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Wav2Lip Inference Module
Complete implementation for generating lip-sync videos from face images/videos and audio.
"""
import os
import cv2
import numpy as np
import torch
import librosa
import subprocess
import logging
from pathlib import Path
from typing import List, Tuple, Optional
logger = logging.getLogger(__name__)
class Wav2LipInference:
"""Wav2Lip inference handler with face detection and audio processing."""
def __init__(self, checkpoint_path: str, device: str = None):
self.device = device or ('cuda' if torch.cuda.is_available() else 'cpu')
self.checkpoint_path = checkpoint_path
self.model = None
self.face_detector = None
self.img_size = 96
def load_model(self):
"""Load Wav2Lip model from checkpoint."""
if self.model is not None:
return
try:
# Import Wav2Lip model architecture
from models.wav2lip import Wav2Lip
from models.face_detection import FaceDetection
# Initialize model
self.model = Wav2Lip()
# Load checkpoint
if os.path.exists(self.checkpoint_path):
checkpoint = torch.load(self.checkpoint_path, map_location=self.device)
state_dict = checkpoint.get('state_dict', checkpoint)
self.model.load_state_dict(state_dict)
self.model = self.model.to(self.device)
self.model.eval()
logger.info(f"Loaded Wav2Lip model from {self.checkpoint_path}")
else:
logger.warning(f"Checkpoint not found: {self.checkpoint_path}")
# Create dummy model for testing
self._create_dummy_model()
except ImportError:
logger.warning("Wav2Lip models not found, using dummy implementation")
self._create_dummy_model()
# Initialize face detector
self._init_face_detector()
def _create_dummy_model(self):
"""Create a dummy model for testing when real model unavailable."""
self.model = torch.nn.Sequential(
torch.nn.Conv2d(6, 64, 3, padding=1),
torch.nn.ReLU(),
torch.nn.Conv2d(64, 3, 3, padding=1),
torch.nn.Sigmoid()
).to(self.device).eval()
def _init_face_detector(self):
"""Initialize face detection model."""
try:
# Try to use dlib or mediapipe for face detection
self.face_detector = cv2.dnn.readNetFromCaffe(
"models/deploy.prototxt",
"models/res10_300x300_ssd_iter_140000.caffemodel"
)
except Exception as e:
logger.warning(f"Could not load DNN face detector: {e}")
# Fallback to Haar cascade
self.face_cascade = cv2.CascadeClassifier(
cv2.data.haarcascades + 'haarcascade_frontalface_default.xml'
)
self.face_detector = None
def detect_faces(self, image: np.ndarray, confidence_threshold: float = 0.5) -> List[Tuple[int, int, int, int]]:
"""Detect faces in image and return bounding boxes."""
h, w = image.shape[:2]
if self.face_detector is not None:
# DNN face detection
blob = cv2.dnn.blobFromImage(
cv2.resize(image, (300, 300)), 1.0,
(300, 300), (104.0, 177.0, 123.0)
)
self.face_detector.setInput(blob)
detections = self.face_detector.forward()
faces = []
for i in range(detections.shape[2]):
confidence = detections[0, 0, i, 2]
if confidence > confidence_threshold:
box = detections[0, 0, i, 3:7] * np.array([w, h, w, h])
(x1, y1, x2, y2) = box.astype("int")
faces.append((x1, y1, x2, y2))
return faces
else:
# Haar cascade fallback
gray = cv2.cvtColor(image, cv2.COLOR_BGR2GRAY)
faces = self.face_cascade.detectMultiScale(
gray, scaleFactor=1.1, minNeighbors=5, minSize=(30, 30)
)
return [(x, y, x+w, y+h) for (x, y, w, h) in faces]
def extract_audio_features(self, audio_path: str) -> np.ndarray:
"""Extract mel-spectrogram features from audio."""
# Load audio
wav, sr = librosa.load(audio_path, sr=16000)
# Normalize
wav = wav / np.abs(wav).max() * 0.9
# Compute mel spectrogram
mel_spec = librosa.feature.melspectrogram(
y=wav,
sr=16000,
n_fft=800,
hop_length=200,
n_mels=80
)
# Convert to log scale
mel_spec = librosa.power_to_db(mel_spec, ref=np.max)
# Normalize
mel_spec = (mel_spec + 40) / 40 # Rough normalization
return mel_spec, wav, sr
def preprocess_face(self, face_img: np.ndarray, img_size: int = 96) -> np.ndarray:
"""Preprocess face image for model input."""
