Output Projection in Seq2Seq model Tensorflow - python

I'm going through the translation code implemented by tensorflow using seq2seq model. I'm following tensorflow tutorial about seq2seq model.
In that tutorial there is a part explaining a concept called output projection which they have implemented in seq2seq_model.py code. I understand the code. But I don't understand what this output projection part is doing.
It would be great if someone can explain me what is going on behind this output projection thing..?
Thank You!!

Internally, a neural network operates on dense vectors of some size, often 256, 512 or 1024 floats (let's say 512 for here). But at the end it needs to predict a word from the vocabulary which is often much larger, e.g., 40000 words. Output projection is the final linear layer that converts (projects) from the internal representation to the larger one. So, for example, it can consist of a 512 x 40000 parameter matrix and a 40000 parameter for the bias vector. The reason it is kept separate in seq2seq code is that some loss functions (e.g., the sampled softmax loss) need direct access to the final 512-sized vector and the output projection matrix. Hope that helps!

Related

Can I use a convolution filter instead of a dense layer for clasification?

I was reading a decent paper S-DCNet and I fell upon a section (page3,table1,classifier) where a convolution layer has been used on the feature map in order to produce a binary classification output as part of an internal process. Since I am a noob and when someone talks to me about classification I automatically make a synapse relating to FCs combined with softmax, I started wondering ... Is this a possible thing to do? Can indeed a convolutional layer be used to classify a binary outcome? The whole concept triggered my imagination so much that I insist on getting answers...
Honestly, how does this actually work? What is the difference between using a convolution filter instead of a fully connected layer for classification purposes?
Edit (Uncertain answer on how does it work): I asked a colleague and he told me that using a filter of the same shape as the length-width shape of the feature map at the current stage, may lead to a learnable binary output (considering that you also reduce the #channels of the feature map to a single channel). But I still don't understand the motivations behind such a technique ..
Using convolutions as FCs can be done (for example) with filters of spatial size (1,1) and with depth of the same size as the FC input size.
The resulting feature map would be of the same size as the input feature map, but each pixel would be the output of a "FC" layer whose weights are the weights of the shared 1x1 conv filter.
This kind of thing is used mainly for semantic segmentation, meaning classification per pixel. U-net is a good example if memory serves.
Also see this.
Also note that 1x1 convolutions have other uses as well.
paperswithcode probably some of the nets there use this trick.

Can CNN autoencoders have different input and output dimensions?

I am working on a problem which requires me to build a deep learning model that based on certain input image it has to output another image. It is worth noting that these two images are conceptually related but they don't have the same dimensions.
At first I thought that a classical CNN with a final dense layer whose argument is the multiplication of the height and width of the output image would suit this case, but when training it was giving strange figures such as accuracy of 0.
While looking for some answers on the Internet I discovered the concepts of CNN autoencoders and I was wondering if this approach could help me solve my problem. Among all the examples I saw, the input and output of an autoencoder had the same size and dimensions.
At this point I wanted to ask if there was a type of CNN autoencoders that produce an output image that has different dimension compared to input image.
Auto-encoder (AE) is an architecture that tries to encode your image into a lower-dimensional representation by learning to reconstruct the data from such representation simultaniously. Therefore AE rely on a unsupervised (don't need labels) data that is used both as an input and as the target (used in the loss).
You can try using a U-net based architecture for your usecase. A U-net would forward intermediate data representations to later layers of the network which should assist with faster learning/mapping of the inputs into a new domain..
You can also experiment with a simple architecture containing a few ResNet blocks without any downsampling layers, which might or might not be enough for your use-case.
If you want to dig a little deeper you can look into Disco-GAN and related methods.They explicitly try to map image into a new domain while maintaining image information.

Keras LSTM Input Transpose

I'm training a timeseries LSTM model using Keras. I understand that the input to the model has to be in the format: [samples, timesteps, features].
However, when I reverse transpose each input element, so the input now is in the format: [samples, features, timesteps] my model accuracy improves significantly, and training time is reduced quite a bit as well. Does anyone have an explanation as to why?
For reference, here are the stats on my training data:
samples: 720
timesteps: 256
features: 4
So my input tensor should have the shape [720, 256, 4] but reshaping to [720, 4, 256] produces better results. Why?
As I said in my rather long comments, the answer is "because you are not learning the same thing". Frameworks like tensorflow and keras attempt to make training and using networks convenient, so as long as your inputs are at least approximately the right shape, the framework will try its best to feed the data into the network. But the network has no way to interpret the data you feed into it. It is up to you to make sure that what you are feeding into the network makes sense in the context of your data. No matter what data you send and what labels you use, the network will do its best to learn a mapping between the data and the labels. And it might succeed. Just because the pattern you are trying to learn makes no sense doesn't mean it cannot be learned. So to answer your question, you need to figure out what is the meaning of your input transposition and given that LSTM will treat your data as a sequence of consecutive datapoints, what sequences did you end up learning.

