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#Importing all the necessary packages
import nltk
import librosa
import torch
import gradio as gr
from transformers import Wav2Vec2Tokenizer, Wav2Vec2ForCTC
nltk.download("punkt")
#Loading the pre-trained model and the tokenizer
model_name = "shizukanabasho/north2"
tokenizer = Wav2Vec2Tokenizer.from_pretrained(model_name)
model = Wav2Vec2ForCTC.from_pretrained(model_name)
def load_data(input_file):
#reading the file
speech, sample_rate = librosa.load(input_file)
#make it 1-D
if len(speech.shape) > 1:
speech = speech[:,0] + speech[:,1]
#Resampling the audio at 16KHz
if sample_rate !=16000:
speech = librosa.resample(speech, sample_rate,16000)
return speech
def correct_casing(input_sentence):
sentences = nltk.sent_tokenize(input_sentence)
return (''.join([s.replace(s[0],s[0].capitalize(),1) for s in sentences]))
def asr_transcript(input_file):
speech = load_data(input_file)
#Tokenize
input_values = tokenizer(speech, return_tensors="pt").input_values
#Take logits
logits = model(input_values).logits
#Take argmax
predicted_ids = torch.argmax(logits, dim=-1)
#Get the words from predicted word ids
transcription = tokenizer.decode(predicted_ids[0])
#Correcting the letter casing
# transcription = correct_casing(transcription.lower())
return transcription
gr.Interface(
asr_transcript,
inputs=[
gr.Audio(source="microphone", type="filepath", optional=True),
gr.Audio(source="upload", type="filepath", optional=True),
title="ASR using Wav2Vec2.0",
description = "This application displays transcribed text for given audio input",
],
outputs="text",
).launch() |