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import numpy as np | |
import cvxpy as cp | |
import re | |
import copy | |
import concurrent.futures | |
import gradio as gr | |
from datetime import datetime | |
import random | |
import moviepy | |
from transformers import pipeline | |
from transformers.pipelines.audio_utils import ffmpeg_read | |
from moviepy.editor import ( | |
ImageClip, | |
VideoFileClip, | |
TextClip, | |
CompositeVideoClip, | |
CompositeAudioClip, | |
AudioFileClip, | |
concatenate_videoclips, | |
concatenate_audioclips | |
) | |
from PIL import Image, ImageDraw, ImageFont | |
from moviepy.audio.AudioClip import AudioArrayClip | |
import subprocess | |
import json | |
import logging | |
import whisperx | |
import time | |
import os | |
import openai | |
from openai import OpenAI | |
import traceback | |
from TTS.api import TTS | |
import torch | |
from pydub import AudioSegment | |
from pyannote.audio import Pipeline | |
import wave | |
import librosa | |
import noisereduce as nr | |
import soundfile as sf | |
from paddleocr import PaddleOCR | |
import cv2 | |
from rapidfuzz import fuzz | |
from tqdm import tqdm | |
import threading | |
logger = logging.getLogger(__name__) | |
# Configure logging | |
logging.basicConfig(level=logging.DEBUG, format="%(asctime)s - %(levelname)s - %(message)s") | |
logger = logging.getLogger(__name__) | |
logger.info(f"MoviePy Version: {moviepy.__version__}") | |
# Accept license terms for Coqui XTTS | |
os.environ["COQUI_TOS_AGREED"] = "1" | |
# torch.serialization.add_safe_globals([XttsConfig]) | |
logger.info(gr.__version__) | |
client = OpenAI( | |
api_key= os.environ.get("openAI_api_key"), # This is the default and can be omitted | |
) | |
hf_api_key = os.environ.get("hf_token") | |
def silence(duration, fps=44100): | |
""" | |
Returns a silent AudioClip of the specified duration. | |
""" | |
return AudioArrayClip(np.zeros((int(fps*duration), 2)), fps=fps) | |
def count_words_or_characters(text): | |
# Count non-Chinese words | |
non_chinese_words = len(re.findall(r'\b[a-zA-Z0-9]+\b', text)) | |
# Count Chinese characters | |
chinese_chars = len(re.findall(r'[\u4e00-\u9fff]', text)) | |
return non_chinese_words + chinese_chars | |
# Define the passcode | |
PASSCODE = "show_feedback_db" | |
css = """ | |
/* Adjust row height */ | |
.dataframe-container tr { | |
height: 50px !important; | |
} | |
/* Ensure text wrapping and prevent overflow */ | |
.dataframe-container td { | |
white-space: normal !important; | |
word-break: break-word !important; | |
} | |
/* Set column widths */ | |
[data-testid="block-container"] .scrolling-dataframe th:nth-child(1), | |
[data-testid="block-container"] .scrolling-dataframe td:nth-child(1) { | |
width: 6%; /* Start column */ | |
} | |
[data-testid="block-container"] .scrolling-dataframe th:nth-child(2), | |
[data-testid="block-container"] .scrolling-dataframe td:nth-child(2) { | |
width: 47%; /* Original text */ | |
} | |
[data-testid="block-container"] .scrolling-dataframe th:nth-child(3), | |
[data-testid="block-container"] .scrolling-dataframe td:nth-child(3) { | |
width: 47%; /* Translated text */ | |
} | |
[data-testid="block-container"] .scrolling-dataframe th:nth-child(4), | |
[data-testid="block-container"] .scrolling-dataframe td:nth-child(4) { | |
display: none !important; | |
} | |
""" | |
# Function to save feedback or provide access to the database file | |
def handle_feedback(feedback): | |
feedback = feedback.strip() # Clean up leading/trailing whitespace | |
if not feedback: | |
return "Feedback cannot be empty.", None | |
if feedback == PASSCODE: | |
# Provide access to the feedback.db file | |
return "Access granted! Download the database file below.", "feedback.db" | |
else: | |
# Save feedback to the database | |
with sqlite3.connect("feedback.db") as conn: | |
cursor = conn.cursor() | |
cursor.execute("CREATE TABLE IF NOT EXISTS studio_feedback (id INTEGER PRIMARY KEY, comment TEXT)") | |
cursor.execute("INSERT INTO studio_feedback (comment) VALUES (?)", (feedback,)) | |
conn.commit() | |
return "Thank you for your feedback!", None | |
def segment_background_audio(audio_path, background_audio_path="background_segments.wav", speech_audio_path="speech_segment.wav"): | |
""" | |
Uses Demucs to separate audio and extract background (non-vocal) parts. | |
Merges drums, bass, and other stems into a single background track. | |
""" | |
# Step 1: Run Demucs using the 4-stem model | |
subprocess.run([ | |
"demucs", | |
"-n", "htdemucs", # 4-stem model | |
audio_path | |
], check=True) | |
# Step 2: Locate separated stem files | |
filename = os.path.splitext(os.path.basename(audio_path))[0] | |
stem_dir = os.path.join("separated", "htdemucs", filename) | |
# Step 3: Load and merge background stems | |
vocals = AudioSegment.from_wav(os.path.join(stem_dir, "vocals.wav")) | |
drums = AudioSegment.from_wav(os.path.join(stem_dir, "drums.wav")) | |
bass = AudioSegment.from_wav(os.path.join(stem_dir, "bass.wav")) | |
other = AudioSegment.from_wav(os.path.join(stem_dir, "other.wav")) | |
background = drums.overlay(bass).overlay(other) | |
# Step 4: Export the merged background | |
background.export(background_audio_path, format="wav") | |
vocals.export(speech_audio_path, format="wav") | |
return background_audio_path, speech_audio_path | |
def transcribe_video_with_speakers(video_path): | |
# Extract audio from video | |
video = VideoFileClip(video_path) | |
audio_path = "audio.wav" | |
video.audio.write_audiofile(audio_path) | |
logger.info(f"Audio extracted from video: {audio_path}") | |
segment_result, speech_audio_path = segment_background_audio(audio_path) | |
print(f"Saved non-speech (background) audio to local") | |
# Set up device | |
device = "cuda" if torch.cuda.is_available() else "cpu" | |
logger.info(f"Using device: {device}") | |
try: | |
# Load a medium model with float32 for broader compatibility | |
model = whisperx.load_model("large-v3", device=device, compute_type="float32") | |
logger.info("WhisperX model loaded") | |
# Transcribe | |
result = model.transcribe(speech_audio_path, chunk_size=4, print_progress = True) | |
logger.info("Audio transcription completed") | |
# Get the detected language | |
detected_language = result["language"] | |
logger.debug(f"Detected language: {detected_language}") | |
# Alignment | |
# model_a, metadata = whisperx.load_align_model(language_code=result["language"], device=device) | |
# result = whisperx.align(result["segments"], model_a, metadata, speech_audio_path, device) | |
# logger.info("Transcription alignment completed") | |
# Diarization (works independently of Whisper model size) | |
diarize_model = whisperx.DiarizationPipeline(use_auth_token=hf_api_key, device=device) | |
