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Browse files- .gitattributes +2 -0
- NotoSansSC-Regular.ttf +3 -0
- README.md +5 -4
- app.py +741 -0
- apt.txt +9 -0
- requirements.txt +22 -0
- speaker_default_sample.wav +3 -0
.gitattributes
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@@ -33,3 +33,5 @@ saved_model/**/* filter=lfs diff=lfs merge=lfs -text
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*.zip filter=lfs diff=lfs merge=lfs -text
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*.zst filter=lfs diff=lfs merge=lfs -text
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*tfevents* filter=lfs diff=lfs merge=lfs -text
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*.zip filter=lfs diff=lfs merge=lfs -text
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*.zst filter=lfs diff=lfs merge=lfs -text
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*tfevents* filter=lfs diff=lfs merge=lfs -text
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NotoSansSC-Regular.ttf filter=lfs diff=lfs merge=lfs -text
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speaker_default_sample.wav filter=lfs diff=lfs merge=lfs -text
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NotoSansSC-Regular.ttf
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version https://git-lfs.github.com/spec/v1
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oid sha256:5cf8b2a0576d5680284ab03a7a8219499d59bbe981a79bb3dc0031f251c39736
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size 10560616
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README.md
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@@ -1,12 +1,13 @@
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---
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-
title:
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emoji:
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colorFrom:
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colorTo:
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sdk: gradio
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sdk_version: 5.23.3
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app_file: app.py
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pinned: false
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---
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Check out the configuration reference at https://huggingface.co/docs/hub/spaces-config-reference
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---
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title: studio_V1
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emoji: 🔥
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colorFrom: pink
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colorTo: red
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sdk: gradio
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sdk_version: 5.23.3
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app_file: app.py
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pinned: false
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short_description: Studio
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---
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Check out the configuration reference at https://huggingface.co/docs/hub/spaces-config-reference
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app.py
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1 |
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import numpy as np
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2 |
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import cvxpy as cp
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3 |
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import re
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4 |
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import concurrent.futures
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5 |
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import gradio as gr
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6 |
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from datetime import datetime
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7 |
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import random
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8 |
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import moviepy
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9 |
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from transformers import pipeline
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10 |
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from transformers.pipelines.audio_utils import ffmpeg_read
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from moviepy.editor import (
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ImageClip,
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VideoFileClip,
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14 |
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TextClip,
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15 |
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CompositeVideoClip,
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16 |
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CompositeAudioClip,
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17 |
+
AudioFileClip,
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18 |
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concatenate_videoclips,
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19 |
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concatenate_audioclips
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20 |
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)
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21 |
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from PIL import Image, ImageDraw, ImageFont
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from moviepy.audio.AudioClip import AudioArrayClip
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23 |
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import subprocess
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import speech_recognition as sr
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import json
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26 |
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from nltk.tokenize import sent_tokenize
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27 |
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import logging
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28 |
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import whisperx
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import time
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30 |
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import os
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31 |
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import openai
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32 |
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from openai import OpenAI
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33 |
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import traceback
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34 |
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from TTS.api import TTS
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35 |
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import torch
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36 |
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from pydub import AudioSegment
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37 |
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from pyannote.audio import Pipeline
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38 |
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import traceback
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39 |
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import wave
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40 |
+
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logger = logging.getLogger(__name__)
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42 |
+
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43 |
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# Configure logging
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44 |
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logging.basicConfig(level=logging.DEBUG, format="%(asctime)s - %(levelname)s - %(message)s")
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45 |
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logger = logging.getLogger(__name__)
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46 |
+
logger.info(f"MoviePy Version: {moviepy.__version__}")
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47 |
+
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48 |
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# Accept license terms for Coqui XTTS
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49 |
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os.environ["COQUI_TOS_AGREED"] = "1"
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50 |
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# torch.serialization.add_safe_globals([XttsConfig])
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51 |
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52 |
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logger.info(gr.__version__)
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53 |
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54 |
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client = OpenAI(
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api_key= os.environ.get("openAI_api_key"), # This is the default and can be omitted
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56 |
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)
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57 |
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hf_api_key = os.environ.get("hf_token")
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58 |
+
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59 |
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def silence(duration, fps=44100):
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"""
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61 |
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Returns a silent AudioClip of the specified duration.
