diff --git a/deep_speech_2/README.md b/deep_speech_2/README.md
new file mode 100644
index 0000000000..48ee9f9a1b
--- /dev/null
+++ b/deep_speech_2/README.md
@@ -0,0 +1,58 @@
+# Deep Speech 2 on PaddlePaddle
+
+## Quick Start
+
+### Installation
+
+Please replace `$PADDLE_INSTALL_DIR` with your paddle installation directory.
+
+```
+pip install -r requirements.txt
+export LD_LIBRARY_PATH=$PADDLE_INSTALL_DIR/Paddle/third_party/install/warpctc/lib:$LD_LIBRARY_PATH
+```
+
+For some machines, we also need to install libsndfile1. Details to be added.
+
+### Preparing Dataset(s)
+
+```
+python librispeech.py
+```
+
+More help for arguments:
+
+```
+python librispeech.py --help
+```
+
+### Traininig
+
+For GPU Training:
+
+```
+CUDA_VISIBLE_DEVICES=0,1,2,3 python train.py --trainer_count 4
+```
+
+For CPU Training:
+
+```
+python train.py --trainer_count 8 --use_gpu False
+```
+
+More help for arguments:
+
+```
+python train.py --help
+```
+
+### Inferencing
+
+```
+python infer.py
+```
+
+More help for arguments:
+
+```
+python infer.py --help
+```
diff --git a/deep_speech_2/audio_data_utils.py b/deep_speech_2/audio_data_utils.py
new file mode 100644
index 0000000000..c717bcf182
--- /dev/null
+++ b/deep_speech_2/audio_data_utils.py
@@ -0,0 +1,383 @@
+"""
+    Providing basic audio data preprocessing pipeline, and offering
+    both instance-level and batch-level data reader interfaces.
+"""
+import paddle.v2 as paddle
+import logging
+import json
+import random
+import soundfile
+import numpy as np
+import os
+
+RANDOM_SEED = 0
+logger = logging.getLogger(__name__)
+
+
+class DataGenerator(object):
+    """
+    DataGenerator provides basic audio data preprocessing pipeline, and offers
+    both instance-level and batch-level data reader interfaces.
+    Normalized FFT are used as audio features here.
+
+    :param vocab_filepath: Vocabulary file path for indexing tokenized
+                           transcriptions.
+    :type vocab_filepath: basestring
+    :param normalizer_manifest_path: Manifest filepath for collecting feature
+                                     normalization statistics, e.g. mean, std.
+    :type normalizer_manifest_path: basestring
+    :param normalizer_num_samples: Number of instances sampled for collecting
+                                   feature normalization statistics.
+                                   Default is 100.
+    :type normalizer_num_samples: int
+    :param max_duration: Audio clips with duration (in seconds) greater than
+                         this will be discarded. Default is 20.0.
+    :type max_duration: float
+    :param min_duration: Audio clips with duration (in seconds) smaller than
+                         this will be discarded. Default is 0.0.
+    :type min_duration: float
+    :param stride_ms: Striding size (in milliseconds) for generating frames.
+                      Default is 10.0. 
+    :type stride_ms: float
+    :param window_ms: Window size (in milliseconds) for frames. Default is 20.0.
+    :type window_ms: float
+    :param max_frequency: Maximun frequency for FFT features. FFT features of
+                          frequency larger than this will be discarded.
+                          If set None, all features will be kept.
+                          Default is None.
+    :type max_frequency: float
+    """
+
+    def __init__(self,
+                 vocab_filepath,
+                 normalizer_manifest_path,
+                 normalizer_num_samples=100,
+                 max_duration=20.0,
+                 min_duration=0.0,
+                 stride_ms=10.0,
+                 window_ms=20.0,
+                 max_frequency=None):
+        self.__max_duration__ = max_duration
+        self.__min_duration__ = min_duration
+        self.__stride_ms__ = stride_ms
+        self.__window_ms__ = window_ms
+        self.__max_frequency__ = max_frequency
+        self.__random__ = random.Random(RANDOM_SEED)
+        # load vocabulary (dictionary)
+        self.__vocab_dict__, self.__vocab_list__ = \
+            self.__load_vocabulary_from_file__(vocab_filepath)
+        # collect normalizer statistics
+        self.__mean__, self.__std__ = self.__collect_normalizer_statistics__(
+            manifest_path=normalizer_manifest_path,
+            num_samples=normalizer_num_samples)
+
+    def __audio_featurize__(self, audio_filename):
+        """
+        Preprocess audio data, including feature extraction, normalization etc..
+        """
+        features = self.__audio_basic_featurize__(audio_filename)
+        return self.__normalize__(features)
+
+    def __text_featurize__(self, text):
+        """
+        Preprocess text data, including tokenizing and token indexing etc..
+        """
+        return self.__convert_text_to_char_index__(
+            text=text, vocabulary=self.__vocab_dict__)
+
+    def __audio_basic_featurize__(self, audio_filename):
+        """
+        Compute basic (without normalization etc.) features for audio data.
