Spatial Pyramid Pooling (SPP) is a technique which allows Convolutional Neural Networks (CNNs) to use input images of any size, not only $224\text{px} \times 224\text{px}$ as most architectures do. (However, there is a lower bound for the size of the input image).

## Idea

* Convolutional layers operate on any size, but fully connected layers need fixed-size inputs
* Solution:
  * Add a new SPP layer on top of the last convolutional layer, before the fully connected layer
  * Use an approach similar to bag of words (BoW), but maintain the spatial information. The BoW approach is used for text classification, where the order of the words is discarded and only the number of occurences is kept.
  * The SPP layer operates on each feature map independently.
  * The output of the SPP layer is of dimension $k \cdot M$, where $k$ is the number of feature maps the SPP layer got as input and $M$ is the number of bins.

Example: We could use spatial pyramid pooling with 21 bins:

* 1 bin which is the max of the complete feature map
* 4 bins which divide the image into 4 regions of equal size (depending on the input size) and rectangular shape. Each bin gets the max of its region.
* 16 bins which divide the image into 4 regions of equal size (depending on the input size) and rectangular shape. Each bin gets the max of its region.

## Evaluation

* Pascal VOC 2007, Caltech101: state-of-the-art, without finetuning
* ImageNet 2012: Boosts accuracy for various CNN architectures
* ImageNet Large Scale Visual Recognition Challenge (ILSVRC) 2014: Rank #2


## Code

The paper claims that the code is [here](http://research.microsoft.com/en-us/um/people/kahe/), but this seems not to be the case any more.

People have tried to implement it with Tensorflow ([1](http://stackoverflow.com/q/40913794/562769), [2](https://github.com/fchollet/keras/issues/2080), [3](https://github.com/tensorflow/tensorflow/issues/6011)), but by now no public working implementation is available.


## Related papers

* [Atrous Convolution](https://arxiv.org/abs/1606.00915)

SSD aims to solve the major problem with most of the current state of the art object detectors namely Faster RCNN and like. All the object detection algortihms have same  methodology 

- Train 2 different nets - Region Proposal Net (RPN) and advanced classifier to detect class of an object and bounding box separately.
- During inference, run the test image at different scales to detect object at multiple scales to account for invariance

This makes the nets extremely slow. Faster RCNN could operate at **7 FPS with 73.2% mAP** while SSD could achieve **59 FPS with 74.3% mAP ** on VOC 2007 dataset.

#### Methodology

SSD uses a single net for predict object class and bounding box. However it doesn't do that directly. It uses a mechanism for choosing ROIs, training end-to-end for predicting class and boundary shift for that ROI.

##### ROI selection

Borrowing from FasterRCNNs SSD uses the  concept of anchor boxes for generating ROIs from the feature maps of last layer of shared conv layer. For each pixel in layer of feature maps, k default boxes with different aspect ratios are chosen around every pixel in the map. So if there are feature maps each of m x n resolutions - that's *mnk* ROIs for a single feature layer. Now SSD uses multiple feature layers (with differing resolutions) for generating such ROIs primarily to capture size invariance of objects. But because earlier layers in deep conv net tends to capture low level features, it uses features after certain levels and layers henceforth.

##### ROI labelling
Any ROI that matches to Ground Truth for a class after applying appropriate transforms and having Jaccard overlap greater than 0.5 is positive. Now, given all feature maps are at different resolutions and each boxes are at different aspect ratios, doing that's not simple. SDD uses simple scaling and aspect ratios to get to the appropriate  ground truth dimensions for calculating Jaccard overlap for default boxes for each pixel at the given resolution

##### ROI classification

SSD uses single convolution kernel of 3*3 receptive fields to predict for each ROI the 4 offsets (centre-x offset, centre-y offset, height offset , width offset) from the Ground Truth box for each RoI, along with class confidence scores for each class. So that is if there are c classes (including background), there are (c+4) filters for each convolution kernels that looks at a ROI. 

So summarily we have convolution kernels  that look at ROIs (which are default boxes around each pixel in feature map layer) to generate (c+4) scores for each RoI. Multiple feature map layers with different resolutions are used for generating such ROIs. Some ROIs are positive and some negative depending on jaccard overlap after ground box has scaled appropriately taking resolution differences in input image and feature map into consideration.

Here's how it looks :
![](https://i.imgur.com/HOhsPZh.png)

##### Training

For each ROI a combined loss is calculated as a combination of localisation error and classification error. The details are best explained in the figure. 
![](https://i.imgur.com/zEDuSgi.png)

##### Inference
For each ROI predictions a small threshold is used to first filter out irrelevant predictions, Non Maximum Suppression (nms) with jaccard overlap of 0.45 per class is applied then on the remaining candidate ROIs and the top 200 detections per image are kept.

For further understanding of the intuitions regarding the paper and the results obtained please consider giving the full paper a read. 

The open sourced code is available at this [Github repo](https://github.com/weiliu89/caffe/tree/ssd)
 

(See also a more thorough summary in [a LaTeX PDF][1].)

This paper has some nice clear theory which bridges maximum likelihood (supervised) learning and standard reinforcement learning. It focuses on *structured prediction* tasks, where we want to learn to predict $p_\theta(y \mid x)$ where $y$ is some object with complex internal structure.

We can agree on some deficiencies of maximum likelihood learning:

- ML training fails to assign **partial credit**. Models are trained to maximize the likelihood of the ground-truth outputs in the dataset, and all other outputs are equally wrong. This is an increasingly important problem as the space of possible solutions grows.
- ML training is potentially disconnected from **downstream task reward**. In machine translation, we usually want to optimize relatively complex metrics like BLEU or TER. Since these metrics are non-differentiable, we have to settle for optimizing proxy losses that we hope are related to the metric of interest.

Reinforcement learning offers an attractive alternative in theory. RL algorithms are designed to optimize non-differentiable (even stochastic) reward functions, which sounds like just what we want. But RL algorithms have their own problems with this sort of structured output space:

- Standard RL algorithms rely on samples from the model we are learning, $p_\theta(y \mid x)$. This becomes intractable when our output space is very complex (e.g. 80-token sequences where each word is drawn from a vocabulary of 80,000 words).
- The reward spaces for problems of interest are extremely sparse. Our metrics will assign 0 reward to most of the 80^80K possible outputs in the translation problem in the paper.
- Vanilla RL doesn't take into account the ground-truth outputs available to us in structured prediction.

This paper designs a solution which combines supervised learning with a reinforcement learning-inspired smoothing method. Concretely, the authors design an **exponentiated payoff distribution** $q(y \mid y^*; \tau)$ which assigns high mass to high-reward outputs $y$ and low mass elsewhere. This distribution is used to effectively smooth the loss function established by the ground-truth outputs in the supervised data. We end up optimizing the following objective:

$$\mathcal L_\text{RML} = - \mathbb E_{x, y^* \sim \mathcal D}\left[ \sum_y q(y \mid y^*; \tau) \log p_\theta(y \mid x) \right]$$

This optimization depends on samples from our dataset $\mathcal D$ and, more importantly, the stationary payoff distribution $q$. This contrasts strongly with standard RL training, where the objective depends on samples from the non-stationary model distribution $p_\theta$. To make that clear, we can rewrite the above with another expectation:

$$\mathcal L_\text{RML} = - \mathbb E_{x, y^* \sim \mathcal D, y \sim q(y \mid y^*; \tau)}\left[ \log p_\theta(y \mid x) \right]$$

### Model details

If you're interested in the low-level details, I wrote up the gist of the math in [this PDF][1].

