Falling back to manual overlap detection…")
timeline_tracks = list(diarization.itertracks(yield_label=True))
for i, (turn1, _, spk1) in enumerate(timeline_tracks):
for j, (turn2, _, spk2) in enumerate(timeline_tracks):
if i >= j or spk1 == spk2:
continue
o_start, o_end = max(turn1.start, turn2.start), min(turn1.end, turn2.end)
if o_start < o_end:
s = max(0, int(o_start * sample_rate))
e = min(audio_length, int(o_end * sample_rate))
if s < e:
both_speaking_mask[s:e] = True
t = np.sum(both_speaking_mask) / sample_rate
xprint(f" Found {t:.1f}s of overlapped speech (manual) ")
return both_speaking_mask
def _configure_vad(self, vad_onset: float, vad_offset: float):
"""Configure VAD parameters efficiently."""
xprint("Applying more sensitive VAD parameters...")
try:
with warnings.catch_warnings():
warnings.filterwarnings("ignore", category=UserWarning)
if hasattr(self.pipeline, '_vad'):
self.pipeline._vad.instantiate({
"onset": vad_onset,
"offset": vad_offset,
"min_duration_on": 0.1,
"min_duration_off": 0.1,
"pad_onset": 0.1,
"pad_offset": 0.1,
})
xprint(f"✓ VAD parameters updated: onset={vad_onset}, offset={vad_offset}")
else:
xprint("⚠ Could not access VAD component directly")
except Exception as e:
xprint(f"⚠ Could not modify VAD parameters: {e}")
def _get_overlap_pipeline(self):
"""
Build a pyannote-3-native OverlappedSpeechDetection pipeline.
uses the open-licence `pyannote/segmentation-3.0` checkpoint
only `min_duration_on/off` can be tuned (API 3.x)
"""
if self._overlap_pipeline is not None:
return None if self._overlap_pipeline is False else self._overlap_pipeline
try:
from pyannote.audio.pipelines import OverlappedSpeechDetection
xprint("Building OverlappedSpeechDetection with segmentation-3.0…")
with warnings.catch_warnings():
warnings.filterwarnings("ignore", category=UserWarning)
# 1) constructor → segmentation model ONLY
ods = OverlappedSpeechDetection(
segmentation="pyannote/segmentation-3.0"
)
# 2) instantiate → **single dict** with the two valid knobs
ods.instantiate({
"min_duration_on": 0.06, # ≈ your previous 0.055 s
"min_duration_off": 0.10, # ≈ your previous 0.098 s
})
if torch.cuda.is_available():
ods.to(torch.device("cuda"))
self._overlap_pipeline = ods
xprint("✓ Overlap pipeline ready (segmentation-3.0)")
return ods
except Exception as e:
xprint(f"⚠ Could not build overlap pipeline ({e}). "
"Falling back to manual pair-wise detection.")
self._overlap_pipeline = False
return None
def _xprint_setup_instructions(self):
"""xprint setup instructions."""
xprint("\nTo use Pyannote 3.1:")
xprint("1. Get token: https://huggingface.co/settings/tokens")
xprint("2. Accept terms: https://huggingface.co/pyannote/speaker-diarization-3.1")
xprint("3. Run with: --token YOUR_TOKEN")
def load_audio(self, audio_path: str) -> Tuple[torch.Tensor, int]:
"""Load and preprocess audio efficiently."""
xprint(f"Loading audio: {audio_path}")
with warnings.catch_warnings():
warnings.filterwarnings("ignore", category=UserWarning)
waveform, sample_rate = torchaudio.load(audio_path)
# Convert to mono efficiently
if waveform.shape[0] > 1:
waveform = torch.mean(waveform, dim=0, keepdim=True)
xprint(f"Audio: {waveform.shape[1]} samples at {sample_rate}Hz")
return waveform, sample_rate
def perform_optimized_diarization(self, audio_path: str) -> object:
"""
Optimized diarization with efficient parameter testing.
"""
xprint("Running optimized Pyannote 3.1 diarization...")
# Optimized strategy order - most likely to succeed first
strategies = [
{"min_speakers": 2, "max_speakers": 2}, # Most common case
{"num_speakers": 2}, # Direct specification
{"min_speakers": 2, "max_speakers": 3}, # Slight flexibility
{"min_speakers": 1, "max_speakers": 2}, # Fallback
{"min_speakers": 2, "max_speakers": 4}, # More flexibility
{} # No constraints
]
for i, params in enumerate(strategies):
try:
xprint(f"Strategy {i+1}: {params}")
# Clear GPU memory before each attempt
if torch.cuda.is_available():
torch.cuda.empty_cache()
with warnings.catch_warnings():
warnings.filterwarnings("ignore", category=UserWarning)
diarization = self.pipeline(audio_path, **params)
speakers = list(diarization.labels())
speaker_count = len(speakers)
xprint(f" → Detected {speaker_count} speakers: {speakers}")
# Accept first successful result with 2+ speakers
if speaker_count >= 2:
xprint(f"✓ Success with strategy {i+1}! Using {speaker_count} speakers")
return diarization
elif speaker_count == 1 and i == 0:
# Store first result as fallback
fallback_diarization = diarization
except Exception as e:
xprint(f" Strategy {i+1} failed: {e}")
continue
# If we only got 1 speaker, try one aggressive attempt
if 'fallback_diarization' in locals():
xprint("Attempting aggressive clustering for single speaker...")
try:
aggressive_diarization = self._try_aggressive_clustering(audio_path)
if aggressive_diarization and len(list(aggressive_diarization.labels())) >= 2:
return aggressive_diarization
except Exception as e:
xprint(f"Aggressive clustering failed: {e}")
xprint("Using single speaker result")
return fallback_diarization
# Last resort - run without constraints
xprint("Last resort: running without constraints...")
with warnings.catch_warnings():
warnings.filterwarnings("ignore", category=UserWarning)
return self.pipeline(audio_path)
def _try_aggressive_clustering(self, audio_path: str) -> object:
"""Try aggressive clustering parameters."""
try:
from pyannote.audio.pipelines.speaker_diarization import SpeakerDiarization
with warnings.catch_warnings():
warnings.filterwarnings("ignore", category=UserWarning)
# Create aggressive pipeline
temp_pipeline = SpeakerDiarization(
segmentation=self.pipeline.segmentation,
embedding=self.pipeline.embedding,
clustering="AgglomerativeClustering"
)
temp_pipeline.instantiate({
"clustering": {
"method": "centroid",
"min_cluster_size": 1,
"threshold": 0.1,
},
"segmentation": {
"min_duration_off": 0.0,
"min_duration_on": 0.1,
}
})
return temp_pipeline(audio_path, min_speakers=2)
except Exception as e:
xprint(f"Aggressive clustering setup failed: {e}")
return None
def create_optimized_speaker_masks(self, diarization, audio_length: int, sample_rate: int) -> Dict[str, np.ndarray]:
"""Optimized mask creation using vectorized operations."""
xprint("Creating optimized speaker masks...")
speakers = list(diarization.labels())
xprint(f"Processing speakers: {speakers}")
# Handle edge cases
if len(speakers) == 0:
xprint("⚠ No speakers detected, creating dummy masks")
return self._create_dummy_masks(audio_length)
if len(speakers) == 1:
xprint("⚠ Only 1 speaker detected, creating temporal split")
return self._create_optimized_temporal_split(diarization, audio_length, sample_rate)
# Extract both-speaking regions from diarization timeline
both_speaking_regions = self._extract_both_speaking_regions(diarization, audio_length, sample_rate)
# Optimized mask creation for multiple speakers
masks = {}
# Batch process all speakers
for speaker in speakers:
# Get all segments for this speaker at once
segments = []
speaker_timeline = diarization.label_timeline(speaker)
for segment in speaker_timeline:
start_sample = max(0, int(segment.start * sample_rate))
end_sample = min(audio_length, int(segment.end * sample_rate))
if start_sample < end_sample:
segments.append((start_sample, end_sample))
# Vectorized mask creation
if segments:
mask = self._create_mask_vectorized(segments, audio_length)
masks[speaker] = mask
speaking_time = np.sum(mask) / sample_rate
xprint(f" {speaker}: {speaking_time:.1f}s speaking time")
else:
masks[speaker] = np.zeros(audio_length, dtype=np.float32)
# Store both-speaking info for later use
self._both_speaking_regions = both_speaking_regions
return masks
def _create_mask_vectorized(self, segments: List[Tuple[int, int]], audio_length: int) -> np.ndarray:
"""Create mask using vectorized operations."""
mask = np.zeros(audio_length, dtype=np.float32)
if not segments:
return mask
# Convert segments to arrays for vectorized operations
segments_array = np.array(segments)
starts = segments_array[:, 0]
ends = segments_array[:, 1]
# Use advanced indexing for bulk assignment
for start, end in zip(starts, ends):
mask[start:end] = 1.0
return mask
def _create_dummy_masks(self, audio_length: int) -> Dict[str, np.ndarray]:
"""Create dummy masks for edge cases."""
return {
"SPEAKER_00": np.ones(audio_length, dtype=np.float32) * 0.5,
"SPEAKER_01": np.ones(audio_length, dtype=np.float32) * 0.5
}
def _create_optimized_temporal_split(self, diarization, audio_length: int, sample_rate: int) -> Dict[str, np.ndarray]:
"""Optimized temporal split with vectorized operations."""
xprint("Creating optimized temporal split...")
# Extract all segments at once
segments = []
for turn, _, speaker in diarization.itertracks(yield_label=True):
segments.append((turn.start, turn.end))
segments.sort()
xprint(f"Found {len(segments)} speech segments")
if len(segments) <= 1:
# Single segment or no segments - simple split
return self._create_simple_split(audio_length)
# Vectorized gap analysis
segment_array = np.array(segments)
gaps = segment_array[1:, 0] - segment_array[:-1, 1] # Vectorized gap calculation
if len(gaps) > 0:
longest_gap_idx = np.argmax(gaps)
longest_gap_duration = gaps[longest_gap_idx]
xprint(f"Longest gap: {longest_gap_duration:.1f}s after segment {longest_gap_idx+1}")
if longest_gap_duration > 1.0:
# Split at natural break
split_point = longest_gap_idx + 1
xprint(f"Splitting at natural break: segments 1-{split_point} vs {split_point+1}-{len(segments)}")
return self._create_split_masks(segments, split_point, audio_length, sample_rate)
# Fallback: alternating assignment
xprint("Using alternating assignment...")
return self._create_alternating_masks(segments, audio_length, sample_rate)
def _create_simple_split(self, audio_length: int) -> Dict[str, np.ndarray]:
"""Simple temporal split in half."""
mid_point = audio_length // 2
masks = {
"SPEAKER_00": np.zeros(audio_length, dtype=np.float32),
"SPEAKER_01": np.zeros(audio_length, dtype=np.float32)
}
masks["SPEAKER_00"][:mid_point] = 1.0
masks["SPEAKER_01"][mid_point:] = 1.0
return masks
def _create_split_masks(self, segments: List[Tuple[float, float]], split_point: int,
audio_length: int, sample_rate: int) -> Dict[str, np.ndarray]:
"""Create masks with split at specific point."""
masks = {
"SPEAKER_00": np.zeros(audio_length, dtype=np.float32),
"SPEAKER_01": np.zeros(audio_length, dtype=np.float32)
}
# Vectorized segment processing
for i, (start_time, end_time) in enumerate(segments):
start_sample = max(0, int(start_time * sample_rate))
end_sample = min(audio_length, int(end_time * sample_rate))
if start_sample < end_sample:
speaker_key = "SPEAKER_00" if i < split_point else "SPEAKER_01"
masks[speaker_key][start_sample:end_sample] = 1.0
return masks
def _create_alternating_masks(self, segments: List[Tuple[float, float]],
audio_length: int, sample_rate: int) -> Dict[str, np.ndarray]:
"""Create masks with alternating assignment."""
masks = {
"SPEAKER_00": np.zeros(audio_length, dtype=np.float32),
"SPEAKER_01": np.zeros(audio_length, dtype=np.float32)
}
for i, (start_time, end_time) in enumerate(segments):
start_sample = max(0, int(start_time * sample_rate))
end_sample = min(audio_length, int(end_time * sample_rate))
if start_sample < end_sample:
speaker_key = f"SPEAKER_0{i % 2}"
masks[speaker_key][start_sample:end_sample] = 1.0
return masks
def apply_optimized_background_preservation(self, masks: Dict[str, np.ndarray],
audio_length: int) -> Dict[str, np.ndarray]:
"""
Heavily optimized background preservation using pure vectorized operations.
"""
xprint("Applying optimized voice separation logic...")
# Ensure exactly 2 speakers
speaker_keys = self._get_top_speakers(masks, audio_length)
# Pre-allocate final masks
final_masks = {
speaker: np.zeros(audio_length, dtype=np.float32)
for speaker in speaker_keys
}
# Get active masks (vectorized)
active_0 = masks.get(speaker_keys[0], np.zeros(audio_length)) > 0.5
active_1 = masks.get(speaker_keys[1], np.zeros(audio_length)) > 0.5
# Vectorized mask assignment
both_active = active_0 & active_1
only_0 = active_0 & ~active_1
only_1 = ~active_0 & active_1
neither = ~active_0 & ~active_1
# Apply assignments (all vectorized)
final_masks[speaker_keys[0]][both_active] = 1.0
final_masks[speaker_keys[1]][both_active] = 1.0
final_masks[speaker_keys[0]][only_0] = 1.0
final_masks[speaker_keys[1]][only_0] = 0.0
