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