diarize_segments = diarize_model(speech_audio_path) | |
logger.info("Speaker diarization completed") | |
# Assign speakers | |
result = whisperx.assign_word_speakers(diarize_segments, result) | |
logger.info("Speakers assigned to transcribed segments") | |
except Exception as e: | |
logger.error(f"❌ WhisperX pipeline failed: {e}") | |
# Extract timestamps, text, and speaker IDs | |
transcript_with_speakers = [ | |
{ | |
"start": segment["start"], | |
"end": segment["end"], | |
"text": segment["text"], | |
"speaker": segment.get("speaker", "SPEAKER_00") | |
} | |
for segment in result["segments"] | |
] | |
# Collect audio for each speaker | |
speaker_audio = {} | |
logger.info("🔎 Start collecting valid audio segments per speaker...") | |
for idx, segment in enumerate(result["segments"]): | |
speaker = segment.get("speaker", "SPEAKER_00") | |
start = segment["start"] | |
end = segment["end"] | |
if end > start and (end - start) > 0.05: # Require >50ms duration | |
if speaker not in speaker_audio: | |
speaker_audio[speaker] = [(start, end)] | |
else: | |
speaker_audio[speaker].append((start, end)) | |
logger.debug(f"Segment {idx}: Added to speaker {speaker} [{start:.2f}s → {end:.2f}s]") | |
else: | |
logger.warning(f"⚠️ Segment {idx} discarded: invalid duration ({start:.2f}s → {end:.2f}s)") | |
# Collapse and truncate speaker audio | |
speaker_sample_paths = {} | |
audio_clip = AudioFileClip(speech_audio_path) | |
logger.info(f"🔎 Found {len(speaker_audio)} speakers with valid segments. Start creating speaker samples...") | |
for speaker, segments in speaker_audio.items(): | |
logger.info(f"🔹 Speaker {speaker}: {len(segments)} valid segments") | |
speaker_clips = [audio_clip.subclip(start, end) for start, end in segments] | |
if not speaker_clips: | |
logger.warning(f"⚠️ No valid audio clips for speaker {speaker}. Skipping sample creation.") | |
continue | |
if len(speaker_clips) == 1: | |
logger.debug(f"Speaker {speaker}: Only one clip, skipping concatenation.") | |
combined_clip = speaker_clips[0] | |
else: | |
logger.debug(f"Speaker {speaker}: Concatenating {len(speaker_clips)} clips.") | |
combined_clip = concatenate_audioclips(speaker_clips) | |
truncated_clip = combined_clip.subclip(0, min(30, combined_clip.duration)) | |
logger.debug(f"Speaker {speaker}: Truncated to {truncated_clip.duration:.2f} seconds.") | |
# Step 4: Save the final result | |
sample_path = f"speaker_{speaker}_sample.wav" | |
truncated_clip.write_audiofile(sample_path) | |
speaker_sample_paths[speaker] = sample_path | |
logger.info(f"✅ Created and saved sample for {speaker}: {sample_path}") | |
# Cleanup | |
logger.info("🧹 Closing audio clip and removing temporary files...") | |
video.close() | |
audio_clip.close() | |
os.remove(speech_audio_path) | |
logger.info("✅ Finished processing all speaker samples.") | |
return transcript_with_speakers, detected_language | |
# Function to get the appropriate translation model based on target language | |
# def get_translation_model(source_language, target_language): | |
# """ | |
# Get the translation model based on the source and target language. | |
# Parameters: | |
# - target_language (str): The language to translate the content into (e.g., 'es', 'fr'). | |
# - source_language (str): The language of the input content (default is 'en' for English). | |
# Returns: | |
# - str: The translation model identifier. | |
# """ | |
# # List of allowable languages | |
# allowable_languages = ["en", "es", "fr", "zh", "de", "it", "pt", "ja", "ko", "ru", "hi", "tr"] | |
# # Validate source and target languages | |
# if source_language not in allowable_languages: | |
# logger.debug(f"Invalid source language '{source_language}'. Supported languages are: {', '.join(allowable_languages)}") | |
# # Return a default model if source language is invalid | |
# source_language = "en" # Default to 'en' | |
# if target_language not in allowable_languages: | |
# logger.debug(f"Invalid target language '{target_language}'. Supported languages are: {', '.join(allowable_languages)}") | |
# # Return a default model if target language is invalid | |
# target_language = "zh" # Default to 'zh' | |
# if source_language == target_language: | |
# source_language = "en" # Default to 'en' | |
# target_language = "zh" # Default to 'zh' | |
# # Return the model using string concatenation | |
# return f"Helsinki-NLP/opus-mt-{source_language}-{target_language}" | |
# def translate_single_entry(entry, translator): | |
# original_text = entry["text"] | |
# translated_text = translator(original_text)[0]['translation_text'] | |
# return { | |
# "start": entry["start"], | |
# "original": original_text, | |
# "translated": translated_text, | |
# "end": entry["end"], | |
# "speaker": entry["speaker"] | |
# } | |
# def translate_text(transcription_json, source_language, target_language): | |
# # Load the translation model for the specified target language | |
# translation_model_id = get_translation_model(source_language, target_language) | |
# logger.debug(f"Translation model: {translation_model_id}") | |
# translator = pipeline("translation", model=translation_model_id) | |
# # Use ThreadPoolExecutor to parallelize translations | |
# with concurrent.futures.ThreadPoolExecutor() as executor: | |
# # Submit all translation tasks and collect results | |
# translate_func = lambda entry: translate_single_entry(entry, translator) | |
# translated_json = list(executor.map(translate_func, transcription_json)) | |
# # Sort the translated_json by start time | |
# translated_json.sort(key=lambda x: x["start"]) | |
# # Log the components being added to translated_json | |
# for entry in translated_json: | |
# logger.debug("Added to translated_json: start=%s, original=%s, translated=%s, end=%s, speaker=%s", | |
# entry["start"], entry["original"], entry["translated"], entry["end"], entry["speaker"]) | |
# return translated_json | |
def update_translations(file, edited_table, source_language, target_language, process_mode): | |
""" | |
Update the translations based on user edits in the Gradio Dataframe. | |
""" | |
output_video_path = "output_video.mp4" | |
logger.debug(f"Editable Table: {edited_table}") | |
if file is None: | |
logger.info("No file uploaded. Please upload a video/audio file.") | |
return None, [], None, "No file uploaded. Please upload a video/audio file." | |
try: | |
start_time = time.time() # Start the timer | |