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62 |
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"""
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63 |
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return AudioArrayClip(np.zeros((int(fps*duration), 2)), fps=fps)
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64 |
+
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65 |
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def count_words_or_characters(text):
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# Count non-Chinese words
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67 |
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non_chinese_words = len(re.findall(r'\b[a-zA-Z0-9]+\b', text))
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68 |
+
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69 |
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# Count Chinese characters
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70 |
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chinese_chars = len(re.findall(r'[\u4e00-\u9fff]', text))
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71 |
+
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72 |
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return non_chinese_words + chinese_chars
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73 |
+
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74 |
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# Define the passcode
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75 |
+
PASSCODE = "show_feedback_db"
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76 |
+
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77 |
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css = """
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78 |
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/* Adjust row height */
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79 |
+
.dataframe-container tr {
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80 |
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height: 50px !important;
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81 |
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}
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82 |
+
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83 |
+
/* Ensure text wrapping and prevent overflow */
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84 |
+
.dataframe-container td {
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85 |
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white-space: normal !important;
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86 |
+
word-break: break-word !important;
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87 |
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}
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88 |
+
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89 |
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/* Set column widths */
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90 |
+
[data-testid="block-container"] .scrolling-dataframe th:nth-child(1),
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91 |
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[data-testid="block-container"] .scrolling-dataframe td:nth-child(1) {
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92 |
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width: 6%; /* Start column */
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93 |
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}
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94 |
+
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95 |
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[data-testid="block-container"] .scrolling-dataframe th:nth-child(2),
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96 |
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[data-testid="block-container"] .scrolling-dataframe td:nth-child(2) {
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97 |
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width: 47%; /* Original text */
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98 |
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}
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99 |
+
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100 |
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[data-testid="block-container"] .scrolling-dataframe th:nth-child(3),
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101 |
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[data-testid="block-container"] .scrolling-dataframe td:nth-child(3) {
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102 |
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width: 47%; /* Translated text */
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103 |
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}
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104 |
+
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105 |
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[data-testid="block-container"] .scrolling-dataframe th:nth-child(4),
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[data-testid="block-container"] .scrolling-dataframe td:nth-child(4) {
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107 |
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display: none !important;
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108 |
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}
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109 |
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"""
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110 |
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111 |
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# Function to save feedback or provide access to the database file
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112 |
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def handle_feedback(feedback):
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113 |
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feedback = feedback.strip() # Clean up leading/trailing whitespace
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114 |
+
if not feedback:
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115 |
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return "Feedback cannot be empty.", None
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116 |
+
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117 |
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if feedback == PASSCODE:
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118 |
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# Provide access to the feedback.db file
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119 |
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return "Access granted! Download the database file below.", "feedback.db"
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120 |
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else:
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121 |
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# Save feedback to the database
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122 |
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with sqlite3.connect("feedback.db") as conn:
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123 |
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cursor = conn.cursor()
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124 |
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cursor.execute("CREATE TABLE IF NOT EXISTS studio_feedback (id INTEGER PRIMARY KEY, comment TEXT)")
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125 |
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cursor.execute("INSERT INTO studio_feedback (comment) VALUES (?)", (feedback,))
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126 |
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conn.commit()
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127 |
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return "Thank you for your feedback!", None
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128 |
+
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129 |
+
def segment_background_audio(audio_path, background_audio_path="background_segments.wav"):
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130 |
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pipeline = Pipeline.from_pretrained("pyannote/voice-activity-detection", use_auth_token=hf_api_key)
