http://wiki.math.uwaterloo.ca/statwiki/api.php?action=feedcontributions&user=Iaoellme&feedformat=atomstatwiki - User contributions [US]2021-11-27T11:52:20ZUser contributionsMediaWiki 1.28.3http://wiki.math.uwaterloo.ca/statwiki/index.php?title=Research_Papers_Classification_System&diff=49343Research Papers Classification System2020-12-06T07:59:00Z<p>Iaoellme: /* Conclusion */</p>
<hr />
<div>= Presented by =<br />
Jill Wang, Junyi (Jay) Yang, Yu Min (Chris) Wu, Chun Kit (Calvin) Li<br />
<br />
= Introduction =<br />
With the increasing advance of computer science and information technology, there is an increasingly overwhelming number of papers that have been published. Because of the mass number of papers, it has become incredibly hard to find and categorize papers. This paper introduces a paper classification system that utilizes the Term Frequency-Inverse Document Frequency (TF-IDF), Latent Dirichlet Allocation (LDA), and K-means clustering. The most important technology the system used to process big data is the Hadoop Distributed File System (HDFS). The system can handle quantitatively complex research paper classification problems efficiently and accurately.<br />
<br />
===General Framework===<br />
<br />
The paper classification system classifies research papers based on the abstracts given that the core of most papers is presented in the abstracts. <br />
<br />
[[ File:Systemflow.png |right|image on right| 400px]]<br />
<ol><li>Paper Crawling <br />
<p>Collects abstracts from research papers published during a given period</p></li><br />
<li>Preprocessing<br />
<p> <ol style="list-style-type:lower-alpha"><li>Removes stop words in the papers crawled, in which only nouns are extracted from the papers</li><br />
<li>generates a keyword dictionary, keeping only the top-N keywords with the highest frequencies</li> </ol><br />
</p></li> <br />
<li>Topic Modelling<br />
<p> Use the LDA to group the keywords into topics</p><br />
</li><br />
<li>Paper Length Calculation<br />
<p> Calculates the total number of occurrences of words to prevent an unbalanced TF values caused by the various length of abstracts using the map-reduce algorithm</p><br />
</li><br />
<li>Word Frequency Calculation<br />
<p> Calculates the Term Frequency (TF) values which represent the frequency of keywords in a research paper</p><br />
</li><br />
<li>Document Frequency Calculation<br />
<p> Calculates the Document Frequency (DF) values which represents the frequency of keywords in a collection of research papers. The higher the DF value, the lower the importance of a keyword.</p><br />
</li><br />
<li>TF-IDF calculation<br />
<p> Calculates the inverse of the DF which represents the importance of a keyword.</p><br />
</li><br />
<li>Paper Classification<br />
<p> Classify papers by topics using the K-means clustering algorithm.</p><br />
</li><br />
</ol><br />
<br />
<br />
===Technologies===<br />
<br />
The HDFS with a Hadoop cluster composed of one master node, one sub-node, and four data nodes is what is used to process the massive paper data. Hadoop-2.6.5 version in Java is what is used to perform the TF-IDF calculation. Spark MLlib is what is used to perform the LDA. The Scikit-learn library is what is used to perform the K-means clustering.<br />
<br />
===HDFS===<br />
<br />
Hadoop Distributed File System (HDFS) was used to process big data in this system. HFDS has been shown to process big data rapidly and stably with high scalability which makes it a perfect choice for this problem. What Hadoop does is to break a big collection of data into different partitions and pass each partition to one individual processor. Each processor will only have information about the partition of data it has received.<br />
<br />
'''In this summary, we are going to focus on introducing the main algorithms of what this system uses, namely LDA, TF-IDF, and K-Means.'''<br />
<br />
=Data Preprocessing=<br />
===Crawling of Abstract Data===<br />
<br />
Under the assumption that audiences tend to first read the abstract of a paper to gain an overall understanding of the material, it is reasonable to assume the abstract section includes “core words” that can be used to effectively classify a paper's subject.<br />
<br />
An abstract is crawled to have its stop words removed. Stop words are words that are usually ignored by search engines, such as “the”, “a”, and etc. Afterward, nouns are extracted, as a more condensed representation for efficient analysis.<br />
<br />
This is managed on HDFS. The TF-IDF value of each paper is calculated through map-reduce, an easy-to-use programming model and implementation for processing and generating large data sets. The user must specify (i) a map procedure, that filters and sorts the input data to produce a set of intermediate key/value pairs and (ii) a reduce function, which performs a summary operation on the intermediate values with the same key and returns a smaller set of output key/value pairs. The MapReduce interface enables this process by grouping the intermediate values with the same key and passing them as input to the reduce function. For example, one could count the number of times various words appear in a large number of documents by setting your map procedure to count the number of occurrences of each word in a single document, and your reduce function to sum all counts of a given word [[https://dl.acm.org/doi/pdf/10.1145/1327452.1327492?casa_token=_Zg_DWxQzKEAAAAA:EHII0CaP36_ojGMT8huqTGLNMSEc-CKzZAoXBxSXe6pr2WB0DCQvEKa30CFQW0NSbB2-CVo8GcBcJAg 1]].<br />
