diff --git a/README.md b/README.md
index 60ce9e92b6..42253940d3 100644
--- a/README.md
+++ b/README.md
@@ -9,6 +9,7 @@ of various sequence-to-sequence models, including:
   - [Gehring et al. (2017): Convolutional Sequence to Sequence Learning](examples/conv_seq2seq/README.md)
   - [Edunov et al. (2018): Classical Structured Prediction Losses for Sequence to Sequence Learning](https://github.com/pytorch/fairseq/tree/classic_seqlevel)
   - [Fan et al. (2018): Hierarchical Neural Story Generation](examples/stories/README.md)
+  - **_New_** [wav2vec: Unsupervised Pre-training for Speech Recognition (Schneider et al., 2019)](examples/wav2vec/README.md)
 - **LightConv and DynamicConv models**
   - **_New_** [Wu et al. (2019): Pay Less Attention with Lightweight and Dynamic Convolutions](examples/pay_less_attention_paper/README.md)
 - **Long Short-Term Memory (LSTM) networks**
@@ -82,6 +83,7 @@ as well as example training and evaluation commands.
 - [Language Modeling](examples/language_model/README.md): convolutional models are available
 
 We also have more detailed READMEs to reproduce results from specific papers:
+- [Schneider et al. (2019): wav2vec: Unsupervised Pre-training for Speech Recognition](examples/wav2vec/README.md)
 - [Shen et al. (2019) Mixture Models for Diverse Machine Translation: Tricks of the Trade](examples/translation_moe/README.md)
 - [Wu et al. (2019): Pay Less Attention with Lightweight and Dynamic Convolutions](examples/pay_less_attention_paper/README.md)
 - [Edunov et al. (2018): Understanding Back-Translation at Scale](examples/backtranslation/README.md)
diff --git a/examples/wav2vec/README.md b/examples/wav2vec/README.md
new file mode 100644
index 0000000000..325e25420f
--- /dev/null
+++ b/examples/wav2vec/README.md
@@ -0,0 +1,31 @@
+# wav2vec
+
+Example to train a wav2vec model as described in [wav2vec: Unsupervised Pre-training for Speech Recognition (Schneider et al., 2019)](https://arxiv.org/abs/1904.05862).
+
+## Training a new model with the CLI tools
+
+Given a directory containing wav files to be used for pretraining (we recommend splitting each file into separate file 10 to 30 seconds in length)
+
+### Prepare training data manifest:
+
+```
+$ python scripts/wav2vec_manifest.py /path/to/waves --dest /manifest/path --ext wav
+```
+
+### Train a wav2vec model:
+
+```
+$ python train.py /manifest/path --save-dir /model/path --num-workers 6 --fp16 --max-update 400000 --save-interval 1 --no-epoch-checkpoints \
+--arch wav2vec --task audio_pretraining --lr 1e-06 --min-lr 1e-09 --optimizer adam --max-lr 0.005 --lr-scheduler cosine \
+--conv-feature-layers [(512, 10, 5), (512, 8, 4), (512, 4, 2), (512, 4, 2), (512, 4, 2), (512, 1, 1), (512, 1, 1)] \
+--conv-aggregator-layers [(512, 2, 1), (512, 3, 1), (512, 4, 1), (512, 5, 1), (512, 6, 1), (512, 7, 1), (512, 8, 1), (512, 9, 1), (512, 10, 1), (512, 11, 1), (512, 12, 1), (512, 13, 1)] \
+--skip-connections-agg --residual-scale 0.5 --log-compression --warmup-updates 500 --warmup-init-lr 1e-07 --criterion binary_cross_entropy --num-negatives 10 \
+--max-sample-size 150000 --max-tokens 1500000 ---skip-invalid-size-inputs-valid-test
+```
+
+### Extract embeddings from the downstream task data:
+
+```
+$ PYTHONPATH /path/to/fairseq python scripts/wav2vec_featurize.py --input /path/to/task/waves --output /path/to/output \
+--model /model/path/checkpoint_best.pt --split train valid test
+```
diff --git a/fairseq/criterions/binary_cross_entropy.py b/fairseq/criterions/binary_cross_entropy.py
new file mode 100644
index 0000000000..06f269692c
--- /dev/null
+++ b/fairseq/criterions/binary_cross_entropy.py
@@ -0,0 +1,73 @@
+# Copyright (c) 2017-present, Facebook, Inc.
+# All rights reserved.
+#
+# This source code is licensed under the license found in the LICENSE file in
+# the root directory of this source tree. An additional grant of patent rights
+# can be found in the PATENTS file in the same directory.
+
+import math
+import torch
+import torch.nn.functional as F
+
+from fairseq import utils
+
+from . import FairseqCriterion, register_criterion
+
+
+@register_criterion('binary_cross_entropy')
+class BinaryCrossEntropyCriterion(FairseqCriterion):
+
+    def __init__(self, args, task):
+        super().__init__(args, task)
+
+    def forward(self, model, sample, reduce=True):
+        """Compute the loss for the given sample.
+
+        Returns a tuple with three elements:
+        1) the loss
+        2) the sample size, which is used as the denominator for the gradient
+        3) logging outputs to display while training
+        """
+        net_output = model(**sample['net_input'])
+        logits = model.get_logits(net_output).float()
+        target = model.get_targets(sample, net_output, expand_steps=False).float()
+
+        if hasattr(model, 'get_target_weights'):
+            weights = model.get_target_weights(target, net_output)
+            if torch.is_tensor(weights):
+                weights = weights.float()
+        else:
+            weights = 1.
+
+        loss = F.binary_cross_entropy_with_logits(logits, target, reduce=False)
+
+        loss = loss * weights
+
+        if reduce:
+            loss = loss.sum()
+
+        sample_size = target.numel()
+        logging_output = {
+            'loss': utils.item(loss.data) if reduce else loss.data,
+            'ntokens': sample_size,
+            'nsentences': logits.size(0),
+            'sample_size': sample_size,
+        }
+        return loss, sample_size, logging_output
+
+    @staticmethod
+    def aggregate_logging_outputs(logging_outputs):
+        """Aggregate logging outputs from data parallel training."""