final_masks[speaker_keys[0]][only_1] = 0.0
final_masks[speaker_keys[1]][only_1] = 1.0
# Handle ambiguous regions efficiently
if np.any(neither):
ambiguous_assignments = self._compute_ambiguous_assignments_vectorized(
masks, speaker_keys, neither, audio_length
)
# Apply ambiguous assignments
final_masks[speaker_keys[0]][neither] = (ambiguous_assignments == 0).astype(np.float32) * 0.5
final_masks[speaker_keys[1]][neither] = (ambiguous_assignments == 1).astype(np.float32) * 0.5
# xprint statistics (vectorized)
sample_rate = 16000 # Assume 16kHz for timing
xprint(f" Both speaking clearly: {np.sum(both_active)/sample_rate:.1f}s")
xprint(f" {speaker_keys[0]} only: {np.sum(only_0)/sample_rate:.1f}s")
xprint(f" {speaker_keys[1]} only: {np.sum(only_1)/sample_rate:.1f}s")
xprint(f" Ambiguous (assigned): {np.sum(neither)/sample_rate:.1f}s")
# Apply minimum duration smoothing to prevent rapid switching
final_masks = self._apply_minimum_duration_smoothing(final_masks, sample_rate)
return final_masks
def _get_top_speakers(self, masks: Dict[str, np.ndarray], audio_length: int) -> List[str]:
"""Get top 2 speakers by speaking time."""
speaker_keys = list(masks.keys())
if len(speaker_keys) > 2:
# Vectorized speaking time calculation
speaking_times = {k: np.sum(v) for k, v in masks.items()}
speaker_keys = sorted(speaking_times.keys(), key=lambda x: speaking_times[x], reverse=True)[:2]
xprint(f"Keeping top 2 speakers: {speaker_keys}")
elif len(speaker_keys) == 1:
speaker_keys.append("SPEAKER_SILENT")
return speaker_keys
def _compute_ambiguous_assignments_vectorized(self, masks: Dict[str, np.ndarray],
speaker_keys: List[str],
ambiguous_mask: np.ndarray,
audio_length: int) -> np.ndarray:
"""Compute speaker assignments for ambiguous regions using vectorized operations."""
ambiguous_indices = np.where(ambiguous_mask)[0]
if len(ambiguous_indices) == 0:
return np.array([])
# Get speaker segments efficiently
speaker_segments = {}
for speaker in speaker_keys:
if speaker in masks and speaker != "SPEAKER_SILENT":
mask = masks[speaker] > 0.5
# Find segments using vectorized operations
diff = np.diff(np.concatenate(([False], mask, [False])).astype(int))
starts = np.where(diff == 1)[0]
ends = np.where(diff == -1)[0]
speaker_segments[speaker] = np.column_stack([starts, ends])
else:
speaker_segments[speaker] = np.array([]).reshape(0, 2)
# Vectorized distance calculations
distances = {}
for speaker in speaker_keys:
segments = speaker_segments[speaker]
if len(segments) == 0:
distances[speaker] = np.full(len(ambiguous_indices), np.inf)
else:
# Compute distances to all segments at once
distances[speaker] = self._compute_distances_to_segments(ambiguous_indices, segments)
# Assign based on minimum distance with late-audio bias
assignments = self._assign_based_on_distance(
distances, speaker_keys, ambiguous_indices, audio_length
)
return assignments
def _apply_minimum_duration_smoothing(self, masks: Dict[str, np.ndarray],
sample_rate: int, min_duration_ms: int = 600) -> Dict[str, np.ndarray]:
"""
Apply minimum duration smoothing with STRICT timer enforcement.
Uses original both-speaking regions from diarization.
"""
xprint(f"Applying STRICT minimum duration smoothing ({min_duration_ms}ms)...")
min_samples = int(min_duration_ms * sample_rate / 1000)
speaker_keys = list(masks.keys())
if len(speaker_keys) != 2:
return masks
mask0 = masks[speaker_keys[0]]
mask1 = masks[speaker_keys[1]]
# Use original both-speaking regions from diarization
both_speaking_original = getattr(self, '_both_speaking_regions', np.zeros(len(mask0), dtype=bool))
# Identify regions based on original diarization info
ambiguous_original = (mask0 < 0.3) & (mask1 < 0.3) & ~both_speaking_original
# Clear dominance: one speaker higher, and not both-speaking or ambiguous
remaining_mask = ~both_speaking_original & ~ambiguous_original
speaker0_dominant = (mask0 > mask1) & remaining_mask
speaker1_dominant = (mask1 > mask0) & remaining_mask
# Create preference signal including both-speaking as valid state
# -1=ambiguous, 0=speaker0, 1=speaker1, 2=both_speaking
preference_signal = np.full(len(mask0), -1, dtype=int)
preference_signal[speaker0_dominant] = 0
preference_signal[speaker1_dominant] = 1
preference_signal[both_speaking_original] = 2
# STRICT state machine enforcement
smoothed_assignment = np.full(len(mask0), -1, dtype=int)
corrections = 0
# State variables
current_state = -1 # -1=unset, 0=speaker0, 1=speaker1, 2=both_speaking
samples_remaining = 0 # Samples remaining in current state's lock period
# Process each sample with STRICT enforcement
for i in range(len(preference_signal)):
preference = preference_signal[i]
# If we're in a lock period, enforce the current state
if samples_remaining > 0:
# Force current state regardless of preference
smoothed_assignment[i] = current_state
samples_remaining -= 1
# Count corrections if this differs from preference
if preference >= 0 and preference != current_state:
corrections += 1
else:
# Lock period expired - can consider new state
if preference >= 0:
# Clear preference available (including both-speaking)
if current_state != preference:
# Switch to new state and start new lock period
current_state = preference
samples_remaining = min_samples - 1 # -1 because we use this sample
smoothed_assignment[i] = current_state
else:
# Ambiguous preference
if current_state >= 0:
# Continue with current state if we have one
smoothed_assignment[i] = current_state
else:
# No current state and ambiguous - leave as ambiguous
smoothed_assignment[i] = -1
# Convert back to masks based on smoothed assignment
smoothed_masks = {}
for i, speaker in enumerate(speaker_keys):
new_mask = np.zeros_like(mask0)
# Assign regions where this speaker is dominant
speaker_regions = smoothed_assignment == i
new_mask[speaker_regions] = 1.0
# Assign both-speaking regions (state 2) to both speakers
both_speaking_regions = smoothed_assignment == 2
new_mask[both_speaking_regions] = 1.0
# Handle ambiguous regions that remain unassigned
unassigned_ambiguous = smoothed_assignment == -1
if np.any(unassigned_ambiguous):
# Use original ambiguous values only for truly unassigned regions
original_ambiguous_mask = ambiguous_original & unassigned_ambiguous
new_mask[original_ambiguous_mask] = masks[speaker][original_ambiguous_mask]
smoothed_masks[speaker] = new_mask
# Calculate and xprint statistics
both_speaking_time = np.sum(smoothed_assignment == 2) / sample_rate
speaker0_time = np.sum(smoothed_assignment == 0) / sample_rate
speaker1_time = np.sum(smoothed_assignment == 1) / sample_rate
ambiguous_time = np.sum(smoothed_assignment == -1) / sample_rate
xprint(f" Both speaking clearly: {both_speaking_time:.1f}s")
xprint(f" {speaker_keys[0]} only: {speaker0_time:.1f}s")
xprint(f" {speaker_keys[1]} only: {speaker1_time:.1f}s")
xprint(f" Ambiguous (assigned): {ambiguous_time:.1f}s")
xprint(f" Enforced minimum duration on {corrections} samples ({corrections/sample_rate:.2f}s)")
return smoothed_masks
def _compute_distances_to_segments(self, indices: np.ndarray, segments: np.ndarray) -> np.ndarray:
"""Compute minimum distances from indices to segments (vectorized)."""
if len(segments) == 0:
return np.full(len(indices), np.inf)
# Broadcast for vectorized computation
indices_expanded = indices[:, np.newaxis] # Shape: (n_indices, 1)
starts = segments[:, 0] # Shape: (n_segments,)
ends = segments[:, 1] # Shape: (n_segments,)
# Compute distances to all segments
dist_to_start = np.maximum(0, starts - indices_expanded) # Shape: (n_indices, n_segments)
dist_from_end = np.maximum(0, indices_expanded - ends) # Shape: (n_indices, n_segments)
# Minimum of distance to start or from end for each segment
distances = np.minimum(dist_to_start, dist_from_end)
# Return minimum distance to any segment for each index
return np.min(distances, axis=1)
def _assign_based_on_distance(self, distances: Dict[str, np.ndarray],
speaker_keys: List[str],
ambiguous_indices: np.ndarray,
audio_length: int) -> np.ndarray:
"""Assign speakers based on distance with late-audio bias."""
speaker_0_distances = distances[speaker_keys[0]]
speaker_1_distances = distances[speaker_keys[1]]
# Basic assignment by minimum distance
assignments = (speaker_1_distances < speaker_0_distances).astype(int)
# Apply late-audio bias (vectorized)
late_threshold = int(audio_length * 0.6)
late_indices = ambiguous_indices > late_threshold
if np.any(late_indices) and len(speaker_keys) > 1:
# Simple late-audio bias: prefer speaker 1 in later parts
assignments[late_indices] = 1
return assignments
def _save_outputs_optimized(self, waveform: torch.Tensor, masks: Dict[str, np.ndarray],
sample_rate: int, audio_path: str, output1, output2) -> Dict[str, str]:
"""Optimized output saving with parallel processing."""
output_paths = {}
def save_speaker_audio(speaker_mask_pair, output):
speaker, mask = speaker_mask_pair
# Convert mask to tensor efficiently
mask_tensor = torch.from_numpy(mask).unsqueeze(0)
# Apply mask
masked_audio = waveform * mask_tensor
with warnings.catch_warnings():
warnings.filterwarnings("ignore", category=UserWarning)
torchaudio.save(output, masked_audio, sample_rate)
xprint(f"✓ Saved {speaker}: {output}")
return speaker, output
# Use ThreadPoolExecutor for parallel saving
with ThreadPoolExecutor(max_workers=2) as executor:
results = list(executor.map(save_speaker_audio, masks.items(), [output1, output2]))
output_paths = dict(results)
return output_paths
def print_summary(self, audio_path: str):
"""xprint diarization summary."""
with warnings.catch_warnings():
warnings.filterwarnings("ignore", category=UserWarning)
diarization = self.perform_optimized_diarization(audio_path)
xprint("\n=== Diarization Summary ===")
for turn, _, speaker in diarization.itertracks(yield_label=True):
xprint(f"{speaker}: {turn.start:.1f}s - {turn.end:.1f}s")
def extract_dual_audio(audio, output1, output2, verbose = False):
global verbose_output
verbose_output = verbose
separator = OptimizedPyannote31SpeakerSeparator(
None,
None,
vad_onset=0.2,
vad_offset=0.8
)
# Separate audio
import time
start_time = time.time()
outputs = separator.separate_audio(audio, output1, output2)
elapsed_time = time.time() - start_time
xprint(f"\n=== SUCCESS (completed in {elapsed_time:.2f}s) ===")
for speaker, path in outputs.items():
xprint(f"{speaker}: {path}")
def main():
parser = argparse.ArgumentParser(description="Optimized Pyannote 3.1 Speaker Separator")
parser.add_argument("--audio", required=True, help="Input audio file")
parser.add_argument("--output", required=True, help="Output directory")
parser.add_argument("--token", help="Hugging Face token")
parser.add_argument("--local-model", help="Path to local 3.1 model")
parser.add_argument("--summary", action="store_true", help="xprint summary")
# VAD sensitivity parameters
parser.add_argument("--vad-onset", type=float, default=0.2,
help="VAD onset threshold (lower = more sensitive to speech start, default: 0.2)")
parser.add_argument("--vad-offset", type=float, default=0.8,
help="VAD offset threshold (higher = keeps speech longer, default: 0.8)")
args = parser.parse_args()
xprint("=== Optimized Pyannote 3.1 Speaker Separator ===")
xprint("Performance optimizations: vectorized operations, memory management, parallel processing")
xprint(f"Audio: {args.audio}")
xprint(f"Output: {args.output}")
xprint(f"VAD onset: {args.vad_onset}")
xprint(f"VAD offset: {args.vad_offset}")
xprint()
if not os.path.exists(args.audio):
xprint(f"ERROR: Audio file not found: {args.audio}")
return
try:
# Initialize with VAD parameters
separator = OptimizedPyannote31SpeakerSeparator(
args.token,
args.local_model,
vad_onset=args.vad_onset,
vad_offset=args.vad_offset
)
# print summary if requested
if args.summary:
separator.print_summary(args.audio)
# Separate audio
import time
start_time = time.time()
audio_name = Path(args.audio).stem
output_filename = f"{audio_name}_speaker0.wav"
output_filename1 = f"{audio_name}_speaker1.wav"
output_path = os.path.join(args.output, output_filename)
output_path1 = os.path.join(args.output, output_filename1)
outputs = separator.separate_audio(args.audio, output_path, output_path1)
elapsed_time = time.time() - start_time
xprint(f"\n=== SUCCESS (completed in {elapsed_time:.2f}s) ===")
for speaker, path in outputs.items():
xprint(f"{speaker}: {path}")
except Exception as e:
xprint(f"ERROR: {e}")
return 1
if __name__ == "__main__":
exit(main())

