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import torch
from datasets import load_dataset
from transformers import pipeline
device = "cuda:0" if torch.cuda.is_available() else "cpu"
pipe = pipeline(
"automatic-speech-recognition", model="openai/whisper-base", device=device
)
def translate(audio):
outputs = pipe(audio, max_new_tokens=256, generate_kwargs={"task": "transcribe", "language": "it"})
return outputs["text"]
from transformers import SpeechT5Processor, SpeechT5ForTextToSpeech, SpeechT5HifiGan
processor = SpeechT5Processor.from_pretrained("burraco135/speecht5_finetuned_voxpopuli_it")
model = SpeechT5ForTextToSpeech.from_pretrained("burraco135/speecht5_finetuned_voxpopuli_it")
vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan")
model.to(device)
vocoder.to(device)
embeddings_dataset = load_dataset("Matthijs/cmu-arctic-xvectors", split="validation")
speaker_embeddings = torch.tensor(embeddings_dataset[7306]["xvector"]).unsqueeze(0)
def synthesise(text):
inputs = processor(text=text, return_tensors="pt")
speech = model.generate_speech(
inputs["input_ids"].to(device), speaker_embeddings.to(device), vocoder=vocoder
)
return speech.cpu()
import numpy as np
target_dtype = np.int16
max_range = np.iinfo(target_dtype).max
def speech_to_speech_translation(audio):
translated_text = translate(audio)
synthesised_speech = synthesise(translated_text)
synthesised_speech = (synthesised_speech.numpy() * max_range).astype(np.int16)
return 16000, synthesised_speech
import gradio as gr
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"),
)
with demo:
gr.TabbedInterface([mic_translate, file_translate], ["Microphone", "Audio File"])
demo.launch(debug=True)