+        """
+        return self.__spectrogram_from_file__(
+            filename=audio_filename,
+            stride_ms=self.__stride_ms__,
+            window_ms=self.__window_ms__,
+            max_freq=self.__max_frequency__)
+
+    def __collect_normalizer_statistics__(self, manifest_path, num_samples=100):
+        """
+        Compute feature normalization statistics, i.e. mean and stddev.
+        """
+        # read manifest
+        manifest = self.__read_manifest__(
+            manifest_path=manifest_path,
+            max_duration=self.__max_duration__,
+            min_duration=self.__min_duration__)
+        # sample for statistics
+        sampled_manifest = self.__random__.sample(manifest, num_samples)
+        # extract spectrogram feature
+        features = []
+        for instance in sampled_manifest:
+            spectrogram = self.__audio_basic_featurize__(
+                instance["audio_filepath"])
+            features.append(spectrogram)
+        features = np.hstack(features)
+        mean = np.mean(features, axis=1).reshape([-1, 1])
+        std = np.std(features, axis=1).reshape([-1, 1])
+        return mean, std
+
+    def __normalize__(self, features, eps=1e-14):
+        """
+        Normalize features to be of zero mean and unit stddev.
+        """
+        return (features - self.__mean__) / (self.__std__ + eps)
+
+    def __spectrogram_from_file__(self,
+                                  filename,
+                                  stride_ms=10.0,
+                                  window_ms=20.0,
+                                  max_freq=None,
+                                  eps=1e-14):
+        """
+        Laod audio data and calculate the log of spectrogram by FFT.
+        Refer to utils.py in https://github.com/baidu-research/ba-dls-deepspeech
+        """
+        audio, sample_rate = soundfile.read(filename)
+        if audio.ndim >= 2:
+            audio = np.mean(audio, 1)
+        if max_freq is None:
+            max_freq = sample_rate / 2
+        if max_freq > sample_rate / 2:
+            raise ValueError("max_freq must be greater than half of "
+                             "sample rate.")
+        if stride_ms > window_ms:
+            raise ValueError("Stride size must not be greater than "
+                             "window size.")
+        stride_size = int(0.001 * sample_rate * stride_ms)
+        window_size = int(0.001 * sample_rate * window_ms)
+        spectrogram, freqs = self.__extract_spectrogram__(
+            audio,
+            window_size=window_size,
+            stride_size=stride_size,
+            sample_rate=sample_rate)
+        ind = np.where(freqs <= max_freq)[0][-1] + 1
+        return np.log(spectrogram[:ind, :] + eps)
+
+    def __extract_spectrogram__(self, samples, window_size, stride_size,
+                                sample_rate):
+        """
+        Compute the spectrogram by FFT for a discrete real signal.
+        Refer to utils.py in https://github.com/baidu-research/ba-dls-deepspeech
+        """
+        # extract strided windows
+        truncate_size = (len(samples) - window_size) % stride_size
+        samples = samples[:len(samples) - truncate_size]
+        nshape = (window_size, (len(samples) - window_size) // stride_size + 1)
+        nstrides = (samples.strides[0], samples.strides[0] * stride_size)
+        windows = np.lib.stride_tricks.as_strided(
+            samples, shape=nshape, strides=nstrides)
+        assert np.all(
+            windows[:, 1] == samples[stride_size:(stride_size + window_size)])
+        # window weighting, squared Fast Fourier Transform (fft), scaling
+        weighting = np.hanning(window_size)[:, None]
+        fft = np.fft.rfft(windows * weighting, axis=0)
+        fft = np.absolute(fft)**2
+        scale = np.sum(weighting**2) * sample_rate
+        fft[1:-1, :] *= (2.0 / scale)
+        fft[(0, -1), :] /= scale
+        # prepare fft frequency list
+        freqs = float(sample_rate) / window_size * np.arange(fft.shape[0])
+        return fft, freqs
+
+    def __load_vocabulary_from_file__(self, vocabulary_path):
+        """
+        Load vocabulary from file.
+        """
+        if not os.path.exists(vocabulary_path):
+            raise ValueError("Vocabulary file %s not found.", vocabulary_path)
+        vocab_lines = []
+        with open(vocabulary_path, 'r') as file:
+            vocab_lines.extend(file.readlines())
+        vocab_list = [line[:-1] for line in vocab_lines]
+        vocab_dict = dict(
+            [(token, id) for (id, token) in enumerate(vocab_list)])
+        return vocab_dict, vocab_list
+
+    def __convert_text_to_char_index__(self, text, vocabulary):
+        """
+        Convert text string to a list of character index integers.
+        """
+        return [vocabulary[w] for w in text]
+
+    def __read_manifest__(self, manifest_path, max_duration, min_duration):
+        """
+        Load and parse manifest file.
