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#References: 1. https://www.kdnuggets.com/2021/03/speech-text-wav2vec.html | |
#2. https://www.youtube.com/watch?v=4CoVcsxZphE | |
#3. https://www.analyticsvidhya.com/blog/2021/02/hugging-face-introduces-the-first-automatic-speech-recognition-model-wav2vec2/ | |
#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 model and the tokenizer | |
model_name = "bofenghuang/asr-wav2vec2-ctc-french" #"wasertech/wav2vec2-cv-fr-9" | |
tokenizer = Wav2Vec2Tokenizer.from_pretrained(model_name) | |
model = Wav2Vec2ForCTC.from_pretrained(model_name) | |
def load_data(input_file): | |
""" Function for resampling to ensure that the speech input is sampled at 16KHz. | |
""" | |
#read the file | |
speech, sample_rate = librosa.load(input_file) | |
#make it 1-D | |
if len(speech.shape) > 1: | |
speech = speech[:,0] + speech[:,1] | |
#Resampling at 16KHz since wav2vec2-base-960h is pretrained and fine-tuned on speech audio sampled at 16 KHz. | |
if sample_rate !=16000: | |
speech = librosa.resample(speech, orig_sr=sample_rate, target_sr=16000) | |
return speech | |
def correct_casing(input_sentence): | |
""" This function is for correcting the casing of the generated transcribed text | |
""" | |
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): | |
"""This function generates transcripts for the provided audio input | |
""" | |
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]) | |
#Output is all upper case | |
transcription = correct_casing(transcription.lower()) | |
return transcription | |
gr.Interface(asr_transcript, | |
inputs = gr.inputs.Audio(source="microphone", type="filepath", optional=True, label="Démarrer l'enregistrement"), | |
outputs = gr.outputs.Textbox(label="Transcription"), | |
title="🎙️ Parlez, on vous écoute !", | |
description = "Enregistrez un audio ou utilisez les examples pour interagir avec notre dernier modèle.", | |
examples = [["wav/1.wav"], ["wav/2.wav"], ["wav/3.wav"], ["wav/4.wav"], ["wav/5.wav"]], theme="grass").launch() | |