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import gradio as gr |
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import numpy as np |
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import torch |
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from transformers import pipeline |
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from transformers import VitsModel, VitsTokenizer, FSMTForConditionalGeneration, FSMTTokenizer, Wav2Vec2ForCTC, Wav2Vec2Processor, MarianMTModel, MarianTokenizer, T5ForConditionalGeneration, T5Tokenizer |
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device = "cuda:0" if torch.cuda.is_available() else "cpu" |
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asr_pipe = pipeline("automatic-speech-recognition", model="asapp/sew-d-tiny-100k-ft-ls100h", device=device) |
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translation_en_to_rus = pipeline("translation", model="Helsinki-NLP/opus-mt-en-ru") |
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model = VitsModel.from_pretrained("facebook/mms-tts-rus") |
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tokenizer = VitsTokenizer.from_pretrained("facebook/mms-tts-rus") |
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def translate(audio): |
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en_text = asr_pipe(audio, max_new_tokens=256, generate_kwargs={"task": "translate"}) |
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translated_text = translation_en_to_rus(en_text["text"]) |
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return translated_text[0]['translation_text'] |
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def synthesise(text): |
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inputs = tokenizer(text, return_tensors="pt") |
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with torch.no_grad(): |
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speech = model(**inputs).waveform |
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return speech.cpu() |
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def speech_to_speech_translation(audio): |
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translated_text = translate(audio) |
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synthesised_speech = synthesise(translated_text) |
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synthesised_speech = (synthesised_speech.numpy() * 32767).astype(np.int16) |
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return 16000, synthesised_speech[0] |
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demo = gr.Blocks() |
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mic_translate = gr.Interface( |
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fn=speech_to_speech_translation, |
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inputs=gr.Audio(source="microphone", type="filepath"), |
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outputs=gr.Audio(label="Generated Speech", type="numpy") |
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) |
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file_translate = gr.Interface( |
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fn=speech_to_speech_translation, |
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inputs=gr.Audio(source="upload", type="filepath"), |
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outputs=gr.Audio(label="Generated Speech", type="numpy"), |
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examples=[["./example.wav"]] |
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) |
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with demo: |
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gr.TabbedInterface([mic_translate, file_translate], ["Microphone", "Audio File"]) |
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demo.launch() |