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import nltk | |
import librosa | |
import torch | |
import kenlm | |
import gradio as gr | |
from pyctcdecode import build_ctcdecoder | |
from transformers import Wav2Vec2Processor, AutoModelForCTC | |
nltk.download("punkt") | |
wav2vec2processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-base-960h") | |
wav2vec2model = AutoModelForCTC.from_pretrained("facebook/wav2vec2-base-960h") | |
hubertprocessor = Wav2Vec2Processor.from_pretrained("facebook/hubert-large-ls960-ft") | |
hubertmodel = AutoModelForCTC.from_pretrained("facebook/hubert-large-ls960-ft") | |
def return_processor_and_model(model_name): | |
return Wav2Vec2Processor.from_pretrained(model_name), AutoModelForCTC.from_pretrained(model_name) | |
def load_and_fix_data(input_file): | |
speech, sample_rate = librosa.load(input_file) | |
if len(speech.shape) > 1: | |
speech = speech[:,0] + speech[:,1] | |
if sample_rate !=16000: | |
speech = librosa.resample(speech, sample_rate,16000) | |
return speech | |
def fix_transcription_casing(input_sentence): | |
sentences = nltk.sent_tokenize(input_sentence) | |
return (' '.join([s.replace(s[0],s[0].capitalize(),1) for s in sentences])) | |
def predict_and_ctc_decode(input_file, model_name): | |
processor, model = return_processor_and_model(model_name) | |
speech = load_and_fix_data(input_file) | |
input_values = processor(speech, return_tensors="pt", sampling_rate=16000).input_values | |
logits = model(input_values).logits.cpu().detach().numpy()[0] | |
vocab_list = list(processor.tokenizer.get_vocab().keys()) | |
decoder = build_ctcdecoder(vocab_list) | |
pred = decoder.decode(logits) | |
transcribed_text = fix_transcription_casing(pred.lower()) | |
return transcribed_text | |
def predict_and_ctc_lm_decode(input_file, model_name): | |
processor, model = return_processor_and_model(model_name) | |
speech = load_and_fix_data(input_file) | |
input_values = processor(speech, return_tensors="pt", sampling_rate=16000).input_values | |
logits = model(input_values).logits.cpu().detach().numpy()[0] | |
vocab_list = list(processor.tokenizer.get_vocab().keys()) | |
vocab_dict = processor.tokenizer.get_vocab() | |
sorted_dict = {k.lower(): v for k, v in sorted(vocab_dict.items(), key=lambda item: item[1])} | |
decoder = build_ctcdecoder( | |
list(sorted_dict.keys()), | |
"4gram_small.arpa.gz", | |
) | |
pred = decoder.decode(logits) | |
transcribed_text = fix_transcription_casing(pred.lower()) | |
return transcribed_text | |
def predict_and_greedy_decode(input_file, model_name): | |
processor, model = return_processor_and_model(model_name) | |
speech = load_and_fix_data(input_file) | |
input_values = processor(speech, return_tensors="pt", sampling_rate=16000).input_values | |
logits = model(input_values).logits | |
predicted_ids = torch.argmax(logits, dim=-1) | |
pred = processor.batch_decode(predicted_ids) | |
transcribed_text = fix_transcription_casing(pred[0].lower()) | |
return transcribed_text | |
def return_all_predictions(input_file, model_name): | |
return predict_and_ctc_decode(input_file, model_name), predict_and_ctc_lm_decode(input_file, model_name), predict_and_greedy_decode(input_file, model_name) | |
gr.Interface(return_all_predictions, | |
inputs = [gr.inputs.Audio(source="microphone", type="filepath", label="Record/ Drop audio"), gr.inputs.Dropdown(["facebook/wav2vec2-base-960h", "facebook/hubert-large-ls960-ft"], label="Model Name")], | |
outputs = [gr.outputs.Textbox(label="Beam CTC decoding"), gr.outputs.Textbox(label="Beam CTC decoding w/ LM"), gr.outputs.Textbox(label="Greedy decoding")], | |
title="ASR using Wav2Vec2/ Hubert & pyctcdecode", | |
description = "Comparing greedy decoder with beam search CTC decoder, record/ drop your audio!", | |
layout = "horizontal", | |
examples = [["test1.wav", "facebook/wav2vec2-base-960h"], ["test2.wav", "facebook/hubert-large-ls960-ft"]], | |
theme="huggingface", | |
enable_queue=True).launch() |