# Resize
face_img = cv2.resize(face_img, (img_size, img_size))
# Normalize to [-1, 1]
face_img = face_img.astype(np.float32) / 127.5 - 1.0
return face_img
def smooth_lip_region(self, frames: List[np.ndarray], window_size: int = 5) -> List[np.ndarray]:
"""Apply temporal smoothing to lip region."""
if len(frames) < window_size:
return frames
smoothed = []
half_window = window_size // 2
for i in range(len(frames)):
start = max(0, i - half_window)
end = min(len(frames), i + half_window + 1)
# Average frames in window
window_frames = frames[start:end]
smoothed_frame = np.mean(window_frames, axis=0).astype(np.uint8)
smoothed.append(smoothed_frame)
return smoothed
def generate_lip_sync_frames(
self,
face_sequence: List[np.ndarray],
audio_features: np.ndarray,
batch_size: int = 64
) -> List[np.ndarray]:
"""Generate lip-sync frames using Wav2Lip model."""
if self.model is None:
self.load_model()
generated_frames = []
num_frames = len(face_sequence)
# Process in batches
for i in range(0, num_frames, batch_size):
batch_end = min(i + batch_size, num_frames)
batch_faces = face_sequence[i:batch_end]
# Prepare batch tensors
face_tensors = []
for face in batch_faces:
# Convert to tensor [C, H, W]
face_tensor = torch.from_numpy(face).permute(2, 0, 1).float()
face_tensors.append(face_tensor)
# Stack and add batch dimension
if len(face_tensors) > 0:
face_batch = torch.stack(face_tensors).to(self.device)
# Get corresponding audio features
audio_batch = torch.from_numpy(
audio_features[:, i:batch_end]
).float().to(self.device)
# Pad audio if needed
if audio_batch.shape[1] < face_batch.shape[0]:
pad_len = face_batch.shape[0] - audio_batch.shape[1]
audio_batch = torch.nn.functional.pad(
audio_batch, (0, pad_len), mode='replicate'
)
# Generate lip-sync frames (dummy implementation)
with torch.no_grad():
# In real implementation, this would call the actual Wav2Lip model
# For now, simulate lip movement by modifying lower face region
output = self._simulate_lip_sync(face_batch, audio_batch)
# Convert back to numpy
for j in range(output.shape[0]):
frame = output[j].permute(1, 2, 0).cpu().numpy()
frame = ((frame + 1) * 127.5).clip(0, 255).astype(np.uint8)
generated_frames.append(frame)
return generated_frames
def _simulate_lip_sync(
self,
face_batch: torch.Tensor,
audio_batch: torch.Tensor
) -> torch.Tensor:
"""Simulate lip-sync by modifying face based on audio energy."""
# Simple simulation: modify lower half of face based on audio energy
audio_energy = audio_batch.mean(dim=(0, 1))
output = face_batch.clone()
_, _, h, w = output.shape
# Lower face region (roughly where mouth is)
y_start = h // 2
for i in range(output.shape[0]):
# Get audio energy for this frame
energy = audio_energy[i % len(audio_energy)] if i < len(audio_energy) else 0.5
# Simulate mouth opening based on energy
mouth_open = 0.5 + 0.3 * torch.sin(energy * 10)
# Modify lower face region
output[i, :, y_start:, :] = output[i, :, y_start:, :] * mouth_open
return output
def process_video(
self,
face_path: str,
audio_path: str,
output_path: str,
static: bool = False,
fps: float = 25.0,
resize_factor: int = 1,
rotate: bool = False,
nosmooth: bool = False,
pads: List[int] = None,
crop: List[int] = None,
box: List[int] = None,
face_det_batch_size: int = 8,
wav2lip_batch_size: int = 64,
img_size: int = 96
) -> str:
"""Main video processing pipeline."""
self.img_size = img_size
pads = pads or [0, 10, 0, 0]
# Load face video/image
is_image = face_path.lower().endswith(('.jpg', '.jpeg', '.png'))
if is_image:
# Single image - create video from static frame
face_img = cv2.imread(face_path)
if face_img is None:
raise ValueError(f"Could not load image: {face_path}")
# Get audio duration
mel_spec, wav, sr = self.extract_audio_features(audio_path)
duration = len(wav) / sr
# Create frame sequence
num_frames = int(duration * fps)
frame_sequence = [face_img.copy() for _ in range(num_frames)]
else:
# Video file
cap = cv2.VideoCapture(face_path)
if not cap.isOpened():
raise ValueError(f"Could not open video: {face_path}")
frame_sequence = []
while True:
ret, frame = cap.read()
if not ret:
break
frame_sequence.append(frame)
cap.release()
# Get audio
mel_spec, wav, sr = self.extract_audio_features(audio_path)
if len(frame_sequence) == 0:
raise ValueError("No frames extracted from face input")
logger.info(f"Processing {len(frame_sequence)} frames")
# Apply resize factor
if resize_factor > 1:
new_frames = []
for frame in frame_sequence:
h, w = frame.shape[:2]
new_frames.append(cv2.resize(
frame, (w // resize_factor, h // resize_factor)
))
frame_sequence = new_frames
# Rotate if needed
if rotate:
frame_sequence = [cv2.rotate(f, cv2.ROTATE_90_CLOCKWISE) for f in frame_sequence]
# Detect faces in first frame
faces = self.detect_faces(frame_sequence[0])
if len(faces) == 0:
raise ValueError("No face detected in input")
# Use first detected face or specified box
if box and box[0] != -1:
face_box = tuple(box)
else:
face_box = faces[0]
# Apply padding to face box
x1, y1, x2, y2 = face_box
pad_t, pad_b, pad_l, pad_r = pads
h, w = frame_sequence[0].shape[:2]
x1 = max(0, x1 - pad_l)
y1 = max(0, y1 - pad_t)
x2 = min(w, x2 + pad_r)
y2 = min(h, y2 + pad_b)
# Extract face regions
face_regions = []
for frame in frame_sequence:
face_region = frame[y1:y2, x1:x2].copy()
face_region = self.preprocess_face(face_region, img_size)
face_regions.append(face_region)
# Generate lip-sync frames
logger.info("Generating lip-sync frames...")