View File

@ -38,7 +38,10 @@ pygame>=2.1.0
sounddevice>=0.4.0
# rembg==2.0.65
torchdiffeq >= 0.2.5
# 'nitrous-ema',
# 'hydra_colorlog',
tensordict >= 0.6.1
open_clip_torch >= 2.29.0
pyloudnorm
misaki
soundfile
# num2words
# spacy

View File

@ -1,4 +1,3 @@
from . import configs, distributed, modules
from .image2video import WanI2V
from .text2video import WanT2V
from .any2video import WanAny2V
from .diffusion_forcing import DTT2V

View File

@ -13,6 +13,7 @@ import torch
import torch.nn as nn
import torch.cuda.amp as amp
import torch.distributed as dist
import numpy as np
from tqdm import tqdm
from PIL import Image
import torchvision.transforms.functional as TF
@ -21,14 +22,15 @@ from .distributed.fsdp import shard_model
from .modules.model import WanModel
from .modules.t5 import T5EncoderModel
from .modules.vae import WanVAE
from .modules.clip import CLIPModel
from .utils.fm_solvers import (FlowDPMSolverMultistepScheduler,
get_sampling_sigmas, retrieve_timesteps)
from .utils.fm_solvers_unipc import FlowUniPCMultistepScheduler
from wan.modules.posemb_layers import get_rotary_pos_embed
from .utils.vace_preprocessor import VaceVideoProcessor
from wan.utils.basic_flowmatch import FlowMatchScheduler
from wan.utils.utils import get_outpainting_frame_location
from wgp import update_loras_slists
from wan.utils.utils import get_outpainting_frame_location, resize_lanczos, calculate_new_dimensions
from .multitalk.multitalk_utils import MomentumBuffer, adaptive_projected_guidance
def optimized_scale(positive_flat, negative_flat):
@ -43,14 +45,20 @@ def optimized_scale(positive_flat, negative_flat):
return st_star
def timestep_transform(t, shift=5.0, num_timesteps=1000 ):
t = t / num_timesteps
# shift the timestep based on ratio
new_t = shift * t / (1 + (shift - 1) * t)
new_t = new_t * num_timesteps
return new_t
class WanT2V:
class WanAny2V:
def __init__(
self,
config,
checkpoint_dir,
rank=0,
model_filename = None,
model_type = None,
base_model_type = None,
@ -63,7 +71,7 @@ class WanT2V:
):
self.device = torch.device(f"cuda")
self.config = config
self.rank = rank
self.VAE_dtype = VAE_dtype
self.dtype = dtype
self.num_train_timesteps = config.num_train_timesteps
self.param_dtype = config.param_dtype
@ -76,6 +84,14 @@ class WanT2V:
tokenizer_path=os.path.join(checkpoint_dir, config.t5_tokenizer),
shard_fn= None)
if hasattr(config, "clip_checkpoint"):
self.clip = CLIPModel(
dtype=config.clip_dtype,
device=self.device,
checkpoint_path=os.path.join(checkpoint_dir ,
config.clip_checkpoint),
tokenizer_path=os.path.join(checkpoint_dir , config.clip_tokenizer))
self.vae_stride = config.vae_stride
self.patch_size = config.patch_size
@ -83,22 +99,27 @@ class WanT2V:
vae_pth=os.path.join(checkpoint_dir, config.vae_checkpoint), dtype= VAE_dtype,
device=self.device)
logging.info(f"Creating WanModel from {model_filename[-1]}")
from mmgp import offload
# model_filename = "c:/temp/vace1.3/diffusion_pytorch_model.safetensors"
# model_filename = "Vacefusionix_quanto_fp16_int8.safetensors"
# model_filename = "c:/temp/t2v/diffusion_pytorch_model-00001-of-00006.safetensors"
# config_filename= "c:/temp/t2v/t2v.json"
# xmodel_filename = "c:/ml/multitalk/multitalk.safetensors"
# config_filename= "configs/multitalk.json"
# import json
# with open(config_filename, 'r', encoding='utf-8') as f:
# config = json.load(f)
# from mmgp import safetensors2
# sd = safetensors2.torch_load_file(xmodel_filename)
base_config_file = f"configs/{base_model_type}.json"
forcedConfigPath = base_config_file if len(model_filename) > 1 else None
forcedConfigPath = base_config_file if len(model_filename) > 1 or base_model_type in ["flf2v_720p"] else None
# model_filename[1] = xmodel_filename
self.model = offload.fast_load_transformers_model(model_filename, modelClass=WanModel,do_quantize= quantizeTransformer and not save_quantized, writable_tensors= False, defaultConfigPath=base_config_file , forcedConfigPath= forcedConfigPath)
# self.model = offload.load_model_data(self.model, xmodel_filename )
# offload.load_model_data(self.model, "c:/temp/Phantom-Wan-1.3B.pth")
# self.model.to(torch.bfloat16)
# self.model.cpu()
self.model.lock_layers_dtypes(torch.float32 if mixed_precision_transformer else dtype)
# dtype = torch.bfloat16
# offload.load_model_data(self.model, "ckpts/Wan14BT2VFusioniX_fp16.safetensors")
offload.change_dtype(self.model, dtype, True)
# offload.save_model(self.model, "multitalkbf16.safetensors", config_file_path=base_config_file, filter_sd=sd)
# offload.save_model(self.model, "multitalk_quanto_fp16.safetensors", do_quantize= True, config_file_path=base_config_file, filter_sd=sd)
# offload.save_model(self.model, "wan2.1_selforcing_fp16.safetensors", config_file_path=base_config_file)
# offload.save_model(self.model, "wan2.1_text2video_14B_mbf16.safetensors", config_file_path=base_config_file)
# offload.save_model(self.model, "wan2.1_text2video_14B_quanto_mfp16_int8.safetensors", do_quantize=True, config_file_path=base_config_file)
@ -109,7 +130,7 @@ class WanT2V:
self.sample_neg_prompt = config.sample_neg_prompt
if base_model_type in ["vace_14B", "vace_1.3B"]:
if self.model.config.get("vace_in_dim", None) != None:
self.vid_proc = VaceVideoProcessor(downsample=tuple([x * y for x, y in zip(config.vae_stride, self.patch_size)]),
min_area=480*832,
max_area=480*832,
@ -121,6 +142,9 @@ class WanT2V:
self.adapt_vace_model()
self.num_timesteps = 1000
self.use_timestep_transform = True
def vace_encode_frames(self, frames, ref_images, masks=None, tile_size = 0, overlapped_latents = None):
if ref_images is None:
ref_images = [None] * len(frames)
@ -134,10 +158,11 @@ class WanT2V:
reactive = [i * m + 0 * (1 - m) for i, m in zip(frames, masks)]
inactive = self.vae.encode(inactive, tile_size = tile_size)
if overlapped_latents != None and False :
if overlapped_latents != None and False : # disabled as quality seems worse
# inactive[0][:, 0:1] = self.vae.encode([frames[0][:, 0:1]], tile_size = tile_size)[0] # redundant
for t in inactive:
t[:, 1:overlapped_latents.shape[1] + 1] = overlapped_latents
overlapped_latents[: 0:1] = inactive[0][: 0:1]
reactive = self.vae.encode(reactive, tile_size = tile_size)
latents = [torch.cat((u, c), dim=0) for u, c in zip(inactive, reactive)]
@ -277,12 +302,14 @@ class WanT2V:
image_sizes.append(src_video[i].shape[2:])
for k, keep in enumerate(keep_video_guide_frames):
if not keep:
src_video[i][:, k:k+1] = 0
src_mask[i][:, k:k+1] = 1
pos = prepend_count + k
src_video[i][:, pos:pos+1] = 0
src_mask[i][:, pos:pos+1] = 1
for k, frame in enumerate(inject_frames):
if frame != None:
src_video[i][:, k:k+1], src_mask[i][:, k:k+1] = self.fit_image_into_canvas(frame, image_size, 0, device, True, outpainting_dims, return_mask= True)
pos = prepend_count + k
src_video[i][:, pos:pos+1], src_mask[i][:, pos:pos+1] = self.fit_image_into_canvas(frame, image_size, 0, device, True, outpainting_dims, return_mask= True)
self.background_mask = None
@ -323,12 +350,15 @@ class WanT2V:
return torch.cat(ref_vae_latents, dim=1)
def generate(self,
input_prompt,
input_frames= None,
input_masks = None,
input_ref_images = None,
input_video=None,
image_start = None,
image_end = None,
denoising_strength = 1.0,
target_camera=None,
context_scale=None,
@ -342,7 +372,6 @@ class WanT2V:
guide_scale=5.0,
n_prompt="",
seed=-1,
offload_model=True,
callback = None,
enable_RIFLEx = None,
VAE_tile_size = 0,
@ -352,128 +381,31 @@ class WanT2V:
slg_end = 1.0,
cfg_star_switch = True,
cfg_zero_step = 5,
audio_scale=None,
audio_cfg_scale=None,
audio_proj=None,
audio_context_lens=None,
overlapped_latents = None,
return_latent_slice = None,
overlap_noise = 0,
conditioning_latents_size = 0,
keep_frames_parsed = [],
model_filename = None,
model_type = None,
loras_slists = None,
offloadobj = None,
apg_switch = False,
**bbargs
):
r"""
Generates video frames from text prompt using diffusion process.
Args:
input_prompt (`str`):
Text prompt for content generation
size (tupele[`int`], *optional*, defaults to (1280,720)):
Controls video resolution, (width,height).
frame_num (`int`, *optional*, defaults to 81):
How many frames to sample from a video. The number should be 4n+1
shift (`float`, *optional*, defaults to 5.0):
Noise schedule shift parameter. Affects temporal dynamics
sample_solver (`str`, *optional*, defaults to 'unipc'):
Solver used to sample the video.
sampling_steps (`int`, *optional*, defaults to 40):
Number of diffusion sampling steps. Higher values improve quality but slow generation
guide_scale (`float`, *optional*, defaults 5.0):
Classifier-free guidance scale. Controls prompt adherence vs. creativity
n_prompt (`str`, *optional*, defaults to ""):
Negative prompt for content exclusion. If not given, use `config.sample_neg_prompt`
seed (`int`, *optional*, defaults to -1):
Random seed for noise generation. If -1, use random seed.
offload_model (`bool`, *optional*, defaults to True):
If True, offloads models to CPU during generation to save VRAM
Returns:
torch.Tensor:
Generated video frames tensor. Dimensions: (C, N H, W) where:
- C: Color channels (3 for RGB)
- N: Number of frames (81)
- H: Frame height (from size)
- W: Frame width from size)
"""
# preprocess
vace = "Vace" in model_filename
if n_prompt == "":
n_prompt = self.sample_neg_prompt
seed = seed if seed >= 0 else random.randint(0, sys.maxsize)
seed_g = torch.Generator(device=self.device)
seed_g.manual_seed(seed)
if self._interrupt:
return None
context = self.text_encoder([input_prompt], self.device)[0]
context_null = self.text_encoder([n_prompt], self.device)[0]
context = context.to(self.dtype)
context_null = context_null.to(self.dtype)
input_ref_images_neg = None
phantom = False
if target_camera != None:
width = input_video.shape[2]
height = input_video.shape[1]
input_video = input_video.to(dtype=self.dtype , device=self.device)
input_video = input_video.permute(3, 0, 1, 2).div_(127.5).sub_(1.)
source_latents = self.vae.encode([input_video])[0] #.to(dtype=self.dtype, device=self.device)
del input_video
# Process target camera (recammaster)
from wan.utils.cammmaster_tools import get_camera_embedding
cam_emb = get_camera_embedding(target_camera)
cam_emb = cam_emb.to(dtype=self.dtype, device=self.device)
if denoising_strength < 1. and input_frames != None:
height, width = input_frames.shape[-2:]
source_latents = self.vae.encode([input_frames])[0]
if vace :
# vace context encode
input_frames = [u.to(self.device) for u in input_frames]
input_ref_images = [ None if u == None else [v.to(self.device) for v in u] for u in input_ref_images]
input_masks = [u.to(self.device) for u in input_masks]
if self.background_mask != None: self.background_mask = [m.to(self.device) for m in self.background_mask]
previous_latents = None
# if overlapped_latents != None:
# input_ref_images = [u[-1:] for u in input_ref_images]
z0 = self.vace_encode_frames(input_frames, input_ref_images, masks=input_masks, tile_size = VAE_tile_size, overlapped_latents = overlapped_latents )
m0 = self.vace_encode_masks(input_masks, input_ref_images)
