Feature Pyramid Networks (FPNs) build on top of the state-of-the-art implementation for object detection net - Faster RCNN. Faster RCNN faces a major problem in training for scale-invariance as the computations  can be memory-intensive and extremely slow. So FRCNN only applies multi-scale approach while testing. 

On the other hand, feature pyramids were mainstream when hand-generated features were used -primarily to counter scale-invariance. Feature pyramids are collections of features computed at multi-scale versions of the same image. Improving on a similar idea presented in *DeepMask*, FPN brings back feature pyramids using different feature maps of conv layers with differing spatial resolutions with predictiosn happening on all levels of pyramid. Using feature maps directly as it is, would be tough as initial layers tend to contain lower level representations and poor semantics but good localisation whereas deeper layers tend to constitute higher level representations with rich semantics but suffer poor localisation due to multiple subsampling.

##### Methodology

FPN can be used with any normal conv architecture, that's used for classification. In such an architecture all layers have progressively decreasing spatial resolutions (say C1, C2,..C5). FPN would now take C5 and convolve with 1x1 kernel to reduce filters to give P5. Next, P5 is upsampled and merged it to C4 (C4 is convolved with 1x1 kernel to decrease filter size in order to match that of upsampled P5) by adding element wise to produce P4. Similarly P4 is upsampled and merged with C3(in a similar way) to give P3 and so on. The final set of feature maps, in this case {P2 .. P5} are used as feature pyramids. 

This is how pyramids would look like 
![](https://i.imgur.com/oHFmpww.png)

*Usage of combination of {P2,..P5} as compared to only P2* : P2 produces highest resolution, most semantic features and could as well be the default choice but because of shared weights across rest of feature layers and the learned scale invariance makes the pyramidal variant more robust to generating false ROIs

For next steps, it could be RPN or RCNN, the regression and classifier would share weights across for all *anchors* (of varying aspect ratios) at each level of the feature pyramids. This step is similar to [Single Shot Detector (SSD) Networks ](http://www.shortscience.org/paper?bibtexKey=conf/eccv/LiuAESRFB16)

##### Observation
The FPN was used in FRCNN in both parts of RPN and RCNN separately and then combined FPN in both parts and produced state-of-the-art result in MS COCO challenges bettering results of COCO '15 & '16 winner models ( Faster RCNN +++ & GMRI) for mAP. FPN also can be used for instance segmentation by using fully convolutional layers on top of the image pyramids. FPN outperforms results from *DeepMask*, *SharpMask*, *InstanceFCN*

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)

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 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

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!

FaceNet directly maps face images to $\mathbb{R}^{128}$ where distances directly correspond to a measure of face similarity. They use a triplet loss function. The triplet is (face of person A, other face of person A, face of person which is not A). Later, this is called (anchor, positive, negative).

The loss function is learned and inspired by LMNN. The idea is to minimize the distance between the two images of the same person and maximize the distance to the other persons image.

## LMNN

Large Margin Nearest Neighbor (LMNN) is learning a pseudo-metric

$$d(x, y) = (x -y) M  (x -y)^T$$

where $M$ is a positive-definite matrix. The only difference between a pseudo-metric and a metric is that $d(x, y) = 0 \Leftrightarrow x = y$ does not hold.

## Curriculum Learning: Triplet selection

Show simple examples first, then increase the difficulty. This is done by selecting the triplets.

They use the triplets which are *hard*. For the positive example, this means the distance between the anchor and the positive example is high. For the negative example this means the distance between the anchor and the negative example is low.

They want to have

$$||f(x_i^a) - f(x_i^p)||_2^2 + \alpha < ||f(x_i^a) - f(x_i^n)||_2^2$$

where $\alpha$ is a margin and $x_i^a$ is the anchor, $x_i^p$ is the positive face example and $x_i^n$ is the negative example. They increase $\alpha$ over time. It is crucial that $f$ maps the images not in the complete $\mathbb{R}^{128}$, but on the unit sphere. Otherwise one could double $\alpha$ by simply making $f' = 2 \cdot f$.

## Tasks

* **Face verification**: Is this the same person?
* **Face recognition**: Who is this person?