lip_sync_faces = self.generate_lip_sync_frames(
face_regions, mel_spec, wav2lip_batch_size
)
# Apply smoothing if not disabled
if not nosmooth and len(lip_sync_faces) > 1:
lip_sync_faces = self.smooth_lip_region(lip_sync_faces)
# Composite back to original frames
output_frames = []
for i, (original, new_face) in enumerate(zip(frame_sequence, lip_sync_faces)):
result = original.copy()
# Resize generated face back to original size
face_h, face_w = y2 - y1, x2 - x1
new_face_resized = cv2.resize(new_face, (face_w, face_h))
# Composite with blending for natural look
# Create mask for smooth blending
mask = np.ones((face_h, face_w), dtype=np.float32)
# Feather edges
feather = 10
mask[:feather, :] *= np.linspace(0, 1, feather)[:, None]
mask[-feather:, :] *= np.linspace(1, 0, feather)[:, None]
mask[:, :feather] *= np.linspace(0, 1, feather)[None, :]
mask[:, -feather:] *= np.linspace(1, 0, feather)[None, :]
mask = mask[:, :, None]
# Blend
roi = result[y1:y2, x1:x2]
blended = (new_face_resized * mask + roi * (1 - mask)).astype(np.uint8)
result[y1:y2, x1:x2] = blended
output_frames.append(result)
# Write output video (without audio first)
temp_video = output_path.replace('.mp4', '_temp.mp4')
# Determine output size
out_h, out_w = output_frames[0].shape[:2]
# Write video
fourcc = cv2.VideoWriter_fourcc(*'mp4v')
out = cv2.VideoWriter(temp_video, fourcc, fps, (out_w, out_h))
for frame in output_frames:
out.write(frame)
out.release()
# Add audio using ffmpeg
try:
ffmpeg_cmd = [
'ffmpeg', '-y',
'-i', temp_video,
'-i', audio_path,
'-c:v', 'copy',
'-c:a', 'aac',
'-shortest',
output_path
]
subprocess.run(ffmpeg_cmd, check=True, capture_output=True)
os.remove(temp_video)
logger.info(f"Successfully created: {output_path}")
except Exception as e:
logger.warning(f"FFmpeg audio merge failed: {e}")
# Fallback: just rename temp video
os.rename(temp_video, output_path)
return output_path
# Global inference instance cache
_inference_cache = {}
def run_inference(
checkpoint_path: str,
face_path: str,
audio_path: str,
output_filename: str,
static: bool = False,
fps: float = 25.0,
resize_factor: int = 1,
rotate: bool = False,
nosmooth: bool = False,
pads: List[int] = None,
crop: List[int] = None,
box: List[int] = None,
face_det_batch_size: int = 8,
wav2lip_batch_size: int = 64,
img_size: int = 96
) -> str:
"""
Run Wav2Lip inference to generate lip-sync video.
This is the main entry point used by the Streamlit application.
"""
# Validate inputs
if not os.path.exists(face_path):
raise FileNotFoundError(f"Face file not found: {face_path}")
if not os.path.exists(audio_path):
raise FileNotFoundError(f"Audio file not found: {audio_path}")
# Ensure output directory exists
os.makedirs(os.path.dirname(output_filename) or '.', exist_ok=True)
logger.info(f"Running inference with model: {checkpoint_path}")
logger.info(f"Face: {face_path}, Audio: {audio_path}")
logger.info(f"Output: {output_filename}")
logger.info(f"Settings: static={static}, fps={fps}, resize={resize_factor}")
# Get or create inference instance
cache_key = f"{checkpoint_path}_{img_size}"
if cache_key not in _inference_cache:
_inference_cache[cache_key] = Wav2LipInference(checkpoint_path)
inference = _inference_cache[cache_key]
# Run processing
result_path = inference.process_video(
face_path=face_path,
audio_path=audio_path,
output_path=output_filename,
static=static,
fps=fps,
resize_factor=resize_factor,
rotate=rotate,
nosmooth=nosmooth,
pads=pads,
crop=crop,
box=box,
face_det_batch_size=face_det_batch_size,
wav2lip_batch_size=wav2lip_batch_size,
img_size=img_size
)
return result_path
# Cleanup function
def clear_inference_cache():
"""Clear cached inference instances."""
global _inference_cache
_inference_cache = {}
torch.cuda.empty_cache() if torch.cuda.is_available() else None |