if self.background_mask != None:
zbg = self.vace_encode_frames([ref_img[0] for ref_img in input_ref_images], None, masks=self.background_mask, tile_size = VAE_tile_size )
mbg = self.vace_encode_masks(self.background_mask, None)
for zz0, mm0, zzbg, mmbg in zip(z0, m0, zbg, mbg):
zz0[:, 0:1] = zzbg
mm0[:, 0:1] = mmbg
self.background_mask = zz0 = mm0 = zzbg = mmbg = None
z = self.vace_latent(z0, m0)
target_shape = list(z0[0].shape)
target_shape[0] = int(target_shape[0] / 2)
else:
if input_ref_images != None: # Phantom Ref images
phantom = True
input_ref_images = self.get_vae_latents(input_ref_images, self.device)
input_ref_images_neg = torch.zeros_like(input_ref_images)
F = frame_num
target_shape = (self.vae.model.z_dim, (F - 1) // self.vae_stride[0] + 1 + (input_ref_images.shape[1] if input_ref_images != None else 0),
height // self.vae_stride[1],
width // self.vae_stride[2])
seq_len = math.ceil((target_shape[2] * target_shape[3]) /
(self.patch_size[1] * self.patch_size[2]) *
target_shape[1])
if self._interrupt:
return None
noise = [ torch.randn( *target_shape, dtype=torch.float32, device=self.device, generator=seed_g) ]
# evaluation mode
if sample_solver == 'causvid':
if sample_solver =="euler":
# prepare timesteps
timesteps = list(np.linspace(self.num_timesteps, 1, sampling_steps, dtype=np.float32))
timesteps.append(0.)
timesteps = [torch.tensor([t], device=self.device) for t in timesteps]
if self.use_timestep_transform:
timesteps = [timestep_transform(t, shift=shift, num_timesteps=self.num_timesteps) for t in timesteps][:-1]
sample_scheduler = None
elif sample_solver == 'causvid':
sample_scheduler = FlowMatchScheduler(num_inference_steps=sampling_steps, shift=shift, sigma_min=0, extra_one_step=True)
timesteps = torch.tensor([1000, 934, 862, 756, 603, 410, 250, 140, 74])[:sampling_steps].to(self.device)
sample_scheduler.timesteps =timesteps
@ -496,20 +428,158 @@ class WanT2V:
else:
raise NotImplementedError(f"Unsupported Scheduler {sample_solver}")
seed_g = torch.Generator(device=self.device)
seed_g.manual_seed(seed)
# sample videos
latents = noise[0]
del noise
kwargs = {'pipeline': self, 'callback': callback}
if self._interrupt:
return None
# Text Encoder
if n_prompt == "":
n_prompt = self.sample_neg_prompt
context = self.text_encoder([input_prompt], self.device)[0]
context_null = self.text_encoder([n_prompt], self.device)[0]
context = context.to(self.dtype)
context_null = context_null.to(self.dtype)
# from mmgp import offload
# offloadobj.unload_all()
if self._interrupt:
return None
vace = model_type in ["vace_1.3B","vace_14B", "vace_multitalk_14B"]
phantom = model_type in ["phantom_1.3B", "phantom_14B"]
fantasy = model_type in ["fantasy"]
multitalk = model_type in ["multitalk", "vace_multitalk_14B"]
ref_images_count = 0
trim_frames = 0
extended_overlapped_latents = None
# image2video
lat_frames = int((frame_num - 1) // self.vae_stride[0]) + 1
if image_start != None:
any_end_frame = False
if input_frames != None:
_ , preframes_count, height, width = input_frames.shape
lat_h, lat_w = height // self.vae_stride[1], width // self.vae_stride[2]
clip_context = self.clip.visual([input_frames[:, -1:]]) #.to(self.param_dtype)
input_frames = input_frames.to(device=self.device).to(dtype= self.VAE_dtype)
enc = torch.concat( [input_frames, torch.zeros( (3, frame_num-preframes_count, height, width),
device=self.device, dtype= self.VAE_dtype)],
dim = 1).to(self.device)
input_frames = None
else:
preframes_count = 1
image_start = TF.to_tensor(image_start)
any_end_frame = image_end != None
add_frames_for_end_image = any_end_frame and model_type not in ["fun_inp_1.3B", "fun_inp", "i2v_720p"]
if any_end_frame:
image_end = TF.to_tensor(image_end)
if add_frames_for_end_image:
frame_num +=1
lat_frames = int((frame_num - 2) // self.vae_stride[0] + 2)
trim_frames = 1
h, w = image_start.shape[1:]
h, w = calculate_new_dimensions(height, width, h, w, fit_into_canvas)
width, height = w, h
lat_h = round(
h // self.vae_stride[1] //
self.patch_size[1] * self.patch_size[1])
lat_w = round(
w // self.vae_stride[2] //
self.patch_size[2] * self.patch_size[2])
h = lat_h * self.vae_stride[1]
w = lat_w * self.vae_stride[2]
clip_image_size = self.clip.model.image_size
img_interpolated = resize_lanczos(image_start, h, w).sub_(0.5).div_(0.5).unsqueeze(0).transpose(0,1).to(self.device) #, self.dtype
image_start = resize_lanczos(image_start, clip_image_size, clip_image_size)
image_start = image_start.sub_(0.5).div_(0.5).to(self.device) #, self.dtype
if image_end!= None:
img_interpolated2 = resize_lanczos(image_end, h, w).sub_(0.5).div_(0.5).unsqueeze(0).transpose(0,1).to(self.device) #, self.dtype
image_end = resize_lanczos(image_end, clip_image_size, clip_image_size)
image_end = image_end.sub_(0.5).div_(0.5).to(self.device) #, self.dtype
if image_end != None and model_type == "flf2v_720p":
clip_context = self.clip.visual([image_start[:, None, :, :], image_end[:, None, :, :]])
else:
clip_context = self.clip.visual([image_start[:, None, :, :]])
if any_end_frame:
enc= torch.concat([
img_interpolated,
torch.zeros( (3, frame_num-2, h, w), device=self.device, dtype= self.VAE_dtype),
img_interpolated2,
], dim=1).to(self.device)
else:
enc= torch.concat([
img_interpolated,
torch.zeros( (3, frame_num-1, h, w), device=self.device, dtype= self.VAE_dtype)
], dim=1).to(self.device)
image_start = image_end = img_interpolated = img_interpolated2 = None
msk = torch.ones(1, frame_num, lat_h, lat_w, device=self.device)
if any_end_frame:
msk[:, preframes_count: -1] = 0
if add_frames_for_end_image:
msk = torch.concat([ torch.repeat_interleave(msk[:, 0:1], repeats=4, dim=1), msk[:, 1:-1], torch.repeat_interleave(msk[:, -1:], repeats=4, dim=1) ], dim=1)
else:
msk = torch.concat([ torch.repeat_interleave(msk[:, 0:1], repeats=4, dim=1), msk[:, 1:] ], dim=1)
else:
msk[:, preframes_count:] = 0
msk = torch.concat([ torch.repeat_interleave(msk[:, 0:1], repeats=4, dim=1), msk[:, 1:] ], dim=1)
msk = msk.view(1, msk.shape[1] // 4, 4, lat_h, lat_w)
msk = msk.transpose(1, 2)[0]
lat_y = self.vae.encode([enc], VAE_tile_size, any_end_frame= any_end_frame and add_frames_for_end_image)[0]
overlapped_latents_frames_num = int(1 + (preframes_count-1) // 4)
if overlapped_latents != None:
# disabled because looks worse
if False and overlapped_latents_frames_num > 1: lat_y[:, 1:overlapped_latents_frames_num] = overlapped_latents[:, 1:]
extended_overlapped_latents = lat_y[:, :overlapped_latents_frames_num].clone()
y = torch.concat([msk, lat_y])
lat_y = None
kwargs.update({'clip_fea': clip_context, 'y': y})
# Recam Master
if target_camera != None:
width = input_video.shape[2]
height = input_video.shape[1]
input_video = input_video.to(dtype=self.dtype , device=self.device)
input_video = input_video.permute(3, 0, 1, 2).div_(127.5).sub_(1.)
source_latents = self.vae.encode([input_video])[0] #.to(dtype=self.dtype, device=self.device)
del input_video
# Process target camera (recammaster)
from wan.utils.cammmaster_tools import get_camera_embedding
cam_emb = get_camera_embedding(target_camera)
cam_emb = cam_emb.to(dtype=self.dtype, device=self.device)
kwargs['cam_emb'] = cam_emb
# Video 2 Video
if denoising_strength < 1. and input_frames != None:
height, width = input_frames.shape[-2:]
source_latents = self.vae.encode([input_frames])[0]
injection_denoising_step = 0
inject_from_start = False
if denoising_strength < 1 and input_frames != None:
if len(keep_frames_parsed) == 0 or all(keep_frames_parsed): keep_frames_parsed = []
if input_frames != None and denoising_strength < 1 :
if overlapped_latents != None:
overlapped_latents_frames_num = overlapped_latents.shape[1]
overlapped_frames_num = (overlapped_latents_frames_num-1) * 4 + 1
else:
overlapped_latents_frames_num = overlapped_frames_num = 0
if len(keep_frames_parsed) == 0 or (overlapped_frames_num + len(keep_frames_parsed)) == input_frames.shape[1] and all(keep_frames_parsed) : keep_frames_parsed = []
injection_denoising_step = int(sampling_steps * (1. - denoising_strength) )
latent_keep_frames = []
if source_latents.shape[1] < latents.shape[1] or len(keep_frames_parsed) > 0:
if source_latents.shape[1] < lat_frames or len(keep_frames_parsed) > 0:
inject_from_start = True
if len(keep_frames_parsed) >0 :
if overlapped_frames_num > 0: keep_frames_parsed = [True] * overlapped_frames_num + keep_frames_parsed
latent_keep_frames =[keep_frames_parsed[0]]
for i in range(1, len(keep_frames_parsed), 4):
latent_keep_frames.append(all(keep_frames_parsed[i:i+4]))
@ -519,32 +589,77 @@ class WanT2V:
if hasattr(sample_scheduler, "sigmas"): sample_scheduler.sigmas= sample_scheduler.sigmas[injection_denoising_step:]
injection_denoising_step = 0
# Phantom
if phantom:
input_ref_images_neg = None
if input_ref_images != None: # Phantom Ref images
input_ref_images = self.get_vae_latents(input_ref_images, self.device)
input_ref_images_neg = torch.zeros_like(input_ref_images)
ref_images_count = input_ref_images.shape[1] if input_ref_images != None else 0
# Vace
if vace :
# vace context encode
input_frames = [u.to(self.device) for u in input_frames]
input_ref_images = [ None if u == None else [v.to(self.device) for v in u] for u in input_ref_images]
input_masks = [u.to(self.device) for u in input_masks]
if self.background_mask != None: self.background_mask = [m.to(self.device) for m in self.background_mask]
z0 = self.vace_encode_frames(input_frames, input_ref_images, masks=input_masks, tile_size = VAE_tile_size, overlapped_latents = overlapped_latents )
m0 = self.vace_encode_masks(input_masks, input_ref_images)
if self.background_mask != None:
zbg = self.vace_encode_frames([ref_img[0] for ref_img in input_ref_images], None, masks=self.background_mask, tile_size = VAE_tile_size )
mbg = self.vace_encode_masks(self.background_mask, None)
for zz0, mm0, zzbg, mmbg in zip(z0, m0, zbg, mbg):
zz0[:, 0:1] = zzbg
mm0[:, 0:1] = mmbg
self.background_mask = zz0 = mm0 = zzbg = mmbg = None
z = self.vace_latent(z0, m0)
ref_images_count = len(input_ref_images[0]) if input_ref_images != None and input_ref_images[0] != None else 0
context_scale = context_scale if context_scale != None else [1.0] * len(z)
kwargs.update({'vace_context' : z, 'vace_context_scale' : context_scale, "ref_images_count": ref_images_count })
if overlapped_latents != None :
overlapped_latents_size = overlapped_latents.shape[1]
extended_overlapped_latents = z[0][0:16, 0:overlapped_latents_size + ref_images_count].clone()
target_shape = list(z0[0].shape)
target_shape[0] = int(target_shape[0] / 2)
lat_h, lat_w = target_shape[-2:]
height = self.vae_stride[1] * lat_h
width = self.vae_stride[2] * lat_w
else:
target_shape = (self.vae.model.z_dim, lat_frames + ref_images_count, height // self.vae_stride[1], width // self.vae_stride[2])
if multitalk and audio_proj != None:
from wan.multitalk.multitalk import get_target_masks
audio_proj = [audio.to(self.dtype) for audio in audio_proj]
human_no = len(audio_proj[0])
token_ref_target_masks = get_target_masks(human_no, lat_h, lat_w, height, width, face_scale = 0.05, bbox = None).to(self.dtype) if human_no > 1 else None