### Analysis

#### Relationship to label smoothing

This training approach is mathematically equivalent to label smoothing, applied here to structured output problems. In next-word prediction language modeling, a popular trick involves smoothing the target distributions by combining the ground-truth output with some simple base model, e.g. a unigram word frequency distribution. (This just means we take a weighted sum of the one-hot vector from our supervised data and a normalized frequency vector calculated on some corpus.) Mathematically, the cross entropy with label smoothing is

$$\mathcal L_\text{ML-smooth} = - \mathbb E_{x, y^* \sim \mathcal D} \left[ \sum_y p_\text{smooth}(y; y^*) \log p_\theta(y \mid x) \right]$$

(The equation above leaves out a constant entropy term.)

The gradient of this objective looks exactly the same as the reward-augmented ML gradient from the paper:

$$\nabla_\theta \mathcal L_\text{ML-smooth} = \mathbb E_{x, y^* \sim \mathcal D, y \sim p_\text{smooth}} \left[ \log p_\theta(y \mid x) \right]$$

So reward-augmented likelihood is equivalent to label smoothing, where our smoothing distribution is log-proportional to our downstream reward function.

#### Relationship to distillation

Optimizing the reward-augmented maximum likelihood is equivalent to minimizing the KL divergence $$D_\text{KL}(q(y \mid y^*; \tau) \mid\mid p_\theta(y \mid x))$$

This divergence reaches zero iff $q = p$. We can say, then, that the effect of optimizing on $\mathcal L_\text{RML}$ is to **distill** the reward function (which parameterizes $q$) into the model parameters $\theta$ (which parameterize $p_\theta$).

It's exciting to think about other sorts of more complex models that we might be able to distill in this framework. The unfortunate (?) restriction is that the "source" model of the distillation ($q$ in this paper) must admit to efficient sampling.

#### Relationship to adversarial training

We can also view reward-augmented maximum likelihood training as a data augmentation technique: it synthesizes new "partially correct" examples using the reward function as a guide. We then train on all of the original and synthesized data, again weighting the gradients based on the reward function.

Adversarial training is a similar data augmentation technique which generates examples that force the model to be robust to changes in its input space (robust to changes of $x$). Both adversarial training and the RML objective encourage the model to be robust "near" the ground-truth supervised data. A high-level comparison:

- Adversarial training can be seen as data augmentation in the input space; RML training performs data augmentation in the output space.
- Adversarial training is a **model-based data augmentation**: the samples are generated from a process that depends on the current parameters during training. RML training performs **data-based augmentation**, which could in theory be done independent of the actual training process.

---

Thanks to Andrej Karpathy, Alec Radford, and Tim Salimans for interesting discussion which contributed to this summary.

[1]: https://drive.google.com/file/d/0B3Rdm_P3VbRDVUQ4SVBRYW82dU0/view

The last two years have seen a number of improvements in the field of language model pretraining, and BERT - Bidirectional Encoder Representations from Transformers - is the most recent entry into this canon. The general problem posed by language model pretraining is: can we leverage huge amounts of raw text, which aren’t labeled for any specific classification task, to help us train better models for supervised language tasks (like translation, question answering, logical entailment, etc)? Mechanically, this works by either 1) training word embeddings and then using those embeddings as input feature representations for supervised models, or 2) treating the problem as a transfer learning problem, and fine-tune to a supervised task - similar to how you’d fine-tune a model trained on ImageNet by carrying over parameters, and then training on your new task. Even though the text we’re learning on is strictly speaking unsupervised (lacking a supervised label), we need to design a task on which we calculate gradients in order to train our representations. For unsupervised language modeling, that task is typically structured as predicting a word in a sequence given prior words in that sequence. Intuitively, in order for a model to do a good job at predicting the word that comes next in a sentence, it needs to have learned patterns about language, both on grammatical and semantic levels. A notable change recently has been the shift from learning unconditional word vectors (where the word’s representation is the same globally) to contextualized ones, where the representation of the word is dependent on the sentence context it’s found in. All the baselines discussed here are of this second type. 

The two main baselines that the BERT model compares itself to are OpenAI’s GPT, and Peters et al’s ELMo. The GPT model uses a self-attention-based Transformer architecture, going through each word in the sequence, and predicting the next word by calculating an attention-weighted representation of all prior words. (For those who aren’t familiar, attention works by multiplying a “query” vector with every word in a variable-length sequence, and then putting the outputs of those multiplications into a softmax operator, which inherently gets you a weighting scheme that adds to one). ELMo uses models that gather context in both directions, but in a fairly simple way: it learns one deep LSTM that goes from left to right,  predicting word k using words 0-k-1, and a second LSTM that goes from right to left, predicting word k using words k+1 onward. These two predictions are combined (literally: just summed together) to get a representation for the word at position k. 

https://i.imgur.com/2329e3L.png

BERT differs from prior work in this area in several small ways, but one primary one: instead of representing a word using only information from words before it, or a simple sum of prior information and subsequent information, it uses the full context from before and after the word in each of its multiple layers. It also uses an attention-based Transformer structure, but instead of incorporating just prior context, it pulls in information from the full sentence. To allow for a model that actually uses both directions of context at a time in its unsupervised prediction task, the authors of BERT slightly changed the nature of that task: it replaces the word being predicted with the “mask” token, so that even with multiple layers of context aggregation on both sides, the model doesn’t have any way of knowing what the token is. By contrast, if it weren’t masked, after the first layer of context aggregation, the representations of other words in the sequence would incorporate information about the predicted word k, making it trivial, if another layer were applied on top of that first one, for the model to directly have access to the value it’s trying to predict. This problem can either be solved by using multiple layers, each of which can only see prior context (like GPT), by learning fully separate L-R and R-L models, and combining them at the final layer (like ELMo) or by masking tokens, and predicting the value of the masked tokens using the full remainder of the context. 

This task design crucially allows for a multi-layered bidirectional architecture, and consequently a much richer representation of context in each word’s pre-trained representation. BERT demonstrates dramatic improvements over prior work when fine tuned on a small amount of supervised data, suggesting that this change added substantial value.

This paper shows exciting results on using Model-based RL for Atari. 

Model-based RL has shown impressive improvements in sample efficiency on Mujoco tasks ([Chua et. al, 2018](https://arxiv.org/abs/1805.12114)), so its nice to see that the sample efficiency improvements carry over to Pixel-based envs like Atari too. 

Specifically, the authors show that their model-based method can do well on several Atari games after training on only 100K env steps (400K frames with FrameSkip 4) which roughly corresponds to 2 hours of game play. They compare to SOTA model-free variants (Rainbow, PPO) after similar number of frames and show that the model-based version achieves much better scores. 

The overall training procedure has a very Dyna like flavor. The algorithm, termed SimPLe follows an iterative scheme of:
* Collect experience from the real environment using a policy (initialized to random). 
* Use this experience to train the world model (a next-step frame prediction model, and a reward prediction model). This amounts to supervised learning on `{(s, a) -> s’}` and `{(s, a) -> r}` pairs. 
* Generate rollouts using the world model, and learn a policy with these rollouts using PPO.

https://i.imgur.com/SZLmdME.png

**Countering distributional shift:**

A key issue when training models is compounding errors when doing multi-step rollouts. This is similar to the problem of making predictions with RNNs trained via teacher-forcing, and hence it's natural to leverage existing techniques from that literature. 