## Datasets

* 99.63% accuracy on Labeled FAces in the Wild (LFW)
* 95.12% accuracy on YouTube Faces DB

## Network

Two models are evaluated: The [Zeiler & Fergus model](http://www.shortscience.org/paper?bibtexKey=journals/corr/ZeilerF13)  and an architecture based on the [Inception model](http://www.shortscience.org/paper?bibtexKey=journals/corr/SzegedyLJSRAEVR14).

## See also

* [DeepFace](http://www.shortscience.org/paper?bibtexKey=conf/cvpr/TaigmanYRW14#martinthoma)

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)
 

In the paper, authors  Bengio et al. , use the DenseNet for semantic segmentation. DenseNets iteratively concatenates input feature maps to output feature maps. The biggest contribution was the use of a novel upsampling path - given conventional upsampling would've caused severe memory cruch.

#### Background

All fully convolutional semantic segmentation nets generally follow a conventional path - a downsampling path which acts as feature extractor, an upsampling path that restores the locational information of every feature extracted in the downsampling path. 

As opposed to Residual Nets (where input feature maps are added to the output) , in DenseNets,the output is concatenated to input which has some interesting implications:  
- DenseNets  are efficient in the parameter usage, since all the feature maps are reused
- DenseNets perform deep supervision thanks to short path to all feature maps in the architecture 

Using DenseNets for segmentation though had an issue with upsampling in the conventional way of concatenating feature maps through skip connections as feature maps could easily go beyond 1-1.5 K. So Bengio et al. suggests a novel way - wherein only feature maps produced in the last Dense layer are only upsampled and not the entire feature maps. Post upsampling, the output is concatenated with feature maps of same resolution from downsampling path through skip connection. That way, the information lost during pooling in the downsampling path can be recovered.

#### Methodology & Architecture

In the downsampling path, the input is concatenated with the output of a dense block, whereas for upsampling the output of dense block is upsampled (without concatenating it with the input) and then concatenated with the same resolution output of downsampling path. 

Here's the overall architecture 
![](https://i.imgur.com/tqsPj72.png)

Here's how a Dense Block looks like 
![](https://i.imgur.com/MMqosoj.png)

#### Results
The 103 Conv layer based DenseNet (FC-DenseNet103) performed better than shallower networks when compared on CamVid dataset. Though the FC-DenseNets were not pre-trained or used any post-processing like CRF or temporal smoothening etc.  When comparing to other nets FC-DenseNet architectures achieve state-of-the-art, improving upon models with 10 times more parameters. It is also worth mentioning that  small model FC-DenseNet56 already outperforms  popular  architectures  with  at  least 100 times more parameters. 

## **Keywords**

Progressive GAN , High resolution generator

---

## **Summary**

1.  **Introduction**
    1.  **Goal of the paper**
        1.  Generation of very high quality images using progressively increasing size of the generator and discriminator.
        1.  Improved training and stability of GANs.
        1.  New metric for evaluating GAN results.
        1.  A high quality version of CELEBA-HQ dataset.
    1. **Previous Research**
        1.  Generative methods help to produce new samples from higher-dimensional data distributions such as images .
        1.  The common approaches for generative methods are :
            1.  Autoregressive models : Produce sharp images and are slow to evaluate. eg PixelCNN
            1.  Variational Autoencoders : Easy to train but produces blurry images.
            1.  Generative Adversarial Neural Network : Produces sharp images at small resolutions but are highly unstable.
1.  **Method**
    1.  **Basic GAN architecture**
        1.  Gan consists of two major parts :
            1.  _Generator_ : Creates a sample image from latent code which look very close to the training images.
            1.  _Discriminator_: Discriminator is trained to assess how close the sample image looks to the training image.
        1.  To measure the overlap between the training and the generated distributions many methods are used like Jensen-Shannon divergence , least-squares divergence and Wasserstein Distance.
        1.  Larger resolution generations cause problems because it becomes difficult for both the training and the generated networks amplifying the gradient problem. Larger resolutions also require large memory and can cause problems.
        1.  A mechanism is also proposed to stop the generator from participating in escalation that causes mode collapse problem.

