app.py

from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
import soundfile as sf
import torch
import gradio as gr


# load model and processor
processor = Wav2Vec2Processor.from_pretrained("h4d35/Wav2Vec2-hi")
model = Wav2Vec2ForCTC.from_pretrained("h4d35/Wav2Vec2-hi")

# define function to read in sound file
def map_to_array(file):
   speech, _ = sf.read(file)
   return speech
 


# tokenize
def inference(audio):
   input_values = processor(map_to_array(audio.name), return_tensors="pt", padding="longest").input_values  # Batch size 1

   # retrieve logits
   logits = model(input_values).logits

   # take argmax and decode
   predicted_ids = torch.argmax(logits, dim=-1)
   transcription = processor.batch_decode(predicted_ids)
   return transcription[0]

inputs = gr.inputs.Audio(label="Input Audio", type="file")
outputs =  gr.outputs.Textbox(label="Output Text")

title = "HindiASR"
description = "HindiASR using Wav2Vec2.0"


#examples=[['poem.wav']]
gr.Interface(inference, inputs, outputs, title=title, description=description).launch()