+        loss_sum = sum(log.get('loss', 0) for log in logging_outputs)
+        ntokens = sum(log.get('ntokens', 0) for log in logging_outputs)
+        nsentences = sum(log.get('nsentences', 0) for log in logging_outputs)
+        sample_size = sum(log.get('sample_size', 0) for log in logging_outputs)
+        agg_output = {
+            'loss': loss_sum / sample_size / math.log(2),
+            'ntokens': ntokens,
+            'nsentences': nsentences,
+            'sample_size': sample_size,
+        }
+        if sample_size != ntokens:
+            agg_output['nll_loss'] = loss_sum / ntokens / math.log(2)
+        return agg_output
\ No newline at end of file
diff --git a/fairseq/criterions/fairseq_criterion.py b/fairseq/criterions/fairseq_criterion.py
index 5c910f4e65..fbe5afc8e8 100644
--- a/fairseq/criterions/fairseq_criterion.py
+++ b/fairseq/criterions/fairseq_criterion.py
@@ -13,7 +13,7 @@ class FairseqCriterion(_Loss):
     def __init__(self, args, task):
         super().__init__()
         self.args = args
-        self.padding_idx = task.target_dictionary.pad()
+        self.padding_idx = task.target_dictionary.pad() if task.target_dictionary is not None else -100
 
     @staticmethod
     def add_args(parser):
diff --git a/fairseq/data/__init__.py b/fairseq/data/__init__.py
index e4ef89541a..ffee78429e 100644
--- a/fairseq/data/__init__.py
+++ b/fairseq/data/__init__.py
@@ -10,6 +10,7 @@
 
 from .fairseq_dataset import FairseqDataset
 
+from .audio.raw_audio_dataset import RawAudioDataset
 from .backtranslation_dataset import BacktranslationDataset
 from .block_pair_dataset import BlockPairDataset
 from .concat_dataset import ConcatDataset
@@ -51,6 +52,7 @@
     'MMapIndexedDataset',
     'MonolingualDataset',
     'NoisingDataset',
+    'RawAudioDataset',
     'RoundRobinZipDatasets',
     'ShardedIterator',
     'TokenBlockDataset',
diff --git a/fairseq/data/audio/raw_audio_dataset.py b/fairseq/data/audio/raw_audio_dataset.py
new file mode 100644
index 0000000000..7fb25cc6c3
--- /dev/null
+++ b/fairseq/data/audio/raw_audio_dataset.py
@@ -0,0 +1,128 @@
+# Copyright (c) 2017-present, Facebook, Inc.
+# All rights reserved.
+#
+# This source code is licensed under the license found in the LICENSE file in
+# the root directory of this source tree. An additional grant of patent rights
+# can be found in the PATENTS file in the same directory.
+
+
+import os
+import numpy as np
+import sys
+import torch
+import torch.nn.functional as F
+
+from .. import FairseqDataset
+
+
+class RawAudioDataset(FairseqDataset):
+
+    def __init__(self, manifest_path, sample_rate, max_sample_size=None, min_sample_size=None,
+                 shuffle=True):
+        super().__init__()
+
+        self.sample_rate = sample_rate
+        self.fnames = []
+        self.sizes = []
+        self.max_sample_size = max_sample_size if max_sample_size is not None else sys.maxsize
+        self.min_sample_size = min_sample_size if min_sample_size is not None else self.max_sample_size
+
+        with open(manifest_path, 'r') as f:
+            self.root_dir = f.readline().strip()
+            for line in f:
+                items = line.strip().split('\t')
+                assert len(items) == 2, line
+                self.fnames.append(items[0])
+                self.sizes.append(int(items[1]))
+        self.shuffle = shuffle
+
+    def __getitem__(self, index):
+        fname = os.path.join(self.root_dir, self.fnames[index])
+        import soundfile as sf
+
+        wav, curr_sample_rate = sf.read(fname)
+        feats = torch.from_numpy(wav).float()
+
+        if feats.dim() == 2:
+            feats = feats.mean(-1)
+
+        if curr_sample_rate != self.sample_rate:
+            factor = self.sample_rate / curr_sample_rate
+            feats = self.resample(feats, factor)
+
+        assert feats.dim() == 1, feats.dim()
+
+        return {
+            'id': index,
+            'source': feats,
+        }
+
+    def resample(self, x, factor):
+        return F.interpolate(x.view(1, 1, -1), scale_factor=factor).squeeze()
+
+    def __len__(self):
+        return len(self.fnames)
+
+    def collater(self, samples):
+        if len(samples) == 0:
+            return {}
+
+        sources = [s['source'] for s in samples]
+        sizes = [len(s) for s in sources]
+        target_size = min(min(sizes), self.max_sample_size)
+
+        if self.min_sample_size < target_size:
+            target_size = np.random.randint(self.min_sample_size, target_size + 1)
+
+        collated_sources = sources[0].new(len(sources), target_size)
+        for i, (source, size) in enumerate(zip(sources, sizes)):
+            diff = size - target_size
+            assert diff >= 0
+            if diff == 0:
+                collated_sources[i] = source
+            else:
+                start = np.random.randint(0, diff + 1)
+                end = size - diff + start
+                collated_sources[i] = source[start:end]
+
+        return {
+            'id': torch.LongTensor([s['id'] for s in samples]),
+            'net_input': {
+                'source': collated_sources,
+            },
+        }
+
+    def get_dummy_batch(
+            self, num_tokens, max_positions, src_len=2048, tgt_len=128,
+    ):
+        """Return a dummy batch with a given number of tokens."""