+        """
+        manifest = []
+        for json_line in open(manifest_path):
+            try:
+                json_data = json.loads(json_line)
+            except Exception as e:
+                raise ValueError("Error reading manifest: %s" % str(e))
+            if (json_data["duration"] <= max_duration and
+                    json_data["duration"] >= min_duration):
+                manifest.append(json_data)
+        return manifest
+
+    def __padding_batch__(self, batch, padding_to=-1, flatten=False):
+        """
+        Padding audio part of features (only in the time axis -- column axis)
+        with zeros, to make each instance in the batch share the same
+        audio feature shape.
+
+        If `padding_to` is set -1, the maximun column numbers in the batch will
+        be used as the target size. Otherwise, `padding_to` will be the target
+        size. Default is -1.
+
+        If `flatten` is set True, audio data will be flatten to be a 1-dim
+        ndarray. Default is False.
+        """
+        new_batch = []
+        # get target shape
+        max_length = max([audio.shape[1] for audio, text in batch])
+        if padding_to != -1:
+            if padding_to < max_length:
+                raise ValueError("If padding_to is not -1, it should be greater"
+                                 " or equal to the original instance length.")
+            max_length = padding_to
+        # padding
+        for audio, text in batch:
+            padded_audio = np.zeros([audio.shape[0], max_length])
+            padded_audio[:, :audio.shape[1]] = audio
+            if flatten:
+                padded_audio = padded_audio.flatten()
+            new_batch.append((padded_audio, text))
+        return new_batch
+
+    def instance_reader_creator(self,
+                                manifest_path,
+                                sort_by_duration=True,
+                                shuffle=False):
+        """
+        Instance reader creator for audio data. Creat a callable function to
+        produce instances of data.
+
+        Instance: a tuple of a numpy ndarray of audio spectrogram and a list of
+        tokenized and indexed transcription text.
+
+        :param manifest_path: Filepath of manifest for audio clip files.
+        :type manifest_path: basestring
+        :param sort_by_duration: Sort the audio clips by duration if set True
+                                 (for SortaGrad).
+        :type sort_by_duration: bool
+        :param shuffle: Shuffle the audio clips if set True.
+        :type shuffle: bool
+        :return: Data reader function.
+        :rtype: callable
+        """
+        if sort_by_duration and shuffle:
+            sort_by_duration = False
+            logger.warn("When shuffle set to true, "
+                        "sort_by_duration is forced to set False.")
+
+        def reader():
+            # read manifest
+            manifest = self.__read_manifest__(
+                manifest_path=manifest_path,
+                max_duration=self.__max_duration__,
+                min_duration=self.__min_duration__)
+            # sort (by duration) or shuffle manifest
+            if sort_by_duration:
+                manifest.sort(key=lambda x: x["duration"])
+            if shuffle:
+                self.__random__.shuffle(manifest)
+            # extract spectrogram feature
+            for instance in manifest:
+                spectrogram = self.__audio_featurize__(
+                    instance["audio_filepath"])
+                transcript = self.__text_featurize__(instance["text"])
+                yield (spectrogram, transcript)
+
+        return reader
+
+    def batch_reader_creator(self,
+                             manifest_path,
+                             batch_size,
+                             padding_to=-1,
+                             flatten=False,
+                             sort_by_duration=True,
+                             shuffle=False):
+        """
+        Batch data reader creator for audio data. Creat a callable function to
+        produce batches of data.
+        
+        Audio features will be padded with zeros to make each instance in the
+        batch to share the same audio feature shape.
+
+        :param manifest_path: Filepath of manifest for audio clip files.
+        :type manifest_path: basestring
+        :param batch_size: Instance number in a batch.
+        :type batch_size: int
+        :param padding_to:  If set -1, the maximun column numbers in the batch
+                            will be used as the target size for padding.
+                            Otherwise, `padding_to` will be the target size.
+                            Default is -1.
+        :type padding_to: int
+        :param flatten: If set True, audio data will be flatten to be a 1-dim
+                        ndarray. Otherwise, 2-dim ndarray. Default is False.
+        :type flatten: bool
+        :param sort_by_duration: Sort the audio clips by duration if set True
+                                 (for SortaGrad).
+        :type sort_by_duration: bool
+        :param shuffle: Shuffle the audio clips if set True.
+        :type shuffle: bool
+        :return: Batch reader function, producing batches of data when called.
+        :rtype: callable
+        """
+
+        def batch_reader():
+            instance_reader = self.instance_reader_creator(
+                manifest_path=manifest_path,
+                sort_by_duration=sort_by_duration,
+                shuffle=shuffle)
+            batch = []
+            for instance in instance_reader():
+                batch.append(instance)
+                if len(batch) == batch_size:
+                    yield self.__padding_batch__(batch, padding_to, flatten)
+                    batch = []
+            if len(batch) > 0:
+                yield self.__padding_batch__(batch, padding_to, flatten)
+
+        return batch_reader
+
+    def vocabulary_size(self):
+        """
+        Get vocabulary size.