# Convert the edited_table (list of lists) back to list of dictionaries | |
updated_translations = [ | |
{ | |
"start": row["start"], # Access by column name | |
"original": row["original"], | |
"translated": row["translated"], | |
"end": row["end"] | |
} | |
for _, row in edited_table.iterrows() | |
] | |
translated_json = apply_adaptive_speed(updated_translations, source_language, target_language) | |
# Call the function to process the video with updated translations | |
add_transcript_voiceover(file.name, translated_json, output_video_path, process_mode) | |
# Calculate elapsed time | |
elapsed_time = time.time() - start_time | |
elapsed_time_display = f"Updates applied successfully in {elapsed_time:.2f} seconds." | |
return output_video_path, elapsed_time_display | |
except Exception as e: | |
raise ValueError(f"Error updating translations: {e}") | |
def create_subtitle_clip_pil(text, start_time, end_time, video_width, video_height, font_path): | |
try: | |
subtitle_width = int(video_width * 0.8) | |
aspect_ratio = video_height / video_width | |
subtitle_font_size = int(video_width // 22 if aspect_ratio > 1.2 else video_height // 24) | |
font = ImageFont.truetype(font_path, subtitle_font_size) | |
dummy_img = Image.new("RGBA", (subtitle_width, 1), (0, 0, 0, 0)) | |
draw = ImageDraw.Draw(dummy_img) | |
# Word wrapping | |
lines = [] | |
line = "" | |
for word in text.split(): | |
test_line = f"{line} {word}".strip() | |
bbox = draw.textbbox((0, 0), test_line, font=font) | |
w = bbox[2] - bbox[0] | |
if w <= subtitle_width - 10: | |
line = test_line | |
else: | |
lines.append(line) | |
line = word | |
lines.append(line) | |
outline_width=2 | |
line_heights = [draw.textbbox((0, 0), l, font=font)[3] - draw.textbbox((0, 0), l, font=font)[1] for l in lines] | |
total_height = sum(line_heights) + (len(lines) - 1) * 5 + 6 * outline_width | |
img = Image.new("RGBA", (subtitle_width, total_height), (0, 0, 0, 0)) | |
draw = ImageDraw.Draw(img) | |
def draw_text_with_outline(draw, pos, text, font, fill="yellow", outline="black", outline_width = outline_width): | |
x, y = pos | |
# Draw outline | |
for dx in range(-outline_width, outline_width + 1): | |
for dy in range(-outline_width, outline_width + 1): | |
if dx != 0 or dy != 0: | |
draw.text((x + dx, y + dy), text, font=font, fill=outline) | |
# Draw main text | |
draw.text((x, y), text, font=font, fill=fill) | |
y = 0 | |
for idx, line in enumerate(lines): | |
bbox = draw.textbbox((0, 0), line, font=font) | |
w = bbox[2] - bbox[0] | |
x = (subtitle_width - w) // 2 | |
draw_text_with_outline(draw, (x, y), line, font) | |
y += line_heights[idx] + 5 | |
img_np = np.array(img) | |
margin = int(video_height * 0.05) | |
img_clip = ImageClip(img_np) # Create the ImageClip first | |
image_height = img_clip.size[1] | |
txt_clip = ( | |
img_clip # Use the already created clip | |
.set_start(start_time) | |
.set_duration(end_time - start_time) | |
.set_position(("center", video_height - image_height - margin)) | |
.set_opacity(0.9) | |
) | |
return txt_clip | |
except Exception as e: | |
logger.error(f"❌ Failed to create subtitle clip: {e}") | |
return None | |
def solve_optimal_alignment(original_segments, generated_durations, total_duration): | |
""" | |
Aligns speech segments using quadratic programming. If optimization fails, | |
applies greedy fallback: center shorter segments, stretch longer ones. | |
Logs alignment results for traceability. | |
""" | |
N = len(original_segments) | |
d = np.array(generated_durations) | |
m = np.array([(seg['start'] + seg['end']) / 2 for seg in original_segments]) | |
if N == 0 or len(generated_durations) == 0: | |
logger.warning("⚠️ Alignment skipped: empty segments or durations.") | |
return original_segments # or raise an error, depending on your app logic | |
try: | |
s = cp.Variable(N) | |
objective = cp.Minimize(cp.sum_squares(s + d / 2 - m)) | |
constraints = [s[0] >= 0] | |
for i in range(N - 1): | |
constraints.append(s[i] + d[i] <= s[i + 1]) | |
constraints.append(s[N - 1] + d[N - 1] <= total_duration) | |
problem = cp.Problem(objective, constraints) | |
problem.solve() | |
if s.value is None: | |
raise ValueError("Solver failed") | |
for i in range(N): | |
original_segments[i]['start'] = round(s.value[i], 3) | |
original_segments[i]['end'] = round(s.value[i] + d[i], 3) | |
logger.info( | |
f"[OPT] Segment {i}: duration={d[i]:.2f}s | start={original_segments[i]['start']:.2f}s | " | |
f"end={original_segments[i]['end']:.2f}s | mid={m[i]:.2f}s" | |
) | |
except Exception as e: | |
logger.warning(f"⚠️ Optimization failed: {e}, falling back to greedy alignment.") | |
for i in range(N): | |
orig_start = original_segments[i]['start'] | |
orig_end = original_segments[i]['end'] | |
orig_mid = (orig_start + orig_end) / 2 | |
gen_duration = generated_durations[i] | |
orig_duration = orig_end - orig_start | |
if gen_duration <= orig_duration: | |
new_start = orig_mid - gen_duration / 2 | |
new_end = orig_mid + gen_duration / 2 | |
else: | |
extra = (gen_duration - orig_duration) / 2 | |
new_start = orig_start - extra | |
new_end = orig_end + extra | |
if i > 0: | |
prev_end = original_segments[i - 1]['end'] | |
new_start = max(new_start, prev_end + 0.01) | |
if i < N - 1: | |
next_start = original_segments[i + 1]['start'] | |
new_end = min(new_end, next_start - 0.01) | |
if new_end <= new_start: | |
new_start = orig_start | |
new_end = orig_start + gen_duration | |
original_segments[i]['start'] = round(new_start, 3) | |
original_segments[i]['end'] = round(new_end, 3) | |
logger.info( | |
f"[FALLBACK] Segment {i}: duration={gen_duration:.2f}s | start={new_start:.2f}s | " | |
f"end={new_end:.2f}s | original_mid={orig_mid:.2f}s" | |
) | |
return original_segments | |