    1.  **Progressive growing of GANs**
        1.  The primary method for the GAN training is to start off from a low resolution image and add extra layers in each step of the training process.
        1.  Lower resolution images are more stable as they have very less class information and as the resolution of the image increases further smaller details and features are added to the image.
        1.  This leads to a smooth increase in the quality of image instead of the network learning lot of details in one single step.
    1.  **Mini-batch separation**
        1.  GANs tend to capture only a very small set of features from the image.
        1.  "Minibatch discrimination" is used to generate feature vector for each individual image along with one for the the mini batch of images also.
        
          ![alt_text](https://i.imgur.com/dHFl5OV.png "image_tooltip")
1.  **Conclusion**
    1.  Higher resolution images are able to be generated which are robust and efficient.
    1.  Improved quality of the generated images is given.
    1.  Reduced training time for a comparable result and output quality and resolution.



---



## **Notes**



*   Gradient Problem : At higher resolutions it becomes easier to tell the differences between the training and the testing images [1]. This is referred to as the gradient problem.
*   Mode Collapse : The generator is incapable of creating a large variety of samples and get stuck.


## **Open research questions**



1.  Improved methods for a true photorealism generation of images.
1.  Improved semantic sensibility and improved understanding of the dataset.

## **References**



1.  [https://blog.acolyer.org/2018/05/10/progressive-growing-of-gans-for-improved-quality-stability-and-variation/](https://blog.acolyer.org/2018/05/10/progressive-growing-of-gans-for-improved-quality-stability-and-variation/) 
1.  [https://medium.com/@jonathan_hui/gan-why-it-is-so-hard-to-train-generative-advisory-networks-819a86b3750b](https://medium.com/@jonathan_hui/gan-why-it-is-so-hard-to-train-generative-advisory-networks-819a86b3750b)

This paper describes how to apply the idea of batch normalization (BN) successfully to recurrent neural networks, specifically to LSTM networks. The technique involves the 3 following ideas:

**1) Careful initialization of the BN scaling parameter.** While standard practice is to initialize it to 1 (to have unit variance), they show that this situation creates problems with the gradient flow through time, which vanishes quickly. A value around 0.1 (used in the experiments) preserves gradient flow much better.

**2) Separate BN for the "hiddens to hiddens pre-activation and for the "inputs to hiddens" pre-activation.** In other words, 2 separate BN operators are applied on each contributions to the pre-activation, before summing and passing through the tanh and sigmoid non-linearities.

**3) Use of largest time-step BN statistics for longer test-time sequences.** Indeed, one issue with applying BN to RNNs is that if the input sequences have varying length, and if one uses per-time-step mean/variance statistics in the BN transformation (which is the natural thing to do), it hasn't been clear how do deal with the last time steps of longer sequences seen at test time, for which BN has no statistics from the training set. The paper shows evidence that the pre-activation statistics tend to gradually converge to stationary values over time steps, which supports the idea of simply using the training set's last time step statistics.

Among these ideas, I believe the most impactful idea is 1). The papers mentions towards the end that improper initialization of the BN scaling parameter probably explains previous failed attempts to apply BN to recurrent networks.

Experiments on 4 datasets confirms the method's success.

**My two cents**

This is an excellent development for LSTMs. BN has had an important impact on our success in training deep neural networks, and this approach might very well have a similar impact on the success of LSTMs in practice.

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 is follow-up work to the ResNets paper. It studies the propagation formulations behind the connections of deep residual networks and performs ablation experiments. A residual block can be represented with the equations $y_l = h(x_l) + F(x_l, W_l); x_{l+1} = f(y_l)$. $x_l$ is the input to the l-th unit and $x_{l+1}$ is the output of the l-th unit. In the original ResNets paper, $h(x_l) = x_l$, $f$ is ReLu, and F consists of 2-3 convolutional layers (bottleneck architecture) with BN and ReLU in between. In this paper, they propose a residual block with both $h(x)$ and $f(x)$ as identity mappings, which trains faster and performs better than their earlier baseline. Main contributions:

- Identity skip connections work much better than other multiplicative interactions that they experiment with:
    - Scaling $(h(x) = \lambda x)$: Gradients can explode or vanish depending on whether modulating scalar \lambda > 1 or < 1.
    - Gating ($1-g(x)$ for skip connection and $g(x)$ for function F):
    For gradients to propagate freely, $g(x)$ should approach 1, but
    F gets suppressed, hence suboptimal. This is similar to highway
    networks. $g(x)$ is a 1x1 convolutional layer.
    - Gating (shortcut-only): Setting high biases pushes initial $g(x)$
    towards identity mapping, and test error is much closer to baseline.
    - 1x1 convolutional shortcut: These work well for shallower networks
    (~34 layers), but training error becomes high for deeper networks,
    probably because they impede gradient propagation.

