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import numpy as np
import re
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 speech_recognition as sr
import json
from nltk.tokenize import sent_tokenize
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 traceback
import wave
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"):
"""
Detects and extracts non-speech (background) segments from audio using pyannote VAD.
Parameters:
- audio_path (str): Path to input audio (.wav).
- segment_audio_path (str): Path to save the output non-speech audio.
- hf_token (str): Hugging Face auth token for pyannote.
Returns:
- List of non-speech timestamp tuples (start, end) in seconds.
"""
# Step 1: Load pipeline
pipeline = Pipeline.from_pretrained("pyannote/voice-activity-detection", use_auth_token=hf_api_key)
# Step 2: Apply VAD to get speech segments
vad_result = pipeline(audio_path)
print("βœ… Speech segments detected.")
# Step 3: Get full duration of the audio
full_audio = AudioSegment.from_wav(audio_path)
full_duration_sec = len(full_audio) / 1000.0
# Step 4: Compute non-speech segments
background_segments = []
current_time = 0.0
for segment in vad_result.itersegments():
if current_time < segment.start:
background_segments.append((current_time, segment.start))
current_time = segment.end
if current_time < full_duration_sec:
background_segments.append((current_time, full_duration_sec))
print(f"πŸ•’ Non-speech segments: {background_segments}")
# Step 5: Extract and combine non-speech segments
non_speech_audio = AudioSegment.empty()
for start, end in background_segments:
segment = full_audio[int(start * 1000):int(end * 1000)]
non_speech_audio += segment
# Step 6: Export the non-speech audio
non_speech_audio.export(background_audio_path, format="wav")
print(f"🎡 Non-speech audio saved to: {background_audio_path}")
return background_segments
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=5, 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["speaker"]
}
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"]
# 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
if aspect_ratio > 1.2: # Portrait video
subtitle_font_size = int(video_width // 18)
else: # Landscape video
subtitle_font_size = int(video_height // 20)
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)
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)
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
img = Image.new("RGBA", (subtitle_width, total_height), (0, 0, 0, 0))
draw = ImageDraw.Draw(img)
y = 0
for idx, line in enumerate(lines):
bbox = draw.textbbox((0, 0), line, font=font)
w = bbox[2] - bbox[0]
draw.text(((subtitle_width - w) // 2, y), line, font=font, fill="yellow")
y += line_heights[idx] + 5
img_np = np.array(img) # <- βœ… Fix: convert to NumPy
txt_clip = ImageClip(img_np).set_start(start_time).set_duration(end_time - start_time).set_position("bottom").set_opacity(0.8)
return txt_clip
except Exception as e:
logger.error(f"\u274c Failed to create subtitle clip: {e}")
return None
def process_entry(entry, i, tts_model, video_width, video_height, process_mode, target_language, font_path, use_clone, 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
if process_mode > 1:
try:
segment_audio_path = f"segment_{i}_voiceover.wav"
desired_duration = entry["end"] - entry["start"]
desired_speed = calibrated_speed(entry['translated'], desired_duration)
speaker = entry.get("speaker", "default")
speaker_wav_path = f"speaker_{speaker}_sample.wav"
if use_clone and speaker_wav_path and os.path.exists(speaker_wav_path):
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)
logger.debug(f"Audio clip duration: {audio_clip.duration}, Desired duration: {desired_duration}")
if audio_clip.duration < desired_duration:
silence_duration = desired_duration - audio_clip.duration
audio_clip = concatenate_audioclips([audio_clip, silence(duration=silence_duration)])
logger.info(f"Padded audio with {silence_duration} seconds of silence.")
audio_segment = audio_clip.set_start(entry["start"]).set_duration(desired_duration)
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, 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 = []
error_messages = []
if process_mode == 3:
global tts_model
if tts_model is None:
try:
print("πŸ”„ Loading XTTS model...")
tts_model = TTS(model_name="tts_models/multilingual/multi-dataset/your_tts")
print("βœ… XTTS model loaded successfully.")
except Exception as e:
print("❌ Error loading XTTS model:")
traceback.print_exc()
return f"Error loading XTTS model: {e}"
## Need to implmenet backup option.
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, use_clone, 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, error = future.result()
results.append((i, txt_clip, audio_segment))
if error:
error_messages.append(f"[Entry {i}] {error}")
except Exception as e:
err = f"❌ Unexpected error in future result: {e}"
logger.error(err)
error_messages.append(err)
# Sort by entry index to ensure order
results.sort(key=lambda x: x[0])
text_clips = [clip for _, clip, _ in results if clip]
if process_mode>1:
audio_segments = [segment for _, _, segment in results if 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])
logger.info("βœ… Background audio loaded and merged with voiceover.")
else:
final_audio = voice_audio
logger.info("⚠️ No background audio found. Using voiceover only.")
final_video = final_video.set_audio(final_audio)
except Exception as e:
logger.error(f"❌ Failed to set audio: {e}")
logger.info(f"Saving the final video to: {output_path}")
final_video.write_videofile(output_path, codec="libx264", audio_codec="aac")
logger.info("Video processing completed successfully.")
if error_messages:
logger.warning("⚠️ Errors encountered during processing:")
for msg in error_messages:
logger.warning(msg)
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 = "βœ… Voice cloning completed successfully."
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 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 > 30:
return 2
else:
slope = (2 - 1.0) / (30 - 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}")
# Step 2: Translate the transcription
logger.info(f"Translating transcription from {source_language} to {target_language}...")
translated_json = translate_text(transcription_json, source_language, target_language)
logger.info(f"Translation completed. Number of translated segments: {len(translated_json)}")
# 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()