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"): """ 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 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") return background_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 = 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(audio_path, chunk_size=6, 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, 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(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 = {} for segment in result["segments"]: speaker = segment["speaker"] if speaker not in speaker_audio: speaker_audio[speaker] = [] speaker_audio[speaker].append((segment["start"], segment["end"])) # Collapse and truncate speaker audio speaker_sample_paths = {} audio_clip = AudioFileClip(audio_path) for speaker, segments in speaker_audio.items(): speaker_clips = [audio_clip.subclip(start, end) for start, end in segments] combined_clip = concatenate_audioclips(speaker_clips) truncated_clip = combined_clip.subclip(0, min(30, combined_clip.duration)) sample_path = f"speaker_{speaker}_sample.wav" truncated_clip.write_audiofile(sample_path) speaker_sample_paths[speaker] = sample_path logger.info(f"Created sample for {speaker}: {sample_path}") # Clean up video.close() audio_clip.close() os.remove(audio_path) 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, 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() ] # Call the function to process the video with updated translations add_transcript_voiceover(file.name, updated_translations, 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]) 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): frame_idx, frame_time, frame = args init_ocr_model() # Ensure model is loaded once per process if frame is None or frame.size == 0: return {"time": frame_time, "text": ""} if not isinstance(frame, np.ndarray): return {"time": frame_time, "text": ""} if frame.dtype != np.uint8: frame = frame.astype(np.uint8) try: subtitle_crop, _ = find_best_subtitle_region(frame, ocr_model) result = ocr_model.ocr(subtitle_crop, cls=True) texts = [line[1][0] for line in result[0]] if result[0] else [] 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 else: collapsed.append(current) current = {"start": time, "end": time, "text": text} if current: collapsed.append(current) # Log collapsed OCR summary 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 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: Refine existing WhisperX segments (Phase 1) merged_segments = [] used_ocr_indices = set() for entry_idx, entry in enumerate(transcription_json): start = entry.get("start", 0) end = entry.get("end", 0) base_text = entry.get("text", "") best_match_idx = None best_score = -1 for ocr_idx, ocr in enumerate(collapsed_ocr): time_overlap = not (ocr["end"] < start - time_tolerance or ocr["start"] > end + time_tolerance) if not time_overlap: continue sim = fuzz.ratio(ocr["text"], base_text) if sim > best_score: best_score = sim best_match_idx = ocr_idx updated_entry = entry.copy() if best_match_idx is not None and best_score >= text_similarity_threshold: updated_entry["text"] = collapsed_ocr[best_match_idx]["text"] updated_entry["ocr_matched"] = True updated_entry["ocr_similarity"] = best_score used_ocr_indices.add(best_match_idx) else: updated_entry["ocr_matched"] = False updated_entry["ocr_similarity"] = best_score if best_score >= 0 else None merged_segments.append(updated_entry) # Step 4: Insert unused OCR segments (Phase 2) inserted_segments = [] for ocr_idx, ocr in enumerate(collapsed_ocr): if ocr_idx in used_ocr_indices: continue # Check for fuzzy duplicates in WhisperX duplicate = False for whisper_seg in transcription_json: if abs(ocr["start"] - whisper_seg["start"]) < time_tolerance or abs(ocr["end"] - whisper_seg["end"]) < time_tolerance: sim = fuzz.ratio(ocr["text"], whisper_seg["text"]) if sim >= text_similarity_threshold: duplicate = True break if duplicate: logger.debug(f"🟡 Skipping near-duplicate OCR: '{ocr['text']}'") continue # Infer speaker from nearest WhisperX entry nearby = sorted(transcription_json, key=lambda x: abs(x["start"] - ocr["start"])) speaker_guess = nearby[0].get("speaker", "unknown") if nearby else "unknown" inserted_segment = { "start": ocr["start"], "end": ocr["end"], "text": ocr["text"], "speaker": speaker_guess } inserted_segments.append(inserted_segment) # Step 5: Combine and sort final_segments = merged_segments + inserted_segments final_segments = sorted(final_segments, key=lambda x: x["start"]) print(f"✅ Post-editing completed: {len(final_segments)} total segments " f"({len(inserted_segments)} OCR-inserted segments)") return final_segments 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) 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] # Align using optimization (modifies translated_json in-place) translated_json = solve_optimal_alignment(translated_json, generated_durations, video.duration) # 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) 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}") transcription_json_merged = post_edit_transcribed_segments(transcription_json, file.name) # Step 2: Translate the transcription logger.info(f"Translating transcription from {source_language} to {target_language}...") translated_json_raw = translate_text(transcription_json_merged, source_language, target_language) 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 editable_table, output_video_path, elapsed_time_display except Exception as e: logger.error(f"An error occurred: {str(e)}") return [], 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") # Language codes 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, process_mode], outputs=[processed_video_output, elapsed_time_display] ) submit_button.click( upload_and_manage, inputs=[file_input, language_input, process_mode], outputs=[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()