Tag Archives: TensorFlow

How Machine Learning with TensorFlow Enabled Mobile Proof-Of-Purchase at Coca-Cola

In this guest editorial, Patrick Brandt of The Coca-Cola Company tells us how they're using AI and TensorFlow to achieve frictionless proof-of-purchase.

Coca-Cola's core loyalty program launched in 2006 as MyCokeRewards.com. The "MCR.com" platform included the creation of unique product codes for every Coca-Cola, Sprite, Fanta, and Powerade product sold in 20oz bottles and cardboard "fridge-packs" purchasable at grocery stores and other retail outlets. Users could enter these product codes at MyCokeRewards.com to participate in promotional campaigns.

Fast-forward to 2016: Coke's loyalty programs are still hugely popular with millions of product codes having been entered for promotions and sweepstakes. However, mobile browsing went from non-existent in 2006 to over 50% share by the end of 2016. The launch of Coke.com as a mobile-first web experience (replacing MCR.com) was a response to these changes in browsing behavior. Thumb-entering 14-character codes into a mobile device could be a difficult enough user experience to impact the success of our programs. We want to provide our mobile audience the best possible experience, and recent advances in artificial intelligence opened new opportunities.

The quest for frictionless proof-of-purchase

For years Coke attempted to use off-the-shelf optical character recognition (OCR) libraries and services to read product codes with little success. Our printing process typically uses low-resolution dot-matrix fonts with the cap or fridge-pack media running under the printhead at very high speeds. All of this translates into a low-fidelity string of characters that defeats off-the-shelf OCR offerings (and can sometimes be hard to read with the human eye as well). OCR is critical to simplifying the code-entry process for mobile users: they should be able to take a picture of a code and automatically have the purchase registered for a promotional entry. We needed a purpose-built OCR system to recognize our product codes.

Bottlecap and fridge-pack examples

Our research led us to a promising solution: Convolutional Neural Networks. CNNs are one of a family of "deep learning" neural networks that are at the heart of modern artificial intelligence products. Google has used CNNs to extract street address numbers from StreetView images. CNNs also perform remarkably well at recognizing handwritten digits. These number-recognition use-cases were a perfect proxy for the type of problem we were trying to solve: extracting strings from images that contain small character sets with lots of variance in the appearance of the characters.

CNNs with TensorFlow

In the past, developing deep neural networks like CNNs has been a challenge because of the complexity of available training and inference libraries. TensorFlow, a machine learning framework that was open sourced by Google in November 2015, is designed to simplify the development of deep neural networks.

TensorFlow provides high-level interfaces to different kinds of neuron layers and popular loss functions, which makes it easier to implement different CNN model architectures. The ability to rapidly iterate over different model architectures dramatically reduced the time required to build Coke's custom OCR solution because different models could be developed, trained, and tested in a matter of days. TensorFlow models are also portable: the framework supports model execution natively on mobile devices ("AI on the edge") or in servers hosted remotely in the cloud. This enables a "create once, run anywhere" approach for model execution across many different platforms, including web-based and mobile.

Machine learning: practice makes perfect

Any neural network is only as good as the data used to train it. We knew that we needed a large set of labeled product-code images to train a CNN that would achieve our performance goals. Our training set would be built in three phases:

  1. Pre-launch simulated images
  2. Pre-launch real-world images
  3. Images labeled by our users in production

The pre-launch training phase began by programmatically generating millions of simulated product-code images. These simulated images included variations in tilt, lighting, shadows, and blurriness. The prediction accuracy (i.e. how often all 14 characters were correctly predicted within the top-10 predictions) was at 50% against real-world images when the model was trained using only simulated images. This provided a baseline for transfer-learning: a model initially trained with simulated images was the foundation for a more accurate model that would be trained against real-world images.

The challenge now turned to enriching the simulated images with enough real-world images to hit our performance goals. We created a purpose-built training app for iOS and Android devices that "trainers" could use to take pictures of codes and label them; these labeled images were then transferred to cloud storage for training. We did a production run of several thousand product codes on bottle caps and fridge-packs and distributed these to multiple suppliers who used the app to create the initial real-world training set.

Even with an augmented and enriched training set, there is no substitute for images created by end-users in a variety of environmental conditions. We knew that scans would sometimes result in an inaccurate code prediction, so we needed to provide a user-experience that would allow users to quickly correct these predictions. Two components are essential to delivering this experience: a product-code validation service that has been in use since the launch of our original loyalty platform in 2006 (to verify that a predicted code is an actual code) and a prediction algorithm that performs a regression to determine a per-character confidence at each one of the 14 character positions. If a predicted code is invalid, the top prediction as well as the confidence levels for each character are returned to the user interface. Low-confidence characters are visually highlighted to guide the user to update characters that need attention.

Error correction user interface lets users correct invalid predictions and generate useful training data

This user interface innovation enables an active learning process: a feedback loop allows the model to gradually improve by returning corrected predictions to the training pipeline. In this way, our users organically improve the accuracy of the character recognition model over time.

Product-code recognition pipeline

Optimizing for maximum performance

To meet user expectations around performance, we established a few ambitious requirements for the product-code OCR pipeline:

  • It had to be fast: we needed a one-second average processing time once the image of the product-code was sent into the OCR pipeline
  • It had to be accurate: our goal was to achieve 95% string recognition accuracy at launch with the guarantee that the model could be improved over time via active learning
  • It had to be small: the OCR pipeline needs to be small enough to be distributed directly to mobile apps and accommodate over-the-air updates as the model improves over time
  • It had to handle diverse product code media: dozens of different combinations of font types, bottlecaps, and cardboard fridge-pack media

We initially explored an architecture that used a single CNN for all product-code media. This approach created a model that was too large to be distributed to mobile apps and the execution time was longer than desired. Our applied-AI partners at Quantiphi, Inc.began iterating on different model architectures, eventually landing on one that used multiple CNNs.

This new architecture reduced the model size dramatically without sacrificing accuracy, but it was still on the high end of what we needed in order to support over-the-air updates to mobile apps. We next used TensorFlow's prebuilt quantization module to reduce the model size by reducing the fidelity of the weights between connected neurons. Quantization reduced the model size by a factor of 4, but a dramatic reduction in model size occurred when Quantiphi had a breakthrough using a new approach called SqueezeNet.