This paper uses one such technique: scheduled sampling, that is during training randomly replace some frames of the input by the prediction from the previous step. This seems like a natural way to make the model robust to slight distributional changes. 


**Commentary / possible future work:** 

* The paper evaluated only on 26 out of 60 Atari games in ALE. I would have really liked if the authors showed performance numbers on all the games even if they weren’t good.  
* Related: I suspect the method would not work well when the initial diversity of frames given by the random policy is not sufficient (ex. Sparse reward games like Montezuma’s revenge/Pitfall). Using sample efficient exploration algorithms to augment model learning would be really interesting. 
* The trained world-model is able to rollout only for 50 time-steps (compounding errors don't allow for longer rollouts), it might be worthwhile to explore models that can do long-horizon predictions [(TD-VAE?)](https://openreview.net/forum?id=S1x4ghC9tQ). 
* Apart from sample-efficiency gains, one reason I am excited about models is their potential ability to generalize to different tasks in the same environment. Benchmarking their generalization capability should thus be an exciting next step. 

Finally, props to authors for open-sourcing the code: [tensor2tensor/tensor2tensor/rl at master · tensorflow/tensor2tensor · GitHub](https://github.com/tensorflow/tensor2tensor/tree/master/tensor2tensor/rl) and providing detailed instructions to run. 

The paper discusses a number of extensions to the Skip-gram model previously proposed by Mikolov et al (citation [7] in the paper): which learns linear word embeddings that are particularly useful for analogical reasoning type tasks. The extensions proposed (namely, negative sampling and sub-sampling of high frequency words) enable extremely fast training of the model on large scale datasets. This also results in significantly improved performance as compared to previously proposed techniques based on neural networks. The authors also provide a method for training phrase level embeddings by slightly tweaking the original training algorithm. 

This paper proposes 3 improvements for the skip-gram model which allows for learning embeddings for words. The first improvement is subsampling frequent word, the second is the use of a simplified version of noise constrastive estimation (NCE) and finally they propose a method to learn idiomatic phrase embeddings. In all three cases the improvements are somewhat ad-hoc. In practice, both the subsampling and negative samples help to improve generalization substantially on an analogical reasoning task. The paper reviews related work and furthers the interesting topic of additive compositionality in embeddings. 

The article does not propose any explanation as to why the negative sampling produces better results than NCE which it is suppose to loosely approximate. In fact it doesn't explain why besides the obvious generalization gain the negative sampling scheme should be preferred to NCE since they achieve similar speeds. 

This paper models object detection as a regression problem for bounding
boxes and object class probabilities with a single pass through the CNN. The
main contribution is the idea of dividing the image into a 7x7 grid, and having
each cell predict a distribution over class labels as well as a bounding box
for the object whose center falls into it. It's much faster than R-CNN and
Fast R-CNN, as the additional step of extracting region proposals has been
removed.

## Strengths

- Works real-time. Base model runs at 45fps and a faster version goes up to
150fps, and they claim that it's more than twice as fast as other works on
real-time detection.

- End-to-end model; Localization and classification errors can be jointly
optimized.

- YOLO makes more localization errors and fewer background mistakes than
Fast R-CNN, so using YOLO to eliminate false background detections from
Fast R-CNN results in ~3% mAP gain (without much computational time as R-CNN
is much slower).

## Weaknesses / Notes

- Results fall short of state-of-the-art: 57.9% v/s 70.4% mAP (Faster R-CNN).

- Performs worse at detecting small objects, as at most one object per grid
cell can be detected.

This paper deals with an important problem where a deep classification system is made explainable. After the (continuing) success of Deep Networks, researchers are trying to open the blackbox and this work is one of the foremosts. The authors explored the strength of a deep learning method (vision-language model) to explain the performance of another deep learning model (image classification). The approach jointly predicts a class label and explains why it predicted so in natural language.

The paper starts with a very important differentiation between two basic schools of *explnation* systems - the *introspection* explanation system and the *justification* explanation system. The introspection system looks into the model to get an explanation (e.g., "This is a Western Grebe because filter 2 has a high activation..."). On the other hand, a justification system justifies the decision by producing sentence details on how visual evidence is compatible with the system output (e.g., "This is a Western Grebe because it has red eyes..."). The paper focuses on *justification* explanation system and proposes a novel one.

The authors argue that unlike a description of an image or a sentence defining a class (not necessarily in presence of an image), visual explanation, conditioned on an input image, provides much more of an explanatory text on why the image is classified as a certain category mentioning only image relevant features. The broad outline of the approach is given in Fig (2) of the paper.
https://i.imgur.com/tta2qDp.png
The first stage consists of a deep convolutional network for classification which generates a softmax distribution over the classes. As the task handles fine-grained bird species classification, it uses a compact bilinear feature representation known to work well for the fine-grained classification tasks. The second stage is a stacked LSTM which generates natural language sentences or explanations justifying the decision of the first stage. The first LSTM of the stack receives the previously generated word. The second LSTM receives the output of the first LSTM along with image features and predicted label distribution from the classification network. This LSTM produces the sequence of output words until an "end-of-sentence" token is generated. The intuition behind using predicted label distribution for explanation is that it would inform the explanation generation model which words and attributes are more likely to occur in the description.

Two kinds of losses are used for the second stage *i.e.*, the language model. The first one is termed as the *Relevance Loss* which is the typical sentence generation loss that is seen in literature. This is the sum of cross-entropy losses of the generated words with respect to the ground truth words. Its role is to optimize the alignment between generated and ground truth sentences. However, this loss is not very effective in producing sentences which include class discriminative information. class specificity is a global sentence property. This is illustrated with the following example - *whereas a sentence "This is an all black bird with a bright red eye" is class specific to a "Bronzed Cowbird", words and phrases in the sentence, such as "black" or "red eye" are less class discriminative on their own.* As a result, cross entropy loss on individual words turns out to be less effective in capturing the global sentence property of which class specifity is an example. The authors address this issue by proposing an addiitonal loss, termed as the *Discriminative Loss* which is based on a reinforcement learning paradigm. Before computing the loss, a sentence is sampled. The sentence is passed through a LSTM-based classification network whose task is to produce the ground truth category $C$ given only the sampled sentence. The reward for this operation is simply the probability of the ground truth category $C$ given only the sentence. The intuition is - for the model to produce an output with a large reward, the generated sentence must include enough information to classify the original image properly. The *Discriminative Loss* is the expectation of the negative of this reward and a wieghted linear combination of the two losses is optimized during training.

My experience in reinforcement learning is limited. However, I must say I did not quite get why is sampling of the sentences required (which called for the special algorithm for backpropagation). If the idea is to see whether a generated sentence can be used to get at the ground truth category, could the last internal state of one of the stacked LSTM not be used? It would have been better to get some more intution behind the sampling operation. Another thing which (is fairly obvious but still I felt) is missing is not mentioning the loss used in the fine grained classification network.