if fantasy and audio_proj != None:
kwargs.update({ "audio_proj": audio_proj.to(self.dtype), "audio_context_lens": audio_context_lens, })
if self._interrupt:
return None
# Ropes
batch_size = 1
if target_camera != None:
shape = list(latents.shape[1:])
shape = list(target_shape[1:])
shape[0] *= 2
freqs = get_rotary_pos_embed(shape, enable_RIFLEx= False)
else:
freqs = get_rotary_pos_embed(latents.shape[1:], enable_RIFLEx= enable_RIFLEx)
freqs = get_rotary_pos_embed(target_shape[1:], enable_RIFLEx= enable_RIFLEx)
kwargs = {'freqs': freqs, 'pipeline': self, 'callback': callback}
if target_camera != None:
kwargs.update({'cam_emb': cam_emb})
if vace:
ref_images_count = len(input_ref_images[0]) if input_ref_images != None and input_ref_images[0] != None else 0
context_scale = context_scale if context_scale != None else [1.0] * len(z)
kwargs.update({'vace_context' : z, 'vace_context_scale' : context_scale})
if overlapped_latents != None :
overlapped_latents_size = overlapped_latents.shape[1] + 1
# overlapped_latents_size = 3
z_reactive = [ zz[0:16, 0:overlapped_latents_size + ref_images_count].clone() for zz in z]
kwargs["freqs"] = freqs
# Steps Skipping
cache_type = self.model.enable_cache
if cache_type != None:
x_count = 3 if phantom else 2
x_count = 3 if phantom or fantasy or multitalk else 2
self.model.previous_residual = [None] * x_count
if cache_type == "tea":
self.model.compute_teacache_threshold(self.model.cache_start_step, timesteps, self.model.cache_multiplier)
@ -552,6 +667,7 @@ class WanT2V:
self.model.compute_magcache_threshold(self.model.cache_start_step, timesteps, self.model.cache_multiplier)
self.model.accumulated_err, self.model.accumulated_steps, self.model.accumulated_ratio = [0.0] * x_count, [0] * x_count, [1.0] * x_count
self.model.one_for_all = x_count > 2
if callback != None:
callback(-1, None, True)
@ -560,15 +676,29 @@ class WanT2V:
if chipmunk:
self.model.setup_chipmunk()
# init denoising
updated_num_steps= len(timesteps)
if callback != None:
from wgp import update_loras_slists
update_loras_slists(self.model, loras_slists, updated_num_steps)
callback(-1, None, True, override_num_inference_steps = updated_num_steps)
if sample_scheduler != None:
scheduler_kwargs = {} if isinstance(sample_scheduler, FlowMatchScheduler) else {"generator": seed_g}
latents = torch.randn( *target_shape, dtype=torch.float32, device=self.device, generator=seed_g)
if apg_switch != 0:
apg_momentum = -0.75
apg_norm_threshold = 55
text_momentumbuffer = MomentumBuffer(apg_momentum)
audio_momentumbuffer = MomentumBuffer(apg_momentum)
# denoising
for i, t in enumerate(tqdm(timesteps)):
timestep = [t]
offload.set_step_no_for_lora(self.model, i)
timestep = torch.stack([t])
kwargs.update({"t": timestep, "current_step": i})
kwargs["slg_layers"] = slg_layers if int(slg_start * sampling_steps) <= i < int(slg_end * sampling_steps) else None
if denoising_strength < 1 and input_frames != None and i <= injection_denoising_step:
sigma = t / 1000
@ -585,78 +715,94 @@ class WanT2V:
latents = noise * sigma + (1 - sigma) * source_latents
noise = None
if overlapped_latents != None :
overlap_noise_factor = overlap_noise / 1000
if extended_overlapped_latents != None:
latent_noise_factor = t / 1000
for zz, zz_r, ll in zip(z, z_reactive, [latents, None]): # extra None for second control net
zz[0:16, ref_images_count:overlapped_latents_size + ref_images_count] = zz_r[:, ref_images_count:] * (1.0 - overlap_noise_factor) + torch.randn_like(zz_r[:, ref_images_count:] ) * overlap_noise_factor
if ll != None:
ll[:, 0:overlapped_latents_size + ref_images_count] = zz_r * (1.0 - latent_noise_factor) + torch.randn_like(zz_r ) * latent_noise_factor
latents[:, 0:extended_overlapped_latents.shape[1]] = extended_overlapped_latents * (1.0 - latent_noise_factor) + torch.randn_like(extended_overlapped_latents ) * latent_noise_factor
if vace:
overlap_noise_factor = overlap_noise / 1000
for zz in z:
zz[0:16, ref_images_count:extended_overlapped_latents.shape[1] ] = extended_overlapped_latents[:, ref_images_count:] * (1.0 - overlap_noise_factor) + torch.randn_like(extended_overlapped_latents[:, ref_images_count:] ) * overlap_noise_factor
if target_camera != None:
latent_model_input = torch.cat([latents, source_latents], dim=1)
else:
latent_model_input = latents
kwargs["slg_layers"] = slg_layers if int(slg_start * sampling_steps) <= i < int(slg_end * sampling_steps) else None
offload.set_step_no_for_lora(self.model, i)
timestep = torch.stack(timestep)
kwargs["current_step"] = i
kwargs["t"] = timestep
if guide_scale == 1:
noise_pred = self.model( [latent_model_input], x_id = 0, context = [context], **kwargs)[0]
if self._interrupt:
return None
elif joint_pass:
if phantom:
pos_it, pos_i, neg = self.model(
[ torch.cat([latent_model_input[:,:-input_ref_images.shape[1]], input_ref_images], dim=1) ] * 2 +
[ torch.cat([latent_model_input[:,:-input_ref_images_neg.shape[1]], input_ref_images_neg], dim=1)],
context = [context, context_null, context_null], **kwargs)
gen_args = {
"x" : ([ torch.cat([latent_model_input[:,:-ref_images_count], input_ref_images], dim=1) ] * 2 +
[ torch.cat([latent_model_input[:,:-ref_images_count], input_ref_images_neg], dim=1)]),
"context": [context, context_null, context_null] ,
}
elif fantasy:
gen_args = {
"x" : [latent_model_input, latent_model_input, latent_model_input],
"context" : [context, context_null, context_null],
"audio_scale": [audio_scale, None, None ]
}
elif multitalk:
gen_args = {
"x" : [latent_model_input, latent_model_input, latent_model_input],
"context" : [context, context_null, context_null],
"multitalk_audio": [audio_proj, audio_proj, [torch.zeros_like(audio_proj[0][-1:]), torch.zeros_like(audio_proj[1][-1:])]],
"multitalk_masks": [token_ref_target_masks, token_ref_target_masks, None]
}
else:
noise_pred_cond, noise_pred_uncond = self.model(
[latent_model_input, latent_model_input], context = [context, context_null], **kwargs)
if self._interrupt:
return None
else:
if phantom:
pos_it = self.model(
[ torch.cat([latent_model_input[:,:-input_ref_images.shape[1]], input_ref_images], dim=1) ], x_id = 0, context = [context], **kwargs
)[0]
if self._interrupt:
return None
pos_i = self.model(
[ torch.cat([latent_model_input[:,:-input_ref_images.shape[1]], input_ref_images], dim=1) ], x_id = 1, context = [context_null],**kwargs
)[0]
if self._interrupt:
return None
neg = self.model(
[ torch.cat([latent_model_input[:,:-input_ref_images_neg.shape[1]], input_ref_images_neg], dim=1) ], x_id = 2, context = [context_null], **kwargs
)[0]
if self._interrupt:
return None
else:
noise_pred_cond = self.model(
[latent_model_input], x_id = 0, context = [context], **kwargs)[0]
if self._interrupt:
return None
noise_pred_uncond = self.model(
[latent_model_input], x_id = 1, context = [context_null], **kwargs)[0]
if self._interrupt:
return None
gen_args = {
"x" : [latent_model_input, latent_model_input],
"context": [context, context_null]
}
# del latent_model_input
# CFG Zero *. Thanks to https://github.com/WeichenFan/CFG-Zero-star/
if joint_pass and guide_scale > 1:
ret_values = self.model( **gen_args , **kwargs)
if self._interrupt:
return None
else:
size = 1 if guide_scale == 1 else len(gen_args["x"])
ret_values = [None] * size
for x_id in range(size):
sub_gen_args = {k : [v[x_id]] for k, v in gen_args.items() }
ret_values[x_id] = self.model( **sub_gen_args, x_id= x_id , **kwargs)[0]
if self._interrupt:
return None
sub_gen_args = None
if guide_scale == 1:
pass
noise_pred = ret_values[0]
elif phantom:
guide_scale_img= 5.0
guide_scale_text= guide_scale #7.5
pos_it, pos_i, neg = ret_values
noise_pred = neg + guide_scale_img * (pos_i - neg) + guide_scale_text * (pos_it - pos_i)
pos_it = pos_i = neg = None
elif fantasy:
noise_pred_cond, noise_pred_noaudio, noise_pred_uncond = ret_values
noise_pred = noise_pred_uncond + guide_scale * (noise_pred_noaudio - noise_pred_uncond) + audio_cfg_scale * (noise_pred_cond - noise_pred_noaudio)
noise_pred_noaudio = None
elif multitalk:
noise_pred_cond, noise_pred_drop_text, noise_pred_uncond = ret_values
if apg_switch != 0:
noise_pred = noise_pred_cond + (guide_scale - 1) * adaptive_projected_guidance(noise_pred_cond - noise_pred_drop_text,
noise_pred_cond,
momentum_buffer=text_momentumbuffer,
norm_threshold=apg_norm_threshold) \
+ (audio_cfg_scale - 1) * adaptive_projected_guidance(noise_pred_drop_text - noise_pred_uncond,
noise_pred_cond,
momentum_buffer=audio_momentumbuffer,
norm_threshold=apg_norm_threshold)
else:
noise_pred = noise_pred_uncond + guide_scale * (noise_pred_cond - noise_pred_drop_text) + audio_cfg_scale * (noise_pred_drop_text - noise_pred_uncond)
noise_pred_uncond = noise_pred_cond = noise_pred_drop_text = None
else:
noise_pred_cond, noise_pred_uncond = ret_values
if apg_switch != 0:
noise_pred = noise_pred_cond + (guide_scale - 1) * adaptive_projected_guidance(noise_pred_cond - noise_pred_uncond,
noise_pred_cond,
momentum_buffer=text_momentumbuffer,
norm_threshold=apg_norm_threshold)
else:
noise_pred_text = noise_pred_cond
if cfg_star_switch:
# CFG Zero *. Thanks to https://github.com/WeichenFan/CFG-Zero-star/
positive_flat = noise_pred_text.view(batch_size, -1)
negative_flat = noise_pred_uncond.view(batch_size, -1)
@ -668,12 +814,17 @@ class WanT2V:
else:
noise_pred_uncond *= alpha
noise_pred = noise_pred_uncond + guide_scale * (noise_pred_text - noise_pred_uncond)
noise_pred_uncond, noise_pred_cond, noise_pred_text, pos_it, pos_i, neg = None, None, None, None, None, None
ret_values = noise_pred_uncond = noise_pred_cond = noise_pred_text = neg = None
if sample_solver == "euler":
dt = timesteps[i] if i == len(timesteps)-1 else (timesteps[i] - timesteps[i + 1])
dt = dt / self.num_timesteps
latents = latents - noise_pred * dt[:, None, None, None]
else:
temp_x0 = sample_scheduler.step(
noise_pred[:, :target_shape[1]].unsqueeze(0),
t,
latents.unsqueeze(0),
# return_dict=False,
**scheduler_kwargs)[0]
latents = temp_x0.squeeze(0)
del temp_x0
@ -684,23 +835,19 @@ class WanT2V:
x0 = [latents]
if chipmunk:
self.model.release_chipmunk() # need to add it at every exit when in prof
self.model.release_chipmunk() # need to add it at every exit when in prod
if return_latent_slice != None:
if overlapped_latents != None:
# latents [:, 1:] = self.toto
for zz, zz_r, ll in zip(z, z_reactive, [latents]):
ll[:, 0:overlapped_latents_size + ref_images_count] = zz_r
latent_slice = latents[:, return_latent_slice].clone()
if input_frames == None:
if phantom:
# phantom post processing
x0 = [x0_[:,:-input_ref_images.shape[1]] for x0_ in x0]
videos = self.vae.decode(x0, VAE_tile_size)
else:
if vace:
# vace post processing
videos = self.decode_latent(x0, input_ref_images, VAE_tile_size)
else:
if phantom and input_ref_images != None:
trim_frames = input_ref_images.shape[1]
if trim_frames > 0: x0 = [x0_[:,:-trim_frames] for x0_ in x0]
videos = self.vae.decode(x0, VAE_tile_size)
if return_latent_slice != None:
return { "x" : videos[0], "latent_slice" : latent_slice }
return videos[0]

