diff --git a/codes/models/asr/__init__.py b/codes/models/asr/__init__.py
new file mode 100644
index 00000000..e69de29b
diff --git a/codes/models/asr/w2v_wrapper.py b/codes/models/asr/w2v_wrapper.py
new file mode 100644
index 00000000..153a8385
--- /dev/null
+++ b/codes/models/asr/w2v_wrapper.py
@@ -0,0 +1,108 @@
+from itertools import groupby
+
+import torch
+import torch.nn as nn
+import torch.nn.functional as F
+from transformers import Wav2Vec2ForCTC, Wav2Vec2CTCTokenizer
+
+from data.audio.unsupervised_audio_dataset import load_audio
+from models.tacotron2.text import symbols, sequence_to_text
+from trainer.networks import register_model
+from utils.util import opt_get
+
+
+class Wav2VecWrapper(nn.Module):
+    """
+    Basic wrapper class that makes Wav2Vec2 usable by DLAS.
+    """
+    def __init__(self, vocab_size=148, basis_model='facebook/wav2vec2-large', freeze_transformer=False, output_wer=True):
+        super().__init__()
+        self.w2v = Wav2Vec2ForCTC.from_pretrained(basis_model)
+        # Perform some surgery to get the model we actually want.
+        self.w2v.wav2vec2.encoder.gradient_checkpointing = True
+        self.w2v.lm_head = nn.Linear(self.w2v.config.hidden_size, vocab_size)
+        self.w2v.config.vocab_size = vocab_size
+        self.w2v.config.pad_token_id = 0
+        self.w2v.config.ctc_loss_reduction = 'mean'
+        self.w2v.config.apply_spec_augment = True
+
+        # We always freeze the feature extractor, which needs some special operations in DLAS
+        for p in self.w2v.wav2vec2.feature_extractor.parameters():
+            p.requires_grad = False
+            p.DO_NOT_TRAIN = True
+        if freeze_transformer:
+            # Also freeze the encoder here.
+            for p in list(self.w2v.wav2vec2.encoder.parameters()) + list(self.w2v.wav2vec2.feature_projection.parameters()):
+                p.requires_grad = False
+                p.DO_NOT_TRAIN = True
+
+        self.output_wer = output_wer
+        if output_wer:
+            self.last_pred = []
+            self.last_labels = []
+
+    def forward(self, audio, unaligned_tokens, wav_lengths, text_lengths):
+        audio = audio[:, :, :wav_lengths.max()]
+        unaligned_tokens = unaligned_tokens[:, :text_lengths.max()]
+        attention_mask = torch.ones_like(audio).squeeze(1)
+        for b in range(audio.shape[0]):
+            attention_mask[b, wav_lengths[b]:] = 0
+            unaligned_tokens[b, text_lengths[b]:] = -100
+
+        audio_norm = (audio - audio.mean()) / torch.sqrt(audio.var() + 1e-7)
+        outputs = self.w2v(input_values=audio_norm.squeeze(1), attention_mask=attention_mask, labels=unaligned_tokens)
+
+        if self.output_wer:
+            self.last_pred.append(torch.argmax(outputs.logits, dim=-1))
+            if len(self.last_pred) > 10:
+                self.last_pred = self.last_pred[1:]
+            self.last_labels.append(unaligned_tokens)
+            if len(self.last_labels) > 10:
+                self.last_labels = self.last_labels[1:]
+        return outputs.loss
+
+    def decode_ctc(self, output):
+        if isinstance(output, torch.Tensor):
+            output = output.tolist()
+        tokens = [token_group[0] for token_group in groupby(output)]
+        filtered_tokens = list(filter(lambda token: token != 0, tokens))
+        return filtered_tokens
+
+    def get_debug_values(self, step, net_name):
+        res = {}
+        if self.output_wer and step % 100 == 0:
+            from datasets import load_metric
+            wer_metric = load_metric("wer")
+            label_strings = []
+            pred_strings = []
+            for last_labels, last_pred in zip(self.last_labels, self.last_pred):
+                last_labels[last_labels == -100] = 0
+                label_strings.extend([sequence_to_text(lbl) for lbl in last_labels])
+                pred_strings.extend([sequence_to_text(self.decode_ctc(pred)) for pred in last_pred])
+            wer = wer_metric.compute(predictions=pred_strings, references=label_strings)
+            res['wer'] = wer
+            print(f"Sample prediction: {pred_strings[0]} <=> {label_strings[0]}")
+        return res
+
+    def inference(self, audio):
+        audio_norm = (audio - audio.mean()) / torch.sqrt(audio.var() + 1e-7)
+        logits = self.w2v(input_values=audio_norm.squeeze(1)).logits
+        pred = logits.argmax(dim=-1)
+        return self.decode_ctc(pred)
+
+
+@register_model
+def register_wav2vec2_finetune(opt_net, opt):
+    return Wav2VecWrapper(**opt_get(opt_net, ['kwargs'], {}))
+
+
+if __name__ == '__main__':
+    w2v = Wav2VecWrapper(basis_model='facebook/wav2vec2-large-960h', freeze_transformer=True)
+    loss = w2v(torch.randn(2,1,50000), torch.randint(0,40,(2,70)), torch.tensor([20000, 30000]), torch.tensor([35, 50]))
+    w2v.get_debug_values(0,"")
+
+    sd = torch.load('../experiments/train_wav2vec_mass_archived_r0/models/19500_wav2vec.pth')
+    w2v.load_state_dict(sd)
+    pred = w2v.inference(load_audio('Y:\\clips\\books1\\754_Dan Simmons - The Rise Of Endymion 356 of 450\\00026.wav', 16000).unsqueeze(0))
+    res = sequence_to_text(pred[0])
+    print(res)
diff --git a/codes/train.py b/codes/train.py
index 334ffab2..5d443872 100644
--- a/codes/train.py
+++ b/codes/train.py
@@ -316,7 +316,7 @@ class Trainer:
 
 if __name__ == '__main__':
     parser = argparse.ArgumentParser()
-    parser.add_argument('-opt', type=str, help='Path to option YAML file.', default='../options/train_encoder_build_ctc_alignments.yml')
+    parser.add_argument('-opt', type=str, help='Path to option YAML file.', default='../options/train_wav2vec_mass.yml')
     parser.add_argument('--launcher', choices=['none', 'pytorch'], default='none', help='job launcher')
     parser.add_argument('--local_rank', type=int, default=0)
     args = parser.parse_args()