Multilabel classification using LSTM on variable length signal using Keras

I have recently started working on ECG signal classification in to various classes. It is basically multi label classification task (Total 4 classes). I am new to Deep Learning, LSTM and Keras that why i am confused in few things.
I am thinking about giving normalized original signal as input to the network, is this a good approach?
I also need to understand training input shape for LSTM as ECG signals are of variable length (9000 to 18000 samples) and usually classifier need fixed variable input. How can i handle such type of input in case of LSTM.
Finally what should be structure of deep LSTM network for such lengthy input and how many layers should i use.
Thanks for your time.
Regards
I am thinking about giving normalized original signal as input to the network, is this a good approach?
Yes this is a good approach. It is actually quite standard for Deep Learning algorithms to give them your input normalized or rescaled.
This usually helps your model converge faster, as now you are inside smaller range (i.e.: [-1, 1]) instead of greater un-normalized ranges from your original input (say [0, 1000]). It also helps you get better, more precise results, as it helps solve problems like the vanishing gradient as well as adapting better to modern activation and optimizer functions.
I also need to understand training input shape for LSTM as ECG signals are of variable length (9000 to 18000 samples) and usually classifier need fixed variable input. How can i handle such type of input in case of LSTM.
This part is really important. You are correct, LSTM expects to receive inputs with a fixed shape, one that you know beforehand (in fact, any Deep Learning layer expects fixed shape inputs). This is also explained in the keras docs on Recurrent Layers where they say:
Input shape
3D tensor with shape (batch_size, timesteps, input_dim).
As we can see, it expects your data to have a number of timesteps as well as a dimension on each one of those timesteps (batch size is usually 1). To exemplify, suppose your input data consists of elements like: [[1,4],[2,3],[3,2],[4,1]]. Then, using a batch_size of 1, the shape of your data would be (1,4,2). As you have 4 timesteps, each with 2 features.
So bottom line, you have to make sure that you pre-process you data so it has a fixed shape you can then pass to your LSTM layers. This one you will have to find out by yourself, as you know your data and problem better than we do.
Maybe you can fix the samples you obtain from your signal, discarding some and keeping others so every signal is of the same length (if you say your signals are between 9k and 18k choosing 9000 could be the logical choice, discarding samples from the others you get). You could even do some other conversion to your data in a way that you can map from inputs of 9000-18000 to a fixed size.
Finally what should be structure of deep LSTM network for such lengthy input and how many layers should i use.
This one is really quite broad and doesn't have a unique answer. It would depend on the nature of your problem, and determining those parameters a priori is not so straightforward.
What I recommend you do is to start with a simple model first, and then add layers and blocks (neurons) incrementally until you are satisfied with the results.
Try just one hidden layer first, train and test your model and check your performance. You can then add more blocks and see if your performance improved. You can also add more layers and check for the same until you are satisfied.
This is a good way to create Deep Learning models, as you will arrive to the results you want while keeping your Network as lean as possible, which in turn helps your execution time and complexity. Good luck with your coding, hope you find this useful.

What is the key feature in MNIST Dataset that is used to classify images

I was recently learning about neural networks and came across MNIST data set. i understood that a sigmoid cost function is used to reduce the loss. Also, weights and biases gets adjusted and an optimum weights and biases are found after the training. the thing i did not understand is, on what basis the images are classified. For example, to classify whether a patient has cancer or not, data like age, location, etc., becomes features. in MNIST dataset, i did not find any of that. Am i missing something here. Please help me with this
First of all the Network pipeline consists of 3 main parts:
Input Manipulation:
Parameters that effect the finding of minimum:
Parameters like your descission function in your interpretation
layer (often fully connected layer)
In contrast to your regular machine learning pipeline where you have to extract features manually a CNN uses filters. (Filters like in edge detection or viola and jones).
If a filter runs across the images and is convolved with pixels it Produces an output.
This output is then interpreted by a neuron. If the output is above a threshold it is considered as valid (Step function counts 1 if valid or in case of Sigmoid it has a value on the sigmoid function).
The next steps are the same as before.
This is progressed until the interpretation layer (often softmax). This layer interprets your computation (if the filters are good adapted to your problem you will get a good predicted label) which means you have a low difference between (y_guess - y_true_label).
Now you can see that for the guess of y we have multiplied the input x with many weights w and also used functions on it. This can be seen like a chain rule in analysis.
To get better results the effect of a single weight on the input must be known. Therefore, you use Backpropagation which is a derivative of the Error with respect to all w. The Trick is that you can reuse derivatives which is more or less Backpropagation and it becomes easier since you can use Matrix vector notation.
If you have your gradient, you can use the normal concept of minimization where you walk along the steepest descent. (There are also many other gradient methods like adagrad or adam etc).
The steps will repeat until convergence or until you reach the maximum epochs.
So the answer is: THE COMPUTED WEIGHTS (FILTERS) ARE THE KEY TO DETECT NUMBERS AND DIGITS :)

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