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131 |
+
vad_result = pipeline(audio_path)
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132 |
+
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133 |
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full_audio = AudioSegment.from_wav(audio_path)
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134 |
+
full_duration_sec = len(full_audio) / 1000.0
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135 |
+
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136 |
+
current_time = 0.0
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137 |
+
result_audio = AudioSegment.empty()
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138 |
+
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139 |
+
for segment in vad_result.itersegments():
|
140 |
+
# Background segment before the speech
|
141 |
+
if current_time < segment.start:
|
142 |
+
bg = full_audio[int(current_time * 1000):int(segment.start * 1000)]
|
143 |
+
result_audio += bg
|
144 |
+
# Add silence for the speech duration
|
145 |
+
silence_duration = segment.end - segment.start
|
146 |
+
result_audio += AudioSegment.silent(duration=int(silence_duration * 1000))
|
147 |
+
current_time = segment.end
|
148 |
+
|
149 |
+
# Handle any remaining background after the last speech
|
150 |
+
if current_time < full_duration_sec:
|
151 |
+
result_audio += full_audio[int(current_time * 1000):]
|
152 |
+
|
153 |
+
result_audio.export(background_audio_path, format="wav")
|
154 |
+
return background_audio_path
|
155 |
+
|
156 |
+
def transcribe_video_with_speakers(video_path):
|
157 |
+
# Extract audio from video
|
158 |
+
video = VideoFileClip(video_path)
|
159 |
+
audio_path = "audio.wav"
|
160 |
+
video.audio.write_audiofile(audio_path)
|
161 |
+
logger.info(f"Audio extracted from video: {audio_path}")
|
162 |
+
|
163 |
+
segment_result = segment_background_audio(audio_path)
|
164 |
+
print(f"Saved non-speech (background) audio to local")
|
165 |
+
|
166 |
+
# Set up device
|
167 |
+
device = "cuda" if torch.cuda.is_available() else "cpu"
|
168 |
+
logger.info(f"Using device: {device}")
|
169 |
+
|
170 |
+
try:
|
171 |
+
# Load a medium model with float32 for broader compatibility
|
172 |
+
model = whisperx.load_model("large-v3", device=device, compute_type="float32")
|
173 |
+
logger.info("WhisperX model loaded")
|
174 |
+
|
175 |
+
# Transcribe
|
176 |
+
result = model.transcribe(audio_path, chunk_size=6, print_progress = True)
|
177 |
+
logger.info("Audio transcription completed")
|
178 |
+
|
179 |
+
# Get the detected language
|
180 |
+
detected_language = result["language"]
|
181 |
+
logger.debug(f"Detected language: {detected_language}")
|
182 |
+
# Alignment
|
183 |
+
model_a, metadata = whisperx.load_align_model(language_code=result["language"], device=device)
|
184 |
+
result = whisperx.align(result["segments"], model_a, metadata, audio_path, device)
|
185 |
+
logger.info("Transcription alignment completed")
|
186 |
+
|
187 |
+
# Diarization (works independently of Whisper model size)
|
188 |
+
diarize_model = whisperx.DiarizationPipeline(use_auth_token=hf_api_key, device=device)
|
189 |
+
diarize_segments = diarize_model(audio_path)
|
190 |
+
logger.info("Speaker diarization completed")
|
191 |
+
|
192 |
+
# Assign speakers
|
193 |
+
result = whisperx.assign_word_speakers(diarize_segments, result)
|
194 |
+
logger.info("Speakers assigned to transcribed segments")
|
195 |
+
|
196 |
+
except Exception as e:
|
197 |
+
logger.error(f"❌ WhisperX pipeline failed: {e}")
|
198 |
+
|
199 |
+
# Extract timestamps, text, and speaker IDs
|
200 |
+
transcript_with_speakers = [
|
201 |
+
{
|
202 |
+
"start": segment["start"],
|
203 |
+
"end": segment["end"],
|
204 |
+
"text": segment["text"],
|
205 |
+
"speaker": segment["speaker"]
|
206 |
+
}
|
207 |
+
for segment in result["segments"]
|
208 |
+
]
|
209 |
+
|
210 |
+
# Collect audio for each speaker
|
211 |
+
speaker_audio = {}
|
212 |
+
for segment in result["segments"]:
|
213 |
+
speaker = segment["speaker"]
|
214 |
+
if speaker not in speaker_audio:
|
215 |
+
speaker_audio[speaker] = []
|
216 |
+
speaker_audio[speaker].append((segment["start"], segment["end"]))
|
217 |
+
|
218 |
+
# Collapse and truncate speaker audio
|
219 |
+
speaker_sample_paths = {}
|
220 |
+
audio_clip = AudioFileClip(audio_path)
|
221 |
+
for speaker, segments in speaker_audio.items():
|
222 |
+
speaker_clips = [audio_clip.subclip(start, end) for start, end in segments]
|
223 |
+
combined_clip = concatenate_audioclips(speaker_clips)
|
224 |
+
truncated_clip = combined_clip.subclip(0, min(30, combined_clip.duration))
|
225 |
+
sample_path = f"speaker_{speaker}_sample.wav"
|
226 |
+
truncated_clip.write_audiofile(sample_path)
|
227 |
+
speaker_sample_paths[speaker] = sample_path
|
228 |
+
logger.info(f"Created sample for {speaker}: {sample_path}")
|
229 |
+
|
230 |
+
# Clean up
|
231 |
+
video.close()
|
232 |
+
audio_clip.close()
|
233 |
+
os.remove(audio_path)
|
234 |
+
|
235 |
+
return transcript_with_speakers, detected_language
|
236 |
+
|
237 |
+
# Function to get the appropriate translation model based on target language
|
238 |
+
def get_translation_model(source_language, target_language):
|
239 |
+
"""
|
240 |
+
Get the translation model based on the source and target language.
|
241 |
+
|
242 |
+
Parameters:
|
243 |
+
- target_language (str): The language to translate the content into (e.g., 'es', 'fr').
|
244 |
+
- source_language (str): The language of the input content (default is 'en' for English).
|
245 |
+
|
246 |
+
Returns:
|
247 |
+
- str: The translation model identifier.
|
248 |
+
"""
|
249 |
+
# List of allowable languages
|
250 |
+
allowable_languages = ["en", "es", "fr", "zh", "de", "it", "pt", "ja", "ko", "ru"]
|
251 |
+
|
252 |
+
# Validate source and target languages
|
253 |
+
if source_language not in allowable_languages:
|
254 |
+
logger.debug(f"Invalid source language '{source_language}'. Supported languages are: {', '.join(allowable_languages)}")
|
255 |
+
# Return a default model if source language is invalid
|
256 |
+
source_language = "en" # Default to 'en'
|
257 |
+
|
258 |
+
if target_language not in allowable_languages:
|
259 |
+
logger.debug(f"Invalid target language '{target_language}'. Supported languages are: {', '.join(allowable_languages)}")
|
260 |
+
# Return a default model if target language is invalid
|
261 |
+
target_language = "zh" # Default to 'zh'
|
262 |
+
|
263 |
+
if source_language == target_language:
|
264 |
+
source_language = "en" # Default to 'en'
|
265 |
+
target_language = "zh" # Default to 'zh'
|
266 |
+
|
267 |
+
# Return the model using string concatenation
|
268 |
+
return f"Helsinki-NLP/opus-mt-{source_language}-{target_language}"
|
269 |
+
|
270 |
+
def translate_single_entry(entry, translator):
|
271 |
+
original_text = entry["text"]
|
272 |
+
translated_text = translator(original_text)[0]['translation_text']
|
273 |
+
return {
|
274 |
+
"start": entry["start"],
|
275 |
+
"original": original_text,
|
276 |
+
"translated": translated_text,
|
277 |
+
"end": entry["end"],
|
278 |
+
"speaker": entry["speaker"]
|
279 |
+
}
|
280 |
+
|
281 |
+
def translate_text(transcription_json, source_language, target_language):
|
282 |
+
# Load the translation model for the specified target language
|
283 |
+
translation_model_id = get_translation_model(source_language, target_language)
|
284 |
+
logger.debug(f"Translation model: {translation_model_id}")
|
285 |
+
translator = pipeline("translation", model=translation_model_id)
|
286 |
+
|
287 |
+
# Use ThreadPoolExecutor to parallelize translations
|
288 |
+
with concurrent.futures.ThreadPoolExecutor() as executor:
|
289 |
+
# Submit all translation tasks and collect results
|
290 |
+
translate_func = lambda entry: translate_single_entry(entry, translator)
|
291 |
+
translated_json = list(executor.map(translate_func, transcription_json))
|
292 |
+
|
293 |
+
# Sort the translated_json by start time
|
294 |
+
translated_json.sort(key=lambda x: x["start"])
|
295 |
+
|
296 |
+
# Log the components being added to translated_json
|
297 |
+
for entry in translated_json:
|
298 |
+
logger.debug("Added to translated_json: start=%s, original=%s, translated=%s, end=%s, speaker=%s",
|
299 |
+
entry["start"], entry["original"], entry["translated"], entry["end"], entry["speaker"])
|
300 |
+
|
301 |
+
return translated_json
|
302 |
+
|
303 |
+
def update_translations(file, edited_table, process_mode):
|
304 |
+
"""
|
305 |
+
Update the translations based on user edits in the Gradio Dataframe.