The SqueezeNet model was published by a team of researchers from UC Berkeley and Stanford in November of 2016. It uses a small but highly complex design to achieve accuracy levels on par with much larger models against popular benchmarks such as Imagenet. After re-architecting our character recognition models to use a SqueezeNet CNN, Quantiphi was able to reduce the model size of certain media types by a factor of 100. Since the SqueezeNet model was inherently smaller, a richer feature detection architecture could be constructed, achieving much higher accuracy at much smaller sizes compared to our first batch of models trained without SqueezeNet. We now have a highly accurate model that can be easily updated on remote devices; the recognition success rate of our final model before active learning was close to 96%, which translates into a 99.7% character recognition accuracy (just 3 misses for every 1000 character predictions).

Valid product-code recognition examples with different types of occlusion, translation, and camera focus issues

Crossing boundaries with AI

Advances in artificial intelligence and the maturity of TensorFlow enabled us to finally achieve a long-sought proof-of-purchase capability. Since launching in late February 2017, our product code recognition platform has fueled more than a dozen promotions and resulted in over 180,000 scanned codes; it is now a core component for all of Coca-Cola North America's web-based promotions.

Moving to an AI-enabled product-code recognition platform has been valuable for two key reasons:

  • Frictionless proof-of-purchase was enabled in a timely fashion, corresponding to our overall move to a mobile-first marketing platform.
  • Coke saved millions of dollars by avoiding the requirement to update printers in our production lines to support higher-fidelity fonts that would work with existing off-the-shelf OCR software.

Our product-code recognition platform is the first execution of new AI-enabled capabilities at scale within Coca-Cola. We're now exploring AI applications across multiple lines of business, from new product development to ecommerce retail optimization.

Introduction to TensorFlow Datasets and Estimators

Posted by The TensorFlow Team

TensorFlow 1.3 introduces two important features that you should try out:

  • Datasets: A completely new way of creating input pipelines (that is, reading data into your program).
  • Estimators: A high-level way to create TensorFlow models. Estimators include pre-made models for common machine learning tasks, but you can also use them to create your own custom models.

Below you can see how they fit in the TensorFlow architecture. Combined, they offer an easy way to create TensorFlow models and to feed data to them:

Our Example Model

To explore these features we're going to build a model and show you relevant code snippets. The complete code is available here, including instructions for getting the training and test files. Note that the code was written to demonstrate how Datasets and Estimators work functionally, and was not optimized for maximum performance.

The trained model categorizes Iris flowers based on four botanical features (sepal length, sepal width, petal length, and petal width). So, during inference, you can provide values for those four features and the model will predict that the flower is one of the following three beautiful variants:

From left to right: Iris setosa(by Radomil, CC BY-SA 3.0), Iris versicolor (by Dlanglois, CC BY-SA 3.0), and Iris virginica(by Frank Mayfield, CC BY-SA 2.0).

We're going to train a Deep Neural Network Classifier with the below structure. All input and output values will be float32, and the sum of the output values will be 1 (as we are predicting the probability for each individual Iris type):

For example, an output result might be 0.05 for Iris Setosa, 0.9 for Iris Versicolor, and 0.05 for Iris Virginica, which indicates a 90% probability that this is an Iris Versicolor.

Alright! Now that we have defined the model, let's look at how we can use Datasets and Estimators to train it and make predictions.

Introducing The Datasets

Datasets is a new way to create input pipelines to TensorFlow models. This API is much more performant than using feed_dict or the queue-based pipelines, and it's cleaner and easier to use. Although Datasets still resides in tf.contrib.data at 1.3, we expect to move this API to core at 1.4, so it's high time to take it for a test drive.

At a high-level, the Datasets consists of the following classes:

Where:

  • Dataset: Base class containing methods to create and transform datasets. Also allows you initialize a dataset from data in memory, or from a Python generator.
  • TextLineDataset: Reads lines from text files.
  • TFRecordDataset: Reads records from TFRecord files.
  • FixedLengthRecordDataset: Reads fixed size records from binary files.
  • Iterator: Provides a way to access one dataset element at a time.

Our dataset

To get started, let's first look at the dataset we will use to feed our model. We'll read data from a CSV file, where each row will contain five values-the four input values, plus the label:

The label will be:

  • 0 for Iris Setosa
  • 1 for Versicolor
  • 2 for Virginica.

Representing our dataset

To describe our dataset, we first create a list of our features:


feature_names = [
'SepalLength',
'SepalWidth',
'PetalLength',
'PetalWidth']

When we train our model, we'll need a function that reads the input file and returns the feature and label data. Estimators requires that you create a function of the following format:


def input_fn():
...<code>...
return ({ 'SepalLength':[values], ..<etc>.., 'PetalWidth':[values] },
[IrisFlowerType])

The return value must be a two-element tuple organized as follows: :

  • The first element must be a dict in which each input feature is a key, and then a list of values for the training batch.
  • The second element is a list of labels for the training batch.

Since we are returning a batch of input features and training labels, it means that all lists in the return statement will have equal lengths. Technically speaking, whenever we referred to "list" here, we actually mean a 1-d TensorFlow tensor.

To allow simple reuse of the input_fn we're going to add some arguments to it. This allows us to build input functions with different settings. The arguments are pretty straightforward:

  • file_path: The data file to read.
  • perform_shuffle: Whether the record order should be randomized.
  • repeat_count: The number of times to iterate over the records in the dataset. For example, if we specify 1, then each record is read once. If we specify None, iteration will continue forever.

Here's how we can implement this function using the Dataset API. We will wrap this in an "input function" that is suitable when feeding our Estimator model later on:

def my_input_fn(file_path, perform_shuffle=False, repeat_count=1):
def decode_csv(line):
parsed_line = tf.decode_csv(line, [[0.], [0.], [0.], [0.], [0]])
label = parsed_line[-1:] # Last element is the label
del parsed_line[-1] # Delete last element
features = parsed_line # Everything (but last element) are the features
d = dict(zip(feature_names, features)), label
return d

dataset = (tf.contrib.data.TextLineDataset(file_path) # Read text file
.skip(1) # Skip header row
.map(decode_csv)) # Transform each elem by applying decode_csv fn
if perform_shuffle:
# Randomizes input using a window of 256 elements (read into memory)
dataset = dataset.shuffle(buffer_size=256)
dataset = dataset.repeat(repeat_count) # Repeats dataset this # times
dataset = dataset.batch(32) # Batch size to use
iterator = dataset.make_one_shot_iterator()
batch_features, batch_labels = iterator.get_next()
return batch_features, batch_labels

Note the following: :

  • TextLineDataset: The Dataset API will do a lot of memory management for you when you're using its file-based datasets. You can, for example, read in dataset files much larger than memory or read in multiple files by specifying a list as argument.
  • shuffle: Reads buffer_size records, then shuffles (randomizes) their order.
  • map: Calls the decode_csv function with each element in the dataset as an argument (since we are using TextLineDataset, each element will be a line of CSV text). Then we apply decode_csv to each of the lines.
  • decode_csv: Splits each line into fields, providing the default values if necessary. Then returns a dict with the field keys and field values. The map function updates each elem (line) in the dataset with the dict.