+        if isinstance(max_positions, float) or isinstance(max_positions, int):
+            src_len = min(src_len, max_positions)
+        bsz = num_tokens // src_len
+        return self.collater([
+            {
+                'id': i,
+                'source': torch.rand(src_len),
+            }
+            for i in range(bsz)
+        ])
+
+    def num_tokens(self, index):
+        return self.size(index)
+
+    def size(self, index):
+        """Return an example's size as a float or tuple. This value is used when
+        filtering a dataset with ``--max-positions``."""
+        return min(self.sizes[index], self.max_sample_size)
+
+    def ordered_indices(self):
+        """Return an ordered list of indices. Batches will be constructed based
+        on this order."""
+
+        if self.shuffle:
+            order = [np.random.permutation(len(self))]
+        else:
+            order = [np.arange(len(self))]
+
+        order.append(self.sizes)
+        return np.lexsort(order)
diff --git a/fairseq/models/wav2vec.py b/fairseq/models/wav2vec.py
new file mode 100644
index 0000000000..e89ede158b
--- /dev/null
+++ b/fairseq/models/wav2vec.py
@@ -0,0 +1,475 @@
+# Copyright (c) 2017-present, Facebook, Inc.
+# All rights reserved.
+#
+# This source code is licensed under the license found in the LICENSE file in
+# the root directory of this source tree. An additional grant of patent rights
+# can be found in the PATENTS file in the same directory.
+
+import sys
+
+import math
+
+import torch
+import torch.nn as nn
+import torch.nn.functional as F
+
+from . import (
+    BaseFairseqModel, register_model, register_model_architecture
+)
+
+
+@register_model('wav2vec')
+class Wav2VecModel(BaseFairseqModel):
+
+    @staticmethod
+    def add_args(parser):
+        """Add model-specific arguments to the parser."""
+        parser.add_argument('--prediction-steps', type=int, metavar='N', help='number of steps ahead to predict')
+        parser.add_argument('--sample-distance', type=int, metavar='N',
+                            help='sample distance from target. does not work properly with cross-sampling')
+        parser.add_argument('--cross-sample-negatives', action='store_true',
+                            help='whether to sample negatives across examples in the same batch')
+        parser.add_argument('--num-negatives', type=int, metavar='N', help='number of negative examples')
+        parser.add_argument('--conv-feature-layers', type=str, metavar='EXPR',
+                            help='convolutional feature extraction layers [(dim, kernel_size, stride), ...]')
+        parser.add_argument('--conv-aggregator-layers', type=str, metavar='EXPR',
+                            help='convolutional feature extraction layers [(dim, kernel_size, stride), ...]')
+        parser.add_argument('--dropout', type=float, metavar='D', help='dropout to apply within the model')
+        parser.add_argument('--dropout-features', type=float, metavar='D', help='dropout to apply to the features')
+        parser.add_argument('--dropout-agg', type=float, metavar='D', help='dropout to apply after aggregation step')
+        parser.add_argument('--encoder', type=str, choices=['cnn'], help='type of encoder to use')
+        parser.add_argument('--aggregator', type=str, choices=['cnn', 'gru'],
+                            help='type of aggregator to use')
+        parser.add_argument('--gru-dim', type=int, metavar='N', help='GRU dimensionality')
+
+        parser.add_argument('--no-conv-bias', action='store_true',
+                            help='if set, does not learn bias for conv layers')
+        parser.add_argument('--agg-zero-pad', action='store_true',
+                            help='if set, zero pads in aggregator instead of repl pad')
+
+        parser.add_argument('--skip-connections-feat', action='store_true',
+                            help='if set, adds skip connections to the feature extractor')
+        parser.add_argument('--skip-connections-agg', action='store_true',
+                            help='if set, adds skip connections to the aggregator')
+        parser.add_argument('--residual-scale', type=float, metavar='D',
+                            help='scales residual by sqrt(value)')
+
+        parser.add_argument('--log-compression', action='store_true',
+                            help='if set, adds a log compression to feature extractor')
+
+        parser.add_argument('--balanced-classes', action='store_true',
+                            help='if set, loss is scaled to balance for number of negatives')
+        parser.add_argument('--project-features', choices=['none', 'same', 'new'],
+                            help='if not none, features are projected using the (same or new) aggregator')
+
+        parser.add_argument('--non-affine-group-norm', action='store_true',
+                            help='if set, group norm is not affine')
+
+        parser.add_argument('--offset', help='if set, introduces an offset from target to predictions. '
+                                             'if set to "auto", it is computed automatically from the receptive field')
+
+    @classmethod
+    def build_model(cls, args, task):
+        """Build a new model instance."""