- Experiments on activations.
    - BN after addition messes up information flow, and performs considerably
    worse.
    - ReLU before addition forces the signal to be non-negative, so the signal is monotonically increasing, while ideally a residual function should be free to take values in (-inf, inf).
    - BN + ReLU pre-activation works best. This also prevents overfitting, due
    to BN's regularizing effect. Input signals to all weight layers are normalized.

## Strengths

- Thorough set of experiments to show that identity shortcut connections
are easiest for the network to learn. Activation of any deeper unit can
be written as the sum of the activation of a shallower unit and a residual
function. This also implies that gradients can be directly propagated to
shallower units. This is in contrast to usual feedforward networks, where
gradients are essentially a series of matrix-vector products, that may vanish, as networks grow deeper.

- Improved accuracies than their previous ResNets paper.

## Weaknesses / Notes

- Residual units are useful and share the same core idea that worked in
LSTM units. Even though stacked non-linear layers are capable of asymptotically
approximating any arbitrary function, it is clear from recent work that
residual functions are much easier to approximate than the complete function.
The [latest Inception paper](http://arxiv.org/abs/1602.07261) also reports
that training is accelerated and performance is improved by using identity
skip connections across Inception modules.

- It seems like the degradation problem, which serves as motivation for
residual units, exists in the first place for non-idempotent activation
functions such as sigmoid, hyperbolic tan. This merits further
investigation, especially with recent work on function-preserving transformations such as [Network Morphism](http://arxiv.org/abs/1603.01670), which expands the Net2Net idea to sigmoid, tanh, by using parameterized activations, initialized to identity mappings.

TLDR; The authors use an attention mechanism in image caption generation, allowing the decoder RNN focus on specific parts of the image. In order find the correspondence between words and image patches, the RNN uses a lower convolutional layer as its input (before pooling). The authors propose both a "hard" attention (trained using sampling methods) and "soft" attention (trained end-to-end) mechanism, and show qualitatively that the decoder focuses on sensible regions while generating text, adding an additional layer of interpretability to the model. The attention-based models achieve state-of-the art on Flickr8k, Flickr30 and MS Coco.

#### Key Points

- To find image correspondence use lower convolutional layers to attend to.
- Two attention mechanisms: Soft and hard. Depending on evaluation metric (BLEU vs. METERO) one or the other performs better.
- Largest data set (MS COCO) takes 3 days to train on Titan Black GPU. Oxford VGG.
- Soft attention is same as for seq2seq models.
- Attention weights are visualized by upsampling and applying a Gaussian

#### Notes/Questions

- Would've liked to see an explanation of when/how soft vs. hard attention does better.
- What is the computational overhead of using the attention mechanism? Is it significant?

The authors have a dataset of 780 electronic health records and they use it to detect various medical events such as adverse drug events, drug dosage, etc. The task is done by assigning a label to each word in the document.

https://i.imgur.com/bZ7yM0z.png

Annotation statistics for the corpus of health records.

They look at CRFs, LSTMs and GRUs. Both LSTMs and GRUs outperform the CRF, but the best performance is achieved by a GRU trained on whole documents.

At NIPS 2017, Ali Rahimi was invited on stage to give a keynote after a paper he was on received the “Test of Time” award. While there, in front of several thousand researchers, he gave an impassioned argument for more rigor: more small problems to validate our assumptions, more visibility into why our optimization algorithms work the way they do. The now-famous catchphrase of the talk was “alchemy”; he argued that the machine learning community has been effective at finding things that work, but less effective at understanding why the techniques we use work. A central example he used in his talk is that of Batch Normalization: a now nearly-universal step in optimizing deep nets, but one where our accepted explanation of “reducing internal covariate shift” is less rigorous than one might hope. 

