Update README.md

README.md

@@ -53,7 +53,8 @@ Larger models trained with more data are on the way.
 # Model Usage
 
 ```python
-from transformers import Wav2Vec2Processor, HubertModel
+from transformers import Wav2Vec2Processor
+from transformers import AutoModel
 import torch
 from torch import nn
 from datasets import load_dataset
@@ -62,10 +63,10 @@ from datasets import load_dataset
 dataset = load_dataset("hf-internal-testing/librispeech_asr_demo", "clean", split="validation")
 dataset = dataset.sort("id")
 sampling_rate = dataset.features["audio"].sampling_rate
-processor = Wav2Vec2Processor.from_pretrained("facebook/hubert-large-ls960-ft")
+processor = Wav2Vec2Processor.from_pretrained("m-a-p/MERT-v0")
 
 # loading our model weights
-model = HubertModel.from_pretrained("m-a-p/MERT-v0")
+model = AutoModel.from_pretrained("m-a-p/MERT-v0")
 
 # audio file is decoded on the fly
 inputs = processor(dataset[0]["audio"]["array"], sampling_rate=sampling_rate, return_tensors="pt")