|
306 |
+
"""
|
307 |
+
output_video_path = "output_video.mp4"
|
308 |
+
logger.debug(f"Editable Table: {edited_table}")
|
309 |
+
|
310 |
+
if file is None:
|
311 |
+
logger.info("No file uploaded. Please upload a video/audio file.")
|
312 |
+
return None, [], None, "No file uploaded. Please upload a video/audio file."
|
313 |
+
|
314 |
+
try:
|
315 |
+
start_time = time.time() # Start the timer
|
316 |
+
|
317 |
+
# Convert the edited_table (list of lists) back to list of dictionaries
|
318 |
+
updated_translations = [
|
319 |
+
{
|
320 |
+
"start": row["start"], # Access by column name
|
321 |
+
"original": row["original"],
|
322 |
+
"translated": row["translated"],
|
323 |
+
"end": row["end"]
|
324 |
+
}
|
325 |
+
for _, row in edited_table.iterrows()
|
326 |
+
]
|
327 |
+
|
328 |
+
# Call the function to process the video with updated translations
|
329 |
+
add_transcript_voiceover(file.name, updated_translations, output_video_path, process_mode)
|
330 |
+
|
331 |
+
# Calculate elapsed time
|
332 |
+
elapsed_time = time.time() - start_time
|
333 |
+
elapsed_time_display = f"Updates applied successfully in {elapsed_time:.2f} seconds."
|
334 |
+
|
335 |
+
return output_video_path, elapsed_time_display
|
336 |
+
|
337 |
+
except Exception as e:
|
338 |
+
raise ValueError(f"Error updating translations: {e}")
|
339 |
+
|
340 |
+
def create_subtitle_clip_pil(text, start_time, end_time, video_width, video_height, font_path):
|
341 |
+
try:
|
342 |
+
subtitle_width = int(video_width * 0.8)
|
343 |
+
aspect_ratio = video_height / video_width
|
344 |
+
if aspect_ratio > 1.2: # Portrait video
|
345 |
+
subtitle_font_size = int(video_width // 22)
|
346 |
+
else: # Landscape video
|
347 |
+
subtitle_font_size = int(video_height // 24)
|
348 |
+
|
349 |
+
font = ImageFont.truetype(font_path, subtitle_font_size)
|
350 |
+
|
351 |
+
dummy_img = Image.new("RGBA", (subtitle_width, 1), (0, 0, 0, 0))
|
352 |
+
draw = ImageDraw.Draw(dummy_img)
|
353 |
+
|
354 |
+
lines = []
|
355 |
+
line = ""
|
356 |
+
for word in text.split():
|
357 |
+
test_line = f"{line} {word}".strip()
|
358 |
+
bbox = draw.textbbox((0, 0), test_line, font=font)
|
359 |
+
w = bbox[2] - bbox[0]
|
360 |
+
if w <= subtitle_width - 10:
|
361 |
+
line = test_line
|
362 |
+
else:
|
363 |
+
lines.append(line)
|
364 |
+
line = word
|
365 |
+
lines.append(line)
|
366 |
+
|
367 |
+
line_heights = [draw.textbbox((0, 0), l, font=font)[3] - draw.textbbox((0, 0), l, font=font)[1] for l in lines]
|
368 |
+
total_height = sum(line_heights) + (len(lines) - 1) * 5
|
369 |
+
img = Image.new("RGBA", (subtitle_width, total_height), (0, 0, 0, 0))
|
370 |
+
draw = ImageDraw.Draw(img)
|
371 |
+
|
372 |
+
y = 0
|
373 |
+
for idx, line in enumerate(lines):
|
374 |
+
bbox = draw.textbbox((0, 0), line, font=font)
|
375 |
+
w = bbox[2] - bbox[0]
|
376 |
+
draw.text(((subtitle_width - w) // 2, y), line, font=font, fill="yellow")
|
377 |
+
y += line_heights[idx] + 5
|
378 |
+
|
379 |
+
img_np = np.array(img) # <- ✅ Fix: convert to NumPy
|
380 |
+
txt_clip = ImageClip(img_np).set_start(start_time).set_duration(end_time - start_time).set_position("bottom").set_opacity(0.8)
|
381 |
+
return txt_clip
|
382 |
+
except Exception as e:
|
383 |
+
logger.error(f"\u274c Failed to create subtitle clip: {e}")
|
384 |
+
return None
|
385 |
+
|
386 |
+
def solve_optimal_alignment(original_segments, generated_durations, total_duration):
|
387 |
+
"""
|
388 |
+
Robust version: Aligns generated speech segments, falls back to greedy allocation if solver fails.