+
+        :return: Vocabulary size.
+        :rtype: int
+        """
+        return len(self.__vocab_list__)
+
+    def vocabulary_dict(self):
+        """
+        Get vocabulary in dict.
+
+        :return: Vocabulary in dict.
+        :rtype: dict
+        """
+        return self.__vocab_dict__
+
+    def vocabulary_list(self):
+        """
+        Get vocabulary in list.
+
+        :return: Vocabulary in list
+        :rtype: list
+        """
+        return self.__vocab_list__
+
+    def data_name_feeding(self):
+        """
+        Get feeddings (data field name and corresponding field id).
+
+        :return: Feeding dict.
+        :rtype: dict
+        """
+        feeding = {
+            "audio_spectrogram": 0,
+            "transcript_text": 1,
+        }
+        return feeding
diff --git a/deep_speech_2/eng_vocab.txt b/deep_speech_2/eng_vocab.txt
new file mode 100644
index 0000000000..8268f3f330
--- /dev/null
+++ b/deep_speech_2/eng_vocab.txt
@@ -0,0 +1,28 @@
+'
+ 
+a
+b
+c
+d
+e
+f
+g
+h
+i
+j
+k
+l
+m
+n
+o
+p
+q
+r
+s
+t
+u
+v
+w
+x
+y
+z
diff --git a/deep_speech_2/infer.py b/deep_speech_2/infer.py
new file mode 100644
index 0000000000..1c52c98fd8
--- /dev/null
+++ b/deep_speech_2/infer.py
@@ -0,0 +1,144 @@
+"""
+   Inference for a simplifed version of Baidu DeepSpeech2 model.
+"""
+
+import paddle.v2 as paddle
+from itertools import groupby
+import distutils.util
+import argparse
+import gzip
+from audio_data_utils import DataGenerator
+from model import deep_speech2
+
+parser = argparse.ArgumentParser(
+    description='Simplified version of DeepSpeech2 inference.')
+parser.add_argument(
+    "--num_samples",
+    default=10,
+    type=int,
+    help="Number of samples for inference. (default: %(default)s)")
+parser.add_argument(
+    "--num_conv_layers",
+    default=2,
+    type=int,
+    help="Convolution layer number. (default: %(default)s)")
+parser.add_argument(
+    "--num_rnn_layers",
+    default=3,
+    type=int,
+    help="RNN layer number. (default: %(default)s)")
+parser.add_argument(
+    "--rnn_layer_size",
+    default=512,
+    type=int,
+    help="RNN layer cell number. (default: %(default)s)")
+parser.add_argument(
+    "--use_gpu",
+    default=True,
+    type=distutils.util.strtobool,
+    help="Use gpu or not. (default: %(default)s)")
+parser.add_argument(
+    "--normalizer_manifest_path",
+    default='./manifest.libri.train-clean-100',
+    type=str,
+    help="Manifest path for normalizer. (default: %(default)s)")
+parser.add_argument(
+    "--decode_manifest_path",
+    default='./manifest.libri.test-clean',
+    type=str,
+    help="Manifest path for decoding. (default: %(default)s)")
+parser.add_argument(
+    "--model_filepath",
+    default='./params.tar.gz',
+    type=str,
+    help="Model filepath. (default: %(default)s)")
+args = parser.parse_args()
+
+
+def remove_duplicate_and_blank(id_list, blank_id):
+    """
+    Postprocessing for max-ctc-decoder.
+    - remove consecutive duplicate tokens.
+    - remove blanks.
+    """
+    # remove consecutive duplicate tokens
+    id_list = [x[0] for x in groupby(id_list)]
+    # remove blanks
+    return [id for id in id_list if id != blank_id]
+
+
+def best_path_decode():
+    """
+    Max-ctc-decoding for DeepSpeech2.