That's an introduction to Datasets! Just for fun, we can now use this function to print the first batch:

next_batch = my_input_fn(FILE, True) # Will return 32 random elements

# Now let's try it out, retrieving and printing one batch of data.
# Although this code looks strange, you don't need to understand
# the details.
with tf.Session() as sess:
first_batch = sess.run(next_batch)
print(first_batch)

# Output
({'SepalLength': array([ 5.4000001, ...<repeat to 32 elems>], dtype=float32),
'PetalWidth': array([ 0.40000001, ...<repeat to 32 elems>], dtype=float32),
...
},
[array([[2], ...<repeat to 32 elems>], dtype=int32) # Labels
)

That's actually all we need from the Dataset API to implement our model. Datasets have a lot more capabilities though; please see the end of this post where we have collected more resources.

Introducing Estimators

Estimators is a high-level API that reduces much of the boilerplate code you previously needed to write when training a TensorFlow model. Estimators are also very flexible, allowing you to override the default behavior if you have specific requirements for your model.

There are two possible ways you can build your model using Estimators:

  • Pre-made Estimator - These are predefined estimators, created to generate a specific type of model. In this blog post, we will use the DNNClassifier pre-made estimator.
  • Estimator (base class) - Gives you complete control of how your model should be created by using a model_fn function. We will cover how to do this in a separate blog post.

Here is the class diagram for Estimators:

We hope to add more pre-made Estimators in future releases.

As you can see, all estimators make use of input_fn that provides the estimator with input data. In our case, we will reuse my_input_fn, which we defined for this purpose.

The following code instantiates the estimator that predicts the Iris flower type:

# Create the feature_columns, which specifies the input to our model.
# All our input features are numeric, so use numeric_column for each one.
feature_columns = [tf.feature_column.numeric_column(k) for k in feature_names]

# Create a deep neural network regression classifier.
# Use the DNNClassifier pre-made estimator
classifier = tf.estimator.DNNClassifier(
feature_columns=feature_columns, # The input features to our model
hidden_units=[10, 10], # Two layers, each with 10 neurons
n_classes=3,
model_dir=PATH) # Path to where checkpoints etc are stored

We now have a estimator that we can start to train.

Training the model

Training is performed using a single line of TensorFlow code:

# Train our model, use the previously function my_input_fn
# Input to training is a file with training example
# Stop training after 8 iterations of train data (epochs)
classifier.train(
input_fn=lambda: my_input_fn(FILE_TRAIN, True, 8))

But wait a minute... what is this "lambda: my_input_fn(FILE_TRAIN, True, 8)" stuff? That is where we hook up Datasets with the Estimators! Estimators needs data to perform training, evaluation, and prediction, and it uses the input_fn to fetch the data. Estimators require an input_fn with no arguments, so we create a function with no arguments using lambda, which calls our input_fn with the desired arguments: the file_path, shuffle setting, and repeat_count. In our case, we use our my_input_fn, passing it:

  • FILE_TRAIN, which is the training data file.
  • True, which tells the Estimator to shuffle the data.
  • 8, which tells the Estimator to and repeat the dataset 8 times.

Evaluating Our Trained Model

Ok, so now we have a trained model. How can we evaluate how well it's performing? Fortunately, every Estimator contains an evaluatemethod:

# Evaluate our model using the examples contained in FILE_TEST
# Return value will contain evaluation_metrics such as: loss & average_loss
evaluate_result = estimator.evaluate(
input_fn=lambda: my_input_fn(FILE_TEST, False, 4)
print("Evaluation results")
for key in evaluate_result:
print(" {}, was: {}".format(key, evaluate_result[key]))

In our case, we reach an accuracy of about ~93%. There are various ways of improving this accuracy, of course. One way is to simply run the program over and over. Since the state of the model is persisted (in model_dir=PATH above), the model will improve the more iterations you train it, until it settles. Another way would be to adjust the number of hidden layers or the number of nodes in each hidden layer. Feel free to experiment with this; please note, however, that when you make a change, you need to remove the directory specified in model_dir=PATH, since you are changing the structure of the DNNClassifier.

Making Predictions Using Our Trained Model

And that's it! We now have a trained model, and if we are happy with the evaluation results, we can use it to predict an Iris flower based on some input. As with training, and evaluation, we make predictions using a single function call:

# Predict the type of some Iris flowers.
# Let's predict the examples in FILE_TEST, repeat only once.
predict_results = classifier.predict(
input_fn=lambda: my_input_fn(FILE_TEST, False, 1))
print("Predictions on test file")
for prediction in predict_results:
# Will print the predicted class, i.e: 0, 1, or 2 if the prediction
# is Iris Sentosa, Vericolor, Virginica, respectively.
print prediction["class_ids"][0]

Making Predictions on Data in Memory

The preceding code specified FILE_TEST to make predictions on data stored in a file, but how could we make predictions on data residing in other sources, for example, in memory? As you may guess, this does not actually require a change to our predict call. Instead, we configure the Dataset API to use a memory structure as follows:

# Let create a memory dataset for prediction.
# We've taken the first 3 examples in FILE_TEST.
prediction_input = [[5.9, 3.0, 4.2, 1.5], # -> 1, Iris Versicolor
[6.9, 3.1, 5.4, 2.1], # -> 2, Iris Virginica
[5.1, 3.3, 1.7, 0.5]] # -> 0, Iris Sentosa
def new_input_fn():
def decode(x):
x = tf.split(x, 4) # Need to split into our 4 features
# When predicting, we don't need (or have) any labels
return dict(zip(feature_names, x)) # Then build a dict from them