The experimentation is rigorous. The proposed method is compared with four different baseline and ablation models - description, definition, explanation-label, explanation-discriminative with different permutation and combinations of the presence of two types losses, class precition informations etc. Also the evaluation metrics measure different qualities of the generated exlanations, specifically image and class relevances. To measure image relevance METEOR/CIDEr scores of the generated sentences with the ground truth (image based) explanations are computed. On the other hand, to measure the class relevance, CIDEr scores with class definition (not necessarily based on the images from the dataset) sentences are computed. The proposed approach has continuously shown better performance than any of the baseline or ablation methods. I'd specifically mention about one experiment where the effect of class conditioning is studies (end of Sec 5.2). The finding is quite interesting as it shows that providing or not providing correct class information has drastic effect at the generated explanations. It is seen that giving incorrect class information makes the explanation model hallucinate colors or attributes which are not present in the image but are specific to the class. This raises the question whether it is worth giving the class information when the classifier is poor on the first hand? But, I think the answer lies in the observation that row 5 (with class prediction information) in table 1 is always better than row 4 (no class prediction information). Since, row 5 is better than row 4, this means the classifier is also reasonable and this in turn implies that end-to-end training can improve all the stages of a pipeline which ultimately improves the overall performance of the system too!

In summary, the paper is a very good first step to explain intelligent systems and should encourage a lot more effort in this direction.

This paper builds on top of a bunch of existing ideas for building neural conversational agents so as to control against generic and repetitive responses. 

Their model is the sequence-to-sequence model with attention (Bahdanau et al.), first trained with the usual MLE loss and fine-tuned with policy gradients to optimize for specific conversational properties. Specifically, they define 3 rewards:

1. Ease of answering — Measured as the likelihood of responding to a query with a list of hand-picked dull responses (more negative log likelihood is higher reward). 
2. Information flow — Consecutive responses from the same agent (person) should have different information, measured as negative of log cosine distance (more negative is better).
3. Semantic coherence — Mutual information between source and target (the response should make sense wrt query). $P(a|q) + P(q|a)$ where a is answer, q is question.

The model is pre-trained with the usual supervised objective function, taking source as concatenation of two previous utterances. Then they have two stages of policy gradient training, first with just a mutual information reward and then with a combination of all three. The policy network (sequence-to-sequence model) produces a probability distribution over actions (responses) given state (previous utterances). To estimate the gradient in an iteration, the network is frozen and responses are sampled from the model, the rewards for which are then averaged and gradients are computed for first L tokens of response using MLE and remaining T-L tokens with policy gradients, with L being gradually annealed to zero (moving towards just the long-term reward).

Evaluation is done based on length of dialogue, diversity (distinct unigram, bigrams) and human studies on

1. Which of two outputs has better quality (single turn)
2. Which of two outputs is easier to respond to, and
3. Which of two conversations have better quality (multi turn).

## Strengths

- Interesting results
    - Avoids generic responses
    - 'Ease of responding' reward encourages responses to be question-like
- Adding in hand-engineereed approximate reward functions based on conversational properties and using those to fine-tune a pre-trained network using policy gradients is neat.
- Policy gradient training also encourages two dialogue agents to interact with each other and explore the complete action space (space of responses), which seems desirable to identify modes of the distribution and not converge on a single, high-scoring, generic response.

## Weaknesses / Notes

- Evaluating conversational agents is hard. BLEU / perplexity are intentionally avoided as they don't necessarily reward desirable conversational properties.

#### Introduction

* Introduces a new global log-bilinear regression model which combines the benefits of both global matrix factorization and local context window methods.


#### Global Matrix Factorization Methods

* Decompose large matrices into low-rank approximations.
* eg - Latent Semantic Analysis (LSA)

##### Limitations

* Poor performance on word analogy task
* Frequent words contribute disproportionately high to the similarity measure.

#### Shallow, Local Context-Based Window Methods

* Learn word representations using adjacent words.
* eg - Continous bag-of-words (CBOW) model and skip-gram model.

##### Limitations

* Since they do not operate directly on the global co-occurrence counts, they can not utilise the statistics of the corpus effectively.


#### GloVe Model

* To capture the relationship between words $i$ and $j$, word vector models should use ratios of co-occurene probabilites (with other words $k$) instead of using raw probabilites themselves.
* In most general form:
     * $F(w_{i}, w_{j}, w_{k}^{~} ) = P_{ik}/P_{jk}$
* We want $F$ to encode information in the vector space (which have a linear structure), so we can restrict to the difference of $w_{i}$ and $w_{j}$
    * $F(w_{i} - w_{j}, w_{k}^{~} ) = P_{ik}/P_{jk}$
* Since right hand side is a scalar and left hand side is a vector, we take dot product of the arguments.
    * $F( (w_{i} - w_{j})^{T}, w_{k}^{~} ) = P_{ik}/P_{jk}$
* *F* should be invariant to order of the word pair $i$ and $j$.
    * $F(w_{i}^{T}w_{k}^{~}) = P_{ik}$
* Doing further simplifications and optimisations (refer paper), we get cost function, 
    * $J = \sum_{\text{over all i, j pairs in the vocabulary}}[w_{i}^{T}w_{k}^{~} + b_{i} + b_{k}^{~} - log(X_{ik})]^{2}$
    * $f$ is a weighing function.
    * $f(x) = min((x/x_{max})^{\alpha}, 1)$
    * Typical values, $x_{\max} = 100$ and $\alpha = 3/4$
    * *b* are the bias terms.

##### Complexity

* Depends on a number of non-zero elements in the input matrix.
* Upper bound by the square of vocabulary size
* Since for shallow window-based approaches, complexity depends on $|C|$ (size of the corpus), tighter bounds are needed.
* By modelling number of co-occurrences of words as power law function of frequency rank, the complexity can be shown to be proportional to $|C|^{0.8}$

#### Evaluation

##### Tasks

* Word Analogies
    * a is to b as c is to ___?
    * Both semantic and syntactic pairs
    * Find closest d to $w_{b} - w_{c} + w_{a}$ (using cosine similarity)

* Word Similarity
* Named Entity Recognition

##### Datasets

* Wikipedia Dumps - 2010 and 2014
* Gigaword5
* Combination of Gigaword5 and Wikipedia2014
* CommonCrawl
* 400,000 most frequent words considered from the corpus.

##### Hyperparameters

* Size of context window.
* Whether to distinguish left context from right context.
* $f$ - Word pairs that are $d$ words apart contribute $1/d$ to the total count.
* $xmax = 100$
* $\alpha = 3/4$
* AdaGrad update

##### Models Compared With

* Singular Value Decomposition
* Continous Bag-Of-Words
* Skip-Gram

##### Results

* Glove outperforms all other models significantly.
* Diminishing returns for vectors larger than 200 dimensions.
* Small and asymmetric context windows (context window only to the left) works better for syntactic tasks.
* Long and symmetric context windows (context window to both the sides) works better for semantic tasks.
* Syntactic task benefited from larger corpus though semantic task performed better with Wikipedia instead of Gigaword5 probably due to the comprehensiveness of Wikipedia and slightly outdated nature of Gigaword5.
* Word2vec’s performance decreases if the number of negative samples increases beyond about 10.
* For the same corpus, vocabulary, and window size GloVe consistently achieves better results, faster.

This paper starts by introducing a trick to reduce the variance of stochastic gradient variational Bayes (SGVB) estimators. In neural networks, SGVB consists in learning a variational (e.g. diagonal Gaussian) posterior over the weights and biases of neural networks, through a procedure that (for the most part) alternates between adding (Gaussian) noise to the model's parameters and then performing a model update with backprop. 