With apologies for the long preamble, this is the context in which today’s paper is such a welcome push in the direction of what Rahimi was advocating for - small, focused experimentation that tries to build up knowledge from principles, and, specifically, asks the question: “Does Batch Norm really work via reducing covariate shift”. 

To answer the question of whether internal covariate shift is a likely mechanism of the - empirically very solid - improved performance of Batch Norm, the authors do a few simple experience. First, and most straightforwardly, they train a basic convolutional net with and without BatchNorm, pick a layer, and visualize the activation distribution of that layer over time, both in the Batch Norm and non-Batch Norm case. While they saw the expected performance boost, the Batch Norm case didn’t seem to be meaningfully more stable over time, relative to the normal case. Second, the authors tested what would happen if they added non-zero-mean random noise *after* Batch Norm in the network. The upshot of this was that they were explicitly engineering internal covariate shift, and, if control thereof was the primary useful purpose of Batch Norm, you would expect that to neutralize BN’s good performance. In this experiment, while the authors did indeed see noisier, less stable activation distributions in the noise + BN case (in particular: look at layer 13 activations in the attached image), but noisy BN performed nearly as well as non-noisy, and meaningfully better than the standard model without noise, but also without BN. 

As a final test, they approached the idea of “internal covariate shift” from a different definitional standpoint. Maybe a better way of thinking about it is in terms of stability of your gradients, in the face of updates made by lower layers of the network. That is to say: each parameter of the network pushes itself in the direction of lower loss all else held equal, but in practice, you change lower-level parameters simultaneously, which could cause the directional change the higher-layer parameter thought it needed to be off. So, the authors calculated the “gradient delta” between the gradient the model trains on, and what the gradient would be if you estimated it *after* all of the lower layers of the model had updated, such that the distribution of inputs to that layer has changed. Although the expectation would be that this gradient delta is smaller for batch norm, in fact, the authors found that, if anything, the opposite was true. 

So, in the face of none of these ideas panning out, the authors then introduce the best idea they’ve found for what motivates BN’s improved performance: a smoothing out of the loss function that SGD is optimizing. A smoother curve means, generally speaking, that the magnitudes of your gradients will be smaller, and also that the value of the gradient will change more slowly (i.e. low second derivative). As support for this idea, they show really different results for BN vs standard models in terms of, for example, how predictive a gradient at one point is of a gradient taken after you take a step in the direction of the first gradient. BN has meaningfully more predictive gradients, tied to lower variance in the values of the loss function in the direction of the gradient. The logic for why the mechanism of BN would cause this outcome is a bit tied up in math that’s hard to explain without LaTeX visuals, but basically comes from the idea that Batch Norm decreases the magnitude of the gradient of each layer output with respect to individual weight parameters, by averaging out those magnitudes over the batch. 

As Rahimi said in his initial talk, a lot of modern modeling is “applying brittle optimization techniques to loss surfaces we don’t understand.” And, by and large, that is in fact true: it’s devilishly difficult to get a good handle on what loss surfaces are doing when they’re doing it in several-million-dimensional space. But, it being hard doesn’t mean we should just give up on searching for principles we can build our understanding on, and I think this paper is a really fantastic example of how that can be done well. 

Mask RCNN takes off from where Faster RCNN left, with some augmentations aimed at bettering instance segmentation (which was out of scope for FRCNN). Instance segmentation was achieved remarkably well in *DeepMask* , *SharpMask* and later *Feature Pyramid Networks* (FPN).

Faster RCNN was not designed for pixel-to-pixel alignment between network inputs and outputs. This is most evident in how RoIPool , the de facto core operation for attending to instances, performs coarse spatial quantization for feature extraction. Mask RCNN fixes that by introducing RoIAlign in place of RoIPool.

#### Methodology

Mask RCNN retains most of the architecture of Faster RCNN. It adds the a third branch for segmentation. The third branch takes the output from RoIAlign layer and predicts binary class masks for each class.

##### Major Changes and intutions

**Mask prediction**

Mask prediction segmentation predicts a binary mask for each RoI using fully convolution - and the stark difference being usage of *sigmoid* activation for predicting final mask instead of *softmax*, implies masks don't compete with each other. This *decouples* segmentation from classification. The class prediction branch is used for class prediction and for calculating loss, the mask of predicted loss is used calculating Lmask.