+
+        # make sure all arguments are present in older models
+        base_wav2vec_architecture(args)
+
+        model = Wav2VecModel(args)
+        print(model)
+        return model
+
+    def __init__(self, args):
+        super().__init__()
+
+        self.prediction_steps = args.prediction_steps
+
+        offset = args.offset
+
+        if args.encoder == 'cnn':
+            feature_enc_layers = eval(args.conv_feature_layers)
+            self.feature_extractor = ConvFeatureExtractionModel(
+                conv_layers=feature_enc_layers,
+                dropout=0.,
+                log_compression=args.log_compression,
+                skip_connections=args.skip_connections_feat,
+                residual_scale=args.residual_scale,
+                non_affine_group_norm=args.non_affine_group_norm,
+            )
+            embed = feature_enc_layers[-1][0]
+        else:
+            raise Exception('unknown encoder type ' + args.encoder)
+
+        if args.offset == 'auto':
+            assert args.encoder == 'cnn'
+            jin = 0
+            rin = 0
+            for _, k, stride in feature_enc_layers:
+                if rin == 0:
+                    rin = k
+                rin = rin + (k - 1) * jin
+                if jin == 0:
+                    jin = stride
+                else:
+                    jin *= stride
+            offset = math.ceil(rin / jin)
+
+        offset = int(offset)
+
+        def make_aggregator():
+            if args.aggregator == 'cnn':
+                agg_layers = eval(args.conv_aggregator_layers)
+                agg_dim = agg_layers[-1][0]
+                feature_aggregator = ConvAggegator(
+                    conv_layers=agg_layers,
+                    embed=embed,
+                    dropout=args.dropout,
+                    skip_connections=args.skip_connections_agg,
+                    residual_scale=args.residual_scale,
+                    non_affine_group_norm=args.non_affine_group_norm,
+                    conv_bias=not args.no_conv_bias,
+                    zero_pad=args.agg_zero_pad,
+                )
+            elif args.aggregator == 'gru':
+                agg_dim = args.gru_dim
+                feature_aggregator = nn.Sequential(
+                    TransposeLast(),
+                    nn.GRU(
+                        input_size=embed,
+                        hidden_size=agg_dim,
+                        num_layers=1,
+                        dropout=args.dropout,
+                    ),
+                    TransposeLast(deconstruct_idx=0),
+                )
+            else:
+                raise Exception('unknown aggregator type ' + args.aggregator)
+
+            return feature_aggregator, agg_dim
+
+        self.feature_aggregator, agg_dim = make_aggregator()
+
+        self.wav2vec_predictions = Wav2VecPredictionsModel(
+            in_dim=agg_dim,
+            out_dim=embed,
+            prediction_steps=args.prediction_steps,
+            n_negatives=args.num_negatives,
+            cross_sample_negatives=args.cross_sample_negatives,
+            sample_distance=args.sample_distance,
+            dropout=args.dropout,
+            offset=offset,
+            balanced_classes=args.balanced_classes,
+        )
+
+        self.dropout_feats = nn.Dropout(p=args.dropout_features)
+        self.dropout_agg = nn.Dropout(p=args.dropout_agg)
+
+        if args.project_features == 'none':
+            self.project_features = None
+        elif args.project_features == 'same':
+            self.project_features = self.feature_aggregator
+        elif args.project_features == 'new':
+            self.project_features, _ = make_aggregator()
+
+    def forward(self, source):
+        result = {}
+
+        features = self.feature_extractor(source)
+
+        x = self.dropout_feats(features)
+        x = self.feature_aggregator(x)
+        x = self.dropout_agg(x)
+
+        if self.project_features is not None:
+            features = self.project_features(features)
+        x, targets = self.wav2vec_predictions(x, features)
+        result['cpc_logits'] = x
+        result['cpc_targets'] = targets
+
+        return result
+
+    def upgrade_state_dict_named(self, state_dict, name):
+        return state_dict
+
+    def max_positions(self):
+        """Maximum length supported by the model."""
+        return sys.maxsize
+
+    def get_logits(self, net_output):
+        logits = net_output['cpc_logits']
+        return logits
+
+    def get_targets(self, sample, net_output, expand_steps=True):
+        t = net_output['cpc_targets']
+        return t.contiguous()
+
+    def get_target_weights(self, targets, net_output):
+        targets = net_output['cpc_targets']
+        if isinstance(targets, tuple) and targets[-1] is not None:
+            return targets[-1]
+        return 1.
+
+
+class TransposeLast(nn.Module):
+    def __init__(self, deconstruct_idx=None):
+        super().__init__()
+        self.deconstruct_idx = deconstruct_idx
+
+    def forward(self, x):
+        if self.deconstruct_idx is not None:
+            x = x[self.deconstruct_idx]
+        return x.transpose(-2, -1)
+
+
+class Fp32GroupNorm(nn.GroupNorm):
+    def __init__(self, *args, **kwargs):
+        super().__init__(*args, **kwargs)
+
+    def forward(self, input):
+        output = F.group_norm(
+            input.float(), self.num_groups, self.weight.float() if self.weight is not None else None,
+            self.bias.float() if self.bias is not None else None, self.eps)
+        return output.type_as(input)
+
+
+class Fp32LayerNorm(nn.LayerNorm):
+    def __init__(self, *args, **kwargs):
+        super().__init__(*args, **kwargs)
+
+    def forward(self, input):
+        output = F.layer_norm(
+            input.float(), self.normalized_shape, self.weight.float() if self.weight is not None else None,
+            self.bias.float() if self.bias is not None else None, self.eps)
+        return output.type_as(input)
+
+
+def norm_block(is_layer_norm, dim, affine=True):
+    if is_layer_norm:
+        mod = nn.Sequential(
+            TransposeLast(),
+            Fp32LayerNorm(dim, elementwise_affine=affine),
+            TransposeLast(),
+        )
+    else:
+        mod = Fp32GroupNorm(1, dim, affine=affine)
+
+    return mod
+
+
+class ConvFeatureExtractionModel(nn.Module):
+    def __init__(self, conv_layers, dropout, log_compression, skip_connections, residual_scale, non_affine_group_norm):