The authors present a local reparameterization trick, which exploits the fact that the Gaussian noise added into the weights could instead be added directly into the pre-activation (i.e. before the activation fonction) vectors during forward propagation. This is due to the fact that computing the pre-activation is a linear operation, thus noise at that level is also Gaussian. The advantage of doing so is that, in the context of minibatch training, one can efficiently then add independent noise to the pre-activation vectors for each example of the minibatch. The nature of the local reparameterization trick implies that this is equivalent to using one corrupted version of the weights for each example in the minibatch, something that wouldn't be practical computationally otherwise. This is in fact why, in normal SGVB, previous work would normally use a single corrupted version of the weights for all the minibatch. 

The authors demonstrate that using the local reparameterization trick yields stochastic gradients with lower variance, which should improve the speed of convergence.

Then, the authors demonstrate that the Gaussian version of dropout (one that uses multiplicative Gaussian noise, instead of 0-1 masking noise) can be seen as the local reparameterization trick version of a SGVB objective, with some specific prior and variational posterior. In this SGVB view of Gaussian dropout, the dropout rate is an hyper-parameter of this prior, which can now be tuned by optimizing the variational lower bound of SGVB. In other words, we now have a method to also train the dropout rate! Moreover, it becomes possible to tune an individual dropout rate parameter for each layer, or even each parameter of the model.

Experiments on MNIST confirm that tuning that parameter works and allows to reach good performance of various network sizes, compared to using a default dropout rate. 


##### My two cents

This is another thought provoking connection between Bayesian learning and dropout. Indeed, while Deep GPs have allowed to make a Bayesian connection with regular (binary) dropout learning \cite{journals/corr/GalG15}, this paper sheds light on a neat Bayesian connection for the Gaussian version of dropout. This is great, because it suggests that Gaussian dropout training is another legit way of modeling uncertainty in the parameters of neural networks. It's also nice that that connection also yielded a method for tuning the dropout rate automatically.

I hope future work (by the authors or by others) can evaluate the quality of the corresponding variational posterior in terms of estimating uncertainty in the network and, in particular, in obtaining calibrated output probabilities.

Little detail: I couldn't figure out whether the authors tuned a single dropout rate for the whole network, or used many rates, for instance one per parameter, as they suggest can be done.

This paper presents a variational approach to the maximisation of mutual information in the context of a reinforcement learning agent. Mutual information in this context can provide a learning signal to the agent that is "intrinsically motivated", because it relies solely on the agent's state/beliefs and does not require from the ("outside") user an explicit definition of rewards.

Specifically, the learning objective, for a current state s, is the mutual information between the sequence of K actions a proposed by an exploration distribution $w(a|s)$ and the final state s' of the agent after performing these actions. To understand what the properties of this objective, it is useful to consider the form of this mutual information as a difference of conditional entropies:

$$I(a,s'|s) = H(a|s) - H(a|s',s)$$

Where $I(.|.)$ is the (conditional) mutual information and $H(.|.)$ is the (conditional) entropy. This objective thus asks that the agent find an exploration distribution that explores as much as possible (i.e. has high $H(a|s)$ entropy) but is such that these actions have predictable consequences (i.e. lead to predictable state s' so that $H(a|s',s)$ is low). So one could think of the agent as trying to learn to have control of as much of the environment as possible, thus this objective has also been coined as "empowerment".

The main contribution of this work is to show how to train, on a large scale (i.e. larger state space and action space) with this objective, using neural networks. They build on a variational lower bound on the mutual information and then derive from it a stochastic variational training algorithm for it. The procedure has 3 components: the exploration distribution $w(a|s)$, the environment $p(s'|s,a)$ (can be thought as an encoder, but which isn't modeled and is only interacted with/sampled from) and the planning model $p(a|s',s)$ (which is modeled and can be thought of as a decoder). The main technical contribution is in how to update the exploration distribution (see section 4.2.2 for the technical details).

This approach exploits neural networks of various forms. Neural autoregressive generative models are also used as models for the exploration distribution as well as the decoder or planning distribution. Interestingly, the framework allows to also learn the state representation s as a function of some "raw" representation x of states. For raw states corresponding to images (e.g. the pixels of the screen image in a game), CNNs are used.

## Introduction

* [Link to Paper](http://arxiv.org/pdf/1412.6071v4.pdf)
* Spatial pooling layers are building blocks for Convolutional Neural Networks (CNNs).
* Input to pooling operation is a $N_{in}$ x $N_{in}$ matrix and output is a smaller matrix $N_{out}$ x $N_{out}$.
* Pooling operation divides $N_{in}$ x $N_{in}$ square into $N^2_{out}$ pooling regions $P_{i, j}$.
* $P_{i, j}$ ⊂ $\{1, 2, . . . , N_{in}\}$ $\forall$ $(i, j) \in \{1, . . . , N_{out} \}^2$

## MP2

* Refers to 2x2 max-pooling layer.
* Popular choice for max-pooling operation.

### Advantages of MP2
* Fast.
* Quickly reduces the size of the hidden layer.
* Encodes a degree of invariance with respect to translations and elastic distortions.

### Issues with MP2
* Disjoint nature of pooling regions.
* Since size decreases rapidly, stacks of back-to-back CNNs are needed to build deep networks.

## FMP

* Reduces the spatial size of the image by a factor of *α*, where *α ∈ (1, 2)*.
* Introduces randomness in terms of choice of pooling region.
* Pooling regions can be chosen in a *random* or *pseudorandom* manner.
* Pooling regions can be *disjoint* or *overlapping*.

## Generating Pooling Regions

* Let $a_i$ and $b_i$ be 2 increasing sequences of integers, starting at 1 and ending at $N_{in}$.
* Increments are either 1 or 2.
* For *disjoint regions, $P = [a_{i−1}, a_{i − 1}] × [b_{j−1}, b_{j − 1}]$
* For *overlapping regions, $P = [a_{i−1}, a_i] × [b_{j−1}, b_j 1]$
* Pooling regions can be generated *randomly* by choosing the increment randomly at each step.
* To generate pooling regions in a *peusdorandom* manner, choose $a_i$ = ceil($\alpha | (i+u))$, where $\alpha \in (1, 2)$ with some $u \in (0, 1)$.
* Each FMP layer uses a different pair of sequence.
* An FMP network can be thought of as an ensemble of similar networks, with each different pooling-region configuration defining a different member of the ensemble.

## Observations  

* *Random* FMP is good on its own but may underfit when combined with dropout or training data augmentation.  
* *Pseudorandom* approach generates more stable pooling regions.   
* *Overlapping* FMP performs better than *disjoint* FMP.    

## Weakness

* No justification is provided for the observations mentioned above.
* It needs to be seen how performance is affected if the pooling layer in architectures like GoogLeNet.

# Deep Convolutional Generative Adversarial Nets

## Introduction

* The paper presents Deep Convolutional Generative Adversarial Nets (DCGAN) - a topologically constrained variant of conditional GAN.
* [Link to the paper](https://arxiv.org/abs/1511.06434)

## Benefits

* Stable to train
* Very useful to learn unsupervised image representations.

## Model

* GANs difficult to scale using CNNs.
* Paper proposes following changes to GANs:
    * Replace any pooling layers with strided convolutions (for discriminator) and fractional strided convolutions (for generators).
    * Remove fully connected hidden layers.
    * Use batch normalisation in both generator (all layers except output layer) and discriminator (all layers except input layer).
    * Use LeakyReLU in all layers of the discriminator.
    * Use ReLU activation in all layers of the generator (except output layer which uses Tanh).

## Datasets
    
* Large-Scale Scene Understanding.
* Imagenet-1K.
* Faces dataset.