# ocr_model = None | |
# ocr_lock = threading.Lock() | |
# def init_ocr_model(): | |
# global ocr_model | |
# with ocr_lock: | |
# if ocr_model is None: | |
# ocr_model = PaddleOCR(use_angle_cls=True, lang="ch") | |
# def find_best_subtitle_region(frame, ocr_model, region_height_ratio=0.35, num_strips=5, min_conf=0.5): | |
# """ | |
# Automatically identifies the best subtitle region in a video frame using OCR confidence. | |
# Parameters: | |
# - frame: full video frame (BGR np.ndarray) | |
# - ocr_model: a loaded PaddleOCR model | |
# - region_height_ratio: portion of image height to scan (from bottom up) | |
# - num_strips: how many horizontal strips to evaluate | |
# - min_conf: minimum average confidence to consider a region valid | |
# Returns: | |
# - crop_region: the cropped image region with highest OCR confidence | |
# - region_box: (y_start, y_end) of the region in the original frame | |
# """ | |
# height, width, _ = frame.shape | |
# region_height = int(height * region_height_ratio) | |
# base_y_start = height - region_height | |
# strip_height = region_height // num_strips | |
# best_score = -1 | |
# best_crop = None | |
# best_bounds = (0, height) | |
# for i in range(num_strips): | |
# y_start = base_y_start + i * strip_height | |
# y_end = y_start + strip_height | |
# strip = frame[y_start:y_end, :] | |
# try: | |
# result = ocr_model.ocr(strip, cls=True) | |
# if not result or not result[0]: | |
# continue | |
# total_score = sum(line[1][1] for line in result[0]) | |
# avg_score = total_score / len(result[0]) | |
# if avg_score > best_score: | |
# best_score = avg_score | |
# best_crop = strip | |
# best_bounds = (y_start, y_end) | |
# except Exception as e: | |
# continue # Fail silently on OCR issues | |
# if best_score >= min_conf and best_crop is not None: | |
# return best_crop, best_bounds | |
# else: | |
# # Fallback to center-bottom strip | |
# fallback_y = height - int(height * 0.2) | |
# return frame[fallback_y:, :], (fallback_y, height) | |
# def ocr_frame_worker(args, min_confidence=0.7): | |
# frame_idx, frame_time, frame = args | |
# init_ocr_model() # Load model in thread-safe way | |
# if frame is None or frame.size == 0 or not isinstance(frame, np.ndarray): | |
# return {"time": frame_time, "text": ""} | |
# if frame.dtype != np.uint8: | |
# frame = frame.astype(np.uint8) | |
# try: | |
# result = ocr_model.ocr(frame, cls=True) | |
# lines = result[0] if result else [] | |
# texts = [line[1][0] for line in lines if line[1][1] >= min_confidence] | |
# combined_text = " ".join(texts).strip() | |
# return {"time": frame_time, "text": combined_text} | |
# except Exception as e: | |
# print(f"⚠️ OCR failed at {frame_time:.2f}s: {e}") | |
# return {"time": frame_time, "text": ""} | |
# def frame_is_in_audio_segments(frame_time, audio_segments, tolerance=0.2): | |
# for segment in audio_segments: | |
# start, end = segment["start"], segment["end"] | |
# if (start - tolerance) <= frame_time <= (end + tolerance): | |
# return True | |
# return False | |
# def extract_ocr_subtitles_parallel(video_path, transcription_json, interval_sec=0.5, num_workers=4): | |
# cap = cv2.VideoCapture(video_path) | |
# fps = cap.get(cv2.CAP_PROP_FPS) | |
# frames = [] | |
# frame_idx = 0 | |
# success, frame = cap.read() | |
# while success: | |
# if frame_idx % int(fps * interval_sec) == 0: | |
# frame_time = frame_idx / fps | |
# if frame_is_in_audio_segments(frame_time, transcription_json): | |
# frames.append((frame_idx, frame_time, frame.copy())) | |
# success, frame = cap.read() | |
# frame_idx += 1 | |
# cap.release() | |
# ocr_results = [] | |
# ocr_failures = 0 # Count OCR failures | |
# with concurrent.futures.ThreadPoolExecutor(max_workers=num_workers) as executor: | |
# futures = [executor.submit(ocr_frame_worker, frame) for frame in frames] | |
# for f in tqdm(concurrent.futures.as_completed(futures), total=len(futures)): | |
# try: | |
# result = f.result() | |
# if result["text"]: | |
# ocr_results.append(result) | |
# except Exception as e: | |
# ocr_failures += 1 | |
# logger.info(f"✅ OCR extraction completed: {len(ocr_results)} frames successful, {ocr_failures} frames failed.") | |
# return ocr_results | |
# def collapse_ocr_subtitles(ocr_json, text_similarity_threshold=90): | |
# collapsed = [] | |
# current = None | |
# for entry in ocr_json: | |
# time = entry["time"] | |
# text = entry["text"] | |
# if not current: | |
# current = {"start": time, "end": time, "text": text} | |
# continue | |
# sim = fuzz.ratio(current["text"], text) | |
# if sim >= text_similarity_threshold: | |
# current["end"] = time | |
# logger.debug(f"MERGED: Current end extended to {time:.2f}s for text: '{current['text'][:50]}...' (Similarity: {sim})") | |
# else: | |
# logger.debug(f"NOT MERGING (Similarity: {sim} < Threshold: {text_similarity_threshold}):") | |
# logger.debug(f" Previous segment: {current['start']:.2f}s - {current['end']:.2f}s: '{current['text'][:50]}...'") | |
# logger.debug(f" New segment: {time:.2f}s: '{text[:50]}...'") | |
# collapsed.append(current) | |
# current = {"start": time, "end": time, "text": text} | |
# if current: | |
# collapsed.append(current) | |
# logger.info(f"✅ OCR subtitles collapsed into {len(collapsed)} segments.") | |
# for idx, seg in enumerate(collapsed): | |
# logger.debug(f"[OCR Collapsed {idx}] {seg['start']:.2f}s - {seg['end']:.2f}s: {seg['text'][:50]}...") | |
# return collapsed | |
# def merge_speaker_and_time_from_whisperx( | |
# ocr_json, | |
# whisperx_json, | |
# replace_threshold=90, | |
# time_tolerance=1.0 | |
# ): | |
# merged = [] | |
# used_whisperx = set() | |
# whisperx_used_flags = [False] * len(whisperx_json) | |
# # Step 1: Attempt to match each OCR entry to a WhisperX entry | |
# for ocr in ocr_json: | |
# ocr_start, ocr_end = ocr["start"], ocr["end"] | |
# ocr_text = ocr["text"] | |
# best_match = None | |
# best_score = -1 | |
# best_idx = None | |
# for idx, wx in enumerate(whisperx_json): | |
# wx_start, wx_end = wx["start"], wx["end"] | |
# wx_text = wx["text"] | |
# # Check for time overlap | |
# overlap = not (ocr_end < wx_start - time_tolerance or ocr_start > wx_end + time_tolerance) | |
# if not overlap: | |
# continue | |
# sim = fuzz.ratio(ocr_text, wx_text) | |
# if sim > best_score: | |
# best_score = sim | |
# best_match = wx | |
# best_idx = idx | |
# if best_match and best_score >= replace_threshold: | |
# # Replace WhisperX segment with higher quality OCR text | |
# new_segment = copy.deepcopy(best_match) | |
# new_segment["text"] = ocr_text | |
# new_segment["ocr_replaced"] = True | |
# new_segment["ocr_similarity"] = best_score | |
# whisperx_used_flags[best_idx] = True | |
# merged.append(new_segment) | |
# else: | |
# # No replacement, check if this OCR is outside WhisperX time coverage | |
# covered = any( | |
# abs((ocr_start + ocr_end)/2 - (wx["start"] + wx["end"])/2) < time_tolerance | |
# for wx in whisperx_json | |
# ) | |
# if not covered: | |
# new_segment = copy.deepcopy(ocr) | |
# new_segment["ocr_added"] = True | |
# new_segment["speaker"] = "UNKNOWN" | |
# merged.append(new_segment) | |
# # Step 2: Add untouched WhisperX segments | |
# for idx, wx in enumerate(whisperx_json): | |
# if not whisperx_used_flags[idx]: | |
# merged.append(wx) | |
# # Step 3: Sort all merged segments | |
# merged = sorted(merged, key=lambda x: x["start"]) | |
# return merged | |
# --- Function Definitions --- | |
def process_segment_with_gpt(segment, source_lang, target_lang, model="gpt-4", openai_client=None): | |