+    """
+    # initialize data generator
+    data_generator = DataGenerator(
+        vocab_filepath='eng_vocab.txt',
+        normalizer_manifest_path=args.normalizer_manifest_path,
+        normalizer_num_samples=200,
+        max_duration=20.0,
+        min_duration=0.0,
+        stride_ms=10,
+        window_ms=20)
+    # create network config
+    dict_size = data_generator.vocabulary_size()
+    vocab_list = data_generator.vocabulary_list()
+    audio_data = paddle.layer.data(
+        name="audio_spectrogram",
+        height=161,
+        width=2000,
+        type=paddle.data_type.dense_vector(322000))
+    text_data = paddle.layer.data(
+        name="transcript_text",
+        type=paddle.data_type.integer_value_sequence(dict_size))
+    _, max_id = deep_speech2(
+        audio_data=audio_data,
+        text_data=text_data,
+        dict_size=dict_size,
+        num_conv_layers=args.num_conv_layers,
+        num_rnn_layers=args.num_rnn_layers,
+        rnn_size=args.rnn_layer_size)
+
+    # load parameters
+    parameters = paddle.parameters.Parameters.from_tar(
+        gzip.open(args.model_filepath))
+
+    # prepare infer data
+    feeding = data_generator.data_name_feeding()
+    test_batch_reader = data_generator.batch_reader_creator(
+        manifest_path=args.decode_manifest_path,
+        batch_size=args.num_samples,
+        padding_to=2000,
+        flatten=True,
+        sort_by_duration=False,
+        shuffle=False)
+    infer_data = test_batch_reader().next()
+
+    # run max-ctc-decoding
+    max_id_results = paddle.infer(
+        output_layer=max_id,
+        parameters=parameters,
+        input=infer_data,
+        field=['id'])
+
+    # postprocess
+    instance_length = len(max_id_results) / args.num_samples
+    instance_list = [
+        max_id_results[i * instance_length:(i + 1) * instance_length]
+        for i in xrange(0, args.num_samples)
+    ]
+    for i, instance in enumerate(instance_list):
+        id_list = remove_duplicate_and_blank(instance, dict_size)
+        output_transcript = ''.join([vocab_list[id] for id in id_list])
+        target_transcript = ''.join([vocab_list[id] for id in infer_data[i][1]])
+        print("Target Transcript: %s \nOutput Transcript: %s \n" %
+              (target_transcript, output_transcript))
+
+
+def main():
+    paddle.init(use_gpu=args.use_gpu, trainer_count=1)
+    best_path_decode()
+
+
+if __name__ == '__main__':
+    main()
diff --git a/deep_speech_2/librispeech.py b/deep_speech_2/librispeech.py
new file mode 100644
index 0000000000..676bbec5ce
--- /dev/null
+++ b/deep_speech_2/librispeech.py
@@ -0,0 +1,138 @@
+"""
+   Download, unpack and create manifest for Librespeech dataset.
+
+   Manifest is a json file with each line containing one audio clip filepath,
+   its transcription text string, and its duration. It servers as a unified
+   interfance to organize different data sets.
+"""
+
+import paddle.v2 as paddle
+from paddle.v2.dataset.common import md5file
+import os
+import wget
+import tarfile
+import argparse
+import soundfile
+import json
+
+DATA_HOME = os.path.expanduser('~/.cache2/paddle/dataset/speech')
+
+URL_ROOT = "http://www.openslr.org/resources/12"
+URL_TEST_CLEAN = URL_ROOT + "/test-clean.tar.gz"
+URL_TEST_OTHER = URL_ROOT + "/test-other.tar.gz"
+URL_DEV_CLEAN = URL_ROOT + "/dev-clean.tar.gz"
+URL_DEV_OTHER = URL_ROOT + "/dev-other.tar.gz"
+URL_TRAIN_CLEAN_100 = URL_ROOT + "/train-clean-100.tar.gz"
+URL_TRAIN_CLEAN_360 = URL_ROOT + "/train-clean-360.tar.gz"
+URL_TRAIN_OTHER_500 = URL_ROOT + "/train-other-500.tar.gz"
+
+MD5_TEST_CLEAN = "32fa31d27d2e1cad72775fee3f4849a9"
+MD5_DEV_CLEAN = "42e2234ba48799c1f50f24a7926300a1"
+MD5_TRAIN_CLEAN_100 = "2a93770f6d5c6c964bc36631d331a522"
+MD5_TRAIN_CLEAN_360 = "c0e676e450a7ff2f54aeade5171606fa"
+MD5_TRAIN_CLEAN_500 = "d1a0fd59409feb2c614ce4d30c387708"
+
+parser = argparse.ArgumentParser(
+    description='Downloads and prepare LibriSpeech dataset.')
+parser.add_argument(
+    "--target_dir",
+    default=DATA_HOME + "/Libri",
+    type=str,
+    help="Directory to save the dataset. (default: %(default)s)")
+parser.add_argument(
+    "--manifest_prefix",
+    default="manifest.libri",
+    type=str,
+    help="Filepath prefix for output manifests. (default: %(default)s)")
+args = parser.parse_args()
+
+
+def download(url, md5sum, target_dir):
+    """
+    Download file from url to target_dir, and check md5sum.
+    """
+    if not os.path.exists(target_dir): os.makedirs(target_dir)
+    filepath = os.path.join(target_dir, url.split("/")[-1])
+    if not (os.path.exists(filepath) and md5file(filepath) == md5sum):
+        print("Downloading %s ..." % url)
+        wget.download(url, target_dir)
+        print("\nMD5 Chesksum %s ..." % filepath)
+        assert md5file(filepath) == md5sum, "MD5 checksum failed."
+    return filepath
+
+
+def unpack(filepath, target_dir):
+    """
+    Unpack the file to the target_dir.
+    """
+    print("Unpacking %s ..." % filepath)
+    tar = tarfile.open(filepath)
+    tar.extractall(target_dir)
+    tar.close()
+    return target_dir
+
+
+def create_manifest(data_dir, manifest_path):
+    """
+    Create a manifest file summarizing the dataset (list of filepath and meta
+    data).