View File

@ -1,439 +0,0 @@
# Copyright 2024-2025 The Alibaba Wan Team Authors. All rights reserved.
import gc
import logging
import math
import os
import random
import sys
import types
from contextlib import contextmanager
from functools import partial
import json
import numpy as np
import torch
import torch.cuda.amp as amp
import torch.distributed as dist
import torchvision.transforms.functional as TF
from tqdm import tqdm
from .distributed.fsdp import shard_model
from .modules.clip import CLIPModel
from .modules.model import WanModel
from .modules.t5 import T5EncoderModel
from .modules.vae import WanVAE
from .utils.fm_solvers import (FlowDPMSolverMultistepScheduler,
get_sampling_sigmas, retrieve_timesteps)
from .utils.fm_solvers_unipc import FlowUniPCMultistepScheduler
from wan.modules.posemb_layers import get_rotary_pos_embed
from wan.utils.utils import resize_lanczos, calculate_new_dimensions
from wan.utils.basic_flowmatch import FlowMatchScheduler
def optimized_scale(positive_flat, negative_flat):
# Calculate dot production
dot_product = torch.sum(positive_flat * negative_flat, dim=1, keepdim=True)
# Squared norm of uncondition
squared_norm = torch.sum(negative_flat ** 2, dim=1, keepdim=True) + 1e-8
# st_star = v_cond^T * v_uncond / ||v_uncond||^2
st_star = dot_product / squared_norm
return st_star
class WanI2V:
def __init__(
self,
config,
checkpoint_dir,
model_filename = None,
model_type = None,
base_model_type= None,
text_encoder_filename= None,
quantizeTransformer = False,
dtype = torch.bfloat16,
VAE_dtype = torch.float32,
save_quantized = False,
mixed_precision_transformer = False
):
self.device = torch.device(f"cuda")
self.config = config
self.dtype = dtype
self.VAE_dtype = VAE_dtype
self.num_train_timesteps = config.num_train_timesteps
self.param_dtype = config.param_dtype
# shard_fn = partial(shard_model, device_id=device_id)
self.text_encoder = T5EncoderModel(
text_len=config.text_len,
dtype=config.t5_dtype,
device=torch.device('cpu'),
checkpoint_path=text_encoder_filename,
tokenizer_path=os.path.join(checkpoint_dir, config.t5_tokenizer),
shard_fn=None,
)
self.vae_stride = config.vae_stride
self.patch_size = config.patch_size
self.vae = WanVAE(
vae_pth=os.path.join(checkpoint_dir, config.vae_checkpoint), dtype = VAE_dtype,
device=self.device)
self.clip = CLIPModel(
dtype=config.clip_dtype,
device=self.device,
checkpoint_path=os.path.join(checkpoint_dir ,
config.clip_checkpoint),
tokenizer_path=os.path.join(checkpoint_dir , config.clip_tokenizer))
logging.info(f"Creating WanModel from {model_filename[-1]}")
from mmgp import offload
# fantasy = torch.load("c:/temp/fantasy.ckpt")
# proj_model = fantasy["proj_model"]
# audio_processor = fantasy["audio_processor"]
# offload.safetensors2.torch_write_file(proj_model, "proj_model.safetensors")
# offload.safetensors2.torch_write_file(audio_processor, "audio_processor.safetensors")
# for k,v in audio_processor.items():
# audio_processor[k] = v.to(torch.bfloat16)
# with open("fantasy_config.json", "r", encoding="utf-8") as reader:
# config_text = reader.read()
# config_json = json.loads(config_text)
# offload.safetensors2.torch_write_file(audio_processor, "audio_processor_bf16.safetensors", config=config_json)
# model_filename = [model_filename, "audio_processor_bf16.safetensors"]
# model_filename = "c:/temp/i2v480p/diffusion_pytorch_model-00001-of-00007.safetensors"
# dtype = torch.float16
base_config_file = f"configs/{base_model_type}.json"
forcedConfigPath = base_config_file if len(model_filename) > 1 else None
self.model = offload.fast_load_transformers_model(model_filename, modelClass=WanModel,do_quantize= quantizeTransformer and not save_quantized, writable_tensors= False, defaultConfigPath= base_config_file, forcedConfigPath= forcedConfigPath)
self.model.lock_layers_dtypes(torch.float32 if mixed_precision_transformer else dtype)
offload.change_dtype(self.model, dtype, True)
# offload.save_model(self.model, "wan2.1_image2video_720p_14B_mbf16.safetensors", config_file_path="c:/temp/i2v720p/config.json")
# offload.save_model(self.model, "wan2.1_image2video_720p_14B_quanto_mbf16_int8.safetensors",do_quantize=True, config_file_path="c:/temp/i2v720p/config.json")
# offload.save_model(self.model, "wan2.1_image2video_720p_14B_quanto_mfp16_int8.safetensors",do_quantize=True, config_file_path="c:/temp/i2v720p/config.json")
# offload.save_model(self.model, "wan2.1_Fun_InP_1.3B_bf16_bis.safetensors")
self.model.eval().requires_grad_(False)
if save_quantized:
from wgp import save_quantized_model
save_quantized_model(self.model, model_type, model_filename[0], dtype, base_config_file)
self.sample_neg_prompt = config.sample_neg_prompt
def generate(self,
input_prompt,
image_start,
image_end = None,
height =720,
width = 1280,
fit_into_canvas = True,
frame_num=81,
shift=5.0,
sample_solver='unipc',
sampling_steps=40,
guide_scale=5.0,
n_prompt="",
seed=-1,
callback = None,
enable_RIFLEx = False,
VAE_tile_size= 0,
joint_pass = False,
slg_layers = None,
slg_start = 0.0,
slg_end = 1.0,
cfg_star_switch = True,
cfg_zero_step = 5,
audio_scale=None,
audio_cfg_scale=None,
audio_proj=None,
audio_context_lens=None,
model_filename = None,
offloadobj = None,
**bbargs
):
r"""
Generates video frames from input image and text prompt using diffusion process.
Args:
input_prompt (`str`):
Text prompt for content generation.
image_start (PIL.Image.Image):
Input image tensor. Shape: [3, H, W]
max_area (`int`, *optional*, defaults to 720*1280):
Maximum pixel area for latent space calculation. Controls video resolution scaling
frame_num (`int`, *optional*, defaults to 81):
How many frames to sample from a video. The number should be 4n+1
shift (`float`, *optional*, defaults to 5.0):
Noise schedule shift parameter. Affects temporal dynamics
[NOTE]: If you want to generate a 480p video, it is recommended to set the shift value to 3.0.
sample_solver (`str`, *optional*, defaults to 'unipc'):
Solver used to sample the video.
sampling_steps (`int`, *optional*, defaults to 40):
Number of diffusion sampling steps. Higher values improve quality but slow generation
guide_scale (`float`, *optional*, defaults 5.0):
Classifier-free guidance scale. Controls prompt adherence vs. creativity
n_prompt (`str`, *optional*, defaults to ""):
Negative prompt for content exclusion. If not given, use `config.sample_neg_prompt`
seed (`int`, *optional*, defaults to -1):
Random seed for noise generation. If -1, use random seed
offload_model (`bool`, *optional*, defaults to True):
If True, offloads models to CPU during generation to save VRAM
Returns:
torch.Tensor:
Generated video frames tensor. Dimensions: (C, N H, W) where:
- C: Color channels (3 for RGB)
- N: Number of frames (81)
- H: Frame height (from max_area)
- W: Frame width from max_area)
"""
add_frames_for_end_image = "image2video" in model_filename or "fantasy" in model_filename
image_start = TF.to_tensor(image_start)
lat_frames = int((frame_num - 1) // self.vae_stride[0] + 1)
any_end_frame = image_end !=None
if any_end_frame:
any_end_frame = True
image_end = TF.to_tensor(image_end)
if add_frames_for_end_image:
frame_num +=1
lat_frames = int((frame_num - 2) // self.vae_stride[0] + 2)
h, w = image_start.shape[1:]
h, w = calculate_new_dimensions(height, width, h, w, fit_into_canvas)
lat_h = round(
h // self.vae_stride[1] //
self.patch_size[1] * self.patch_size[1])
lat_w = round(
w // self.vae_stride[2] //
self.patch_size[2] * self.patch_size[2])
h = lat_h * self.vae_stride[1]
w = lat_w * self.vae_stride[2]
clip_image_size = self.clip.model.image_size
img_interpolated = resize_lanczos(image_start, h, w).sub_(0.5).div_(0.5).unsqueeze(0).transpose(0,1).to(self.device) #, self.dtype
image_start = resize_lanczos(image_start, clip_image_size, clip_image_size)
image_start = image_start.sub_(0.5).div_(0.5).to(self.device) #, self.dtype
if image_end!= None:
img_interpolated2 = resize_lanczos(image_end, h, w).sub_(0.5).div_(0.5).unsqueeze(0).transpose(0,1).to(self.device) #, self.dtype
image_end = resize_lanczos(image_end, clip_image_size, clip_image_size)
image_end = image_end.sub_(0.5).div_(0.5).to(self.device) #, self.dtype
max_seq_len = lat_frames * lat_h * lat_w // ( self.patch_size[1] * self.patch_size[2])
seed = seed if seed >= 0 else random.randint(0, sys.maxsize)
seed_g = torch.Generator(device=self.device)
seed_g.manual_seed(seed)
noise = torch.randn(16, lat_frames, lat_h, lat_w, dtype=torch.float32, generator=seed_g, device=self.device)
msk = torch.ones(1, frame_num, lat_h, lat_w, device=self.device)
if any_end_frame:
msk[:, 1: -1] = 0
if add_frames_for_end_image:
msk = torch.concat([ torch.repeat_interleave(msk[:, 0:1], repeats=4, dim=1), msk[:, 1:-1], torch.repeat_interleave(msk[:, -1:], repeats=4, dim=1) ], dim=1)
else:
msk = torch.concat([ torch.repeat_interleave(msk[:, 0:1], repeats=4, dim=1), msk[:, 1:] ], dim=1)
else:
msk[:, 1:] = 0
msk = torch.concat([ torch.repeat_interleave(msk[:, 0:1], repeats=4, dim=1), msk[:, 1:] ], dim=1)
msk = msk.view(1, msk.shape[1] // 4, 4, lat_h, lat_w)
msk = msk.transpose(1, 2)[0]
if n_prompt == "":
n_prompt = self.sample_neg_prompt
if self._interrupt:
return None
# preprocess
context = self.text_encoder([input_prompt], self.device)[0]
context_null = self.text_encoder([n_prompt], self.device)[0]
context = context.to(self.dtype)
context_null = context_null.to(self.dtype)
if self._interrupt:
return None
clip_context = self.clip.visual([image_start[:, None, :, :]])
from mmgp import offload
offloadobj.unload_all()
if any_end_frame:
mean2 = 0
enc= torch.concat([
img_interpolated,
torch.full( (3, frame_num-2, h, w), mean2, device=self.device, dtype= self.VAE_dtype),
img_interpolated2,
], dim=1).to(self.device)
else:
enc= torch.concat([
img_interpolated,
torch.zeros(3, frame_num-1, h, w, device=self.device, dtype= self.VAE_dtype)
], dim=1).to(self.device)
image_start, image_end, img_interpolated, img_interpolated2 = None, None, None, None
lat_y = self.vae.encode([enc], VAE_tile_size, any_end_frame= any_end_frame and add_frames_for_end_image)[0]
y = torch.concat([msk, lat_y])
lat_y = None
# evaluation mode
if sample_solver == 'causvid':
sample_scheduler = FlowMatchScheduler(num_inference_steps=sampling_steps, shift=shift, sigma_min=0, extra_one_step=True)
timesteps = torch.tensor([1000, 934, 862, 756, 603, 410, 250, 140, 74])[:sampling_steps].to(self.device)
sample_scheduler.timesteps =timesteps
sample_scheduler.sigmas = torch.cat([sample_scheduler.timesteps / 1000, torch.tensor([0.], device=self.device)])
elif sample_solver == 'unipc' or sample_solver == "":
sample_scheduler = FlowUniPCMultistepScheduler(
num_train_timesteps=self.num_train_timesteps,
shift=1,
use_dynamic_shifting=False)
sample_scheduler.set_timesteps(
sampling_steps, device=self.device, shift=shift)
timesteps = sample_scheduler.timesteps
elif sample_solver == 'dpm++':
sample_scheduler = FlowDPMSolverMultistepScheduler(
num_train_timesteps=self.num_train_timesteps,
shift=1,
use_dynamic_shifting=False)
sampling_sigmas = get_sampling_sigmas(sampling_steps, shift)
timesteps, _ = retrieve_timesteps(
sample_scheduler,
device=self.device,
sigmas=sampling_sigmas)
else:
raise NotImplementedError("Unsupported scheduler.")
# sample videos
latent = noise
batch_size = 1
freqs = get_rotary_pos_embed(latent.shape[1:], enable_RIFLEx= enable_RIFLEx)
kwargs = { 'clip_fea': clip_context, 'y': y, 'freqs' : freqs, 'pipeline' : self, 'callback' : callback }
if audio_proj != None:
kwargs.update({
"audio_proj": audio_proj.to(self.dtype),
"audio_context_lens": audio_context_lens,
})
cache_type = self.model.enable_cache
if cache_type != None:
x_count = 3 if audio_cfg_scale !=None else 2
self.model.previous_residual = [None] * x_count
if cache_type == "tea":
self.model.compute_teacache_threshold(self.model.cache_start_step, timesteps, self.model.cache_multiplier)
else:
self.model.compute_magcache_threshold(self.model.cache_start_step, timesteps, self.model.cache_multiplier)
self.model.accumulated_err, self.model.accumulated_steps, self.model.accumulated_ratio = [0.0] * x_count, [0] * x_count, [1.0] * x_count
self.model.one_for_all = x_count > 2
# self.model.to(self.device)
if callback != None:
callback(-1, None, True)
latent = latent.to(self.device)
for i, t in enumerate(tqdm(timesteps)):
offload.set_step_no_for_lora(self.model, i)
kwargs["slg_layers"] = slg_layers if int(slg_start * sampling_steps) <= i < int(slg_end * sampling_steps) else None
latent_model_input = latent
timestep = [t]
timestep = torch.stack(timestep).to(self.device)
kwargs.update({
't' :timestep,
'current_step' :i,
})
if guide_scale == 1:
noise_pred = self.model( [latent_model_input], context=[context], audio_scale = None if audio_scale == None else [audio_scale], x_id=0, **kwargs, )[0]
if self._interrupt:
return None
elif joint_pass:
if audio_proj == None:
noise_pred_cond, noise_pred_uncond = self.model(
[latent_model_input, latent_model_input],
context=[context, context_null],
**kwargs)
else:
noise_pred_cond, noise_pred_noaudio, noise_pred_uncond = self.model(
[latent_model_input, latent_model_input, latent_model_input],
context=[context, context, context_null],
audio_scale = [audio_scale, None, None ],
**kwargs)
if self._interrupt:
return None
else:
noise_pred_cond = self.model( [latent_model_input], context=[context], audio_scale = None if audio_scale == None else [audio_scale], x_id=0, **kwargs, )[0]
if self._interrupt:
return None
if audio_proj != None:
noise_pred_noaudio = self.model(
[latent_model_input],
x_id=1,
context=[context],
**kwargs,
)[0]
if self._interrupt:
return None
noise_pred_uncond = self.model(
[latent_model_input],
x_id=1 if audio_scale == None else 2,
context=[context_null],
**kwargs,
)[0]
if self._interrupt:
return None
del latent_model_input
if guide_scale > 1:
# CFG Zero *. Thanks to https://github.com/WeichenFan/CFG-Zero-star/
if cfg_star_switch:
positive_flat = noise_pred_cond.view(batch_size, -1)
negative_flat = noise_pred_uncond.view(batch_size, -1)
alpha = optimized_scale(positive_flat,negative_flat)
alpha = alpha.view(batch_size, 1, 1, 1)
if (i <= cfg_zero_step):
noise_pred = noise_pred_cond*0. # it would be faster not to compute noise_pred...
else:
noise_pred_uncond *= alpha
if audio_scale == None:
noise_pred = noise_pred_uncond + guide_scale * (noise_pred_cond - noise_pred_uncond)
else:
noise_pred = noise_pred_uncond + guide_scale * (noise_pred_noaudio - noise_pred_uncond) + audio_cfg_scale * (noise_pred_cond - noise_pred_noaudio)
noise_pred_uncond, noise_pred_noaudio = None, None
temp_x0 = sample_scheduler.step(
noise_pred.unsqueeze(0),
t,
latent.unsqueeze(0),
return_dict=False,
generator=seed_g)[0]
latent = temp_x0.squeeze(0)
del temp_x0
del timestep
if callback is not None:
callback(i, latent, False)
x0 = [latent]
video = self.vae.decode(x0, VAE_tile_size, any_end_frame= any_end_frame and add_frames_for_end_image)[0]
if any_end_frame and add_frames_for_end_image:
# video[:, -1:] = img_interpolated2
video = video[:, :-1]
del noise, latent
del sample_scheduler
return video