""" | |
Processes a single text segment: restores punctuation and translates using an OpenAI GPT model. | |
""" | |
if openai_client is None: | |
segment_identifier = f"{segment.get('start', 'N/A')}-{segment.get('end', 'N/A')}" | |
logger.error(f"❌ OpenAI client was not provided for segment {segment_identifier}. Cannot process.") | |
return { | |
"start": segment.get("start"), | |
"end": segment.get("end"), | |
"speaker": segment.get("speaker", "SPEAKER_00"), | |
"original": segment["text"], | |
"translated": "[ERROR: OpenAI client not provided]" | |
} | |
original_text = segment["text"] | |
segment_id = f"{segment['start']}-{segment['end']}" # Create a unique ID for this segment for easier log tracking | |
logger.debug( | |
f"Starting processing for segment {segment_id}. " | |
f"Original text preview: '{original_text[:100]}{'...' if len(original_text) > 100 else ''}'" | |
) | |
prompt = ( | |
f"You are a multilingual assistant. Given the following text in {source_lang}, " | |
f"1) restore punctuation, and 2) translate it into {target_lang}.\n\n" | |
f"Text:\n{original_text}\n\n" | |
f"Return in JSON format:\n" | |
f'{{"punctuated": "...", "translated": "..."}}' | |
) | |
try: | |
logger.debug(f"Sending request to OpenAI model '{model}' for segment {segment_id}...") | |
response = openai_client.chat.completions.create( | |
model=model, | |
messages=[{"role": "user", "content": prompt}], | |
temperature=0.3 | |
) | |
content = response.choices[0].message.content.strip() | |
# --- NEW LOGIC: Clean markdown code block fences from the response --- | |
cleaned_content = content | |
if content.startswith("```") and content.endswith("```"): | |
# Attempt to find the actual JSON object within the markdown fence | |
json_start_index = content.find('{') | |
json_end_index = content.rfind('}') | |
if json_start_index != -1 and json_end_index != -1 and json_end_index > json_start_index: | |
cleaned_content = content[json_start_index : json_end_index + 1] | |
logger.debug(f"Removed markdown fences for segment {segment_id}. Extracted JSON portion.") | |
else: | |
logger.warning( | |
f"⚠️ Content starts/ends with '```' but a valid JSON object ({{...}}) was not found within " | |
f"fences for segment {segment_id}. Attempting to parse raw content. Raw content: '{content}'" | |
) | |
# --- END NEW LOGIC --- | |
logger.debug( | |
f"Attempting to parse JSON for segment {segment_id}. " | |
f"Content for parsing preview: '{cleaned_content[:200]}{'...' if len(cleaned_content) > 200 else ''}'" | |
) | |
result_json = {} | |
try: | |
result_json = json.loads(cleaned_content) | |
except json.JSONDecodeError as e: | |
logger.warning( | |
f"⚠️ Failed to parse JSON response for segment {segment_id}. Error: {e}. " | |
f"Content attempted to parse: '{cleaned_content}'" # Log cleaned content here | |
) | |
punctuated_text = original_text | |
translated_text = "" # Return empty translated text on parsing failure | |
else: | |
punctuated_text = result_json.get("punctuated", original_text) | |
translated_text = result_json.get("translated", "") | |
logger.info( | |
f"✅ Successfully processed segment {segment_id}. " | |
f"Punctuated preview: '{punctuated_text[:50]}{'...' if len(punctuated_text) > 50 else ''}', " | |
f"Translated preview: '{translated_text[:50]}{'...' if len(translated_text) > 50 else ''}'" | |
) | |
return { | |
"start": segment["start"], | |
"end": segment["end"], | |
"speaker": segment.get("speaker", "SPEAKER_00"), | |
"original": punctuated_text, | |
"translated": translated_text | |
} | |
except Exception as e: | |
logger.error( | |
f"❌ An unexpected error occurred for segment {segment_id}: {e}", | |
exc_info=True # This logs the full traceback | |
) | |
return { | |
"start": segment["start"], | |
"end": segment["end"], | |
"speaker": segment.get("speaker", "SPEAKER_00"), | |
"original": original_text, | |
"translated": "[ERROR: Processing failed]" | |
} | |
def punctuate_and_translate_parallel(transcription_json, source_lang="zh", target_lang="en", model="gpt-4o", max_workers=5, openai_client=None): | |
""" | |
Orchestrates parallel punctuation restoration and translation of multiple segments | |
using a ThreadPoolExecutor. | |
""" | |
if not transcription_json: | |
logger.warning("No segments provided in transcription_json for parallel processing. Returning an empty list.") | |
return [] | |
logger.info(f"Starting parallel punctuation and translation for {len(transcription_json)} segments.") | |
logger.info( | |
f"Configuration: Model='{model}', Source Language='{source_lang}', " | |
f"Target Language='{target_lang}', Max Workers={max_workers}." | |
) | |
results = [] | |
with concurrent.futures.ThreadPoolExecutor(max_workers=max_workers) as executor: | |
# Submit each segment for processing, ensuring the openai_client is passed to each worker | |
futures = { | |
executor.submit(process_segment_with_gpt, seg, source_lang, target_lang, model, openai_client): seg | |
for seg in transcription_json | |
} | |
logger.info(f"All {len(futures)} segments have been submitted to the thread pool.") | |
# Asynchronously collect results as they complete | |
for i, future in enumerate(concurrent.futures.as_completed(futures)): | |
segment = futures[future] # Retrieve the original segment data for logging context | |
segment_id = f"{segment['start']}-{segment['end']}" | |
try: | |
result = future.result() # This will re-raise any exception from the worker thread | |
results.append(result) | |
logger.debug(f"Collected result for segment {segment_id}. ({i + 1}/{len(futures)} completed)") | |
except Exception as exc: | |
# This catch block is for rare cases where the future itself fails to yield a result, | |
# or an exception was not caught within `process_segment_with_gpt`. | |
logger.error( | |
f"Unhandled exception encountered while retrieving result for segment {segment_id}: {exc}", | |
exc_info=True | |
) | |
# Ensure a placeholder result is added even if future retrieval fails | |
results.append({ | |
"start": segment.get("start"), | |
"end": segment.get("end"), | |
"speaker": segment.get("speaker", "SPEAKER_00"), | |
"original": segment["text"], | |
"translated": "[ERROR: Unhandled exception in parallel processing]" | |
}) | |
logger.info("🎉 Parallel processing complete. All results collected.") | |
return results | |