# The from_tensor_slices function will use a memory structure as input
dataset = tf.contrib.data.Dataset.from_tensor_slices(prediction_input)
dataset = dataset.map(decode)
iterator = dataset.make_one_shot_iterator()
next_feature_batch = iterator.get_next()
return next_feature_batch, None # In prediction, we have no labels

# Predict all our prediction_input
predict_results = classifier.predict(input_fn=new_input_fn)

# Print results
print("Predictions on memory data")
for idx, prediction in enumerate(predict_results):
type = prediction["class_ids"][0] # Get the predicted class (index)
if type == 0:
print("I think: {}, is Iris Sentosa".format(prediction_input[idx]))
elif type == 1:
print("I think: {}, is Iris Versicolor".format(prediction_input[idx]))
else:
print("I think: {}, is Iris Virginica".format(prediction_input[idx])

Dataset.from_tensor_slides() is designed for small datasets that fit in memory. When using TextLineDataset as we did for training and evaluation, you can have arbitrarily large files, as long as your memory can manage the shuffle buffer and batch sizes.

Freebies

Using a pre-made Estimator like DNNClassifier provides a lot of value. In addition to being easy to use, pre-made Estimators also provide built-in evaluation metrics, and create summaries you can see in TensorBoard. To see this reporting, start TensorBoard from your command-line as follows:

# Replace PATH with the actual path passed as model_dir argument when the
# DNNRegressor estimator was created.
tensorboard --logdir=PATH

The following diagrams show some of the data that TensorBoard will provide:

Summary

In this this blogpost, we explored Datasets and Estimators. These are important APIs for defining input data streams and creating models, so investing time to learn them is definitely worthwhile!

For more details, be sure to check out

But it doesn't stop here. We will shortly publish more posts that describe how these APIs work, so stay tuned for that!

Until then, Happy TensorFlow coding!

Build your own Machine Learning Visualizations with the new TensorBoard API



When we open-sourced TensorFlow in 2015, it included TensorBoard, a suite of visualizations for inspecting and understanding your TensorFlow models and runs. Tensorboard included a small, predetermined set of visualizations that are generic and applicable to nearly all deep learning applications such as observing how loss changes over time or exploring clusters in high-dimensional spaces. However, in the absence of reusable APIs, adding new visualizations to TensorBoard was prohibitively difficult for anyone outside of the TensorFlow team, leaving out a long tail of potentially creative, beautiful and useful visualizations that could be built by the research community.

To allow the creation of new and useful visualizations, we announce the release of a consistent set of APIs that allows developers to add custom visualization plugins to TensorBoard. We hope that developers use this API to extend TensorBoard and ensure that it covers a wider variety of use cases.

We have updated the existing dashboards (tabs) in TensorBoard to use the new API, so they serve as examples for plugin creators. For the current listing of plugins included within TensorBoard, you can explore the tensorboard/plugins directory on GitHub. For instance, observe the new plugin that generates precision-recall curves:
The plugin demonstrates the 3 parts of a standard TensorBoard plugin:
  • A TensorFlow summary op used to collect data for later visualization. [GitHub]
  • A Python backend that serves custom data. [GitHub]
  • A dashboard within TensorBoard built with TypeScript and polymer. [GitHub]
Additionally, like other plugins, the “pr_curves” plugin provides a demo that (1) users can look over in order to learn how to use the plugin and (2) the plugin author can use to generate example data during development. To further clarify how plugins work, we’ve also created a barebones TensorBoard “Greeter” plugin. This simple plugin collects greetings (simple strings preceded by “Hello, ”) during model runs and displays them. We recommend starting by exploring (or forking) the Greeter plugin as well as other existing plugins.

A notable example of how contributors are already using the TensorBoard API is Beholder, which was recently created by Chris Anderson while working on his master’s degree. Beholder shows a live video feed of data (e.g. gradients and convolution filters) as a model trains. You can watch the demo video here.
We look forward to seeing what innovations will come out of the community. If you plan to contribute a plugin to TensorBoard’s repository, you should get in touch with us first through the issue tracker with your idea so that we can help out and possibly guide you.

Acknowledgements
Dandelion Mané and William Chargin played crucial roles in building this API.



Transformer: A Novel Neural Network Architecture for Language Understanding



Neural networks, in particular recurrent neural networks (RNNs), are now at the core of the leading approaches to language understanding tasks such as language modeling, machine translation and question answering. In Attention Is All You Need we introduce the Transformer, a novel neural network architecture based on a self-attention mechanism that we believe to be particularly well-suited for language understanding.

In our paper, we show that the Transformer outperforms both recurrent and convolutional models on academic English to German and English to French translation benchmarks. On top of higher translation quality, the Transformer requires less computation to train and is a much better fit for modern machine learning hardware, speeding up training by up to an order of magnitude.
BLEU scores (higher is better) of single models on the standard WMT newstest2014 English to German translation benchmark.
BLEU scores (higher is better) of single models on the standard WMT newstest2014 English to French translation benchmark.
Accuracy and Efficiency in Language Understanding
Neural networks usually process language by generating fixed- or variable-length vector-space representations. After starting with representations of individual words or even pieces of words, they aggregate information from surrounding words to determine the meaning of a given bit of language in context. For example, deciding on the most likely meaning and appropriate representation of the word “bank” in the sentence “I arrived at the bank after crossing the…” requires knowing if the sentence ends in “... road.” or “... river.”

RNNs have in recent years become the typical network architecture for translation, processing language sequentially in a left-to-right or right-to-left fashion. Reading one word at a time, this forces RNNs to perform multiple steps to make decisions that depend on words far away from each other. Processing the example above, an RNN could only determine that “bank” is likely to refer to the bank of a river after reading each word between “bank” and “river” step by step. Prior research has shown that, roughly speaking, the more such steps decisions require, the harder it is for a recurrent network to learn how to make those decisions.

The sequential nature of RNNs also makes it more difficult to fully take advantage of modern fast computing devices such as TPUs and GPUs, which excel at parallel and not sequential processing. Convolutional neural networks (CNNs) are much less sequential than RNNs, but in CNN architectures like ByteNet or ConvS2S the number of steps required to combine information from distant parts of the input still grows with increasing distance.