+
+    Each line of the manifest contains one audio clip filepath, its
+    transcription text string, and its duration. Manifest file servers as a
+    unified interfance to organize data sets.
+    """
+    print("Creating manifest %s ..." % manifest_path)
+    json_lines = []
+    for subfolder, _, filelist in os.walk(data_dir):
+        text_filelist = [
+            filename for filename in filelist if filename.endswith('trans.txt')
+        ]
+        if len(text_filelist) > 0:
+            text_filepath = os.path.join(data_dir, subfolder, text_filelist[0])
+            for line in open(text_filepath):
+                segments = line.strip().split()
+                text = ' '.join(segments[1:]).lower()
+                audio_filepath = os.path.join(data_dir, subfolder,
+                                              segments[0] + '.flac')
+                audio_data, samplerate = soundfile.read(audio_filepath)
+                duration = float(len(audio_data)) / samplerate
+                json_lines.append(
+                    json.dumps({
+                        'audio_filepath': audio_filepath,
+                        'duration': duration,
+                        'text': text
+                    }))
+    with open(manifest_path, 'w') as out_file:
+        for line in json_lines:
+            out_file.write(line + '\n')
+
+
+def prepare_dataset(url, md5sum, target_dir, manifest_path):
+    """
+    Download, unpack and create summmary manifest file.
+    """
+    filepath = download(url, md5sum, target_dir)
+    unpacked_dir = unpack(filepath, target_dir)
+    create_manifest(unpacked_dir, manifest_path)
+
+
+def main():
+    prepare_dataset(
+        url=URL_TEST_CLEAN,
+        md5sum=MD5_TEST_CLEAN,
+        target_dir=os.path.join(args.target_dir, "test-clean"),
+        manifest_path=args.manifest_prefix + ".test-clean")
+    prepare_dataset(
+        url=URL_DEV_CLEAN,
+        md5sum=MD5_DEV_CLEAN,
+        target_dir=os.path.join(args.target_dir, "dev-clean"),
+        manifest_path=args.manifest_prefix + ".dev-clean")
+    prepare_dataset(
+        url=URL_TRAIN_CLEAN_100,
+        md5sum=MD5_TRAIN_CLEAN_100,
+        target_dir=os.path.join(args.target_dir, "train-clean-100"),
+        manifest_path=args.manifest_prefix + ".train-clean-100")
+
+
+if __name__ == '__main__':
+    main()
diff --git a/deep_speech_2/model.py b/deep_speech_2/model.py
new file mode 100644
index 0000000000..6b396900e6
--- /dev/null
+++ b/deep_speech_2/model.py
@@ -0,0 +1,136 @@
+"""
+   A simplifed version of Baidu DeepSpeech2 model.
+"""
+
+import paddle.v2 as paddle
+
+#TODO: add bidirectional rnn.
+
+
+def conv_bn_layer(input, filter_size, num_channels_in, num_channels_out, stride,
+                  padding, act):
+    """
+    Convolution layer with batch normalization.
+    """
+    conv_layer = paddle.layer.img_conv(
+        input=input,
+        filter_size=filter_size,
+        num_channels=num_channels_in,
+        num_filters=num_channels_out,
+        stride=stride,
+        padding=padding,
+        act=paddle.activation.Linear(),
+        bias_attr=False)
+    return paddle.layer.batch_norm(input=conv_layer, act=act)
+
+
+def bidirectional_simple_rnn_bn_layer(name, input, size, act):
+    """
+    Bidirectonal simple rnn layer with sequence-wise batch normalization.
+    The batch normalization is only performed on input-state weights.
+    """
+    # input-hidden weights shared across bi-direcitonal rnn.
+    input_proj = paddle.layer.fc(
+        input=input, size=size, act=paddle.activation.Linear(), bias_attr=False)
+    # batch norm is only performed on input-state projection 
+    input_proj_bn = paddle.layer.batch_norm(
+        input=input_proj, act=paddle.activation.Linear())
+    # forward and backward in time
+    forward_simple_rnn = paddle.layer.recurrent(
+        input=input_proj_bn, act=act, reverse=False)
+    backward_simple_rnn = paddle.layer.recurrent(
+        input=input_proj_bn, act=act, reverse=True)
+    return paddle.layer.concat(input=[forward_simple_rnn, backward_simple_rnn])
+
+
+def conv_group(input, num_stacks):
+    """
+    Convolution group with several stacking convolution layers.