+        super().__init__()
+
+        def block(n_in, n_out, k, stride):
+            return nn.Sequential(
+                nn.Conv1d(n_in, n_out, k, stride=stride, bias=False),
+                nn.Dropout(p=dropout),
+                norm_block(is_layer_norm=False, dim=n_out, affine=not non_affine_group_norm),
+                nn.ReLU(),
+            )
+
+        in_d = 1
+        self.conv_layers = nn.ModuleList()
+        for i, (dim, k, stride) in enumerate(conv_layers):
+            self.conv_layers.append(
+                block(in_d, dim, k, stride))
+            in_d = dim
+
+        self.log_compression = log_compression
+        self.skip_connections = skip_connections
+        self.residual_scale = math.sqrt(residual_scale)
+
+    def forward(self, x):
+        # BxT -> BxCxT
+        x = x.unsqueeze(1)
+
+        for conv in self.conv_layers:
+            residual = x
+            x = conv(x)
+            if self.skip_connections and x.size(1) == residual.size(1):
+                tsz = x.size(2)
+                r_tsz = residual.size(2)
+                residual = residual[..., ::r_tsz // tsz][..., :tsz]
+                x = (x + residual) * self.residual_scale
+
+        if self.log_compression:
+            x = x.abs()
+            x = x + 1
+            x = x.log()
+
+        return x
+
+
+class ZeroPad1d(nn.Module):
+    def __init__(self, pad_left, pad_right):
+        super().__init__()
+        self.pad_left = pad_left
+        self.pad_right = pad_right
+
+    def forward(self, x):
+        return F.pad(x, (self.pad_left, self.pad_right))
+
+
+class ConvAggegator(nn.Module):
+    def __init__(self, conv_layers, embed, dropout, skip_connections, residual_scale, non_affine_group_norm, conv_bias,
+                 zero_pad):
+        super().__init__()
+
+        def block(n_in, n_out, k, stride):
+            # padding dims only really make sense for stride = 1
+            ka = k // 2
+            kb = ka - 1 if k % 2 == 0 else ka
+
+            pad = ZeroPad1d(ka + kb, 0) if zero_pad else nn.ReplicationPad1d((ka + kb, 0))
+
+            return nn.Sequential(
+                pad,
+                nn.Conv1d(n_in, n_out, k, stride=stride, bias=conv_bias),
+                nn.Dropout(p=dropout),
+                norm_block(False, n_out, affine=not non_affine_group_norm),
+                nn.ReLU(),
+            )
+
+        in_d = embed
+        self.conv_layers = nn.ModuleList()
+        self.residual_proj = nn.ModuleList()
+        for i, (dim, k, stride) in enumerate(conv_layers):
+            if in_d != dim and skip_connections:
+                self.residual_proj.append(
+                    nn.Conv1d(in_d, dim, 1, bias=False),
+                )
+            else:
+                self.residual_proj.append(None)
+
+            self.conv_layers.append(
+                block(in_d, dim, k, stride))
+            in_d = dim
+        self.conv_layers = nn.Sequential(*self.conv_layers)
+        self.skip_connections = skip_connections
+        self.residual_scale = math.sqrt(residual_scale)
+
+    def forward(self, x):
+        for rproj, conv in zip(self.residual_proj, self.conv_layers):
+            residual = x
+            x = conv(x)
+            if self.skip_connections:
+                if rproj != None:
+                    residual = rproj(residual)
+                x = (x + residual) * self.residual_scale
+        return x
+
+
+class Wav2VecPredictionsModel(nn.Module):
+    def __init__(self, in_dim, out_dim, prediction_steps, n_negatives, cross_sample_negatives, sample_distance,
+                 dropout, offset, balanced_classes):
+        super().__init__()
+
+        self.n_negatives = n_negatives
+        self.cross_sample_negatives = cross_sample_negatives
+        self.sample_distance = sample_distance
+
+        self.project_to_steps = nn.ConvTranspose2d(in_dim, out_dim, (1, prediction_steps))
+        self.dropout = nn.Dropout(p=dropout)
+        self.offset = offset
+        self.balanced_classes = balanced_classes
+
+    def sample_negatives(self, y):
+        bsz, fsz, tsz = y.shape
+
+        y = y.transpose(0, 1)  # BCT -> CBT
+        y = y.contiguous().view(fsz, -1)  # CBT => C(BxT)
+
+        if self.cross_sample_negatives:
+            high = tsz * bsz
+            assert self.sample_distance is None, 'sample distance is not supported with cross sampling'
+        else:
+            high = tsz if self.sample_distance is None else min(tsz, self.sample_distance)
+
+        neg_idxs = torch.randint(low=0, high=high, size=(bsz, self.n_negatives * tsz))
+
+        if self.sample_distance is not None and self.sample_distance < tsz:
+            neg_idxs += torch.cat(
+                [torch.arange(start=1, end=tsz - self.sample_distance, device=neg_idxs.device, dtype=neg_idxs.dtype),
+                 torch.arange(start=tsz - self.sample_distance, end=tsz - self.sample_distance * 2 - 1, step=-1,
+                              device=neg_idxs.device, dtype=neg_idxs.dtype)])
+
+        if not self.cross_sample_negatives:
+            for i in range(1, bsz):
+                neg_idxs[i] += i * high
+
+        negs = y[..., neg_idxs.view(-1)]
+        negs = negs.view(fsz, bsz, self.n_negatives, tsz).permute(2, 1, 0, 3)  # to NxBxCxT
+
+        return negs
+
+    def forward(self, x, y):
+        negatives = self.sample_negatives(y)
+        y = y.unsqueeze(0)
+        targets = torch.cat([y, negatives], dim=0)
+
+        x = x.unsqueeze(-1)
+        x = self.project_to_steps(x)  # BxCxTxS
+        x = self.dropout(x)
+        x = x.unsqueeze(0).expand(targets.size(0), -1, -1, -1, -1)
+
+        copies, bsz, dim, tsz, steps = x.shape
+        steps = min(steps, tsz - self.offset)
+        predictions = x.new(bsz * copies * (tsz - self.offset + 1) * steps - ((steps + 1) * steps // 2) * copies * bsz)
+        labels = torch.zeros_like(predictions)
+        weights = torch.full_like(labels, 1 / self.n_negatives) if self.balanced_classes else None
+
+        start = end = 0
+        for i in range(steps):
+            offset = i + self.offset
+            end = start + (tsz - offset) * bsz * copies
+            pos_num = (end - start) // copies
+            predictions[start:end] = (x[..., :-offset, i] * targets[..., offset:]).sum(dim=2).flatten()
+            labels[start:start + pos_num] = 1.
+            if weights is not None:
+                weights[start:start + pos_num] = 1.