## Hyperparameters

* Minibatch SGD with minibatch size of 128.
* Weights initialized with 0 centered Normal distribution with standard deviation = 0.02
* Adam    Optimizer
* Slope of leak = 0.2 for LeakyReLU.
* Learning rate = 0.0002, β1 = 0.5

## Observations

* Large-Scale Scene Understanding data
    * Demonstrates that model scales with more data and higher resolution generation.
    * Even though it is unlikely that model would have memorized images (due to low learning rate of minibatch SGD).
* Classifying CIFAR-10 dataset
    * Features
        * Train in Imagenet-1K and test on CIFAR-10.
        * Max pool discriminator's convolutional features (from all layers) to get 4x4 spatial grids.
        * Flatten and concatenate to get a 28672-dimensional vector.
        * Linear L2-SVM classifier trained over the feature vector.
    * 82.8% accuracy, outperforms K-means (80.6%)
* Street View House Number Classifier
    * Similar pipeline as CIFAR-10
    * 22.48% test error.
* The paper contains many examples of images generated by final and intermediate layers of the network.
* Images in the latent space do not show sharp transitions indicating that network did not memorize images.
* DCGAN can learn an interesting hierarchy of features.
* Networks seems to have some success in disentangling image representation from object representation.
* Vector arithmetic can be performed on the Z vectors corresponding to the face samples to get results like `smiling woman - normal woman + normal man = smiling man` visually.

This paper proposes a variant of Neural Turing Machine (NTM) for meta-learning or "learning to learn", in the specific context of few-shot learning (i.e. learning from few examples). Specifically, the proposed model is trained to ingest as input a training set of examples and improve its output predictions as examples are processed, in a purely feed-forward way. This is a form of meta-learning because the model is trained so that its forward pass effectively executes a form of "learning" from the examples it is fed as input.

During training, the model is fed multiples sequences (referred to as episodes) of labeled examples $({\bf x}_1, {\rm null}), ({\bf x}_2, y_1), \dots, ({\bf x}_T, y_{T-1})$, where $T$ is the size of the episode. For instance, if the model is trained to learn how to do 5-class classification from 10 examples per class, $T$ would be $5 \times 10 = 50$. Mainly, the paper presents experiments on the Omniglot dataset, which has 1623 classes. In these experiments, classes are separated into 1200 "training classes" and 423 "test classes", and each episode is generated by randomly selecting 5 classes (each assigned some arbitrary vector representation, e.g. a  one-hot vector that is consistent within the episode, but not across episodes) and constructing a randomly ordered sequence of 50 examples from within the chosen 5 classes. Moreover, the correct label $y_t$ of a given input ${\bf x}_t$ is always provided only at the next time step, but the model is trained to be good at its prediction of the label of ${\bf x}_t$ at the current time step. This is akin to the scenario of online learning on a stream of examples, where the label of an example is revealed only once the model has made a prediction.

The proposed NTM is different from the original NTM of Alex Graves, mostly in how it writes into its memory. The authors propose to focus writing to either the least recently used memory location or the most recently used memory location. Moreover, the least recently used memory location is reset to zero before every write (an operation that seems to be ignored when backpropagating gradients).

Intuitively, the proposed NTM should learn a strategy by which, given a new input, it looks into its memory for information from other examples earlier in the episode (perhaps similarly to what a nearest neighbor classifier would do) to predict the class of the new input.

The paper presents experiments in learning to do multiclass classification on the Omniglot dataset and regression based on functions synthetically generated by a GP. The highlights are that:

1. The proposed model performs much better than an LSTM and better than an NTM with the original write mechanism of Alex Graves (for classification).
2. The proposed model even performs better than a 1st nearest neighbor classifier.
3. The proposed model is even shown to outperform human performance, for the 5-class scenario.
4. The proposed model has decent performance on the regression task, compared to GP predictions using the groundtruth kernel.

**My two cents**

This is probably one of my favorite ICML 2016 papers. I really think meta-learning is a problem that deserves more attention, and this paper presents both an interesting proposal for how to do it and an interesting empirical investigation of it. 

Much like previous work [\[1\]][1] [\[2\]][2], learning is based on automatically generating a meta-learning training set. This is clever I think, since a very large number of such "meta-learning" examples (the episodes) can be constructed, thus transforming what is normally a "small data problem" (few shot learning) into a "big data problem", for which deep learning is more effective.

I'm particularly impressed by how the proposed model outperforms a 1-nearest neighbor classifier. That said, the proposed NTM actually performs 4 reads at each time step, which suggests that a fairer comparison might be with a 4-nearest neighbor classifier. I do wonder how this baseline would compare.

I'm also impressed with the observation that the proposed model surpassed humans.

The paper also proposes to use 5-letter words to describe classes, instead of one-hot vectors. The motivation is that this should make it easier for the model to scale to much more than 5 classes. However, I don't entirely follow the logic as to why one-hot vectors are problematic. In fact, I would think that arbitrarily assigning 5-letter words to classes would instead imply some similarity between classes that share letters that is arbitrary and doesn't reflect true class similarity. 

Also, while I find it encouraging that the performance for regression of the proposed model is decent, I'm curious about how it would compare with a GP approach that incrementally learns the kernel's hyper-parameter (instead of using the groundtruth values, which makes this baseline unrealistically strong).

Finally, I'm still not 100% sure how exactly the NTM is able to implement the type of feed-forward inference I'd expect to be required. I would expect it to learn a memory representation of examples that combines information from the input vector ${\bf x}_t$ *and* its label $y_t$. However, since the label of an input is presented at the following time step in an episode, it is not intuitive to me then how the read/write mechanisms are able to deal with this misalignment. My only guess is that since the controller is an LSTM, then it can somehow remember ${\bf x}_t$ until it gets $y_t$ and appropriately include the combined information into the memory. This could be supported by the fact that using a non-recurrent feed-forward controller is much worse than using an LSTM controller. But I'm not 100% sure of this either.

All the above being said, this is still a really great paper, which I hope will help stimulate more research on meta-learning. Hopefully code for this paper can eventually be released, which would help in popularizing the topic.

[1]: http://snowedin.net/tmp/Hochreiter2001.pdf
[2]: http://www.thespermwhale.com/jaseweston/ram/papers/paper_16.pdf

Everyone has been thinking about how to apply GANs to discrete sequence data for the past year or so. This paper presents the model that I would guess most people thought of as the first-thing-to-try:

1. Build a recurrent generator model which samples from its softmax outputs at each timestep.
2. Pass sampled sequences to a recurrent discriminator model which distinguishes between sampled sequences and real-data sequences.
3. Train the discriminator under the standard GAN loss.
4. Train the generator with a REINFORCE (policy gradient) objective, where each trajectory is assigned a single episodic reward: the score assigned to the generated sequence by the discriminator.

Sounds hacky, right? We're learning a generator with a high-variance model-free reinforcement learning algorithm, in a very seriously non-stationary environment. (Here the "environment" is a discriminator being jointly learned with the generator.)

There's just one trick in this paper on top of that setup: for non-terminal states, the reward is defined as the *expectation* of the discriminator score after stochastically generating from that state forward. To restate using standard (somewhat sloppy) RL syntax, in different terms than the paper: (under stochastic sequential policy $\pi$, with current state $s_t$, trajectory $\tau_{1:T}$ and discriminator $D(\tau)$)

$$r_t = \mathbb E_{\tau_{t+1:T} \sim \pi(s_t)} \left[ D(\tau_{1:T}) \right]$$

The rewards are estimated via Monte Carlo — i.e., just take the mean of $N$ rollouts from each intermediate state. They claim this helps to reduce variance. That makes intuitive sense, but I don't see any results in the paper demonstrating the effect of varying $N$.

