app.py

import gradio as gr
import os
import librosa
from transformers import  Wav2Vec2ProcessorWithLM, AutoModelForCTC, Wav2Vec2CTCTokenizer, Wav2Vec2FeatureExtractor
import torch 

 
model_name = os.getenv("MODEL_NAME")
auth_token = os.getenv("API_TOKEN")

# Load models 
tokenizer = Wav2Vec2CTCTokenizer.from_pretrained(model_name, eos_token=None, bos_token=None, use_auth_token=auth_token)
processor = Wav2Vec2ProcessorWithLM.from_pretrained(model_name, use_auth_token=auth_token)
feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained(model_name, use_auth_token=auth_token)
decoder = processor.decoder
processor = Wav2Vec2ProcessorWithLM(feature_extractor=feature_extractor, tokenizer=tokenizer, decoder=decoder)
model = AutoModelForCTC.from_pretrained(model_name, use_auth_token=auth_token)

def load_data(input_file):

  # 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 
  if sample_rate !=16_000:
    speech = librosa.resample(speech, sample_rate, 16_000)
  return speech



def transcribe(input_file):
    
    audio = load_data(input_file)
    # audio = input_file
    
    # Tokenize
    input_values = processor(audio, return_tensors="pt", sampling_rate=16_000).input_values
    
    # Take logits
    with torch.no_grad():
        logits = model(input_values).logits.cpu().numpy()[0]

    # Decode
    text = decoder.decode(logits, beam_width=30)
   
    return text


examples = [
    ["examples/example1.mp3"],
    ["examples/example2.mp3"],
  ]

gr.Interface(
    title="Rozpoznání mluvené řeči pro český jazyk",
    fn=transcribe, 
    inputs=gr.inputs.Audio(source="upload", type="filepath"), 
    outputs="text",
    examples=examples
    ).launch()