The Transformer
In contrast, the Transformer only performs a small, constant number of steps (chosen empirically). In each step, it applies a self-attention mechanism which directly models relationships between all words in a sentence, regardless of their respective position. In the earlier example “I arrived at the bank after crossing the river”, to determine that the word “bank” refers to the shore of a river and not a financial institution, the Transformer can learn to immediately attend to the word “river” and make this decision in a single step. In fact, in our English-French translation model we observe exactly this behavior.

More specifically, to compute the next representation for a given word - “bank” for example - the Transformer compares it to every other word in the sentence. The result of these comparisons is an attention score for every other word in the sentence. These attention scores determine how much each of the other words should contribute to the next representation of “bank”. In the example, the disambiguating “river” could receive a high attention score when computing a new representation for “bank”. The attention scores are then used as weights for a weighted average of all words’ representations which is fed into a fully-connected network to generate a new representation for “bank”, reflecting that the sentence is talking about a river bank.

The animation below illustrates how we apply the Transformer to machine translation. Neural networks for machine translation typically contain an encoder reading the input sentence and generating a representation of it. A decoder then generates the output sentence word by word while consulting the representation generated by the encoder. The Transformer starts by generating initial representations, or embeddings, for each word. These are represented by the unfilled circles. Then, using self-attention, it aggregates information from all of the other words, generating a new representation per word informed by the entire context, represented by the filled balls. This step is then repeated multiple times in parallel for all words, successively generating new representations.
The decoder operates similarly, but generates one word at a time, from left to right. It attends not only to the other previously generated words, but also to the final representations generated by the encoder.

Flow of Information
Beyond computational performance and higher accuracy, another intriguing aspect of the Transformer is that we can visualize what other parts of a sentence the network attends to when processing or translating a given word, thus gaining insights into how information travels through the network.

To illustrate this, we chose an example involving a phenomenon that is notoriously challenging for machine translation systems: coreference resolution. Consider the following sentences and their French translations:
It is obvious to most that in the first sentence pair “it” refers to the animal, and in the second to the street. When translating these sentences to French or German, the translation for “it” depends on the gender of the noun it refers to - and in French “animal” and “street” have different genders. In contrast to the current Google Translate model, the Transformer translates both of these sentences to French correctly. Visualizing what words the encoder attended to when computing the final representation for the word “it” sheds some light on how the network made the decision. In one of its steps, the Transformer clearly identified the two nouns “it” could refer to and the respective amount of attention reflects its choice in the different contexts.
The encoder self-attention distribution for the word “it” from the 5th to the 6th layer of a Transformer trained on English to French translation (one of eight attention heads).
Given this insight, it might not be that surprising that the Transformer also performs very well on the classic language analysis task of syntactic constituency parsing, a task the natural language processing community has attacked with highly specialized systems for decades.
In fact, with little adaptation, the same network we used for English to German translation outperformed all but one of the previously proposed approaches to constituency parsing.

Next Steps
We are very excited about the future potential of the Transformer and have already started applying it to other problems involving not only natural language but also very different inputs and outputs, such as images and video. Our ongoing experiments are accelerated immensely by the Tensor2Tensor library, which we recently open sourced. In fact, after downloading the library you can train your own Transformer networks for translation and parsing by invoking just a few commands. We hope you’ll give it a try, and look forward to seeing what the community can do with the Transformer.

Acknowledgements
This research was conducted by Ashish Vaswani, Noam Shazeer, Niki Parmar, Jakob Uszkoreit, Llion Jones, Aidan N. Gomez and Łukasz Kaiser. Additional thanks go to David Chenell for creating the animation above.

Kaldi now offers TensorFlow integration

Posted by Raziel Alvarez, Staff Research Engineer at Google and Yishay Carmiel, Founder of IntelligentWire

Automatic speech recognition (ASR) has seen widespread adoption due to the recent proliferation of virtual personal assistants and advances in word recognition accuracy from the application of deep learning algorithms. Many speech recognition teams rely on Kaldi, a popular open-source speech recognition toolkit. We're announcing today that Kaldi now offers TensorFlow integration.

With this integration, speech recognition researchers and developers using Kaldi will be able to use TensorFlow to explore and deploy deep learning models in their Kaldi speech recognition pipelines. This will allow the Kaldi community to build even better and more powerful ASR systems as well as providing TensorFlow users with a path to explore ASR while drawing upon the experience of the large community of Kaldi developers.

Building an ASR system that can understand human speech in every language, accent, environment, and type of conversation is an extremely complex undertaking. A traditional ASR system can be seen as a processing pipeline with many separate modules, where each module operates on the output from the previous one. Raw audio data enters the pipeline at one end and a transcription of recognized speech emerges from the other. In the case of Kaldi, these ASR transcriptions are post processed in a variety of ways to support an increasing array of end-user applications.

Yishay Carmiel and Hainan Xu of Seattle-based IntelligentWire, who led the development of the integration between Kaldi and TensorFlow with support from the two teams, know this complexity first-hand. Their company has developed cloud software to bridge the gap between live phone conversations and business applications. Their goal is to let businesses analyze and act on the contents of the thousands of conversations their representatives have with customers in real-time and automatically handle tasks like data entry or responding to requests. IntelligentWire is currently focused on the contact center market, in which more than 22 million agents throughout the world spend 50 billion hours a year on the phone and about 25 billion hours interfacing with and operating various business applications.

For an ASR system to be useful in this context, it must not only deliver an accurate transcription but do so with very low latency in a way that can be scaled to support many thousands of concurrent conversations efficiently. In situations like this, recent advances in deep learning can help push technical limits, and TensorFlow can be very useful.

In the last few years, deep neural networks have been used to replace many existing ASR modules, resulting in significant gains in word recognition accuracy. These deep learning models typically require processing vast amounts of data at scale, which TensorFlow simplifies. However, several major challenges must still be overcome when developing production-grade ASR systems:

  • Algorithms - Deep learning algorithms give the best results when tailored to the task at hand, including the acoustic environment (e.g. noise), the specific language spoken, the range of vocabulary, etc. These algorithms are not always easy to adapt once deployed.
  • Data - Building an ASR system for different languages and different acoustic environments requires large quantities of multiple types of data. Such data may not always be available or may not be suitable for the use case.
  • Scale - ASR systems that can support massive amounts of usage and many languages typically consume large amounts of computational power.

One of the ASR system modules that exemplifies these challenges is the language model. Language models are a key part of most state-of-the-art ASR systems; they provide linguistic context that helps predict the proper sequence of words and distinguish between words that sound similar. With recent machine learning breakthroughs, speech recognition developers are now using language models based on deep learning, known as neural language models. In particular, recurrent neural language models have shown superior results over classic statistical approaches.