---

Yep, so it turns out that this sort of works.. with a big caveat:

## The big caveat

Graph from appendix:

![](https://www.dropbox.com/s/5fqh6my63sgv5y4/Bildschirmfoto%202016-09-27%20um%2021.34.44.png?raw=1)

SeqGANs don't work without supervised pretraining. Makes sense — with a cold start, the generator just samples a bunch of nonsense and the discriminator overfits. Both the generator and discriminator are pretrained on supervised data in this paper (see Algorithm 1).

I think it must be possible to overcome this with the proper training tricks and enough sweat. But it's probably more worth our time to address the fundamental problem here of developing better RL for structured prediction tasks.

This paper presents a recurrent neural network architecture in which some of the recurrent weights dynamically change during the forward pass, using a hebbian-like rule. They correspond to the matrices $A(t)$ in the figure below:

![Fast weights RNN figure](http://i.imgur.com/DCznSf4.png)

These weights $A(t)$ are referred to as *fast weights*. Comparatively, the recurrent weights $W$ are referred to as slow weights, since they are only changing due to normal training and are otherwise kept constant at test time.

More specifically, the proposed fast weights RNN compute a series of hidden states $h(t)$ over time steps $t$, but, unlike regular RNNs, the transition from $h(t)$ to $h(t+1)$ consists of multiple ($S$) recurrent layers $h_1(t+1), \dots, h_{S-1}(t+1), h_S(t+1)$, defined as follows:

$$h_{s+1}(t+1) = f(W h(t) + C x(t) + A(t) h_s(t+1))$$

where $f$ is an element-wise non-linearity such as the ReLU activation. The next hidden state $h(t+1)$ is simply defined as the last "inner loop" hidden state $h_S(t+1)$, before moving to the next time step. 

As for the fast weights $A(t)$, they too change between time steps, using the hebbian-like rule:

$$A(t+1) = \lambda A(t) + \eta h(t) h(t)^T$$

where $\lambda$ acts as a decay rate (to partially forget some of what's in the past)  and $\eta$ as the fast weight's "learning rate" (not to be confused with the learning rate used during backprop). Thus, the role played by the fast weights is to rapidly adjust to the recent hidden states and remember the recent past.

In fact, the authors show an explicit relation between these fast weights and memory-augmented architectures that have recently been popular. Indeed, by recursively applying and expending the equation for the fast weights, one obtains

$$A(t) = \eta \sum_{\tau = 1}^{\tau = t-1}\lambda^{t-\tau-1} h(\tau) h(\tau)^T$$

*(note the difference with Equation 3 of the paper... I think there was a typo)* which implies that when computing the $A(t) h_s(t+1)$ term in the expression to go from $h_s(t+1)$ to $h_{s+1}(t+1)$, this term actually corresponds to

$$A(t) h_s(t+1) = \eta \sum_{\tau =1}^{\tau = t-1} \lambda^{t-\tau-1} h(\tau) (h(\tau)^T h_s(t+1))$$

i.e. $A(t) h_s(t+1)$ is a weighted sum of all previous hidden states $h(\tau)$, with each hidden states weighted by an "attention weight" $h(\tau)^T h_s(t+1)$. The difference with many recent memory-augmented architectures is thus that the attention weights aren't computed using a softmax non-linearity.

Experimentally, they find it beneficial to use [layer normalization](https://arxiv.org/abs/1607.06450). Good values for $\eta$ and $\lambda$ seem to be 0.5 and 0.9 respectively. I'm not 100% sure, but I also understand that using $S=1$, i.e. using the fast weights only once per time steps, was usually found to be optimal. Also see Figure 3 for the architecture used on the image classification datasets, which is slightly more involved.

The authors present a series 4 experiments, comparing with regular RNNs (IRNNs, which are RNNs with ReLU units and whose recurrent weights are initialized to a scaled identity matrix) and LSTMs (as well as an associative LSTM for a synthetic associative retrieval task and ConvNets for the two image datasets). Generally, experiments illustrate that the fast weights RNN tends to train faster (in number of updates) and better than the other recurrent architectures. Surprisingly, the fast weights RNN can even be competitive with a ConvNet on the two image classification benchmarks, where the RNN traverses glimpses from the image using a fixed policy.

**My two cents**

This is a very thought provoking paper which, based on the comparison with LSTMs, suggests that fast weights RNNs might be a very good alternative. I'd be quite curious to see what would happen if one was to replace LSTMs with them in the myriad of papers using LSTMs (e.g. all the Seq2Seq work). Intuitively, LSTMs seem to be able to do more than just attending to the recent past. But, for a given task, if one was to observe that fast weights RNNs are competitive to LSTMs, it would suggests that the LSTM isn't doing something that much more complex. So it would be interesting to determine what are the tasks where the extra capacity of an LSTM is actually valuable and exploitable. Hopefully the authors will release some code, to facilitate this exploration. 

The discussion at the end of Section 3 on how exploiting the "memory augmented" view of fast weights is useful to allow the use of minibatches is interesting. However, it also suggests that computations in the fast weights RNN scales quadratically with the sequence size (since in this view, the RNN technically must attend to all previous hidden states, since the beginning of the sequence). This is something to keep in mind, if one was to consider applying this to very long sequences (i.e. much longer than the hidden state dimensionality).

Also, I don't quite get the argument that the "memory augmented" view of fast weights is more amenable to mini-batch training. I understand that having an explicit weight matrix $A(t)$ for each minibatch sequence complicates things. However, in the memory augmented view, we also have a "memory matrix" that is different for each sequence, and yet we can handle that fine. The problem I can imagine is that storing a *sequence of arbitrary weight matrices* for each sequence might be storage demanding (and thus perhaps make it impossible to store a forward/backward pass for more than one sequence at a time), while the implicit memory matrix only requires appending a new row at each time step. Perhaps the argument to be made here is more that there's already mini-batch compatible code out there for dealing with the use of a memory matrix of stored previous memory states.

This work strikes some (partial) resemblance to other recent work, which may serve as food for thought here. The use of possibly multiple computation layers between time steps reminds me of [Adaptive Computation Time (ACT) RNN]( http://www.shortscience.org/paper?bibtexKey=journals/corr/Graves16). Also, expressing a backpropable architecture that involves updates to weights (here, hebbian-like updates) reminds me of recent work that does backprop through the updates of a gradient descent procedure (for instance as in [this work]( http://www.shortscience.org/paper?bibtexKey=conf/icml/MaclaurinDA15)). 

Finally, while I was familiar with the notion of fast weights from the work on [Using Fast Weights to Improve Persistent Contrastive Divergence](http://people.ee.duke.edu/~lcarin/FastGibbsMixing.pdf), I didn't realize that this concept dated as far back as the late 80s. So, for young researchers out there looking for inspiration for research ideas, this paper confirms that looking at the older neural network literature for inspiration is probably a very good strategy :-)

To sum up, this is really nice work, and I'm looking forward to the NIPS 2016 oral presentation of it!