+            start = end
+        assert end == predictions.numel(), '{} != {}'.format(end, predictions.numel())
+
+        if weights is not None:
+            labels = (labels, weights)
+
+        return predictions, labels
+
+
+@register_model_architecture('wav2vec', 'wav2vec')
+def base_wav2vec_architecture(args):
+    conv_feature_layers = '[(512, 10, 5)]'
+    conv_feature_layers += ' + [(512, 8, 4)]'
+    conv_feature_layers += ' + [(512, 4, 2)] * 3'
+    args.conv_feature_layers = getattr(args, 'conv_feature_layers', conv_feature_layers)
+
+    args.conv_aggregator_layers = getattr(args, 'conv_aggregator_layers', '[(512, 3, 1)] * 9')
+
+    args.prediction_steps = getattr(args, 'prediction_steps', 12)
+    args.num_negatives = getattr(args, 'num_negatives', 1)
+    args.sample_distance = getattr(args, 'sample_distance', None)
+    args.cross_sample_negatives = getattr(args, 'cross_sample_negatives', False)
+
+    args.dropout = getattr(args, 'dropout', 0.)
+    args.dropout_features = getattr(args, 'dropout_features', 0.)
+    args.dropout_agg = getattr(args, 'dropout_agg', 0.)
+    args.encoder = getattr(args, 'encoder', 'cnn')
+    args.aggregator = getattr(args, 'aggregator', 'cnn')
+
+    args.skip_connections_feat = getattr(args, 'skip_connections_feat', False)
+    args.skip_connections_agg = getattr(args, 'skip_connections_agg', False)
+    args.residual_scale = getattr(args, 'residual_scale', 0.5)
+
+    args.gru_dim = getattr(args, 'gru_dim', 512)
+
+    args.no_conv_bias = getattr(args, 'no_conv_bias', False)
+    args.agg_zero_pad = getattr(args, 'agg_zero_pad', False)
+
+    args.log_compression = getattr(args, 'log_compression', False)
+
+    args.balanced_classes = getattr(args, 'balanced_classes', False)
+    args.project_features = getattr(args, 'project_features', 'none')
+
+    args.non_affine_group_norm = getattr(args, 'non_affine_group_norm', False)
+
+    args.offset = getattr(args, 'offset', 'auto')
diff --git a/fairseq/tasks/audio_pretraining.py b/fairseq/tasks/audio_pretraining.py
new file mode 100644
index 0000000000..ac97c38ada
--- /dev/null
+++ b/fairseq/tasks/audio_pretraining.py
@@ -0,0 +1,60 @@
+# Copyright (c) 2017-present, Facebook, Inc.
+# All rights reserved.
+#
+# This source code is licensed under the license found in the LICENSE file in
+# the root directory of this source tree. An additional grant of patent rights
+# can be found in the PATENTS file in the same directory.
+
+import os
+
+from fairseq.data import RawAudioDataset
+from . import FairseqTask, register_task
+
+
+@register_task('audio_pretraining')
+class AudioPretrainingTask(FairseqTask):
+    """
+
+    """
+
+    @staticmethod
+    def add_args(parser):
+        """Add task-specific arguments to the parser."""
+        parser.add_argument('data', help='path to data directory')
+        parser.add_argument('--sample-rate', default=16000, type=int,
+                            help='target sample rate. audio files will be up/down sampled to this rate')
+        parser.add_argument('--max-sample-size', default=None, type=int,
+                            help='max sample size to crop to for batching. default = min sample length')
+        parser.add_argument('--min-sample-size', default=None, type=int,
+                            help='min sample size to crop to for batching. default = same as --max-sample-size')
+
+    def __init__(self, args):
+        super().__init__(args)
+
+    @classmethod
+    def setup_task(cls, args, **kwargs):
+        """Setup the task (e.g., load dictionaries).
+
+        Args:
+            args (argparse.Namespace): parsed command-line arguments
+        """
+        return cls(args)
+
+    def load_dataset(self, split, **kwargs):
+        """Load a given dataset split.
+
+        Args:
+            split (str): name of the split (e.g., train, valid, test)
+        """
+
+        manifest = os.path.join(self.args.data, '{}.tsv'.format(split))
+        self.datasets[split] = RawAudioDataset(manifest,
+                                                 sample_rate=self.args.sample_rate,
+                                                 max_sample_size=self.args.max_sample_size,
+                                                 min_sample_size=self.args.min_sample_size)
+
+    @property
+    def target_dictionary(self):
+        """Return the :class:`~fairseq.data.Dictionary` for the language
+        model."""