# def merge_speaker_and_time_from_whisperx(ocr_json, whisperx_json, text_sim_threshold=80, replace_threshold=90): | |
# merged = [] | |
# used_whisperx = set() | |
# for ocr in ocr_json: | |
# ocr_start = ocr["start"] | |
# ocr_end = ocr["end"] | |
# ocr_text = ocr["text"] | |
# best_match = None | |
# best_score = -1 | |
# best_idx = None | |
# for idx, wx in enumerate(whisperx_json): | |
# wx_start, wx_end = wx["start"], wx["end"] | |
# wx_text = wx["text"] | |
# if idx in used_whisperx: | |
# continue # Already matched | |
# time_center_diff = abs((ocr_start + ocr_end)/2 - (wx_start + wx_end)/2) | |
# if time_center_diff > 3: | |
# continue | |
# sim = fuzz.ratio(ocr_text, wx_text) | |
# if sim > best_score: | |
# best_score = sim | |
# best_match = wx | |
# best_idx = idx | |
# new_entry = copy.deepcopy(ocr) | |
# if best_match: | |
# new_entry["speaker"] = best_match.get("speaker", "UNKNOWN") | |
# new_entry["ocr_similarity"] = best_score | |
# if best_score >= replace_threshold: | |
# new_entry["start"] = best_match["start"] | |
# new_entry["end"] = best_match["end"] | |
# used_whisperx.add(best_idx) # Mark used | |
# else: | |
# new_entry["speaker"] = "UNKNOWN" | |
# new_entry["ocr_similarity"] = None | |
# merged.append(new_entry) | |
# return merged | |
def realign_ocr_segments(merged_ocr_json, min_gap=0.2): | |
""" | |
Realign OCR segments to avoid overlaps using midpoint-based adjustment. | |
""" | |
merged_ocr_json = sorted(merged_ocr_json, key=lambda x: x["start"]) | |
for i in range(1, len(merged_ocr_json)): | |
prev = merged_ocr_json[i - 1] | |
curr = merged_ocr_json[i] | |
# If current overlaps with previous, adjust | |
if curr["start"] < prev["end"] + min_gap: | |
midpoint = (prev["end"] + curr["start"]) / 2 | |
prev["end"] = round(midpoint - min_gap / 2, 3) | |
curr["start"] = round(midpoint + min_gap / 2, 3) | |
# Prevent negative durations | |
if curr["start"] >= curr["end"]: | |
curr["end"] = round(curr["start"] + 0.3, 3) | |
return merged_ocr_json | |
def post_edit_transcribed_segments(transcription_json, video_path, | |
interval_sec=0.5, | |
text_similarity_threshold=80, | |
time_tolerance=1.0, | |
num_workers=4): | |
""" | |
Given WhisperX transcription (transcription_json) and video, | |
use OCR subtitles to post-correct and safely insert missing captions. | |
""" | |
# Step 1: Extract OCR subtitles (only near audio segments) | |
ocr_json = extract_ocr_subtitles_parallel( | |
video_path, | |
transcription_json, | |
interval_sec=interval_sec, | |
num_workers=num_workers | |
) | |
# Step 2: Collapse repetitive OCR | |
collapsed_ocr = collapse_ocr_subtitles(ocr_json, text_similarity_threshold=90) | |
# Step 3: Merge and realign OCR segments. | |
ocr_merged = merge_speaker_and_time_from_whisperx(collapsed_ocr, transcription_json) | |
ocr_realigned = realign_ocr_segments(ocr_merged) | |
logger.info(f"✅ Final merged and realigned OCR: {len(ocr_realigned)} segments") | |
return ocr_realigned | |
def process_entry(entry, i, tts_model, video_width, video_height, process_mode, target_language, font_path, speaker_sample_paths=None): | |
logger.debug(f"Processing entry {i}: {entry}") | |
error_message = None | |
try: | |
txt_clip = create_subtitle_clip_pil(entry["translated"], entry["start"], entry["end"], video_width, video_height, font_path) | |
except Exception as e: | |
error_message = f"❌ Failed to create subtitle clip for entry {i}: {e}" | |
logger.error(error_message) | |
txt_clip = None | |
audio_segment = None | |
actual_duration = 0.0 | |
if process_mode > 1: | |
try: | |
segment_audio_path = f"segment_{i}_voiceover.wav" | |
desired_duration = entry["end"] - entry["start"] | |
desired_speed = entry['speed'] #calibrated_speed(entry['translated'], desired_duration) | |
speaker = entry.get("speaker", "SPEAKER_00") | |
speaker_wav_path = f"speaker_{speaker}_sample.wav" | |
if process_mode > 2 and speaker_wav_path and os.path.exists(speaker_wav_path) and target_language in tts_model.synthesizer.tts_model.language_manager.name_to_id.keys(): | |
generate_voiceover_clone(entry['translated'], tts_model, desired_speed, target_language, speaker_wav_path, segment_audio_path) | |
else: | |
generate_voiceover_OpenAI(entry['translated'], target_language, desired_speed, segment_audio_path) | |
if not segment_audio_path or not os.path.exists(segment_audio_path): | |
raise FileNotFoundError(f"Voiceover file not generated at: {segment_audio_path}") | |
audio_clip = AudioFileClip(segment_audio_path) | |
actual_duration = audio_clip.duration | |
audio_segment = audio_clip # Do not set start here, alignment happens later | |
except Exception as e: | |
err = f"❌ Failed to generate audio segment for entry {i}: {e}" | |
logger.error(err) | |
error_message = error_message + " | " + err if error_message else err | |
audio_segment = None | |
return i, txt_clip, audio_segment, actual_duration, error_message | |
def add_transcript_voiceover(video_path, translated_json, output_path, process_mode, target_language="en", speaker_sample_paths=None, background_audio_path="background_segments.wav"): | |
video = VideoFileClip(video_path) | |
font_path = "./NotoSansSC-Regular.ttf" | |
text_clips = [] | |
audio_segments = [] | |
actual_durations = [] | |
error_messages = [] | |
if process_mode > 2: | |
global tts_model | |
if tts_model is None: | |
try: | |
print("🔄 Loading XTTS model...") | |
from TTS.api import TTS | |
tts_model = TTS(model_name="tts_models/multilingual/multi-dataset/your_tts") | |
print("✅ XTTS model loaded successfully.") | |
except Exception as e: | |
print("❌ Error loading XTTS model:") | |
traceback.print_exc() | |
return f"Error loading XTTS model: {e}" | |
with concurrent.futures.ThreadPoolExecutor() as executor: | |
# futures = [executor.submit(process_entry, entry, i, tts_model, video.w, video.h, process_mode, target_language, font_path, speaker_sample_paths) | |
# # for i, entry in enumerate(translated_json)] | |
# results = [] | |
# for future in concurrent.futures.as_completed(futures): | |
# try: | |
# i, txt_clip, audio_segment, actual_duration, error = future.result() | |
# results.append((i, txt_clip, audio_segment, actual_duration)) | |
# if error: | |
# error_messages.append(f"[Entry {i}] {error}") | |
# except Exception as e: | |
# err = f"❌ Unexpected error in future result: {e}" | |
# error_messages.append(err) | |
# Use dict as a placeholder, any failure will leave a (None, None, 0) | |
futures = { | |
executor.submit( | |
process_entry, entry, idx, tts_model, video.w, video.h, | |
process_mode, target_language, font_path, speaker_sample_paths | |
): idx | |
for idx, entry in enumerate(translated_json) | |
} | |
# Give each entry a placeholder first to prevent overstepping boundaries | |
result_map = {idx: (None, None, 0) for idx in range(len(translated_json))} | |