|
389 |
+
Modifies and returns the translated_json with updated 'start' and 'end'.
|
390 |
+
"""
|
391 |
+
N = len(original_segments)
|
392 |
+
d = np.array(generated_durations)
|
393 |
+
m = np.array([(seg['start'] + seg['end']) / 2 for seg in original_segments])
|
394 |
+
|
395 |
+
try:
|
396 |
+
s = cp.Variable(N)
|
397 |
+
objective = cp.Minimize(cp.sum_squares(s + d / 2 - m))
|
398 |
+
|
399 |
+
constraints = [s[0] >= 0]
|
400 |
+
for i in range(N - 1):
|
401 |
+
constraints.append(s[i] + d[i] <= s[i + 1])
|
402 |
+
constraints.append(s[N - 1] + d[N - 1] == total_duration)
|
403 |
+
|
404 |
+
problem = cp.Problem(objective, constraints)
|
405 |
+
problem.solve()
|
406 |
+
|
407 |
+
if s.value is None:
|
408 |
+
raise ValueError("Solver failed")
|
409 |
+
|
410 |
+
for i in range(N):
|
411 |
+
original_segments[i]['start'] = round(s.value[i], 3)
|
412 |
+
original_segments[i]['end'] = round(s.value[i] + d[i], 3)
|
413 |
+
|
414 |
+
except Exception as e:
|
415 |
+
print(f"⚠️ Optimization failed: {e}, falling back to greedy alignment.")
|
416 |
+
|
417 |
+
current_time = 0.0
|
418 |
+
for i in range(N):
|
419 |
+
original_segments[i]['start'] = round(current_time, 3)
|
420 |
+
original_segments[i]['end'] = round(current_time + generated_durations[i], 3)
|
421 |
+
current_time += generated_durations[i]
|
422 |
+
|
423 |
+
return original_segments
|
424 |
+
def process_entry(entry, i, tts_model, video_width, video_height, process_mode, target_language, font_path, speaker_sample_paths=None):
|
425 |
+
logger.debug(f"Processing entry {i}: {entry}")
|
426 |
+
error_message = None
|
427 |
+
|
428 |
+
try:
|
429 |
+
txt_clip = create_subtitle_clip_pil(entry["translated"], entry["start"], entry["end"], video_width, video_height, font_path)
|
430 |
+
except Exception as e:
|
431 |
+
error_message = f"❌ Failed to create subtitle clip for entry {i}: {e}"
|
432 |
+
logger.error(error_message)
|
433 |
+
txt_clip = None
|
434 |
+
|
435 |
+
audio_segment = None
|
436 |
+
actual_duration = 0.0
|
437 |
+
if process_mode > 1:
|
438 |
+
try:
|
439 |
+
segment_audio_path = f"segment_{i}_voiceover.wav"
|
440 |
+
desired_duration = entry["end"] - entry["start"]
|
441 |
+
desired_speed = calibrated_speed(entry['translated'], desired_duration)
|
442 |
+
|
443 |
+
speaker = entry.get("speaker", "default")
|
444 |
+
speaker_wav_path = f"speaker_{speaker}_sample.wav"
|
445 |
+
|
446 |
+
supported_languages = tts_model.synthesizer.tts_model.language_manager.name_to_id.keys()
|
447 |
+
|
448 |
+
if process_mode > 2 and speaker_wav_path and os.path.exists(speaker_wav_path) and target_language in supported_languages:
|
449 |
+
generate_voiceover_clone(entry['translated'], tts_model, desired_speed, target_language, speaker_wav_path, segment_audio_path)
|
450 |
+
else:
|
451 |
+
generate_voiceover_OpenAI(entry['translated'], target_language, desired_speed, segment_audio_path)
|
452 |
+
|
453 |
+
if not segment_audio_path or not os.path.exists(segment_audio_path):
|
454 |
+
raise FileNotFoundError(f"Voiceover file not generated at: {segment_audio_path}")
|
455 |
+
|
456 |
+
audio_clip = AudioFileClip(segment_audio_path)
|
457 |
+
actual_duration = audio_clip.duration
|
458 |
+
|
459 |
+
audio_segment = audio_clip # Do not set start here, alignment happens later
|
460 |
+
|
461 |
+
except Exception as e:
|
462 |
+
err = f"❌ Failed to generate audio segment for entry {i}: {e}"
|
463 |
+
logger.error(err)
|
464 |
+
error_message = error_message + " | " + err if error_message else err
|
465 |
+
audio_segment = None
|
466 |
+
|
467 |
+
return i, txt_clip, audio_segment, actual_duration, error_message
|
468 |
+
|
469 |
+
|
470 |
+
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"):
|
471 |
+
|
472 |
+
video = VideoFileClip(video_path)
|
473 |
+
font_path = "./NotoSansSC-Regular.ttf"
|
474 |
+
|
475 |
+
text_clips = []
|
476 |
+
audio_segments = []
|
477 |
+
actual_durations = []
|
478 |
+
error_messages = []
|
479 |
+
|
480 |
+
if process_mode == 3:
|
481 |
+
global tts_model
|
482 |
+
if tts_model is None:
|
483 |
+
try:
|
484 |
+
print("🔄 Loading XTTS model...")