+        return None
\ No newline at end of file
diff --git a/fairseq/trainer.py b/fairseq/trainer.py
index d3e09efcaf..220eb81429 100644
--- a/fairseq/trainer.py
+++ b/fairseq/trainer.py
@@ -233,11 +233,11 @@ def train_step(self, samples, dummy_batch=False, raise_oom=False):
         # forward and backward pass
         logging_outputs, sample_sizes, ooms = [], [], 0
         for i, sample in enumerate(samples):
-            sample = self._prepare_sample(sample)
+            sample = self._prepare_sample(sample, self.args.fp16)
             if sample is None:
                 # when sample is None, run forward/backward on a dummy batch
                 # and ignore the resulting gradients
-                sample = self._prepare_sample(self._dummy_batch)
+                sample = self._prepare_sample(self._dummy_batch, self.args.fp16)
                 ignore_grad = True
             else:
                 ignore_grad = False
@@ -381,9 +381,9 @@ def valid_step(self, sample, raise_oom=False):
             self.model.eval()
             self.criterion.eval()
 
-            sample = self._prepare_sample(sample)
+            sample = self._prepare_sample(sample, self.args.fp16)
             if sample is None:
-                sample = self._prepare_sample(self._dummy_batch)
+                sample = self._prepare_sample(self._dummy_batch, self.args.fp16)
                 ignore_results = True
             else:
                 ignore_results = False
@@ -488,12 +488,19 @@ def set_num_updates(self, num_updates):
         self._num_updates = num_updates
         self.lr_step_update()
 
-    def _prepare_sample(self, sample):
+    def _prepare_sample(self, sample, fp16):
         if sample is None or len(sample) == 0:
             return None
+
         if self.cuda:
             sample = utils.move_to_cuda(sample)
-        return sample
+
+        def apply_half(t):
+            if t.dtype is torch.float32:
+                return t.half()
+            return t
+
+        return utils.apply(apply_half, sample) if fp16 else sample
 
     def _set_seed(self):
         # Set seed based on args.seed and the update number so that we get
diff --git a/fairseq/utils.py b/fairseq/utils.py
index 10e70a8d08..7af1400275 100644
--- a/fairseq/utils.py
+++ b/fairseq/utils.py
@@ -31,24 +31,26 @@ def load_ensemble_for_inference(filenames, task, model_arg_overrides=None):
     )
 
 
-def move_to_cuda(sample):
+def apply(f, sample):
     if len(sample) == 0:
         return {}
+    if torch.is_tensor(sample):
+        return f(sample)
+    elif isinstance(sample, dict):
+        return {
+            key: apply(f, value)
+            for key, value in sample.items()
+        }
+    elif isinstance(sample, list):
+        return [apply(f, x) for x in sample]
+    else:
+        return sample
 
-    def _move_to_cuda(maybe_tensor):
-        if torch.is_tensor(maybe_tensor):
-            return maybe_tensor.cuda()
-        elif isinstance(maybe_tensor, dict):
-            return {
-                key: _move_to_cuda(value)
-                for key, value in maybe_tensor.items()
-            }
-        elif isinstance(maybe_tensor, list):
-            return [_move_to_cuda(x) for x in maybe_tensor]
-        else:
-            return maybe_tensor
-
-    return _move_to_cuda(sample)
+
+def move_to_cuda(sample):
+    def _move_to_cuda(tensor):
+            return tensor.cuda()
+    return apply(_move_to_cuda, sample)
 
 
 INCREMENTAL_STATE_INSTANCE_ID = defaultdict(lambda: 0)
diff --git a/scripts/wav2vec_featurize.py b/scripts/wav2vec_featurize.py
new file mode 100644
index 0000000000..70764d7933
--- /dev/null
+++ b/scripts/wav2vec_featurize.py
@@ -0,0 +1,231 @@
+""" Helper script to pre-compute embeddings for a wav2letter++ dataset
+"""
+
+import glob, os
+import tqdm
+from shutil import copy
+
+import soundfile as sf
+
+import h5py
+import numpy as np
+
+import torch
+from torch import nn
+
+from fairseq.models.wav2vec import Wav2VecModel
+
+import argparse
+
+
+def read_audio(fname):
+    """ Load an audio file and return PCM along with the sample rate """
+
+    wav, sr = sf.read(fname)
+    assert sr == 16e3
+
+    return wav, 16e3
+
+
+class PretrainedWav2VecModel(nn.Module):
+
+    def __init__(self, fname):
+        super().__init__()
+
+        checkpoint = torch.load(fname)
+        self.args = checkpoint["args"]
+        model = Wav2VecModel.build_model(self.args, None)
+        model.load_state_dict(checkpoint["model"])
+        model.eval()
+
+        self.model = model
+
+    def forward(self, x):
+        with torch.no_grad():
+            z = self.model.feature_extractor(x)
+            if isinstance(z, tuple):
+                z = z[0]
+            c = self.model.feature_aggregator(z)
+        return z, c
+
+
+class EmbeddingWriterConfig(argparse.ArgumentParser):
+
+    def __init__(self):
+        super().__init__("Pre-compute embeddings for wav2letter++ datasets")
+
+        kwargs = {"action": "store", "type": str, "required": True}
+
+        self.add_argument("--input", "-i",
+                          help="Input Directory", **kwargs)
+        self.add_argument("--output", "-o",
+                          help="Output Directory", **kwargs)
+        self.add_argument("--model",
+                          help="Path to model checkpoint", **kwargs)
+        self.add_argument("--split",
+                          help="Dataset Splits", nargs='+', **kwargs)
+        self.add_argument("--ext", default="wav", required=False,
+                          help="Audio file extension")
+
+        self.add_argument("--no-copy-labels", action="store_true",
+                          help="Do not copy label files. Useful for large datasets, use --targetdir in wav2letter then.")