However, the training and deployment of neural language models is complicated and highly time-consuming. For IntelligentWire, the integration of TensorFlow into Kaldi has reduced the ASR development cycle by an order of magnitude. If a language model already exists in TensorFlow, then going from model to proof of concept can take days rather than weeks; for new models, the development time can be reduced from months to weeks. Deploying new TensorFlow models into production Kaldi pipelines is straightforward as well, providing big gains for anyone working directly with Kaldi as well as the promise of more intelligent ASR systems for everyone in the future.

Similarly, this integration provides TensorFlow developers with easy access to a robust ASR platform and the ability to incorporate existing speech processing pipelines, such as Kaldi's powerful acoustic model, into their machine learning applications. Kaldi modules that feed the training of a TensorFlow deep learning model can be swapped cleanly, facilitating exploration, and the same pipeline that is used in production can be reused to evaluate the quality of the model.

We hope this Kaldi-TensorFlow integration will bring these two vibrant open-source communities closer together and support a wide variety of new speech-based products and related research breakthroughs. To get started using Kaldi with TensorFlow, please check out the Kaldi repo and also take a look at an example for Kaldi setup running with TensorFlow.

Launching the Speech Commands Dataset



At Google, we’re often asked how to get started using deep learning for speech and other audio recognition problems, like detecting keywords or commands. And while there are some great open source speech recognition systems like Kaldi that can use neural networks as a component, their sophistication makes them tough to use as a guide to a simpler tasks. Perhaps more importantly, there aren’t many free and openly available datasets ready to be used for a beginner’s tutorial (many require preprocessing before a neural network model can be built on them) or that are well suited for simple keyword detection.

To solve these problems, the TensorFlow and AIY teams have created the Speech Commands Dataset, and used it to add training* and inference sample code to TensorFlow. The dataset has 65,000 one-second long utterances of 30 short words, by thousands of different people, contributed by members of the public through the AIY website. It’s released under a Creative Commons BY 4.0 license, and will continue to grow in future releases as more contributions are received. The dataset is designed to let you build basic but useful voice interfaces for applications, with common words like “Yes”, “No”, digits, and directions included. The infrastructure we used to create the data has been open sourced too, and we hope to see it used by the wider community to create their own versions, especially to cover underserved languages and applications.

To try it out for yourself, download the prebuilt set of the TensorFlow Android demo applications and open up “TF Speech”. You’ll be asked for permission to access your microphone, and then see a list of ten words, each of which should light up as you say them.
The results will depend on whether your speech patterns are covered by the dataset, so it may not be perfect — commercial speech recognition systems are a lot more complex than this teaching example. But we’re hoping that as more accents and variations are added to the dataset, and as the community contributes improved models to TensorFlow, we’ll continue to see improvements and extensions.

You can also learn how to train your own version of this model through the new audio recognition tutorial on TensorFlow.org. With the latest development version of the framework and a modern desktop machine, you can download the dataset and train the model in just a few hours. You’ll also see a wide variety of options to customize the neural network for different problems, and to make different latency, size, and accuracy tradeoffs to run on different platforms.

We are excited to see what new applications people are able to build with the help of this dataset and tutorial, so I hope you get a chance to dive in and start recognizing!


* The architecture this network is based on is described in Convolutional Neural Networks for Small-footprint Keyword Spotting, presented at Interspeech 2015.

TensorFlow Serving 1.0

Posted by Kiril Gorovoy, Software Engineer

We've come a long way since our initial open source release in February 2016 of TensorFlow Serving, a high performance serving system for machine learned models, designed for production environments. Today, we are happy to announce the release of TensorFlow Serving 1.0. Version 1.0 is built from TensorFlow head, and our future versions will be minor-version aligned with TensorFlow releases.

For a good overview of the system, watch Noah Fiedel's talk given at Google I/O 2017.

When we first announced the project, it was a set of libraries providing the core functionality to manage a model's lifecycle and serve inference requests. We later introduced a gRPC Model Server binary with a Predict API and an example of how to deploy it on Kubernetes. Since then, we've worked hard to expand its functionality to fit different use cases and to stabilize the API to meet the needs of users. Today there are over 800 projects within Google using TensorFlow Serving in production. We've battle tested the server and the API and have converged on a stable, robust, high-performance implementation.

We've listened to the open source community and are excited to offer a prebuilt binary available through apt-get install. Now, to get started using TensorFlow Serving, you can simply install and run without needing to spend time compiling. As always, a Docker container can still be used to install the server binary on non-Linux systems.

With this release, TensorFlow Serving is also officially deprecating and stopping support for the legacy SessionBundle model format. SavedModel, TensorFlow's model format introduced as part of TensorFlow 1.0 is now the officially supported format.

To get started, please check out the documentation for the project and our tutorial. Enjoy TensorFlow Serving 1.0!

Building Your Own Neural Machine Translation System in TensorFlow



Machine translation – the task of automatically translating between languages – is one of the most active research areas in the machine learning community. Among the many approaches to machine translation, sequence-to-sequence ("seq2seq") models [1, 2] have recently enjoyed great success and have become the de facto standard in most commercial translation systems, such as Google Translate, thanks to its ability to use deep neural networks to capture sentence meanings. However, while there is an abundance of material on seq2seq models such as OpenNMT or tf-seq2seq, there is a lack of material that teaches people both the knowledge and the skills to easily build high-quality translation systems.

Today we are happy to announce a new Neural Machine Translation (NMT) tutorial for TensorFlow that gives readers a full understanding of seq2seq models and shows how to build a competitive translation model from scratch. The tutorial is aimed at making the process as simple as possible, starting with some background knowledge on NMT and walking through code details to build a vanilla system. It then dives into the attention mechanism [3, 4], a key ingredient that allows NMT systems to handle long sentences. Finally, the tutorial provides details on how to replicate key features in the Google’s NMT (GNMT) system [5] to train on multiple GPUs.

The tutorial also contains detailed benchmark results, which users can replicate on their own. Our models provide a strong open-source baseline with performance on par with GNMT results [5]. We achieve 24.4 BLEU points on the popular WMT’14 English-German translation task.
Other benchmark results (English-Vietnamese, German-English) can be found in the tutorial.