for future in concurrent.futures.as_completed(futures): | |
idx = futures[future] | |
try: | |
_idx, txt, aud, dur, err = future.result() | |
result_map[idx] = (txt, aud, dur) | |
if err: | |
error_messages.append(f"[Entry {idx}] {err}") | |
except Exception as e: | |
# Threads that throw errors also need to take up space to prevent the list index from going out of range | |
error_messages.append(f"[Entry {idx}] unexpected error: {e}") | |
# results.sort(key=lambda x: x[0]) | |
# text_clips = [clip for _, clip, _, _ in results if clip] | |
# generated_durations = [dur for _, _, _, dur in results if dur > 0] | |
# Sort and filter together | |
results.sort(key=lambda x: x[0]) | |
filtered = [(translated_json[i], txt, aud, dur) for i, txt, aud, dur in results if dur > 0] | |
translated_json = [entry for entry, _, _, _ in filtered] | |
generated_durations = [dur for _, _, _, dur in filtered] | |
# Align using optimization (modifies translated_json in-place) | |
if generated_durations: | |
translated_json = solve_optimal_alignment(translated_json, generated_durations, video.duration) | |
else: | |
logger.warning("No generated audio; skip alignment optimisation.") | |
# Set aligned timings | |
# audio_segments = [] | |
# for i, entry in enumerate(translated_json): | |
# segment = results[i][2] # AudioFileClip | |
# if segment: | |
# segment = segment.set_start(entry['start']).set_duration(entry['end'] - entry['start']) | |
# audio_segments.append(segment) | |
audio_segments = [] | |
for i, entry in enumerate(translated_json): | |
_, seg, _dur = result_map[i] # seg is AudioFileClip | |
if seg: | |
audio_segments.append( | |
seg.set_start(entry["start"]).set_duration(entry["end"] - entry["start"]) | |
) | |
final_video = CompositeVideoClip([video] + text_clips) | |
if process_mode > 1 and audio_segments: | |
try: | |
voice_audio = CompositeAudioClip(audio_segments).set_duration(video.duration) | |
if background_audio_path and os.path.exists(background_audio_path): | |
background_audio = AudioFileClip(background_audio_path).set_duration(video.duration) | |
final_audio = CompositeAudioClip([voice_audio, background_audio]) | |
else: | |
final_audio = voice_audio | |
final_video = final_video.set_audio(final_audio) | |
except Exception as e: | |
print(f"❌ Failed to set audio: {e}") | |
final_video.write_videofile(output_path, codec="libx264", audio_codec="aac") | |
return error_messages | |
def generate_voiceover_OpenAI(full_text, language, desired_speed, output_audio_path): | |
""" | |
Generate voiceover from translated text for a given language using OpenAI TTS API. | |
""" | |
# Define the voice based on the language (for now, use 'alloy' as default) | |
voice = "alloy" # Adjust based on language if needed | |
# Define the model (use tts-1 for real-time applications) | |
model = "tts-1" | |
max_retries = 3 | |
retry_count = 0 | |
while retry_count < max_retries: | |
try: | |
# Create the speech using OpenAI TTS API | |
response = client.audio.speech.create( | |
model=model, | |
voice=voice, | |
input=full_text, | |
speed=desired_speed | |
) | |
# Save the audio to the specified path | |
with open(output_audio_path, 'wb') as f: | |
for chunk in response.iter_bytes(): | |
f.write(chunk) | |
logging.info(f"Voiceover generated successfully for {output_audio_path}") | |
break | |
except Exception as e: | |
retry_count += 1 | |
logging.error(f"Error generating voiceover (retry {retry_count}/{max_retries}): {e}") | |
time.sleep(5) # Wait 5 seconds before retrying | |
if retry_count == max_retries: | |
raise ValueError(f"Failed to generate voiceover after {max_retries} retries.") | |
def generate_voiceover_clone(full_text, tts_model, desired_speed, target_language, speaker_wav_path, output_audio_path): | |
try: | |
tts_model.tts_to_file( | |
text=full_text, | |
speaker_wav=speaker_wav_path, | |
language=target_language, | |
file_path=output_audio_path, | |
speed=desired_speed, | |
split_sentences=True | |
) | |
msg = ( | |
f"✅ Voice cloning completed successfully. " | |
f"[Speaker Wav: {speaker_wav_path}] [Speed: {desired_speed}]" | |
) | |
logger.info(msg) | |
return output_audio_path, msg, None | |
except Exception as e: | |
generate_voiceover_OpenAI(full_text, target_language, desired_speed, output_audio_path) | |
err_msg = f"❌ An error occurred: {str(e)}, fallback to premium voice" | |
logger.error(traceback.format_exc()) | |
return None, err_msg, err_msg | |
def apply_adaptive_speed(translated_json_raw, source_language, target_language, k=3.0, default_prior_speed=5.0): | |
""" | |
Adds `speed` (relative, 1.0 = normal speed) and `target_duration` (sec) to each segment | |
using shrinkage-based estimation, language stretch ratios, and optional style modifiers. | |
Speeds are clamped to [0.85, 1.7] to avoid unnatural TTS behavior. | |
""" | |
translated_json = copy.deepcopy(translated_json_raw) | |
# Prior average speech speeds by (category, target language) | |
priors = { | |
("drama", "en"): 5.0, | |
("drama", "zh"): 4.5, | |
("tutorial", "en"): 5.2, | |
("tutorial", "zh"): 4.8, | |
("shortplay", "en"): 5.1, | |
("shortplay", "zh"): 4.7, | |
} | |
# Adjustment ratio based on language pair (source → target) | |
lang_ratio = { | |
("zh", "en"): 0.85, | |
("en", "zh"): 1.15, | |
("zh", "jp"): 1.05, | |
("en", "ja"): 0.9, | |
} | |
# Optional style modulation factor | |
style_modifiers = { | |
"dramatic": 0.9, | |
"urgent": 1.1, | |
"neutral": 1.0 | |
} | |
for idx, entry in enumerate(translated_json): | |
start, end = float(entry.get("start", 0)), float(entry.get("end", 0)) | |
duration = max(0.1, end - start) | |
original_text = entry.get("original", "") | |
translated_text = entry.get("translated", "") | |
category = entry.get("category", "drama") | |
source_lang = source_language | |
target_lang = target_language | |
style = entry.get("style", "neutral").lower() | |
# Observed speed from original | |
base_text = original_text or translated_text | |
obs_speed = len(base_text) / duration | |
# Prior speed | |
prior_speed = priors.get((category, target_lang), default_prior_speed) | |
# Shrinkage | |
shrink_speed = (duration * obs_speed + k * prior_speed) / (duration + k) | |
# Language pacing adjustment | |
ratio = lang_ratio.get((source_lang, target_lang), 1.0) | |
adjusted_speed = shrink_speed * ratio | |
# Style modulation | |
mod = style_modifiers.get(style, 1.0) | |
adjusted_speed *= mod | |
# Final relative speed (normalized to prior) | |
relative_speed = adjusted_speed / prior_speed | |