+        self.add_argument("--use-feat", action="store_true",
+                          help="Use the feature vector ('z') instead of context vector ('c') for features")
+        self.add_argument("--gpu",
+                          help="GPU to use", default=0, type=int)
+
+
+class Prediction():
+    """ Lightweight wrapper around a fairspeech embedding model """
+
+    def __init__(self, fname, gpu=0):
+        self.gpu = gpu
+        self.model = PretrainedWav2VecModel(fname).cuda(gpu)
+
+    def __call__(self, x):
+        x = torch.from_numpy(x).float().cuda(self.gpu)
+        with torch.no_grad():
+            z, c = self.model(x.unsqueeze(0))
+
+        return z.squeeze(0).cpu().numpy(), c.squeeze(0).cpu().numpy()
+
+
+class H5Writer():
+    """ Write features as hdf5 file in wav2letter++ compatible format """
+
+    def __init__(self, fname):
+        self.fname = fname
+        os.makedirs(os.path.dirname(self.fname), exist_ok=True)
+
+    def write(self, data):
+        channel, T = data.shape
+
+        with h5py.File(self.fname, "w") as out_ds:
+            data = data.T.flatten()
+            out_ds["features"] = data
+            out_ds["info"] = np.array([16e3 // 160, T, channel])
+
+
+class EmbeddingDatasetWriter(object):
+    """ Given a model and a wav2letter++ dataset, pre-compute and store embeddings
+
+    Args:
+        input_root, str :
+            Path to the wav2letter++ dataset
+        output_root, str :
+            Desired output directory. Will be created if non-existent
+        split, str :
+            Dataset split
+    """
+
+    def __init__(self, input_root, output_root, split,
+                 model_fname,
+                 extension="wav",
+                 gpu=0,
+                 verbose=False,
+                 use_feat=False,
+                 ):
+
+        assert os.path.exists(model_fname)
+
+        self.model_fname = model_fname
+        self.model = Prediction(self.model_fname, gpu)
+
+        self.input_root = input_root
+        self.output_root = output_root
+        self.split = split
+        self.verbose = verbose
+        self.extension = extension
+        self.use_feat = use_feat
+
+        assert os.path.exists(self.input_path), \
+            "Input path '{}' does not exist".format(self.input_path)
+
+    def _progress(self, iterable, **kwargs):
+        if self.verbose:
+            return tqdm.tqdm(iterable, **kwargs)
+        return iterable
+
+    def require_output_path(self, fname=None):
+        path = self.get_output_path(fname)
+        os.makedirs(path, exist_ok=True)
+
+    @property
+    def input_path(self):
+        return self.get_input_path()
+
+    @property
+    def output_path(self):
+        return self.get_output_path()
+
+    def get_input_path(self, fname=None):
+        if fname is None:
+            return os.path.join(self.input_root, self.split)
+        return os.path.join(self.get_input_path(), fname)
+
+    def get_output_path(self, fname=None):
+        if fname is None:
+            return os.path.join(self.output_root, self.split)
+        return os.path.join(self.get_output_path(), fname)
+
+    def copy_labels(self):
+        self.require_output_path()
+
+        labels = list(filter(lambda x: self.extension not in x, glob.glob(self.get_input_path("*"))))
+        for fname in tqdm.tqdm(labels):
+            copy(fname, self.output_path)
+
+    @property
+    def input_fnames(self):
+        return sorted(glob.glob(self.get_input_path("*.{}".format(self.extension))))
+
+    def __len__(self):
+        return len(self.input_fnames)
+
+    def write_features(self):
+
+        paths = self.input_fnames
+
+        fnames_context = map(lambda x: os.path.join(self.output_path, x.replace("." + self.extension, ".h5context")), \
+                             map(os.path.basename, paths))
+
+        for name, target_fname in self._progress(zip(paths, fnames_context), total=len(self)):
+            wav, sr = read_audio(name)
+            z, c = self.model(wav)
+            feat = z if self.use_feat else c
+            writer = H5Writer(target_fname)
+            writer.write(feat)
+
+    def __repr__(self):
+
+        return "EmbeddingDatasetWriter ({n_files} files)\n\tinput:\t{input_root}\n\toutput:\t{output_root}\n\tsplit:\t{split})".format(
+            n_files=len(self), **self.__dict__)
+
+
+if __name__ == "__main__":
+
+    args = EmbeddingWriterConfig().parse_args()
+
+    for split in args.split:
+
+        writer = EmbeddingDatasetWriter(
+            input_root=args.input,
+            output_root=args.output,
+            split=split,
+            model_fname=args.model,
+            gpu=args.gpu,
+            extension=args.ext,
+            use_feat=args.use_feat,
+        )
+
+        print(writer)
+        writer.require_output_path()
+
+        print("Writing Features...")
+        writer.write_features()
+        print("Done.")
+
+        if not args.no_copy_labels:
+            print("Copying label data...")
+            writer.copy_labels()
+            print("Done.")
\ No newline at end of file
diff --git a/scripts/wav2vec_manifest.py b/scripts/wav2vec_manifest.py
new file mode 100644
index 0000000000..949edd58dc
--- /dev/null
+++ b/scripts/wav2vec_manifest.py
@@ -0,0 +1,58 @@
+#!/usr/bin/env python3
+# Copyright (c) 2017-present, Facebook, Inc.
+# All rights reserved.
+#
+# This source code is licensed under the license found in the LICENSE file in
+# the root directory of this source tree. An additional grant of patent rights
+# can be found in the PATENTS file in the same directory.
+"""
+Data pre-processing: build vocabularies and binarize training data.
+"""
+
+import argparse
+import glob
+import os
+import soundfile
+import random
+
+
+def get_parser():
+    parser = argparse.ArgumentParser()
+    parser.add_argument('root', metavar='DIR', help='root directory containing flac files to index')
+    parser.add_argument('--valid-percent', default=0.01, type=float, metavar='D',
+                        help='percentage of data to use as validation set (between 0 and 1)')
+    parser.add_argument('--dest', default='.', type=str, metavar='DIR', help='output directory')
+    parser.add_argument('--ext', default='flac', type=str, metavar='EXT', help='extension to look for')
+    parser.add_argument('--seed', default=42, type=int, metavar='N', help='random seed')
+    parser.add_argument('--path-must-contain', default=None, type=str, metavar='FRAG',
+                        help='if set, path must contain this substring for a file to be included in the manifest')
+    return parser
+
+
+def main(args):
+    assert args.valid_percent >= 0 and args.valid_percent <= 1.
+
+    dir_path = os.path.realpath(args.root)
+    search_path = os.path.join(dir_path, '**/*.' + args.ext)
+    rand = random.Random(args.seed)
+
+    with open(os.path.join(args.dest, 'train.tsv'), 'w') as train_f, open(
+            os.path.join(args.dest, 'valid.tsv'), 'w') as valid_f:
+        print(dir_path, file=train_f)
+        print(dir_path, file=valid_f)
+
+        for fname in glob.iglob(search_path, recursive=True):
+            file_path = os.path.realpath(fname)
+
+            if args.path_must_contain and args.path_must_contain not in file_path:
+                continue
+
+            frames = soundfile.info(fname).frames
+            dest = train_f if rand.random() > args.valid_percent else valid_f
+            print('{}\t{}'.format(os.path.relpath(file_path, dir_path), frames), file=dest)
+
+
+if __name__ == '__main__':
+    parser = get_parser()
+    args = parser.parse_args()
+    main(args)