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import os
import time
import json
import gradio as gr
import torch
import torchaudio
import numpy as np
from denoiser.demucs import Demucs
from pydub import AudioSegment
modelpath = './denoiser/master64.th'
def transcribe(file_upload, microphone):
file = microphone if microphone is not None else file_upload
model = Demucs(hidden=64)
state_dict = torch.load(modelpath, map_location='cpu')
model.load_state_dict(state_dict)
demucs = model
x, sr = torchaudio.load(file)
out = demucs(x[None])[0]
out = out / max(out.abs().max().item(), 1)
torchaudio.save('enhanced.wav', out, sr)
enhanced = AudioSegment.from_wav('enhanced.wav') # 只有去完噪的需要降 bitrate 再做語音識別
enhanced.export('enhanced.wav', format="wav", bitrate="256k")
return "enhanced.wav"
demo = gr.Interface(
fn=transcribe,
inputs=[
gr.Audio(type="filepath", label="語音質檢原始音檔"),
],
outputs=gr.Audio(type="filepath", label="Output"),
title="<h1>語音質檢/噪音去除 (語音增強)</h1>",
description="""<h2><a href='https://www.twman.org' target='_blank'>TonTon Huang Ph.D. @ 2024/04 </a></h2><br>
為了提升語音識別的效果,可以在識別前先進行噪音去除<br>
<a href='https://github.com/Deep-Learning-101' target='_blank'>Deep Learning 101 Github</a> | <a href='http://deeplearning101.twman.org' target='_blank'>Deep Learning 101</a> | <a href='https://www.facebook.com/groups/525579498272187/' target='_blank'>台灣人工智慧社團 FB</a> | <a href='https://www.youtube.com/c/DeepLearning101' target='_blank'>YouTube</a><br>
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<a href='https://github.com/facebookresearch/denoiser' target='_blank'> Real Time Speech Enhancement in the Waveform Domain (Interspeech 2020)</a>""",
allow_flagging="never",
examples=[
["exampleAudio/15s_2020-03-27_sep1.wav"],
["exampleAudio/13s_2020-03-27_sep2.wav"],
["exampleAudio/30s_2020-04-23_sep1.wav"],
["exampleAudio/15s_2020-04-23_sep2.wav"],
],
)
demo.launch(debug=True) |