View File

@ -531,7 +531,7 @@ class CLIPModel:
seq_len=self.model.max_text_len - 2,
clean='whitespace')
def visual(self, videos):
def visual(self, videos,):
# preprocess
size = (self.model.image_size,) * 2
videos = torch.cat([

View File

@ -11,6 +11,7 @@ from typing import Union,Optional
from mmgp import offload
from .attention import pay_attention
from torch.backends.cuda import sdp_kernel
from wan.multitalk.multitalk_utils import get_attn_map_with_target
__all__ = ['WanModel']
@ -175,7 +176,7 @@ class WanSelfAttention(nn.Module):
self.norm_k = WanRMSNorm(dim, eps=eps) if qk_norm else nn.Identity()
def forward(self, xlist, grid_sizes, freqs, block_mask = None):
def forward(self, xlist, grid_sizes, freqs, block_mask = None, ref_target_masks = None, ref_images_count = 0):
r"""
Args:
x(Tensor): Shape [B, L, num_heads, C / num_heads]
@ -190,7 +191,7 @@ class WanSelfAttention(nn.Module):
# query, key, value function
q = self.q(x)
self.norm_q(q)
q = q.view(b, s, n, d) # !!!
q = q.view(b, s, n, d)
k = self.k(x)
self.norm_k(k)
k = k.view(b, s, n, d)
@ -200,6 +201,12 @@ class WanSelfAttention(nn.Module):
del q,k
q,k = apply_rotary_emb(qklist, freqs, head_first=False)
if ref_target_masks != None:
x_ref_attn_map = get_attn_map_with_target(q, k , grid_sizes, ref_target_masks=ref_target_masks, ref_images_count = ref_images_count)
else:
x_ref_attn_map = None
chipmunk = offload.shared_state.get("_chipmunk", False)
if chipmunk and self.__class__ == WanSelfAttention:
q = q.transpose(1,2)
@ -225,30 +232,10 @@ class WanSelfAttention(nn.Module):
)
del q,k,v
# if not self._flag_ar_attention:
# q = rope_apply(q, grid_sizes, freqs)
# k = rope_apply(k, grid_sizes, freqs)
# x = flash_attention(q=q, k=k, v=v, window_size=self.window_size)
# else:
# q = rope_apply(q, grid_sizes, freqs)
# k = rope_apply(k, grid_sizes, freqs)
# q = q.to(torch.bfloat16)
# k = k.to(torch.bfloat16)
# v = v.to(torch.bfloat16)
# with sdp_kernel(enable_flash=True, enable_math=False, enable_mem_efficient=False):
# x = (
# torch.nn.functional.scaled_dot_product_attention(
# q.transpose(1, 2), k.transpose(1, 2), v.transpose(1, 2), attn_mask=block_mask
# )
# .transpose(1, 2)
# .contiguous()
# )
# output
x = x.flatten(2)
x = self.o(x)
return x
return x, x_ref_attn_map
class WanT2VCrossAttention(WanSelfAttention):
@ -375,7 +362,11 @@ class WanAttentionBlock(nn.Module):
cross_attn_norm=False,
eps=1e-6,
block_id=None,
block_no = 0
block_no = 0,
output_dim=0,
norm_input_visual=True,
class_range=24,
class_interval=4,
):
super().__init__()
self.dim = dim
@ -409,6 +400,22 @@ class WanAttentionBlock(nn.Module):
self.modulation = nn.Parameter(torch.randn(1, 6, dim) / dim**0.5)
self.block_id = block_id
if output_dim > 0:
from wan.multitalk.attention import SingleStreamMutiAttention
# init audio module
self.audio_cross_attn = SingleStreamMutiAttention(
dim=dim,
encoder_hidden_states_dim=output_dim,
num_heads=num_heads,
qk_norm=False,
qkv_bias=True,
eps=eps,
norm_layer=WanRMSNorm,
class_range=class_range,
class_interval=class_interval
)
self.norm_x = WanLayerNorm(dim, eps, elementwise_affine=True) if norm_input_visual else nn.Identity()
def forward(
self,
x,
@ -423,6 +430,9 @@ class WanAttentionBlock(nn.Module):
audio_proj= None,
audio_context_lens= None,
audio_scale=None,
multitalk_audio=None,
multitalk_masks=None,
ref_images_count=0,
):
r"""
Args:
@ -466,11 +476,10 @@ class WanAttentionBlock(nn.Module):
xlist = [x_mod.to(attention_dtype)]
del x_mod
y = self.self_attn( xlist, grid_sizes, freqs, block_mask)
y, x_ref_attn_map = self.self_attn( xlist, grid_sizes, freqs, block_mask = block_mask, ref_target_masks = multitalk_masks, ref_images_count = ref_images_count)
y = y.to(dtype)
if cam_emb != None:
y = self.projector(y)
if cam_emb != None: y = self.projector(y)
x, y = reshape_latent(x , latent_frames), reshape_latent(y , latent_frames)
x.addcmul_(y, e[2])
@ -482,6 +491,25 @@ class WanAttentionBlock(nn.Module):
del y
x += self.cross_attn(ylist, context, grid_sizes, audio_proj, audio_scale, audio_context_lens).to(dtype)
if multitalk_audio != None:
# cross attn of multitalk audio
y = self.norm_x(x)
y = y.to(attention_dtype)
if ref_images_count == 0:
x += self.audio_cross_attn(y, encoder_hidden_states=multitalk_audio, shape=grid_sizes, x_ref_attn_map=x_ref_attn_map)
else:
y_shape = y.shape
y = y.reshape(y_shape[0], grid_sizes[0], -1)
y = y[:, ref_images_count:]
y = y.reshape(y_shape[0], -1, y_shape[-1])
grid_sizes_alt = [grid_sizes[0]-ref_images_count, *grid_sizes[1:]]
y = self.audio_cross_attn(y, encoder_hidden_states=multitalk_audio, shape=grid_sizes_alt, x_ref_attn_map=x_ref_attn_map)
y = y.reshape(y_shape[0], grid_sizes[0]-ref_images_count, -1)
x = x.reshape(y_shape[0], grid_sizes[0], -1)
x[:, ref_images_count:] += y
x = x.reshape(y_shape[0], -1, y_shape[-1])
del y
y = self.norm2(x)
y = reshape_latent(y , latent_frames)
@ -518,6 +546,71 @@ class WanAttentionBlock(nn.Module):
x.add_(hint, alpha= scale)
return x
class AudioProjModel(ModelMixin, ConfigMixin):
def __init__(
self,
seq_len=5,
seq_len_vf=12,
blocks=12,
channels=768,
intermediate_dim=512,
output_dim=768,
context_tokens=32,
norm_output_audio=False,
):
super().__init__()
self.seq_len = seq_len
self.blocks = blocks
self.channels = channels
self.input_dim = seq_len * blocks * channels
self.input_dim_vf = seq_len_vf * blocks * channels
self.intermediate_dim = intermediate_dim
self.context_tokens = context_tokens
self.output_dim = output_dim
# define multiple linear layers
self.proj1 = nn.Linear(self.input_dim, intermediate_dim)
self.proj1_vf = nn.Linear(self.input_dim_vf, intermediate_dim)
self.proj2 = nn.Linear(intermediate_dim, intermediate_dim)
self.proj3 = nn.Linear(intermediate_dim, context_tokens * output_dim)
self.norm = nn.LayerNorm(output_dim) if norm_output_audio else nn.Identity()
def forward(self, audio_embeds, audio_embeds_vf):
video_length = audio_embeds.shape[1] + audio_embeds_vf.shape[1]
B, _, _, S, C = audio_embeds.shape
# process audio of first frame
audio_embeds = rearrange(audio_embeds, "bz f w b c -> (bz f) w b c")
batch_size, window_size, blocks, channels = audio_embeds.shape
audio_embeds = audio_embeds.view(batch_size, window_size * blocks * channels)
# process audio of latter frame
audio_embeds_vf = rearrange(audio_embeds_vf, "bz f w b c -> (bz f) w b c")
batch_size_vf, window_size_vf, blocks_vf, channels_vf = audio_embeds_vf.shape
audio_embeds_vf = audio_embeds_vf.view(batch_size_vf, window_size_vf * blocks_vf * channels_vf)
# first projection
audio_embeds = torch.relu(self.proj1(audio_embeds))
audio_embeds_vf = torch.relu(self.proj1_vf(audio_embeds_vf))
audio_embeds = rearrange(audio_embeds, "(bz f) c -> bz f c", bz=B)
audio_embeds_vf = rearrange(audio_embeds_vf, "(bz f) c -> bz f c", bz=B)
audio_embeds_c = torch.concat([audio_embeds, audio_embeds_vf], dim=1)
audio_embeds_vf = audio_embeds = None
batch_size_c, N_t, C_a = audio_embeds_c.shape
audio_embeds_c = audio_embeds_c.view(batch_size_c*N_t, C_a)
# second projection
audio_embeds_c = torch.relu(self.proj2(audio_embeds_c))
context_tokens = self.proj3(audio_embeds_c).reshape(batch_size_c*N_t, self.context_tokens, self.output_dim)
audio_embeds_c = None
# normalization and reshape
context_tokens = self.norm(context_tokens)
context_tokens = rearrange(context_tokens, "(bz f) m c -> bz f m c", f=video_length)
return context_tokens
class VaceWanAttentionBlock(WanAttentionBlock):
@ -595,7 +688,7 @@ class Head(nn.Module):
class MLPProj(torch.nn.Module):
def __init__(self, in_dim, out_dim):
def __init__(self, in_dim, out_dim, flf_pos_emb=False):
super().__init__()
self.proj = torch.nn.Sequential(
@ -603,11 +696,19 @@ class MLPProj(torch.nn.Module):
torch.nn.GELU(), torch.nn.Linear(in_dim, out_dim),
torch.nn.LayerNorm(out_dim))
if flf_pos_emb: # NOTE: we only use this for `flf2v`
FIRST_LAST_FRAME_CONTEXT_TOKEN_NUMBER = 257 * 2
self.emb_pos = nn.Parameter(
torch.zeros(1, FIRST_LAST_FRAME_CONTEXT_TOKEN_NUMBER, 1280))
def forward(self, image_embeds):
if hasattr(self, 'emb_pos'):
bs, n, d = image_embeds.shape
image_embeds = image_embeds.view(-1, 2 * n, d)
image_embeds = image_embeds + self.emb_pos
clip_extra_context_tokens = self.proj(image_embeds)
return clip_extra_context_tokens
class WanModel(ModelMixin, ConfigMixin):
def setup_chipmunk(self):
# from chipmunk.util import LayerCounter
@ -696,45 +797,18 @@ class WanModel(ModelMixin, ConfigMixin):
qk_norm=True,
cross_attn_norm=True,
eps=1e-6,
flf = False,
recammaster = False,
inject_sample_info = False,
fantasytalking_dim = 0,
multitalk_output_dim = 0,
audio_window=5,
intermediate_dim=512,
context_tokens=32,
vae_scale=4, # vae timedownsample scale
norm_input_visual=True,
norm_output_audio=True,
):
r"""
Initialize the diffusion model backbone.
Args:
model_type (`str`, *optional*, defaults to 't2v'):
Model variant - 't2v' (text-to-video) or 'i2v' (image-to-video)
patch_size (`tuple`, *optional*, defaults to (1, 2, 2)):
3D patch dimensions for video embedding (t_patch, h_patch, w_patch)
text_len (`int`, *optional*, defaults to 512):
Fixed length for text embeddings
in_dim (`int`, *optional*, defaults to 16):
Input video channels (C_in)
dim (`int`, *optional*, defaults to 2048):
Hidden dimension of the transformer
ffn_dim (`int`, *optional*, defaults to 8192):
Intermediate dimension in feed-forward network
freq_dim (`int`, *optional*, defaults to 256):
Dimension for sinusoidal time embeddings
text_dim (`int`, *optional*, defaults to 4096):
Input dimension for text embeddings
out_dim (`int`, *optional*, defaults to 16):
Output video channels (C_out)
num_heads (`int`, *optional*, defaults to 16):
Number of attention heads
num_layers (`int`, *optional*, defaults to 32):
Number of transformer blocks
window_size (`tuple`, *optional*, defaults to (-1, -1)):
Window size for local attention (-1 indicates global attention)
qk_norm (`bool`, *optional*, defaults to True):
Enable query/key normalization
cross_attn_norm (`bool`, *optional*, defaults to False):
Enable cross-attention normalization
eps (`float`, *optional*, defaults to 1e-6):
Epsilon value for normalization layers
"""
super().__init__()
@ -760,6 +834,14 @@ class WanModel(ModelMixin, ConfigMixin):
self.block_mask = None
self.inject_sample_info = inject_sample_info
self.norm_output_audio = norm_output_audio
self.audio_window = audio_window
self.intermediate_dim = intermediate_dim
self.vae_scale = vae_scale
multitalk = multitalk_output_dim > 0
self.multitalk = multitalk
# embeddings
self.patch_embedding = nn.Conv3d(
in_dim, dim, kernel_size=patch_size, stride=patch_size)
@ -780,7 +862,7 @@ class WanModel(ModelMixin, ConfigMixin):
cross_attn_type = 't2v_cross_attn' if model_type == 't2v' else 'i2v_cross_attn'
self.blocks = nn.ModuleList([
WanAttentionBlock(cross_attn_type, dim, ffn_dim, num_heads,
window_size, qk_norm, cross_attn_norm, eps, block_no =i)
window_size, qk_norm, cross_attn_norm, eps, block_no =i, output_dim=multitalk_output_dim, norm_input_visual=norm_input_visual)
for i in range(num_layers)
])
@ -790,7 +872,18 @@ class WanModel(ModelMixin, ConfigMixin):
# buffers (don't use register_buffer otherwise dtype will be changed in to())
if model_type == 'i2v':
self.img_emb = MLPProj(1280, dim)
self.img_emb = MLPProj(1280, dim, flf_pos_emb = flf)
if multitalk :
# init audio adapter
self.audio_proj = AudioProjModel(
seq_len=audio_window,
seq_len_vf=audio_window+vae_scale-1,
intermediate_dim=intermediate_dim,
output_dim=multitalk_output_dim,
context_tokens=context_tokens,
norm_output_audio=norm_output_audio,
)
# initialize weights
self.init_weights()
@ -806,7 +899,10 @@ class WanModel(ModelMixin, ConfigMixin):
self.blocks = nn.ModuleList([
WanAttentionBlock('t2v_cross_attn', self.dim, self.ffn_dim, self.num_heads, self.window_size, self.qk_norm,
self.cross_attn_norm, self.eps, block_no =i,
block_id=self.vace_layers_mapping[i] if i in self.vace_layers else None)
block_id=self.vace_layers_mapping[i] if i in self.vace_layers else None,
output_dim=multitalk_output_dim,
norm_input_visual=norm_input_visual,
)
for i in range(self.num_layers)
])
@ -847,6 +943,10 @@ class WanModel(ModelMixin, ConfigMixin):
for block in self.blocks:
layer_list2 += [block.norm3]
if hasattr(self, "audio_proj"):
for block in self.blocks:
layer_list2 += [block.norm_x]
if hasattr(self, "fps_embedding"):
layer_list2 += [self.fps_embedding, self.fps_projection, self.fps_projection[0], self.fps_projection[2]]
@ -1006,6 +1106,9 @@ class WanModel(ModelMixin, ConfigMixin):
audio_proj=None,
audio_context_lens=None,
audio_scale=None,
multitalk_audio = None,
multitalk_masks = None,
ref_images_count = 0,
):
# patch_dtype = self.patch_embedding.weight.dtype
@ -1090,6 +1193,21 @@ class WanModel(ModelMixin, ConfigMixin):
context_clip = self.img_emb(clip_fea) # bs x 257 x dim
context = [ torch.cat( [context_clip, u ], dim=1 ) for u in context ]
if multitalk_audio != None:
multitalk_audio_list = []
for audio in multitalk_audio:
audio = self.audio_proj(*audio)
audio = torch.concat(audio.split(1), dim=2).to(context[0])
multitalk_audio_list.append(audio)
audio = None
else:
multitalk_audio_list = [None] * len(x_list)
if multitalk_masks != None:
multitalk_masks_list = multitalk_masks
else:
multitalk_masks_list = [None] * len(x_list)
context_list = context
if audio_scale != None:
audio_scale_list = audio_scale
@ -1105,6 +1223,7 @@ class WanModel(ModelMixin, ConfigMixin):
block_mask = block_mask,
audio_proj=audio_proj,
audio_context_lens=audio_context_lens,
ref_images_count=ref_images_count,
)
if vace_context == None:
@ -1137,7 +1256,7 @@ class WanModel(ModelMixin, ConfigMixin):
if self.accumulated_err[cur_x_id]<self.magcache_thresh and self.accumulated_steps[cur_x_id]<=self.magcache_K:
skip_forward = True
if i == 0 and x_id == 0: self.cache_skipped_steps += 1
print(f"skip: step={current_step} for x_id={cur_x_id}, accum error {self.accumulated_err[cur_x_id]}")
# print(f"skip: step={current_step} for x_id={cur_x_id}, accum error {self.accumulated_err[cur_x_id]}")
else:
skip_forward = False
self.accumulated_err[cur_x_id], self.accumulated_steps[cur_x_id], self.accumulated_ratio[cur_x_id] = 0, 0, 1.0
@ -1209,11 +1328,11 @@ class WanModel(ModelMixin, ConfigMixin):
continue
x_list[0] = block(x_list[0], context = context_list[0], audio_scale= audio_scale_list[0], e= e0, **kwargs)
else:
for i, (x, context, hints, audio_scale, should_calc) in enumerate(zip(x_list, context_list, hints_list, audio_scale_list, x_should_calc)):
for i, (x, context, hints, audio_scale, multitalk_audio, multitalk_masks, should_calc) in enumerate(zip(x_list, context_list, hints_list, audio_scale_list, multitalk_audio_list, multitalk_masks_list, x_should_calc)):
if should_calc:
x_list[i] = block(x, context = context, hints= hints, audio_scale= audio_scale, e= e0, **kwargs)
x_list[i] = block(x, context = context, hints= hints, audio_scale= audio_scale, multitalk_audio = multitalk_audio, multitalk_masks =multitalk_masks, e= e0, **kwargs)
del x
context, hints = None, None
context = hints = audio_embedding = None
if self.enable_cache != None:
if joint_pass:

