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soni_translate/text_to_speech.py
CHANGED
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@@ -3,28 +3,31 @@ import edge_tts
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import asyncio
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import nest_asyncio
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def make_voice(tts_text, tts_voice, filename):
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try:
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nest_asyncio.apply()
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asyncio.run(edge_tts.Communicate(tts_text, "-".join(tts_voice.split('-')[:-1])).save(filename))
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except
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def make_voice_gradio(tts_text, tts_voice, filename):
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print(tts_text, filename)
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try:
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asyncio.run(edge_tts.Communicate(tts_text, "-".join(tts_voice.split('-')[:-1])).save(filename))
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except
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import asyncio
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import nest_asyncio
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def make_voice(tts_text, tts_voice, filename,language):
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#print(tts_text, filename)
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try:
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nest_asyncio.apply()
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asyncio.run(edge_tts.Communicate(tts_text, "-".join(tts_voice.split('-')[:-1])).save(filename))
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except:
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try:
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tts = gTTS(tts_text, lang=language)
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tts.save(filename)
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print(f'No audio was received. Please change the tts voice for {tts_voice}. USING gTTS.')
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except:
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tts = gTTS('a', lang=language)
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tts.save(filename)
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print('Error: Audio will be replaced.')
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def make_voice_gradio(tts_text, tts_voice, filename, language):
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print(tts_text, filename)
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try:
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asyncio.run(edge_tts.Communicate(tts_text, "-".join(tts_voice.split('-')[:-1])).save(filename))
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except:
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try:
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tts = gTTS(tts_text, lang=language)
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tts.save(filename)
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print(f'No audio was received. Please change the tts voice for {tts_voice}. USING gTTS.')
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except:
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tts = gTTS('a', lang=language)
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tts.save(filename)
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print('Error: Audio will be replaced.')
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soni_translate/translate_segments.py
CHANGED
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@@ -2,9 +2,15 @@ from tqdm import tqdm
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from deep_translator import GoogleTranslator
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def translate_text(segments, TRANSLATE_AUDIO_TO):
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for line in tqdm(range(len(segments))):
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text = segments[line]['text']
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translator = GoogleTranslator(source='auto', target=TRANSLATE_AUDIO_TO)
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translated_line = translator.translate(text.strip())
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segments[line]['text'] = translated_line
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return segments
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from deep_translator import GoogleTranslator
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def translate_text(segments, TRANSLATE_AUDIO_TO):
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if TRANSLATE_AUDIO_TO == "zh":
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TRANSLATE_AUDIO_TO = "zh-CN"
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translator = GoogleTranslator(source='auto', target=TRANSLATE_AUDIO_TO)
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for line in tqdm(range(len(segments))):
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text = segments[line]['text']
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translated_line = translator.translate(text.strip())
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segments[line]['text'] = translated_line
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return segments
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soni_translate/video_dubbing.py
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@@ -0,0 +1,217 @@
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import numpy as np
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import gradio as gr
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import whisperx
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import torch
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from gtts import gTTS
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import librosa
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import edge_tts
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import gc
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from pydub import AudioSegment
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from tqdm import tqdm
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from deep_translator import GoogleTranslator
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import os
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from soni_translate.audio_segments import create_translated_audio
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from soni_translate.text_to_speech import make_voice
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from soni_translate.translate_segments import translate_text
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import time
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def translate_from_video(
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video,
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YOUR_HF_TOKEN,
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preview=False,
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WHISPER_MODEL_SIZE="large-v1",
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batch_size=16,
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compute_type="float16",
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SOURCE_LANGUAGE= "Automatic detection",
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TRANSLATE_AUDIO_TO="en",
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min_speakers=1,
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max_speakers=2,
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tts_voice00="en-AU-WilliamNeural-Male",
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tts_voice01="en-CA-ClaraNeural-Female",
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tts_voice02="en-GB-ThomasNeural-Male",
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tts_voice03="en-GB-SoniaNeural-Female",
