import gradio as gr from transformers import pipeline from datasets import load_dataset import soundfile as sf import torch import os import io import base64 import numpy as np from pydub import AudioSegment os.environ['TRANSFORMERS_CACHE'] = '.cache' print ("----- setting up pipeline -----") synthesiser = pipeline("text-to-speech", "microsoft/speecht5_tts") print ("----- setting up dataset -----") embeddings_dataset = load_dataset("Matthijs/cmu-arctic-xvectors", split="validation") speaker_embedding = torch.tensor(embeddings_dataset[7306]["xvector"]).unsqueeze(0) # You can replace this embedding with your own as well. print ("----- synthetizing audio -----") #speech = synthesiser("Hello, my dog is cooler than you!", forward_params={"speaker_embeddings": speaker_embedding}) #sf.write("speech.wav", speech["audio"], samplerate=speech["sampling_rate"]) def greet(name): return "Hello " + name + "!!" def synthesise_audio(text, forward_params=None): if len(text) > 100: raise ValueError("Error: El texto es demasiado largo. Por favor, limita tu entrada a 100 caracteres.") speech = synthesiser(text, forward_params={"speaker_embeddings": speaker_embedding}) # sf.write("speech.wav", speech["audio"], samplerate=speech["sampling_rate"]) # return "speech.wav" # Convert numpy array to audio #with io.BytesIO() as f: # sf.write(f, speech["audio"], samplerate=speech["sampling_rate"], format='wav') # audio = f.getvalue() # Convert numpy array to audio audio = np.int16(speech["audio"] * 32767) audio_segment = AudioSegment(audio, sample_width=2, frame_rate=speech["sampling_rate"], channels=1) #return speech["audio"] return audio #demo = gr.Interface(fn=greet, inputs="text", outputs="text", description="----- TTS Testing -----") input_text = gr.Textbox(lines=10, label="Enter text here") demo = gr.Interface(fn=synthesise_audio, inputs=input_text, #outputs="audio", outputs = gr.Audio(type="numpy"), description="----- manuai Text To Speech generator -----", allow_flagging = False) demo.launch(debug = True)