|
485 |
+
from TTS.api import TTS
|
486 |
+
tts_model = TTS(model_name="tts_models/multilingual/multi-dataset/your_tts")
|
487 |
+
print("✅ XTTS model loaded successfully.")
|
488 |
+
except Exception as e:
|
489 |
+
print("❌ Error loading XTTS model:")
|
490 |
+
traceback.print_exc()
|
491 |
+
return f"Error loading XTTS model: {e}"
|
492 |
+
|
493 |
+
with concurrent.futures.ThreadPoolExecutor() as executor:
|
494 |
+
futures = [executor.submit(process_entry, entry, i, tts_model, video.w, video.h, process_mode, target_language, font_path, speaker_sample_paths)
|
495 |
+
for i, entry in enumerate(translated_json)]
|
496 |
+
|
497 |
+
results = []
|
498 |
+
for future in concurrent.futures.as_completed(futures):
|
499 |
+
try:
|
500 |
+
i, txt_clip, audio_segment, actual_duration, error = future.result()
|
501 |
+
results.append((i, txt_clip, audio_segment, actual_duration))
|
502 |
+
if error:
|
503 |
+
error_messages.append(f"[Entry {i}] {error}")
|
504 |
+
except Exception as e:
|
505 |
+
err = f"❌ Unexpected error in future result: {e}"
|
506 |
+
error_messages.append(err)
|
507 |
+
|
508 |
+
results.sort(key=lambda x: x[0])
|
509 |
+
text_clips = [clip for _, clip, _, _ in results if clip]
|
510 |
+
generated_durations = [dur for _, _, _, dur in results if dur > 0]
|
511 |
+
|
512 |
+
# Align using optimization (modifies translated_json in-place)
|
513 |
+
translated_json = solve_optimal_alignment(translated_json, generated_durations, video.duration)
|
514 |
+
|
515 |
+
# Set aligned timings
|
516 |
+
audio_segments = []
|
517 |
+
for i, entry in enumerate(translated_json):
|
518 |
+
segment = results[i][2] # AudioFileClip
|
519 |
+
if segment:
|
520 |
+
segment = segment.set_start(entry['start']).set_duration(entry['end'] - entry['start'])
|
521 |
+
audio_segments.append(segment)
|
522 |
+
|
523 |
+
final_video = CompositeVideoClip([video] + text_clips)
|
524 |
+
|
525 |
+
if process_mode > 1 and audio_segments:
|
526 |
+
try:
|
527 |
+
voice_audio = CompositeAudioClip(audio_segments).set_duration(video.duration)
|
528 |
+
|
529 |
+
if background_audio_path and os.path.exists(background_audio_path):
|
530 |
+
background_audio = AudioFileClip(background_audio_path).set_duration(video.duration)
|
531 |
+
final_audio = CompositeAudioClip([voice_audio, background_audio])
|
532 |
+
else:
|
533 |
+
final_audio = voice_audio
|
534 |
+
|
535 |
+
final_video = final_video.set_audio(final_audio)
|
536 |
+
|
537 |
+
except Exception as e:
|
538 |
+
print(f"❌ Failed to set audio: {e}")
|
539 |
+
|
540 |
+
final_video.write_videofile(output_path, codec="libx264", audio_codec="aac")
|
541 |
+
|
542 |
+
return error_messages
|
543 |
+
|
544 |
+
def generate_voiceover_OpenAI(full_text, language, desired_speed, output_audio_path):
|
545 |
+
"""
|
546 |
+
Generate voiceover from translated text for a given language using OpenAI TTS API.
|
547 |
+
"""
|
548 |
+
# Define the voice based on the language (for now, use 'alloy' as default)
|
549 |
+
voice = "alloy" # Adjust based on language if needed
|
550 |
+
|
551 |
+
# Define the model (use tts-1 for real-time applications)
|
552 |
+
model = "tts-1"
|
553 |
+
|
554 |
+
max_retries = 3
|
555 |
+
retry_count = 0
|
556 |
+
|
557 |
+
while retry_count < max_retries:
|
558 |
+
try:
|
559 |
+
# Create the speech using OpenAI TTS API
|
560 |
+
response = client.audio.speech.create(
|
561 |
+
model=model,
|
562 |
+
voice=voice,
|
563 |
+
input=full_text,
|
564 |
+
speed=desired_speed
|
565 |
+
)
|
566 |
+
# Save the audio to the specified path
|
567 |
+
with open(output_audio_path, 'wb') as f:
|
568 |
+
for chunk in response.iter_bytes():
|
569 |
+
f.write(chunk)
|
570 |
+
logging.info(f"Voiceover generated successfully for {output_audio_path}")
|
571 |
+
break
|
572 |
+
|
573 |
+
except Exception as e:
|
574 |
+
retry_count += 1
|
575 |
+
logging.error(f"Error generating voiceover (retry {retry_count}/{max_retries}): {e}")
|
576 |
+
time.sleep(5) # Wait 5 seconds before retrying
|
577 |
+
|
578 |
+
if retry_count == max_retries:
|
579 |
+
raise ValueError(f"Failed to generate voiceover after {max_retries} retries.")