+    """
+    conv = conv_bn_layer(
+        input=input,
+        filter_size=(11, 41),
+        num_channels_in=1,
+        num_channels_out=32,
+        stride=(3, 2),
+        padding=(5, 20),
+        act=paddle.activation.BRelu())
+    for i in xrange(num_stacks - 1):
+        conv = conv_bn_layer(
+            input=conv,
+            filter_size=(11, 21),
+            num_channels_in=32,
+            num_channels_out=32,
+            stride=(1, 2),
+            padding=(5, 10),
+            act=paddle.activation.BRelu())
+    output_num_channels = 32
+    output_height = 160 // pow(2, num_stacks) + 1
+    return conv, output_num_channels, output_height
+
+
+def rnn_group(input, size, num_stacks):
+    """
+    RNN group with several stacking RNN layers.
+    """
+    output = input
+    for i in xrange(num_stacks):
+        output = bidirectional_simple_rnn_bn_layer(
+            name=str(i), input=output, size=size, act=paddle.activation.BRelu())
+    return output
+
+
+def deep_speech2(audio_data,
+                 text_data,
+                 dict_size,
+                 num_conv_layers=2,
+                 num_rnn_layers=3,
+                 rnn_size=256):
+    """
+    The whole DeepSpeech2 model structure (a simplified version).
+
+    :param audio_data: Audio spectrogram data layer.
+    :type audio_data: LayerOutput
+    :param text_data: Transcription text data layer.
+    :type text_data: LayerOutput
+    :param dict_size: Dictionary size for tokenized transcription.
+    :type dict_size: int
+    :param num_conv_layers: Number of stacking convolution layers.
+    :type num_conv_layers: int
+    :param num_rnn_layers: Number of stacking RNN layers.
+    :type num_rnn_layers: int
+    :param rnn_size: RNN layer size (number of RNN cells).
+    :type rnn_size: int
+    :return: Tuple of the cost layer and the max_id decoder layer.
+    :rtype: tuple of LayerOutput
+    """
+    # convolution group
+    conv_group_output, conv_group_num_channels, conv_group_height = conv_group(
+        input=audio_data, num_stacks=num_conv_layers)
+    # convert data form convolution feature map to sequence of vectors
+    conv2seq = paddle.layer.block_expand(
+        input=conv_group_output,
+        num_channels=conv_group_num_channels,
+        stride_x=1,
+        stride_y=1,
+        block_x=1,
+        block_y=conv_group_height)
+    # rnn group
+    rnn_group_output = rnn_group(
+        input=conv2seq, size=rnn_size, num_stacks=num_rnn_layers)
+    # output token distribution
+    fc = paddle.layer.fc(
+        input=rnn_group_output,
+        size=dict_size + 1,
+        act=paddle.activation.Linear(),
+        bias_attr=True)
+    # ctc cost
+    cost = paddle.layer.warp_ctc(
+        input=fc,
+        label=text_data,
+        size=dict_size + 1,
+        blank=dict_size,
+        norm_by_times=True)
+    # max decoder
+    max_id = paddle.layer.max_id(input=fc)
+    return cost, max_id
diff --git a/deep_speech_2/requirements.txt b/deep_speech_2/requirements.txt
new file mode 100644
index 0000000000..58a93debe4
--- /dev/null
+++ b/deep_speech_2/requirements.txt
@@ -0,0 +1,2 @@
+SoundFile==0.9.0.post1
+wget==3.2
diff --git a/deep_speech_2/train.py b/deep_speech_2/train.py
new file mode 100644
index 0000000000..ad6e5ffd1b
--- /dev/null
+++ b/deep_speech_2/train.py
@@ -0,0 +1,190 @@
+"""
+   Trainer for a simplifed version of Baidu DeepSpeech2 model.
+"""
+
+import paddle.v2 as paddle
+import distutils.util
+import argparse
+import gzip
+import time
+import sys
+from model import deep_speech2
+from audio_data_utils import DataGenerator
+import numpy as np
+
+#TODO: add WER metric
+
+parser = argparse.ArgumentParser(
+    description='Simplified version of DeepSpeech2 trainer.')
+parser.add_argument(
+    "--batch_size", default=32, type=int, help="Minibatch size.")
+parser.add_argument(
+    "--num_passes",
+    default=20,
+    type=int,
+    help="Training pass number. (default: %(default)s)")
+parser.add_argument(
+    "--num_conv_layers",
+    default=2,
+    type=int,
+    help="Convolution layer number. (default: %(default)s)")
+parser.add_argument(
+    "--num_rnn_layers",
+    default=3,
+    type=int,
+    help="RNN layer number. (default: %(default)s)")
+parser.add_argument(
+    "--rnn_layer_size",
+    default=512,
+    type=int,
+    help="RNN layer cell number. (default: %(default)s)")
+parser.add_argument(
+    "--adam_learning_rate",
+    default=5e-4,
+    type=float,
+    help="Learning rate for ADAM Optimizer. (default: %(default)s)")
+parser.add_argument(
+    "--use_gpu",
+    default=True,
+    type=distutils.util.strtobool,
+    help="Use gpu or not. (default: %(default)s)")
+parser.add_argument(
+    "--use_sortagrad",
+    default=False,
+    type=distutils.util.strtobool,
+    help="Use sortagrad or not. (default: %(default)s)")
+parser.add_argument(
+    "--trainer_count",
+    default=4,
+    type=int,
+    help="Trainer number. (default: %(default)s)")
+parser.add_argument(
+    "--normalizer_manifest_path",
+    default='./manifest.libri.train-clean-100',
+    type=str,
+    help="Manifest path for normalizer. (default: %(default)s)")
+parser.add_argument(
+    "--train_manifest_path",
+    default='./manifest.libri.train-clean-100',
+    type=str,
+    help="Manifest path for training. (default: %(default)s)")
+parser.add_argument(
+    "--dev_manifest_path",
+    default='./manifest.libri.dev-clean',
+    type=str,
+    help="Manifest path for validation. (default: %(default)s)")
+args = parser.parse_args()
+
+
+def train():
+    """
+    DeepSpeech2 training.