In addition, this tutorial showcases the fully dynamic seq2seq API (released with TensorFlow 1.2) aimed at making building seq2seq models clean and easy:
  • Easily read and preprocess dynamically sized input sequences using the new input pipeline in tf.contrib.data.
  • Use padded batching and sequence length bucketing to improve training and inference speeds.
  • Train seq2seq models using popular architectures and training schedules, including several types of attention and scheduled sampling.
  • Perform inference in seq2seq models using in-graph beam search.
  • Optimize seq2seq models for multi-GPU settings.
We hope this will help spur the creation of, and experimentation with, many new NMT models by the research community. To get started on your own research, check out the tutorial on GitHub!

Core contributors
Thang Luong, Eugene Brevdo, and Rui Zhao.

Acknowledgements
We would like to especially thank our collaborator on the NMT project, Rui Zhao. Without his tireless effort, this tutorial would not have been possible. Additional thanks go to Denny Britz, Anna Goldie, Derek Murray, and Cinjon Resnick for their work bringing new features to TensorFlow and the seq2seq library. Lastly, we thank Lukasz Kaiser for the initial help on the seq2seq codebase; Quoc Le for the suggestion to replicate GNMT; Yonghui Wu and Zhifeng Chen for details on the GNMT systems; as well as the Google Brain team for their support and feedback!

References
[1] Sequence to sequence learning with neural networks, Ilya Sutskever, Oriol Vinyals, and Quoc V. Le. NIPS, 2014.
[2] Learning phrase representations using RNN encoder-decoder for statistical machine translation, Kyunghyun Cho, Bart Van Merrienboer, Caglar Gulcehre, Dzmitry Bahdanau, Fethi Bougares, Holger Schwenk, and Yoshua Bengio. EMNLP 2014.
[3] Neural machine translation by jointly learning to align and translate, Dzmitry Bahdanau, Kyunghyun Cho, and Yoshua Bengio. ICLR, 2015.
[4] Effective approaches to attention-based neural machine translation, Minh-Thang Luong, Hieu Pham, and Christopher D Manning. EMNLP, 2015.
[5] Google’s Neural Machine Translation System: Bridging the Gap between Human and Machine Translation, Yonghui Wu, Mike Schuster, Zhifeng Chen, Quoc V. Le, Mohammad Norouzi, Wolfgang Macherey, Maxim Krikun, Yuan Cao, Qin Gao, Klaus Macherey, Jeff Klingner, Apurva Shah, Melvin Johnson, Xiaobing Liu, Łukasz Kaiser, Stephan Gouws, Yoshikiyo Kato, Taku Kudo, Hideto Kazawa, Keith Stevens, George Kurian, Nishant Patil, Wei Wang, Cliff Young, Jason Smith, Jason Riesa, Alex Rudnick, Oriol Vinyals, Greg Corrado, Macduff Hughes, Jeffrey Dean. Technical Report, 2016.

MultiModel: Multi-Task Machine Learning Across Domains



Over the last decade, the application and performance of Deep Learning has progressed at an astonishing rate. However, the current state of the field is that the neural network architectures are highly specialized to specific domains of application. An important question remains unanswered: Will a convergence between these domains facilitate a unified model capable of performing well across multiple domains?

Today, we present MultiModel, a neural network architecture that draws from the success of vision, language and audio networks to simultaneously solve a number of problems spanning multiple domains, including image recognition, translation and speech recognition. While strides have been made in this direction before, namely in Google’s Multilingual Neural Machine Translation System used in Google Translate, MultiModel is a first step towards the convergence of vision, audio and language understanding into a single network.

The inspiration for how MultiModel handles multiple domains comes from how the brain transforms sensory input from different modalities (such as sound, vision or taste), into a single shared representation and back out in the form of language or actions. As an analog to these modalities and the transformations they perform, MultiModel has a number of small modality-specific sub-networks for audio, images, or text, and a shared model consisting of an encoder, input/output mixer and decoder, as illustrated below.
MultiModel architecture: small modality-specific sub-networks work with a shared encoder, I/O mixer and decoder. Each petal represents a modality, transforming to and from the internal representation.
We demonstrate that MultiModel is capable of learning eight different tasks simultaneously: it can detect objects in images, provide captions, recognize speech, translate between four pairs of languages, and do grammatical constituency parsing at the same time. The input is given to the model together with a very simple signal that determines which output we are requesting. Below we illustrate a few examples taken from a MultiModel trained jointly on these eight tasks1:
When designing MultiModel it became clear that certain elements from each domain of research (vision, language and audio) were integral to the model’s success in related tasks. We demonstrate that these computational primitives (such as convolutions, attention, or mixture-of-experts layers) clearly improve performance on their originally intended domain of application, while not hindering MultiModel’s performance on other tasks. It is not only possible to achieve good performance while training jointly on multiple tasks, but on tasks with limited quantities of data, the performance actually improves. To our surprise, this happens even if the tasks come from different domains that would appear to have little in common, e.g., an image recognition task can improve performance on a language task.

It is important to note that while MultiModel does not establish new performance records, it does provide insight into the dynamics of multi-domain multi-task learning in neural networks, and the potential for improved learning on data-limited tasks by the introduction of auxiliary tasks. There is a longstanding saying in machine learning: “the best regularizer is more data”; in MultiModel, this data can be sourced across domains, and consequently can be obtained more easily than previously thought. MultiModel provides evidence that training in concert with other tasks can lead to good results and improve performance on data-limited tasks.

Many questions about multi-domain machine learning remain to be studied, and we will continue to work on tuning Multimodel and improving its performance. To allow this research to progress quickly, we open-sourced MultiModel as part of the Tensor2Tensor library. We believe that such synergetic models trained on data from multiple domains will be the next step in deep learning and will ultimately solve tasks beyond the reach of current narrowly trained networks.

Acknowledgements
This work is a collaboration between Googlers Łukasz Kaiser, Noam Shazeer, Ashish Vaswani, Niki Parmar, Llion Jones and Jakob Uszkoreit, and Aidan N. Gomez from the University of Toronto. It was performed while Aidan was working with the Google Brain team.



1 The 8 tasks were: (1) speech recognition (WSJ corpus), (2) image classification (ImageNet), (3) image captioning (MS COCO), (4) parsing (Penn Treebank), (5) English-German translation, (6) German-English translation, (7) English-French translation, (8) French-English translation (all using WMT data-sets).