|
580 |
+
|
581 |
+
def generate_voiceover_clone(full_text, tts_model, desired_speed, target_language, speaker_wav_path, output_audio_path):
|
582 |
+
try:
|
583 |
+
|
584 |
+
tts_model.tts_to_file(
|
585 |
+
text=full_text,
|
586 |
+
speaker_wav=speaker_wav_path,
|
587 |
+
language=target_language,
|
588 |
+
file_path=output_audio_path,
|
589 |
+
speed=desired_speed,
|
590 |
+
split_sentences=True
|
591 |
+
)
|
592 |
+
msg = "✅ Voice cloning completed successfully."
|
593 |
+
logger.info(msg)
|
594 |
+
return output_audio_path, msg, None
|
595 |
+
|
596 |
+
except Exception as e:
|
597 |
+
generate_voiceover_OpenAI(full_text, target_language, desired_speed, output_audio_path)
|
598 |
+
err_msg = f"❌ An error occurred: {str(e)}, fallback to premium voice"
|
599 |
+
logger.error(traceback.format_exc())
|
600 |
+
return None, err_msg, err_msg
|
601 |
+
|
602 |
+
def calibrated_speed(text, desired_duration):
|
603 |
+
"""
|
604 |
+
Compute a speed factor to help TTS fit audio into desired duration,
|
605 |
+
using a simple truncated linear function of characters per second.
|
606 |
+
"""
|
607 |
+
char_count = len(text.strip())
|
608 |
+
if char_count == 0 or desired_duration <= 0:
|
609 |
+
return 1.0 # fallback
|
610 |
+
|
611 |
+
cps = char_count / desired_duration # characters per second
|
612 |
+
|
613 |
+
# Truncated linear mapping
|
614 |
+
if cps < 14:
|
615 |
+
return 1.0
|
616 |
+
elif cps > 30:
|
617 |
+
return 2
|
618 |
+
else:
|
619 |
+
slope = (2 - 1.0) / (30 - 14)
|
620 |
+
return 1.0 + slope * (cps - 14)
|
621 |
+
|
622 |
+
def upload_and_manage(file, target_language, process_mode):
|
623 |
+
if file is None:
|
624 |
+
logger.info("No file uploaded. Please upload a video/audio file.")
|
625 |
+
return None, [], None, "No file uploaded. Please upload a video/audio file."
|
626 |
+
|
627 |
+
try:
|
628 |
+
start_time = time.time() # Start the timer
|
629 |
+
logger.info(f"Started processing file: {file.name}")
|
630 |
+
|
631 |
+
# Define paths for audio and output files
|
632 |
+
audio_path = "audio.wav"
|
633 |
+
output_video_path = "output_video.mp4"
|
634 |
+
voiceover_path = "voiceover.wav"
|
635 |
+
logger.info(f"Using audio path: {audio_path}, output video path: {output_video_path}, voiceover path: {voiceover_path}")
|
636 |
+
|
637 |
+
# Step 1: Transcribe audio from uploaded media file and get timestamps
|
638 |
+
logger.info("Transcribing audio...")
|
639 |
+
transcription_json, source_language = transcribe_video_with_speakers(file.name)
|
640 |
+
logger.info(f"Transcription completed. Detected source language: {source_language}")
|
641 |
+
|
642 |
+
# Step 2: Translate the transcription
|
643 |
+
logger.info(f"Translating transcription from {source_language} to {target_language}...")
|
644 |
+
translated_json = translate_text(transcription_json, source_language, target_language)
|
645 |
+
logger.info(f"Translation completed. Number of translated segments: {len(translated_json)}")
|
646 |
+
|
647 |
+
# Step 3: Add transcript to video based on timestamps
|
648 |
+
logger.info("Adding translated transcript to video...")
|
649 |
+
add_transcript_voiceover(file.name, translated_json, output_video_path, process_mode, target_language)
|
650 |
+
logger.info(f"Transcript added to video. Output video saved at {output_video_path}")
|
651 |
+
|
652 |
+
# Convert translated JSON into a format for the editable table
|
653 |
+
logger.info("Converting translated JSON into editable table format...")
|
654 |
+
editable_table = [
|
655 |
+
[float(entry["start"]), entry["original"], entry["translated"], float(entry["end"]), entry["speaker"]]
|
656 |
+
for entry in translated_json
|
657 |
+
]
|
658 |
+
|
659 |
+
# Calculate elapsed time
|
660 |
+
elapsed_time = time.time() - start_time
|
661 |
+
elapsed_time_display = f"Processing completed in {elapsed_time:.2f} seconds."
|
662 |
+
logger.info(f"Processing completed in {elapsed_time:.2f} seconds.")
