asr-pyctcdecode / app.py
Vaibhav Srivastav
adding kenlm to requirements
e3fed65
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()