Accelerating Deep Learning Research with the Tensor2Tensor Library



Deep Learning (DL) has enabled the rapid advancement of many useful technologies, such as machine translation, speech recognition and object detection. In the research community, one can find code open-sourced by the authors to help in replicating their results and further advancing deep learning. However, most of these DL systems use unique setups that require significant engineering effort and may only work for a specific problem or architecture, making it hard to run new experiments and compare the results.

Today, we are happy to release Tensor2Tensor (T2T), an open-source system for training deep learning models in TensorFlow. T2T facilitates the creation of state-of-the art models for a wide variety of ML applications, such as translation, parsing, image captioning and more, enabling the exploration of various ideas much faster than previously possible. This release also includes a library of datasets and models, including the best models from a few recent papers (Attention Is All You Need, Depthwise Separable Convolutions for Neural Machine Translation and One Model to Learn Them All) to help kick-start your own DL research.

Translation Model
Training time
BLEU (difference from baseline)
Transformer (T2T)
3 days on 8 GPU
28.4 (+7.8)
SliceNet (T2T)
6 days on 32 GPUs
26.1 (+5.5)
1 day on 64 GPUs
26.0 (+5.4)
ConvS2S
18 days on 1 GPU
25.1 (+4.5)
GNMT
1 day on 96 GPUs
24.6 (+4.0)
8 days on 32 GPUs
23.8 (+3.2)
MOSES (phrase-based baseline)
N/A
20.6 (+0.0)
BLEU scores (higher is better) on the standard WMT English-German translation task.
As an example of the kind of improvements T2T can offer, we applied the library to machine translation. As you can see in the table above, two different T2T models, SliceNet and Transformer, outperform the previous state-of-the-art, GNMT+MoE. Our best T2T model, Transformer, is 3.8 points better than the standard GNMT model, which itself was 4 points above the baseline phrase-based translation system, MOSES. Notably, with T2T you can approach previous state-of-the-art results with a single GPU in one day: a small Transformer model (not shown above) gets 24.9 BLEU after 1 day of training on a single GPU. Now everyone with a GPU can tinker with great translation models on their own: our github repo has instructions on how to do that.

Modular Multi-Task Training
The T2T library is built with familiar TensorFlow tools and defines multiple pieces needed in a deep learning system: data-sets, model architectures, optimizers, learning rate decay schemes, hyperparameters, and so on. Crucially, it enforces a standard interface between all these parts and implements current ML best practices. So you can pick any data-set, model, optimizer and set of hyperparameters, and run the training to check how it performs. We made the architecture modular, so every piece between the input data and the predicted output is a tensor-to-tensor function. If you have a new idea for the model architecture, you don’t need to replace the whole setup. You can keep the embedding part and the loss and everything else, just replace the model body by your own function that takes a tensor as input and returns a tensor.

This means that T2T is flexible, with training no longer pinned to a specific model or dataset. It is so easy that even architectures like the famous LSTM sequence-to-sequence model can be defined in a few dozen lines of code. One can also train a single model on multiple tasks from different domains. Taken to the limit, you can even train a single model on all data-sets concurrently, and we are happy to report that our MultiModel, trained like this and included in T2T, yields good results on many tasks even when training jointly on ImageNet (image classification), MS COCO (image captioning), WSJ (speech recognition), WMT (translation) and the Penn Treebank parsing corpus. It is the first time a single model has been demonstrated to be able to perform all these tasks at once.

Built-in Best Practices
With this initial release, we also provide scripts to generate a number of data-sets widely used in the research community1, a handful of models2, a number of hyperparameter configurations, and a well-performing implementation of other important tricks of the trade. While it is hard to list them all, if you decide to run your model with T2T you’ll get for free the correct padding of sequences and the corresponding cross-entropy loss, well-tuned parameters for the Adam optimizer, adaptive batching, synchronous distributed training, well-tuned data augmentation for images, label smoothing, and a number of hyper-parameter configurations that worked very well for us, including the ones mentioned above that achieve the state-of-the-art results on translation and may help you get good results too.

As an example, consider the task of parsing English sentences into their grammatical constituency trees. This problem has been studied for decades and competitive methods were developed with a lot of effort. It can be presented as a sequence-to-sequence problem and be solved with neural networks, but it used to require a lot of tuning. With T2T, it took us only a few days to add the parsing data-set generator and adjust our attention transformer model to train on this problem. To our pleasant surprise, we got very good results in only a week:

Parsing Model
F1 score (higher is better)
Transformer (T2T)
91.3
Dyer et al.
91.7
Zhu et al.
90.4
Socher et al.
90.4
Vinyals & Kaiser et al.
88.3
Parsing F1 scores on the standard test set, section 23 of the WSJ. We only compare here models trained discriminatively on the Penn Treebank WSJ training set, see the paper for more results.

Contribute to Tensor2Tensor
In addition to exploring existing models and data-sets, you can easily define your own model and add your own data-sets to Tensor2Tensor. We believe the already included models will perform very well for many NLP tasks, so just adding your data-set might lead to interesting results. By making T2T modular, we also make it very easy to contribute your own model and see how it performs on various tasks. In this way the whole community can benefit from a library of baselines and deep learning research can accelerate. So head to our github repository, try the new models, and contribute your own!

Acknowledgements
The release of Tensor2Tensor was only possible thanks to the widespread collaboration of many engineers and researchers. We want to acknowledge here the core team who contributed (in alphabetical order): Samy Bengio, Eugene Brevdo, Francois Chollet, Aidan N. Gomez, Stephan Gouws, Llion Jones, Łukasz Kaiser, Nal Kalchbrenner, Niki Parmar, Ryan Sepassi, Noam Shazeer, Jakob Uszkoreit, Ashish Vaswani.



1 We include a number of datasets for image classification (MNIST, CIFAR-10, CIFAR-100, ImageNet), image captioning (MS COCO), translation (WMT with multiple languages including English-German and English-French), language modelling (LM1B), parsing (Penn Treebank), natural language inference (SNLI), speech recognition (TIMIT), algorithmic problems (over a dozen tasks from reversing through addition and multiplication to algebra) and we will be adding more and welcome your data-sets too.

2 Including LSTM sequence-to-sequence RNNs, convolutional networks also with separable convolutions (e.g., Xception), recently researched models like ByteNet or the Neural GPU, and our new state-of-the-art models mentioned in this post that we will be actively updating in the repository.