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#!/usr/bin/env python | |
# coding: utf-8 | |
# In[ ]: | |
import soundfile as sf | |
import torch | |
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor | |
import argparse | |
from glob import glob | |
import torchaudio | |
import subprocess | |
import gradio as gr | |
resampler = torchaudio.transforms.Resample(48_000, 16_000) | |
def get_filename(wav_file): | |
filename_local = wav_file.split('/')[-1][:-4] | |
filename_new = '/tmp/'+filename_local+'_16.wav' | |
subprocess.call(["sox {} -r {} -b 16 -c 1 {}".format(wav_file, str(16000), filename_new)], shell=True) | |
return filename_new | |
def parse_transcription(wav_file): | |
# load pretrained model | |
processor = Wav2Vec2Processor.from_pretrained("jonatasgrosman/wav2vec2-large-xlsr-53-english") | |
model = Wav2Vec2ForCTC.from_pretrained("jonatasgrosman/wav2vec2-large-xlsr-53-english") | |
# load audio | |
wav_file = get_filename(wav_file.name) | |
audio_input, sample_rate = sf.read(wav_file) | |
#test_file = resampler(test_file[0]) | |
# pad input values and return pt tensor | |
input_values = processor(audio_input, sampling_rate=16_000, return_tensors="pt").input_values | |
# INFERENCE | |
# retrieve logits & take argmax | |
logits = model(input_values).logits | |
predicted_ids = torch.argmax(logits, dim=-1) | |
# transcribe | |
transcription = processor.decode(predicted_ids[0], skip_special_tokens=True) | |
return transcription | |
# In[ ]: | |
import gradio as gr | |
title = "Speech-to-Text-English" | |
description = "Upload a English audio clip, and let AI do the hard work of transcribing." | |
gr.Interface( | |
parse_transcription, | |
title=title, | |
inputs=gr.inputs.Audio(label="Record Audio File", type="file", source = "microphone"), | |
description=description, outputs = "text").launch(inline = False) | |