+    """
+    # initialize data generator
+    data_generator = DataGenerator(
+        vocab_filepath='eng_vocab.txt',
+        normalizer_manifest_path=args.normalizer_manifest_path,
+        normalizer_num_samples=200,
+        max_duration=20.0,
+        min_duration=0.0,
+        stride_ms=10,
+        window_ms=20)
+
+    # create network config
+    dict_size = data_generator.vocabulary_size()
+    audio_data = paddle.layer.data(
+        name="audio_spectrogram",
+        height=161,
+        width=2000,
+        type=paddle.data_type.dense_vector(322000))
+    text_data = paddle.layer.data(
+        name="transcript_text",
+        type=paddle.data_type.integer_value_sequence(dict_size))
+    cost, _ = deep_speech2(
+        audio_data=audio_data,
+        text_data=text_data,
+        dict_size=dict_size,
+        num_conv_layers=args.num_conv_layers,
+        num_rnn_layers=args.num_rnn_layers,
+        rnn_size=args.rnn_layer_size)
+
+    # create parameters and optimizer
+    parameters = paddle.parameters.create(cost)
+    optimizer = paddle.optimizer.Adam(
+        learning_rate=args.adam_learning_rate, gradient_clipping_threshold=400)
+    trainer = paddle.trainer.SGD(
+        cost=cost, parameters=parameters, update_equation=optimizer)
+
+    # prepare data reader
+    train_batch_reader_sortagrad = data_generator.batch_reader_creator(
+        manifest_path=args.train_manifest_path,
+        batch_size=args.batch_size // args.trainer_count,
+        padding_to=2000,
+        flatten=True,
+        sort_by_duration=True,
+        shuffle=False)
+    train_batch_reader_nosortagrad = data_generator.batch_reader_creator(
+        manifest_path=args.train_manifest_path,
+        batch_size=args.batch_size // args.trainer_count,
+        padding_to=2000,
+        flatten=True,
+        sort_by_duration=False,
+        shuffle=True)
+    test_batch_reader = data_generator.batch_reader_creator(
+        manifest_path=args.dev_manifest_path,
+        batch_size=args.batch_size // args.trainer_count,
+        padding_to=2000,
+        flatten=True,
+        sort_by_duration=False,
+        shuffle=False)
+    feeding = data_generator.data_name_feeding()
+
+    # create event handler
+    def event_handler(event):
+        global start_time
+        global cost_sum
+        global cost_counter
+        if isinstance(event, paddle.event.EndIteration):
+            cost_sum += event.cost
+            cost_counter += 1
+            if event.batch_id % 50 == 0:
+                print "\nPass: %d, Batch: %d, TrainCost: %f" % (
+                    event.pass_id, event.batch_id, cost_sum / cost_counter)
+                cost_sum, cost_counter = 0.0, 0
+                with gzip.open("params.tar.gz", 'w') as f:
+                    parameters.to_tar(f)
+            else:
+                sys.stdout.write('.')
+                sys.stdout.flush()
+        if isinstance(event, paddle.event.BeginPass):
+            start_time = time.time()
+            cost_sum, cost_counter = 0.0, 0
+        if isinstance(event, paddle.event.EndPass):
+            result = trainer.test(reader=test_batch_reader, feeding=feeding)
+            print "\n------- Time: %d sec,  Pass: %d, ValidationCost: %s" % (
+                time.time() - start_time, event.pass_id, result.cost)
+
+    # run train
+    # first pass with sortagrad
+    if args.use_sortagrad:
+        trainer.train(
+            reader=train_batch_reader_sortagrad,
+            event_handler=event_handler,
+            num_passes=1,
+            feeding=feeding)
+        args.num_passes -= 1
+    # other passes without sortagrad
+    trainer.train(
+        reader=train_batch_reader_nosortagrad,
+        event_handler=event_handler,
+        num_passes=args.num_passes,
+        feeding=feeding)
+
+
+def main():
+    paddle.init(use_gpu=args.use_gpu, trainer_count=args.trainer_count)
+    train()
+
+
+if __name__ == '__main__':
+    main()