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tts_voice04="en-NZ-MitchellNeural-Male",
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tts_voice05="en-GB-MaisieNeural-Female",
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video_output="video_dub.mp4"
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):
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if YOUR_HF_TOKEN == "" or YOUR_HF_TOKEN == None:
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YOUR_HF_TOKEN = os.getenv("YOUR_HF_TOKEN")
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if not os.path.exists('audio'):
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os.makedirs('audio')
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if not os.path.exists('audio2/audio'):
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os.makedirs('audio2/audio')
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# Check GPU
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device = "cuda" if torch.cuda.is_available() else "cpu"
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compute_type = "float32" if device == "cpu" else compute_type
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OutputFile = 'Video.mp4'
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audio_wav = "audio.wav"
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Output_name_file = "audio_dub_solo.ogg"
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mix_audio = "audio_mix.mp3"
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os.system("rm Video.mp4")
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os.system("rm audio.webm")
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os.system("rm audio.wav")
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if os.path.exists(video):
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if preview:
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print('Creating preview video, 10 seconds')
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os.system(f'ffmpeg -y -i "{video}" -ss 00:00:20 -t 00:00:10 -c:v libx264 -c:a aac -strict experimental Video.mp4')
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else:
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os.system(f'ffmpeg -y -i "{video}" -c:v libx264 -c:a aac -strict experimental Video.mp4')
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os.system("ffmpeg -y -i Video.mp4 -vn -acodec pcm_s16le -ar 44100 -ac 2 audio.wav")
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else:
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if preview:
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print('Creating preview from link, 10 seconds')
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#https://github.com/yt-dlp/yt-dlp/issues/2220
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mp4_ = f'yt-dlp -f "mp4" --downloader ffmpeg --downloader-args "ffmpeg_i: -ss 00:00:20 -t 00:00:10" --force-overwrites --max-downloads 1 --no-warnings --no-abort-on-error --ignore-no-formats-error --restrict-filenames -o {OutputFile} {video}'
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wav_ = "ffmpeg -y -i Video.mp4 -vn -acodec pcm_s16le -ar 44100 -ac 2 audio.wav"
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os.system(mp4_)
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os.system(wav_)
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else:
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mp4_ = f'yt-dlp -f "mp4" --force-overwrites --max-downloads 1 --no-warnings --no-abort-on-error --ignore-no-formats-error --restrict-filenames -o {OutputFile} {video}'
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wav_ = f'python -m yt_dlp --output {audio_wav} --force-overwrites --max-downloads 1 --no-warnings --no-abort-on-error --ignore-no-formats-error --extract-audio --audio-format wav {video}'
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os.system(wav_)
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for i in range (120):
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time.sleep(1)
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print('process audio')
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if os.path.exists(audio_wav) and not os.path.exists('audio.webm'):
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time.sleep(1)
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os.system(mp4_)
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break
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if i == 119:
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print('Error donwloading the audio')
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return
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print("Set file complete.")
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SOURCE_LANGUAGE = None if SOURCE_LANGUAGE == 'Automatic detection' else SOURCE_LANGUAGE
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# 1. Transcribe with original whisper (batched)
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model = whisperx.load_model(
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WHISPER_MODEL_SIZE,
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device,
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compute_type=compute_type,
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language= SOURCE_LANGUAGE,
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)
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audio = whisperx.load_audio(audio_wav)
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result = model.transcribe(audio, batch_size=batch_size)
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gc.collect(); torch.cuda.empty_cache(); del model
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print("Transcript complete")
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# 2. Align whisper output
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model_a, metadata = whisperx.load_align_model(
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language_code=result["language"],
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device=device
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)
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result = whisperx.align(
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result["segments"],
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model_a,
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metadata,
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audio,
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device,
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return_char_alignments=True,
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)
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gc.collect(); torch.cuda.empty_cache(); del model_a
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print("Align complete")
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| 124 |
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if result['segments'] == []:
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print('No active speech found in audio')
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return
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# 3. Assign speaker labels
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diarize_model = whisperx.DiarizationPipeline(use_auth_token=YOUR_HF_TOKEN, device=device)
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diarize_segments = diarize_model(
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audio_wav,
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min_speakers=min_speakers,
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max_speakers=max_speakers)
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| 135 |
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result_diarize = whisperx.assign_word_speakers(diarize_segments, result)
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| 136 |
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gc.collect(); torch.cuda.empty_cache(); del diarize_model
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| 137 |