<br />
===Managing Paper Data===<br />
<br />
To construct an effective keyword dictionary using abstract data and keywords data in all of the crawled papers, the authors categorized keywords with similar meanings using a single representative keyword. The approach is called stemming, which is common in cleaning data, where words are reduced to their word stem. An example is "running" and "ran" would be reduced to "run". 1394 keyword categories are extracted, which is still too much to compute. Hence, only the top 30 keyword categories are used.<br />
<br />
<div align="center">[[File:table_1_kswf.JPG|700px]]</div><br />
<br />
=Topic Modeling Using LDA=<br />
<br />
Latent Dirichlet allocation (LDA) is a generative probabilistic model that views documents as random mixtures over latent topics. Each topic is a distribution over words, and the goal is to extract these topics from documents.<br />
<br />
LDA estimates the topic-word distribution <math>P\left(t | z\right)</math> (probability of word "z" having topic "t") and the document-topic distribution <math>P\left(z | d\right)</math> (probability of finding word "z" within a given document "d") using Dirichlet priors for the distributions with a fixed number of topics. For each document, obtain a feature vector:<br />
<br />
\[F = \left( P\left(z_1 | d\right), P\left(z_2 | d\right), \cdots, P\left(z_k | d\right) \right)\]<br />
<br />
In the paper, authors extract topics from preprocessed paper to generate three kinds of topic sets, each with 10, 20, and 30 topics respectively. The following is a table of the 10 topic sets of highest frequency keywords.<br />
<br />
<div align="center">[[File:table_2_tswtebls.JPG|700px]]</div><br />
<br />
<br />
===LDA Intuition===<br />
<br />
LDA uses the Dirichlet priors of the Dirichlet distribution, which allows the algorithm to model a probability distribution ''over prior probability distributions of words and topics''. The following picture illustrates 2-simplex Dirichlet distributions with different alpha values, one for each corner of the triangles. <br />
<br />
<div align="center">[[File:dirichlet_dist.png|700px]]</div><br />
<br />
Simplex is a generalization of the notion of a triangle in k-1 dimension where k is the number of classes. For example, if you wish to classify essays into three groups, English, History and Math then the simplex would be a 2 dimension triangle, if you add philosophy as one of your potential class, then we would need a tetrahedron in 3 deminsion. In Dirichlet distribution, each parameter will be represented by a corner in simplex, so adding additional parameters implies increasing the dimensions of simplex. As illustrated, when alphas are smaller than 1 the distribution is dense at the corners. When the alphas are greater than 1 the distribution is dense at the centers.<br />
<br />
The following illustration shows an example LDA with 3 topics, 4 words and 7 documents.<br />
<br />
<div align="center">[[File:LDA_example.png|800px]]</div><br />
<br />
In the left diagram, there are three topics, hence it is a 2-simplex. In the right diagram there are four words, hence it is a 3-simplex. LDA essentially adjusts parameters in Dirichlet distributions and multinomial distributions (represented by the points), such that, in the left diagram, all the yellow points representing documents and, in the right diagram, all the points representing topics, are as close to a corner as possible. In other words, LDA finds topics for documents and also finds words for topics. At the end topic-word distribution <math>P\left(t | z\right)</math> and the document-topic distribution <math>P\left(z | d\right)</math> are produced.<br />
<br />
=Term Frequency Inverse Document Frequency (TF-IDF) Calculation=<br />
<br />
TF-IDF is widely used to evaluate the importance of a set of words in the fields of information retrieval and text mining. It is a combination of term frequency (TF) and inverse document frequency (IDF). The idea behind this combination is<br />
* It evaluates the importance of a word within a document<br />
* It evaluates the importance of the word among the collection of all documents<br />
<br />
The inverse of the document frequency accounts for the fact that term frequency will naturally increase as document frequency increases. Thus IDF is needed to counteract a word's TF to give an accurate representation of a word's importance.<br />
<br />
The TF-IDF formula has the following form:<br />
<br />
\[TF-IDF_{i,j} = TF_{i,j} \times IDF_{i}\]<br />
<br />
where i stands for the <math>i^{th}</math> word and j stands for the <math>j^{th}</math> document.<br />
<br />
===Term Frequency (TF)===<br />
<br />
TF evaluates the percentage of a given word in a document. Thus, TF value indicates the importance of a word. The TF has a positive relation with the importance.<br />
<br />
In this paper, we only calculate TF for words in the keyword dictionary obtained. For a given keyword i, <math>TF_{i,j}</math> is the number of times word i appears in document j divided by the total number of words in document j.<br />
