app.py

import gradio as gr
import torch
import torchaudio
from transformers import Wav2Vec2ForSequenceClassification, Wav2Vec2Processor

# Load model and processor
model = Wav2Vec2ForSequenceClassification.from_pretrained("./")
processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-base")
model.eval()

def classify(audio):
    waveform, sample_rate = torchaudio.load(audio)
    # Resample if needed
    if sample_rate != 16000:
        resampler = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=16000)
        waveform = resampler(waveform)
    
    # Preprocess
    inputs = processor(waveform.squeeze().numpy(), sampling_rate=16000, return_tensors="pt", padding=True)

    with torch.no_grad():
        logits = model(**inputs).logits
        predicted = torch.argmax(logits, dim=-1).item()

    return f"Predicted Keyword: {predicted}"

gr.Interface(
    fn=classify,
    inputs = gr.Audio(type="filepath", label="Record from microphone"),
    outputs="text",
    title="Hey Alpha Keyword Spotting",
).launch()