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print("Diarize complete")
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| 138 |
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result_diarize['segments'] = translate_text(result_diarize['segments'], TRANSLATE_AUDIO_TO)
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| 140 |
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print("Translation complete")
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| 141 |
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| 142 |
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audio_files = []
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| 143 |
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| 144 |
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# Mapping speakers to voice variables
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| 145 |
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speaker_to_voice = {
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| 146 |
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'SPEAKER_00': tts_voice00,
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| 147 |
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'SPEAKER_01': tts_voice01,
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| 148 |
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'SPEAKER_02': tts_voice02,
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| 149 |
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'SPEAKER_03': tts_voice03,
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'SPEAKER_04': tts_voice04,
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| 151 |
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'SPEAKER_05': tts_voice05
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| 152 |
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}
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| 153 |
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| 154 |
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for segment in tqdm(result_diarize['segments']):
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| 155 |
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| 156 |
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text = segment['text']
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| 157 |
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start = segment['start']
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| 158 |
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end = segment['end']
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| 159 |
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| 160 |
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try:
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| 161 |
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speaker = segment['speaker']
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| 162 |
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except KeyError:
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| 163 |
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segment['speaker'] = "SPEAKER_99"
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| 164 |
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speaker = segment['speaker']
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| 165 |
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print("NO SPEAKER DETECT IN SEGMENT")
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| 166 |
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| 167 |
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# make the tts audio
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| 168 |
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filename = f"audio/{start}.ogg"
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| 169 |
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| 170 |
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if speaker in speaker_to_voice and speaker_to_voice[speaker] != 'None':
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| 171 |
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make_voice(text, speaker_to_voice[speaker], filename, TRANSLATE_AUDIO_TO)
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| 172 |
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elif speaker == "SPEAKER_99":
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| 173 |
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try:
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| 174 |
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tts = gTTS(text, lang=TRANSLATE_AUDIO_TO)
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| 175 |
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tts.save(filename)
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| 176 |
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print('Using GTTS')
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| 177 |
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except:
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| 178 |
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tts = gTTS('a', lang=TRANSLATE_AUDIO_TO)
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| 179 |
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tts.save(filename)
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| 180 |
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print('Error: Audio will be replaced.')
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| 181 |
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|
| 182 |
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# duration
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| 183 |
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duration_true = end - start
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| 184 |
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duration_tts = librosa.get_duration(filename=filename)
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| 185 |
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| 186 |
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# porcentaje
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| 187 |
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porcentaje = duration_tts / duration_true
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| 188 |
+
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| 189 |
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if porcentaje > 2.1:
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| 190 |
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porcentaje = 2.1
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| 191 |
+
elif porcentaje <= 1.2 and porcentaje >= 0.8:
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| 192 |
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porcentaje = 1.0
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| 193 |
+
elif porcentaje <= 0.79:
|
| 194 |
+
porcentaje = 0.8
|
| 195 |
+
|
| 196 |
+
# Smoth and round
|
| 197 |
+
porcentaje = round(porcentaje+0.0, 1)
|
| 198 |
+
|
| 199 |
+
# apply aceleration or opposite to the audio file in audio2 folder
|
| 200 |
+
os.system(f"ffmpeg -y -loglevel panic -i {filename} -filter:a atempo={porcentaje} audio2/{filename}")
|
| 201 |
+
|
| 202 |
+
duration_create = librosa.get_duration(filename=f"audio2/{filename}")
|
| 203 |
+
audio_files.append(filename)
|
| 204 |
+
|
| 205 |
+
# replace files with the accelerates
|
| 206 |
+
os.system("mv -f audio2/audio/*.ogg audio/")
|
| 207 |
+
|
| 208 |
+
os.system(f"rm {Output_name_file}")
|
| 209 |
+
create_translated_audio(result_diarize, audio_files, Output_name_file)
|
| 210 |
+
|
| 211 |
+
os.system(f"rm {mix_audio}")
|
| 212 |
+
os.system(f'ffmpeg -i {audio_wav} -i {Output_name_file} -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress=threshold=0.003:ratio=20[bg]; [bg][mix]amerge[final]" -map [final] {mix_audio}')
|
| 213 |
+
|
| 214 |
+
os.system(f"rm {video_output}")
|
| 215 |
+
os.system(f"ffmpeg -i {OutputFile} -i {mix_audio} -c:v copy -c:a copy -map 0:v -map 1:a -shortest {video_output}")
|
| 216 |
+
|
| 217 |
+
return video_output
|