View File

@ -11,7 +11,8 @@
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
import matplotlib
matplotlib.use('TkAgg')
import matplotlib.pyplot as plt
from flask import Flask, request, jsonify, render_template
import os
@ -21,7 +22,6 @@ import torch
import yaml
import matplotlib
import argparse
matplotlib.use('Agg')
app = Flask(__name__, static_folder='static', template_folder='templates')

View File

@ -14,6 +14,9 @@ from PIL import Image
import numpy as np
from rembg import remove, new_session
import random
import ffmpeg
import os
import tempfile
__all__ = ['cache_video', 'cache_image', 'str2bool']
@ -33,9 +36,6 @@ def seed_everything(seed: int):
def resample(video_fps, video_frames_count, max_target_frames_count, target_fps, start_target_frame ):
import math
if video_fps < target_fps :
video_fps = target_fps
video_frame_duration = 1 /video_fps
target_frame_duration = 1 / target_fps
@ -160,8 +160,8 @@ def calculate_new_dimensions(canvas_height, canvas_width, height, width, fit_int
new_width = round( width * scale / block_size) * block_size
return new_height, new_width
def resize_and_remove_background(img_list, budget_width, budget_height, rm_background, fit_into_canvas = False ):
if rm_background > 0:
def resize_and_remove_background(img_list, budget_width, budget_height, rm_background, ignore_first, fit_into_canvas = False ):
if rm_background:
session = new_session()
output_list =[]
@ -183,7 +183,7 @@ def resize_and_remove_background(img_list, budget_width, budget_height, rm_backg
new_height = int( round(height * scale / 16) * 16)
new_width = int( round(width * scale / 16) * 16)
resized_image= img.resize((new_width,new_height), resample=Image.Resampling.LANCZOS)
if rm_background == 1 or rm_background == 2 and i > 0 :
if rm_background and not (ignore_first and i == 0) :
# resized_image = remove(resized_image, session=session, alpha_matting_erode_size = 1,alpha_matting_background_threshold = 70, alpha_foreground_background_threshold = 100, alpha_matting = True, bgcolor=[255, 255, 255, 0]).convert('RGB')
resized_image = remove(resized_image, session=session, alpha_matting_erode_size = 1, alpha_matting = True, bgcolor=[255, 255, 255, 0]).convert('RGB')
output_list.append(resized_image) #alpha_matting_background_threshold = 30, alpha_foreground_background_threshold = 200,
@ -406,3 +406,137 @@ def create_progress_hook(filename):
return progress_hook(block_num, block_size, total_size, filename)
return hook
import ffmpeg
import os
import tempfile
def extract_audio_tracks(source_video, verbose=False, query_only= False):
"""
Extract all audio tracks from source video to temporary files.
Args:
source_video: Path to video with audio to extract
verbose: Enable verbose output (default: False)
Returns:
List of temporary audio file paths, or empty list if no audio tracks
"""
try:
# Check if source video has audio
probe = ffmpeg.probe(source_video)
audio_streams = [s for s in probe['streams'] if s['codec_type'] == 'audio']
if not audio_streams:
if query_only: return 0
if verbose:
print(f"No audio track found in {source_video}")
return []
if query_only: return len(audio_streams)
if verbose:
print(f"Found {len(audio_streams)} audio track(s)")
# Create temporary audio files for each track
temp_audio_files = []
for i in range(len(audio_streams)):
fd, temp_path = tempfile.mkstemp(suffix=f'_track{i}.aac', prefix='audio_')
os.close(fd) # Close file descriptor immediately
temp_audio_files.append(temp_path)
# Extract each audio track
for i, temp_path in enumerate(temp_audio_files):
(ffmpeg
.input(source_video)
.output(temp_path, **{f'map': f'0:a:{i}', 'acodec': 'aac'})
.overwrite_output()
.run(quiet=not verbose))
return temp_audio_files
except ffmpeg.Error as e:
print(f"FFmpeg error during audio extraction: {e}")
return []
except Exception as e:
print(f"Error during audio extraction: {e}")
return []
def combine_video_with_audio_tracks(target_video, audio_tracks, output_video, verbose=False):
"""
Combine video with audio tracks. Output duration matches video length exactly.
Args:
target_video: Path to video to receive the audio
audio_tracks: List of audio file paths to combine
output_video: Path for the output video
verbose: Enable verbose output (default: False)
Returns:
True if successful, False otherwise
"""
if not audio_tracks:
if verbose:
print("No audio tracks to combine")
return False
try:
# Get video duration to ensure exact alignment
video_probe = ffmpeg.probe(target_video)
video_duration = float(video_probe['streams'][0]['duration'])
if verbose:
print(f"Target video duration: {video_duration:.3f} seconds")
# Combine target video with all audio tracks, force video duration
video = ffmpeg.input(target_video).video
audio_inputs = [ffmpeg.input(audio_path).audio for audio_path in audio_tracks]
# Create output with video duration as master timing
inputs = [video] + audio_inputs
(ffmpeg
.output(*inputs, output_video,
vcodec='copy',
acodec='copy',
t=video_duration) # Force exact video duration
.overwrite_output()
.run(quiet=not verbose))
if verbose:
print(f"Successfully created {output_video} with {len(audio_tracks)} audio track(s) aligned to video duration")
return True
except ffmpeg.Error as e:
print(f"FFmpeg error during video combination: {e}")
return False
except Exception as e:
print(f"Error during video combination: {e}")
return False
def cleanup_temp_audio_files(audio_tracks, verbose=False):
"""
Clean up temporary audio files.
Args:
audio_tracks: List of audio file paths to delete
verbose: Enable verbose output (default: False)
Returns:
Number of files successfully deleted
"""
deleted_count = 0
for audio_path in audio_tracks:
try:
if os.path.exists(audio_path):
os.unlink(audio_path)
deleted_count += 1
if verbose:
print(f"Cleaned up {audio_path}")
except PermissionError:
print(f"Warning: Could not delete {audio_path} (file may be in use)")
except Exception as e:
print(f"Warning: Error deleting {audio_path}: {e}")
if verbose and deleted_count > 0:
print(f"Successfully deleted {deleted_count} temporary audio file(s)")
return deleted_count

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