# Clamp relative speed to [0.85, 1.7] | |
relative_speed = max(0.85, min(1.7, relative_speed)) | |
# Compute target duration for synthesis | |
target_chars = len(translated_text) | |
target_duration = round(target_chars / adjusted_speed, 2) | |
# Logging | |
logger.info( | |
f"[Segment {idx}] dur={duration:.2f}s | obs_speed={obs_speed:.2f} | prior={prior_speed:.2f} | " | |
f"shrinked={shrink_speed:.2f} | lang_ratio={ratio} | style_mod={mod} | " | |
f"adj_speed={adjusted_speed:.2f} | rel_speed={relative_speed:.2f} | " | |
f"target_dur={target_duration:.2f}s" | |
) | |
entry["speed"] = round(relative_speed, 3) | |
entry["target_duration"] = target_duration | |
return translated_json | |
def calibrated_speed(text, desired_duration): | |
""" | |
Compute a speed factor to help TTS fit audio into desired duration, | |
using a simple truncated linear function of characters per second. | |
""" | |
char_count = len(text.strip()) | |
if char_count == 0 or desired_duration <= 0: | |
return 1.0 # fallback | |
cps = char_count / desired_duration # characters per second | |
# Truncated linear mapping | |
if cps < 14: | |
return 1.0 | |
elif cps > 25.2: | |
return 1.7 | |
else: | |
slope = (1.7 - 1.0) / (25.2 - 14) | |
return 1.0 + slope * (cps - 14) | |
def upload_and_manage(file, target_language, process_mode): | |
if file is None: | |
logger.info("No file uploaded. Please upload a video/audio file.") | |
return None, [], None, "No file uploaded. Please upload a video/audio file." | |
try: | |
start_time = time.time() # Start the timer | |
logger.info(f"Started processing file: {file.name}") | |
# Define paths for audio and output files | |
audio_path = "audio.wav" | |
output_video_path = "output_video.mp4" | |
voiceover_path = "voiceover.wav" | |
logger.info(f"Using audio path: {audio_path}, output video path: {output_video_path}, voiceover path: {voiceover_path}") | |
# Step 1: Transcribe audio from uploaded media file and get timestamps | |
logger.info("Transcribing audio...") | |
transcription_json, source_language = transcribe_video_with_speakers(file.name) | |
logger.info(f"Transcription completed. Detected source language: {source_language}") | |
translated_json_raw = punctuate_and_translate_parallel(transcription_json, source_language, target_language, openai_client = client) | |
# Step 2: Translate the transcription | |
# logger.info(f"Translating transcription from {source_language} to {target_language}...") | |
# translated_json_raw = translate_text(transcription_json_merged, ) | |
logger.info(f"Translation completed. Number of translated segments: {len(translated_json_raw)}") | |
translated_json = apply_adaptive_speed(translated_json_raw, source_language, target_language) | |
# Step 3: Add transcript to video based on timestamps | |
logger.info("Adding translated transcript to video...") | |
add_transcript_voiceover(file.name, translated_json, output_video_path, process_mode, target_language) | |
logger.info(f"Transcript added to video. Output video saved at {output_video_path}") | |
# Convert translated JSON into a format for the editable table | |
logger.info("Converting translated JSON into editable table format...") | |
editable_table = [ | |
[float(entry["start"]), entry["original"], entry["translated"], float(entry["end"]), entry["speaker"]] | |
for entry in translated_json | |
] | |
# Calculate elapsed time | |
elapsed_time = time.time() - start_time | |
elapsed_time_display = f"Processing completed in {elapsed_time:.2f} seconds." | |
logger.info(f"Processing completed in {elapsed_time:.2f} seconds.") | |
return source_language, editable_table, output_video_path, elapsed_time_display | |
except Exception as e: | |
logger.error(f"An error occurred: {str(e)}") | |
return None, [], None, f"An error occurred: {str(e)}" | |
# Gradio Interface with Tabs | |
def build_interface(): | |
with gr.Blocks(css=css) as demo: | |
gr.Markdown("## Video Localization") | |
with gr.Row(): | |
with gr.Column(scale=4): | |
file_input = gr.File(label="Upload Video/Audio File") | |
language_input = gr.Dropdown(["en", "es", "fr", "zh"], label="Select Language") | |
source_language_display = gr.Textbox(label="Detected Source Language", interactive=False) | |
process_mode = gr.Radio( | |
choices=[("Transcription Only", 1), ("Transcription with Premium Voice", 2), ("Transcription with Voice Clone", 3)], | |
label="Choose Processing Type", | |
value=1 | |
) | |
submit_button = gr.Button("Post and Process") | |
with gr.Column(scale=8): | |
gr.Markdown("## Edit Translations") | |
# Editable JSON Data | |
editable_table = gr.Dataframe( | |
value=[], # Default to an empty list to avoid undefined values | |
headers=["start", "original", "translated", "end", "speaker"], | |
datatype=["number", "str", "str", "number", "str"], | |
row_count=1, # Initially empty | |
col_count=5, | |
interactive=[False, True, True, False, False], # Control editability | |
label="Edit Translations", | |
wrap=True # Enables text wrapping if supported | |
) | |
save_changes_button = gr.Button("Save Changes") | |
processed_video_output = gr.File(label="Download Processed Video", interactive=True) # Download button | |
elapsed_time_display = gr.Textbox(label="Elapsed Time", lines=1, interactive=False) | |
with gr.Column(scale=1): | |
gr.Markdown("**Feedback**") | |
feedback_input = gr.Textbox( | |
placeholder="Leave your feedback here...", | |
label=None, | |
lines=3, | |
) | |
feedback_btn = gr.Button("Submit Feedback") | |
response_message = gr.Textbox(label=None, lines=1, interactive=False) | |
db_download = gr.File(label="Download Database File", visible=False) | |
# Link the feedback handling | |
def feedback_submission(feedback): | |
message, file_path = handle_feedback(feedback) | |
if file_path: | |
return message, gr.update(value=file_path, visible=True) | |
return message, gr.update(visible=False) | |
save_changes_button.click( | |
update_translations, | |
inputs=[file_input, editable_table, source_language_display, language_input, process_mode], | |
outputs=[processed_video_output, elapsed_time_display] | |
) | |
submit_button.click( | |
upload_and_manage, | |
inputs=[file_input, language_input, process_mode], | |
outputs=[source_language_display, editable_table, processed_video_output, elapsed_time_display] | |
) | |
# Connect submit button to save_feedback_db function | |
feedback_btn.click( | |
feedback_submission, | |
inputs=[feedback_input], | |
outputs=[response_message, db_download] | |
) | |
return demo | |
tts_model = None | |
# Launch the Gradio interface | |
demo = build_interface() | |
demo.launch() |