Also, they show that a single class agnostic mask prediction works almost as effective as separate mask for each class, thereby supporting their method of decoupling classification from segmentation

**RoIAlign**

RoIPool first quantizes a floating-number RoI to the discrete granularity of the feature map, this quantized RoI is then subdivided into spatial bins which are themselves quantized, and finally feature values covered by each bin are aggregated (usually by max pooling). Instead of  quantization of the RoI boundaries
or bin bilinear interpolation is used to compute the exact values of the input features at four regularly sampled locations in each RoI bin, and aggregate the result (using max or average).

**Backbone architecture**

Faster RCNN uses a VGG like structure for extracting features from image, weights of which were shared among RPN and region detection layers. Herein, authors experiment with 2 backbone architectures - ResNet based VGG like in FRCNN and ResNet based [FPN](http://www.shortscience.org/paper?bibtexKey=journals/corr/LinDGHHB16) based. FPN uses convolution feature maps from previous layers and recombining them to produce pyramid of feature maps to be used for prediction instead of single-scale feature layer (final output of conv layer before connecting to fc layers was used in Faster RCNN) 

**Training Objective**

The training objective looks like this 
![](https://i.imgur.com/snUq73Q.png)

Lmask is the addition from Faster RCNN. The method to calculate was mentioned above

#### Observation

Mask RCNN performs significantly better than COCO instance segmentation winners *without any bells and whiskers*. Detailed results are available in the paper

Deeper networks should never have a higher **training** error than smaller ones. In the worst case, the layers should "simply" learn identities. It seems as this is not so easy with conventional networks, as they get much worse with more layers. So the idea is to add identity functions which skip some layers. The network only has to learn the **residuals**. 

Advantages:

* Learning the identity becomes learning 0 which is simpler
* Loss in information flow in the forward pass is not a problem anymore
    * No vanishing / exploding gradient
* Identities don't have parameters to be learned

## Evaluation

The learning rate starts at 0.1 and is divided by 10 when the error plateaus. Weight decay of 0.0001 ($10^{-4}$), momentum of 0.9. They use mini-batches of size 128.

* ImageNet ILSVRC 2015: 3.57% (ensemble)
* CIFAR-10: 6.43%
* MS COCO: 59.0% mAp@0.5 (ensemble)
* PASCAL VOC 2007: 85.6% mAp@0.5
* PASCAL VOC 2012: 83.8% mAp@0.5

## See also

* [DenseNets](http://www.shortscience.org/paper?bibtexKey=journals/corr/1608.06993)

This paper presents a method to train a neural network to make predictions for *counterfactual* questions. In short, such questions are questions about what the result of an intervention would have been, had a different choice for the intervention been made (e.g. *Would this patient have lower blood sugar had she received a different medication?*).

One approach to tackle this problem is to collect data of the form $(x_i, t_i, y_i^F)$ where $x_i$ describes a situation (e.g. a patient), $t_i$ describes the intervention made (in this paper $t_i$ is binary, e.g. $t_i = 1$ if a new treatment is used while $t_i = 0$ would correspond to using the current treatment) and $y_i^F$ is the factual outcome of the intervention $t_i$ for $x_i$. From this training data, a predictor $h(x,t)$ taking the pair $(x_i, t_i)$ as input and outputting a prediction for $y_i^F$ could be trained. 

From this predictor, one could imagine answering counterfactual questions by feeding $(x_i, 1-t_i)$ (i.e. a description of the same situation $x_i$ but with the opposite intervention $1-t_i$) to our predictor and comparing the prediction $h(x_i, 1-t_i)$ with $y_i^F$. This would give us an estimate of the change in the outcome, had a different intervention been made, thus providing an answer to our counterfactual question.

The authors point out that this scenario is related to that of domain adaptation (more specifically to the special case of covariate shift) in which the input training distribution (here represented by inputs $(x_i,t_i)$) is different from the distribution of inputs that will be fed at test time to our predictor (corresponding to the inputs $(x_i, 1-t_i)$). If the choice of intervention $t_i$ is evenly spread and chosen independently from $x_i$, the distributions become the same. However, in observational studies, the choice of $t_i$ for some given $x_i$ is often not independent of $x_i$ and made according to some unknown policy. This is the situation of interest in this paper.