This paper presents a per-frame image-to-image translation system enabling copying of a motion of a person from a source video to a target person. For example, a source video might be a professional dancer performing complicated moves, while the target person is you. By utilizing this approach, it is possible to generate a video of you dancing as a professional. Check the authors' [video](https://www.youtube.com/watch?v=PCBTZh41Ris) for the visual explanation.

**Data preparation**

The authors have manually recorded high-resolution video ( at 120fps ) of a person performing various random moves. The video is further decomposed to frames, and person's pose keypoints (body joints, hands, face) are extracted for each frame. These keypoints are further connected to form a person stick figure. In practice, pose estimation is performed by open source project [OpenPose](https://github.com/CMU-Perceptual-Computing-Lab/openpose).

**Training**
https://i.imgur.com/VZCXZMa.png

Once the data is prepared the training is performed in two stages:
1. **Training pix2pixHD model with temporal smoothing**.
    
   The core model is an original [pix2pixHD](https://tcwang0509.github.io/pix2pixHD/)[1] model with temporal smoothing. Specifically, if we were to use vanilla pix2pixHD, the input to the model would be a stick person image, and the target is the person's image corresponding to the pose. The network's objective would be $min_{G} (Loss1 + Loss2 + Loss3)$, where: 

 - $Loss1 = max_{D_1, D_2, D_3} \sum_{k=1,2,3} \alpha_{GAN}(G, D_k)$ is adverserial loss;
 
 - $Loss2 = \lambda_{FM} \sum_{k=1,2,3} \alpha_{FM}(G,D_k)$ is feature matching loss;

  - $Loss3 = \lambda_{VGG}\alpha_{VGG}(G(x),y)]$ is VGG perceptual loss.

 However, this objective does not account for the fact that we want to generate video composed of frames that are temporally coherent. The authors propose to ensure *temporal smoothing* between adjacent frames by including pose, corresponding image, and generated image from the previous step (zero image for the first frame) as shown in the figure below:

  https://i.imgur.com/0NSeBVt.png

   Since the generated output $G(x_t; G(x_{t-1}))$ at time step $t$ is now conditioned on the previously generated frame $G(x_{t-1})$ as well as current stick image $x_t$, better temporal consistency is ensured. Consequently, the discriminator is now trying to determine both correct generation, as well as temporal consitency for a fake sequence $[x_{t-1}; x_t; G(x_{t-1}), G(x_t)]$.

2. **Training FaceGAN model**.

   https://i.imgur.com/mV1xuMi.png
  
   In order to improve face generation, the authors propose to use specialized FaceGAN. In practice, this is another smaller pix2pixHD model (with a global generator only, instead of local+global) which is fed with a cropped face area of a stick image and cropped face area of a corresponding generated image (from previous step 1) and aims to generate a residual which is added to the previously generated full image. 

**Testing**

 During testing, we extract frames from the input video, obtain pose stick image for each frame, normalize the stick pose image and feed it to pix2pixHD (with temporal consistency) and, further, to FaceGAN to produce final generated image with improved face features. Normalization is needed to capture possible pose variation between a source and a target input video.

**Remarks**

While this method produces a visually appealing result, it is not perfect. The are several reasons for being so:

 1. *Quality of a pose stick image*: if the pose detector "misses" the keypoint, the generator might have difficulties to generate a properly rendered image;
 2. *Motion blur*: motion blur causes pose detector to miss keypoints;
 3. *Severe scale change*: if source person is very far, keypoint detector might fail to detect proper keypoints.

Among video rendering challenges, the authors mention self-occlusion, cloth texture generation, video jittering (training-test motion mismatch).

References:

[1] "High-Resolution Image Synthesis and Semantic Manipulation with Conditional GANs"

**TL;DR**: Rearranging the terms in Maximum Mean Discrepancy yields a much better loss function for the discriminator of Generative Adversarial Nets.

**Keywords**: Generative adversarial nets, Maximum Mean Discrepancy, spectral normalization, convolutional neural networks, Gaussian kernel, local stability.

**Summary**

Generative adversarial nets (GANs) are widely used to learn the data sampling process and are notoriously difficult to train. The training of GANs may be improved from three aspects: loss function, network architecture, and training process.

This study focuses on a loss function called the Maximum Mean Discrepancy (MMD), defined as:
$$
MMD^2(P_X,P_G)=\mathbb{E}_{P_X}k_{D}(x,x')+\mathbb{E}_{P_G}k_{D}(y,y')-2\mathbb{E}_{P_X,P_G}k_{D}(x,y)
$$
where $G,D$ are the generator and discriminator networks, $x,x'$ are real samples, $y,y'$ are generated samples, $k_D=k\circ D$ is a learned kernel that calculates the similariy between two samples. Overall, MMD calculates the distance between the real and the generated sample distributions. Thus, traditionally, the generator is trained to minimize $L_G=MMD^2(P_X,P_G)$, while the discriminator minimizes $L_D=-MMD^2(P_X,P_G)$.

This study makes three contributions:
-  It argues that $L_D$ encourages the discriminator to ignores the fine details in real data. By minimizing $L_D$, $D$ attempts to maximize $\mathbb{E}_{P_X}k_{D}(x,x')$, the similarity between real samples scores. Thus, $D$ has to focus on common features shared by real samples rather than fine details that separate them. This may slow down training. Instead, a repulsive loss is proposed, with no additional computational cost to MMD:
$$
L_D^{rep}=\mathbb{E}_{P_X}k_{D}(x,x')-\mathbb{E}_{P_G}k_{D}(y,y')
$$
- Inspired by the hinge loss, this study proposes a bounded Gaussian kernel for the discriminator to facilitate stable training of MMD-GAN.
- The spectral normalization method divides the weight matrix at each layer by its spectral norm to enforce that each layer is Lipschitz continuous. This study proposes a simple method to calculate the spectral norm of a convolutional kernel.

The results show the efficiency of proposed methods on CIFAR-10, STL-10, CelebA and LSUN-bedroom datasets. In Appendix, we prove that MMD-GAN training using gradient method is locally exponentially stable (a property that the Wasserstein loss does not have), and show that the repulsive loss works well with gradient penalty. 

The paper has been accepted at ICLR 2019 ([OpenReview link](https://openreview.net/forum?id=HygjqjR9Km)). The code is available at [GitHub link](https://github.com/richardwth/MMD-GAN). 

This paper presents an approach to initialize a neural network from the parameters of a smaller and previously trained neural network. This is effectively done by increasing the size (in width and/or depth) of the previously trained neural network, in such of a way that the function represented by the network doesn't change (i.e. the output of the larger neural network is still the same). The motivation here is that initializing larger neural networks in this way allows to accelerate their training, since at initialization the neural network will already be quite good.

In a nutshell, neural networks are made wider by adding several copies (selected randomly) of the same hidden units to the hidden layer, for each hidden layer. To ensure that the neural network output remains the same, each incoming connection weight must also be divided by the number of replicas that unit is connected to in the previous layer. If not training using dropout, it is also recommended to add some noise to this initialization, in order to break its initial symmetry (though this will actually break the property that the network's output is the same). As for making a deeper network, layers are added by initializing them to be the identity function. For ReLU units, this is achieved using an identity matrix as the connection weight matrix. For units based on sigmoid or tanh activations, unfortunately it isn't possible to add such identity layers.

In their experiments on ImageNet, the authors show that this initialization allows them to train larger networks faster than if trained from random initialization. More importantly, they were able to outperform their previous validation set ImageNet accuracy by initializing a very large network from their best Inception network.