<br />
The formula for TF has the following form:<br />
<br />
\[TF_{i,j} = \frac{n_{i,j} }{\sum_k n_{k,j} }\]<br />
<br />
where i stands for the <math>i^{th}</math> word, j stands for the <math>j^{th}</math> document, <math>n_{i,j}</math> stands for the number of times words <math>t_i</math> appear in document <math>d_j</math> and <math>\sum_k n_{k,j} </math> stands for total number of occurence of words in document <math>d_j</math>.<br />
<br />
Note that the denominator is the total number of words remaining in document j after crawling.<br />
<br />
===Document Frequency (DF)===<br />
<br />
DF evaluates the percentage of documents that contain a given word over the entire collection of documents. Thus, the higher DF value is, the less important the word is.<br />
<br />
<math>DF_{i}</math> is the number of documents in the collection with word i divided by the total number of documents in the collection. The formula for DF has the following form:<br />
<br />
\[DF_{i} = \frac{|d_k \in D: n_{i,k} > 0|}{|D|}\]<br />
<br />
where <math>n_{i,k}</math> is the number of times word i appears in document k, |D| is the total number of documents in the collection.<br />
<br />
Since DF and the importance of the word have an inverse relation, we use inverse document frequency (IDF) instead of DF.<br />
<br />
===Inverse Document Frequency (IDF)===<br />
<br />
In this paper, IDF is calculated in a log scale. Since we will receive a large number of documents, i.e, we will have a large |D|<br />
<br />
The formula for IDF has the following form:<br />
<br />
\[IDF_{i} = log\left(\frac{|D|}{|\{d_k \in D: n_{i,k} > 0\}|}\right)\]<br />
<br />
As mentioned before, we will use HDFS. The actual formula applied is:<br />
<br />
\[IDF_{i} = log\left(\frac{|D|+1}{|\{d_k \in D: n_{i,k} > 0\}|+1}\right)\]<br />
<br />
The inverse document frequency gives a measure of how rare a certain term is in a given document corpus.<br />
<br />
=Paper Classification Using K-means Clustering=<br />
<br />
The K-means clustering is an unsupervised classification algorithm that groups similar data into the same class. It is an efficient and simple method that can be applied to different types of data attributes. It is also flexible enough to handle various kinds of noise and outliers.<br />
<br><br />
<br />
Given a set of <math>d</math> by <math>n</math> dataset <math>\mathbf{X} = \left[ \mathbf{x}_1 \cdots \mathbf{x}_n \right]</math>, the algorithm will assign each <math>\mathbf{x}_j</math> into <math>k</math> different clusters based on the characteristics of <math>\mathbf{x}_j</math> itself.<br />
<br><br />
<br />
Moreover, when assigning data into a cluster, the algorithm will also try to minimise the distances between the data and the centre of the cluster which the data belongs to. That is, k-means clustering will minimize the sum of square error:<br />
<br />
\begin{align*}<br />
min \sum_{i=1}^{k} \sum_{j \in C_i} ||x_j - \mu_i||^2<br />
\end{align*}<br />
<br />
where<br />
<ul><br />
<li><math>k</math>: the number of clusters</li><br />
<li><math>C_i</math>: the <math>i^th</math> cluster</li><br />
<li><math>x_j</math>: the <math>j^th</math> data in the <math>C_i</math></li><br />
<li><math>mu_i</math>: the centroid of <math>C_i</math></li><br />
<li><math>||x_j - \mu_i||^2</math>: the Euclidean distance between <math>x_j</math> and <math>\mu_i</math></li><br />
</ul><br />
<br><br />
<br />
K-means Clustering algorithm, an unsupervised algorithm, is chosen because of its advantages to deal with different types of attributes, to run with minimal requirement of domain knowledge, to deal with noise and outliers, to realize clusters with similarities. <br />
<br />
<br />
Since the goal for this paper is to classify research papers and group papers with similar topics based on keywords, the paper uses the K-means clustering algorithm. The algorithm first computes the cluster centroid for each group of papers with a specific topic. Then, it will assign a paper into a cluster based on the Euclidean distance between the cluster centroid and the paper’s TF-IDF value.<br />
<br><br />
<br />
However, different values of <math>k</math> (the number of clusters) will return different clustering results. Therefore, it is important to define the number of clusters before clustering. For example, in this paper, the authors choose to use the Elbow scheme to determine the value of <math>k</math>. The Elbow scheme is a somewhat subjective way of choosing an optimal <math>k</math> that involves plotting the average of the squared distances from the cluster centers of the respective clusters (distortion) as a function of <math>k</math> and choosing a <math>k</math> at which point the decrease in distortion is outweighed by the increase in complexity. Also, to measure the performance of clustering, the authors decide to use the Silhouette scheme. The results of clustering are validated if the Silhouette scheme returns a value greater than <math>0.5</math>.<br />
<br />
=System Testing Results=<br />
<br />
In this paper, the dataset has 3264 research papers from the Future Generation Computer System (FGCS) journal between 1984 and 2017. For constructing keyword dictionaries for each paper, the authors have introduced three methods as shown below:<br />
<br />
<div align="center">[[File:table_3_tmtckd.JPG|700px]]</div><br />