Thus, the authors propose an approach inspired by the domain adaptation literature. Specifically, they propose to have the predictor $h(x,t)$ learn a representation of $x$ that is indiscriminate of the intervention $t$ (see Figure 2 for the proposed neural network architecture). Indeed, this is a notion that is [well established][1] in the domain adaptation literature and has been exploited previously using regularization terms based on [adversarial learning][2] and [maximum mean discrepancy][3]. In this paper, the authors used instead a regularization (noted in the paper as $disc(\Phi_{t=0},\Phi_ {t=1})$) based on the so-called discrepancy distance of [Mansour et al.][4], adapting its use to the case of a neural network. 

As an example, imagine that in our dataset, a new treatment ($t=1$) was much more frequently used than not ($t=0$) for men. Thus, for men, relatively insufficient evidence for counterfactual inference is expected to be found in our training dataset. Intuitively, we would thus want our predictor to not rely as much on that "feature" of patients when inferring the impact of the treatment. 

In addition to this term, the authors also propose incorporating an additional regularizer where the prediction $h(x_i,1-t_i)$ on counterfactual inputs is pushed to be as close as possible to the target $y_{j}^F$ of the observation $x_j$ that is closest to $x_i$ **and** actually had the counterfactual intervention $t_j = 1-t_i$. 

The paper first shows a bound relating the counterfactual generalization error to the discrepancy distance. Moreover, experiments simulating counterfactual inference tasks are presented, in which performance is measured by comparing the predicted treatment effects (as estimated by the difference between the observed effect $y_i^F$ for the observed treatment and the predicted effect $h(x_i, 1-t_i)$ for the opposite treatment) with the real effect (known here because the data is simulated). The paper shows that the proposed approach using neural networks outperforms several baselines on this task.

**My two cents**

The connection with domain adaptation presented here is really clever and enlightening. This sounds like a very compelling approach to counterfactual inference, which can exploit a lot of previous work on domain adaptation.

The paper mentions that selecting the hyper-parameters (such as the regularization terms weights) in this scenario is not a trivial task. Indeed, measuring performance here requires knowing the true difference in intervention outcomes, which in practice usually cannot be known (e.g. two treatments usually cannot be given to the same patient once). In the paper, they somewhat "cheat" by using the ground truth difference in outcomes to measure out-of-sample performance, which the authors admit is unrealistic. Thus, an interesting avenue for future work would be to design practical hyper-parameter selection procedures for this scenario. I wonder whether the *reverse cross-validation* approach we used in our work on our adversarial approach to domain adaptation (see [Section 5.1.2][5]) could successfully be used here.

Finally, I command the authors for presenting such a nicely written description of counterfactual inference problem setup in general, I really enjoyed it!

[1]: https://papers.nips.cc/paper/2983-analysis-of-representations-for-domain-adaptation.pdf
[2]: http://arxiv.org/abs/1505.07818
[3]: http://ijcai.org/Proceedings/09/Papers/200.pdf
[4]: http://www.cs.nyu.edu/~mohri/pub/nadap.pdf
[5]: http://arxiv.org/pdf/1505.07818v4.pdf#page=16

Problem
---------
Video prediction with human objects


Contribution
--------------
Instead of the common approach of predicting directly in pixel-space, use explicit knowledge of human motion space to predict the future of the video.

Approach
--------------
1. VAE to model the possible future movements of humans in the pose space
2. Conditional GAN - use pose information for to predict video in pixel space.



https://image.ibb.co/b1omVF/The_pose_knows.png

Szegedy et al. were (to the best of my knowledge) the first to describe the phenomen of adversarial examples as researched today. Specifically, they described the main objective in order to obtain adversarial examples as

$\arg\min_r \|r\|_2$ s.t. $f(x+r)=l$ and $x+r$ being a valid image

where $f$ is the neural network and $l$ the target class (i.e. targeted adversarial example). In the paper, they originally headlined the section by “blind spots in neural networks”. While they give some explanation and provide experiments, also introducing the notion of transferability of adversarial examples and an idea of adversarial examples used as regularization during training, many questions are left open. The given conclusion, that these adversarial examples are highly unlikely and that these examples lie dense within regular training examples are controversial in the literature.