diff --git a/inference.py b/inference.py
new file mode 100644
index 0000000..91c87be
--- /dev/null
+++ b/inference.py
@@ -0,0 +1,108 @@
+"""
+This file gives a sample demonstration of how to use the given functions in Python, for the Voice Safety Classifier model. 
+"""
+
+import torch
+import librosa
+import numpy as np
+import argparse
+from transformers import WavLMForSequenceClassification
+
+
+def feature_extract_simple(
+    wav,
+    sr=16_000,
+    win_len=15.0,
+    win_stride=15.0,
+    do_normalize=False,
+):
+    """simple feature extraction for wavLM
+    Parameters
+    ----------
+    wav : str or array-like
+        path to the wav file, or array-like
+    sr : int, optional
+        sample rate, by default 16_000
+    win_len : float, optional
+        window length, by default 15.0
+    win_stride : float, optional
+        window stride, by default 15.0
+    do_normalize: bool, optional
+        whether to normalize the input, by default False.
+    Returns
+    -------
+    np.ndarray
+        batched input to wavLM
+    """
+    if type(wav) == str:
+        signal, _ = librosa.core.load(wav, sr=sr)
+    else:
+        try:
+            signal = np.array(wav).squeeze()
+        except Exception as e:
+            print(e)
+            raise RuntimeError
+    batched_input = []
+    stride = int(win_stride * sr)
+    l = int(win_len * sr)
+    if len(signal) / sr > win_len:
+        for i in range(0, len(signal), stride):
+            if i + int(win_len * sr) > len(signal):
+                # padding the last chunk to make it the same length as others
+                chunked = np.pad(signal[i:], (0, l - len(signal[i:])))
+            else:
+                chunked = signal[i : i + l]
+            if do_normalize:
+                chunked = (chunked - np.mean(chunked)) / (np.std(chunked) + 1e-7)
+            batched_input.append(chunked)
+            if i + int(win_len * sr) > len(signal):
+                break
+    else:
+        if do_normalize:
+            signal = (signal - np.mean(signal)) / (np.std(signal) + 1e-7)
+        batched_input.append(signal)
+    return np.stack(batched_input)  # [N, T]
+
+
+def infer(model, inputs):
+    output = model(inputs)
+    probs = torch.sigmoid(torch.Tensor(output.logits))
+    return probs
+
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser()
+    parser.add_argument(
+        "--audio_file",
+        type=str,
+        help="File to run inference",
+    )
+    parser.add_argument(
+        "--model_path",
+        type=str,
+        default="roblox/voice-safety-classifier",
+        help="checkpoint file of model",
+    )
+    args = parser.parse_args()
+    labels_name_list = [
+        "Profanity",
+        "DatingAndSexting",
+        "Racist",
+        "Bullying",
+        "Other",
+        "NoViolation",
+    ]
+    # Model is trained on only 16kHz audio
+    audio, _ = librosa.core.load(args.audio_file, sr=16000)
+    input_np = feature_extract_simple(audio, sr=16000)
+    input_pt = torch.Tensor(input_np)
+    model = WavLMForSequenceClassification.from_pretrained(
+        args.model_path, num_labels=len(labels_name_list)
+    )
+    probs = infer(model, input_pt)
+    probs = probs.reshape(-1, 6).detach().tolist()
+    print(f"Probabilities for {args.audio_file} is:")
+    for chunk_idx in range(len(probs)):
+        print(f"\nSegment {chunk_idx}:")
+        for label_idx, label in enumerate(labels_name_list):
+            print(f"{label} : {probs[chunk_idx][label_idx]}")