|
663 |
+
|
664 |
+
return editable_table, output_video_path, elapsed_time_display
|
665 |
+
|
666 |
+
except Exception as e:
|
667 |
+
logger.error(f"An error occurred: {str(e)}")
|
668 |
+
return [], None, f"An error occurred: {str(e)}"
|
669 |
+
|
670 |
+
# Gradio Interface with Tabs
|
671 |
+
def build_interface():
|
672 |
+
with gr.Blocks(css=css) as demo:
|
673 |
+
gr.Markdown("## Video Localization")
|
674 |
+
with gr.Row():
|
675 |
+
with gr.Column(scale=4):
|
676 |
+
file_input = gr.File(label="Upload Video/Audio File")
|
677 |
+
language_input = gr.Dropdown(["en", "es", "fr", "zh"], label="Select Language") # Language codes
|
678 |
+
process_mode = gr.Radio(choices=[("Transcription Only", 1),("Transcription with Premium Voice",2),("Transcription with Voice Clone", 3)],label="Choose Processing Type",value=1)
|
679 |
+
submit_button = gr.Button("Post and Process")
|
680 |
+
|
681 |
+
with gr.Column(scale=8):
|
682 |
+
gr.Markdown("## Edit Translations")
|
683 |
+
|
684 |
+
# Editable JSON Data
|
685 |
+
editable_table = gr.Dataframe(
|
686 |
+
value=[], # Default to an empty list to avoid undefined values
|
687 |
+
headers=["start", "original", "translated", "end", "speaker"],
|
688 |
+
datatype=["number", "str", "str", "number", "str"],
|
689 |
+
row_count=1, # Initially empty
|
690 |
+
col_count=5,
|
691 |
+
interactive=[False, True, True, False, False], # Control editability
|
692 |
+
label="Edit Translations",
|
693 |
+
wrap=True # Enables text wrapping if supported
|
694 |
+
)
|
695 |
+
save_changes_button = gr.Button("Save Changes")
|
696 |
+
processed_video_output = gr.File(label="Download Processed Video", interactive=True) # Download button
|
697 |
+
elapsed_time_display = gr.Textbox(label="Elapsed Time", lines=1, interactive=False)
|
698 |
+
|
699 |
+
with gr.Column(scale=1):
|
700 |
+
gr.Markdown("**Feedback**")
|
701 |
+
feedback_input = gr.Textbox(
|
702 |
+
placeholder="Leave your feedback here...",
|
703 |
+
label=None,
|
704 |
+
lines=3,
|
705 |
+
)
|
706 |
+
feedback_btn = gr.Button("Submit Feedback")
|
707 |
+
response_message = gr.Textbox(label=None, lines=1, interactive=False)
|
708 |
+
db_download = gr.File(label="Download Database File", visible=False)
|
709 |
+
|
710 |
+
# Link the feedback handling
|
711 |
+
def feedback_submission(feedback):
|
712 |
+
message, file_path = handle_feedback(feedback)
|
713 |
+
if file_path:
|
714 |
+
return message, gr.update(value=file_path, visible=True)
|
715 |
+
return message, gr.update(visible=False)
|
716 |
+
|
717 |
+
save_changes_button.click(
|
718 |
+
update_translations,
|
719 |
+
inputs=[file_input, editable_table, process_mode],
|
720 |
+
outputs=[processed_video_output, elapsed_time_display]
|
721 |
+
)
|
722 |
+
|
723 |
+
submit_button.click(
|
724 |
+
upload_and_manage,
|
725 |
+
inputs=[file_input, language_input, process_mode],
|
726 |
+
outputs=[editable_table, processed_video_output, elapsed_time_display]
|
727 |
+
)
|
728 |
+
|
729 |
+
# Connect submit button to save_feedback_db function
|
730 |
+
feedback_btn.click(
|
731 |
+
feedback_submission,
|
732 |
+
inputs=[feedback_input],
|
733 |
+
outputs=[response_message, db_download]
|
734 |
+
)
|
735 |
+
|
736 |
+
return demo
|
737 |
+
|
738 |
+
tts_model = None
|
739 |
+
# Launch the Gradio interface
|
740 |
+
demo = build_interface()
|
741 |
+
demo.launch()
|
apt.txt
ADDED
@@ -0,0 +1,9 @@
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
1 |
+
espeak
|
2 |
+
ffmpeg
|
3 |
+
libsm6
|
4 |
+
libxext6
|
5 |
+
git
|
6 |
+
git-lfs
|
7 |
+
libgl1-mesa-glx
|
8 |
+
cmake
|
9 |
+
rsync
|
requirements.txt
ADDED
@@ -0,0 +1,22 @@
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
1 |
+
# Core compatibility
|
2 |
+
numpy==1.26.4
|
3 |
+
transformers==4.49.0
|
4 |
+
# Coqui TTS (XTTS v2)
|
5 |
+
coqpit-config
|
6 |
+
coqui-tts==0.26.0
|
7 |
+
coqui-tts-trainer==0.2.3
|
8 |
+
torch==2.6.0 # Or the version best suited for your GPU/CPU
|
9 |
+
# Video Processing
|
10 |
+
moviepy==1.0.3
|
11 |
+
# Web Interface
|
12 |
+
gradio==5.23.3
|
13 |
+
# Audio Utilities (optional but often used)
|
14 |
+
soundfile
|
15 |
+
librosa
|
16 |
+
SpeechRecognition
|
17 |
+
whisperx==3.3.1
|
18 |
+
openai
|
19 |
+
pillow
|
20 |
+
cvxpy
|
21 |
+
# pyannote.audio
|
22 |
+
# torchaudio
|
speaker_default_sample.wav
ADDED
@@ -0,0 +1,3 @@
|
|
|
|
|
|
|
|
|
1 |
+
version https://git-lfs.github.com/spec/v1
|
2 |
+
oid sha256:d63e6190a950695c5cfa697f263c230e6f682be8822971ccaea67a8318a2d747
|
3 |
+
size 1800056
|