<br />
<br />
Then, the authors use the Elbow scheme to define the number of clusters for each method with different numbers of keywords before running the K-means clustering algorithm. The results are shown below:<br />
<br />
<div align="center">[[File:table_4_nocobes.JPG|700px]]</div><br />
<br />
According to Table 4, there is a positive correlation between the number of keywords and the number of clusters. In addition, method 3 combines the advantages for both method 1 and method 2; thus, method 3 requires the least clusters in total. On the other hand, the wrong keywords might be presented in papers; hence, it might not be possible to group papers with similar subjects correctly by using method 1 and so method 1 needs the most number of clusters in total.<br />
<br />
<br />
Next, the Silhouette scheme had been used for measuring the performance for clustering. The average of the Silhouette values for each method with different numbers of keywords are shown below:<br />
<br />
<div align="center">[[File:table_5_asv.JPG|700px]]</div><br />
<br />
Since the clustering is validated if the Silhouette’s value is greater than 0.5, for methods with 10 and 30 keywords, the K-means clustering algorithm produces good results.<br />
<br />
<br />
To evaluate the accuracy of the classification system in this paper, the authors use the F-Score. The authors execute 5 times of experiment and use 500 randomly selected research papers for each trial. The following histogram shows the average value of F-Score for the three methods and different numbers of keywords:<br />
<br />
<div align="center">[[File:fig_16_fsvotm.JPG|700px]]</div><br />
<br />
Note that “TFIDF” means method 1, “LDA” means method 2, and “TFIDF-LDA” means method 3. The number 10, 20, and 30 after each method is the number of keywords the method has used.<br />
According to the histogram above, method 3 has the highest F-Score values than the other two methods with different numbers of keywords. Therefore, the classification system is most accurate when using method 3 as it combines the advantages for both method 1 and method 2.<br />
<br />
=Conclusion=<br />
<br />
This paper introduces a classification system that classifies research papers into different topics by using TF-IDF and LDA scheme with K-means clustering algorithm. The experimental results showed that the proposed system can classify the papers with similar subjects according to the keywords extracted from the abstracts of papers. The authors emphasized that the system can be implemented efficiently on high performance computing infrastructure, using industry-standard technologies. This system allows users to search the papers they want quickly and with the most productivity.<br />
<br />
Furthermore, this classification system might be also used in different types of texts (e.g. documents, tweets, etc.) instead of only classifying research papers.<br />
<br />
=Critique=<br />
<br />
In this paper, DF values are calculated within each partition. This results that for each partition, DF value for a given word will vary and may have an inconsistent result for different partition methods. As mentioned above, there might be a divide by zero problem since some partitions do not have documents containing a given word, but this can be solved by introducing a dummy document as the authors did. Another method that might be better at solving inconsistent results and the divide by zero problems is to have all partitions to communicate with their DF value. Then pass the merged DF value to all partitions to do the final IDF and TF-IDF value. Having all partitions to communicate with the DF value will guarantee a consistent DF value across all partitions and helps avoid a divide by zero problem as words in the keyword dictionary must appear in some documents in the whole collection.<br />
<br />
This paper treated the words in the different parts of a document equivalently, it might perform better if it gives different weights to the same word in different parts. For example, if a word appears in the title of the document, it usually shows it's a main topic of this document so we can put more weight on it to categorize.<br />
<br />
When discussing the potential processing advantages of this classification system for other types of text samples, has the effect of processing mixed samples (text and image or text and video) taken into consideration? IF not, in terms of text classification only, does it have an overwhelming advantage over traditional classification models?<br />
<br />
The preprocessing should also include <math>n</math>-gram tokenization for topic modelling because some topics are inherently two words, such as machine learning where if it is seen separately, it implies different topics.<br />
<br />
This system is very compute-intensive due to the large volumes of dictionaries that can be generated by processing large volumes of data. It would be nice to see how much data HDFS had to process and similarly how much time was saved by using Hadoop for data processing as opposed to centralized approach.<br />
<br />
This system can be improved further in terms of computation times by utilizing other big data framework MapReduce, that can also use HDFS, by parallelizing their computation across multiple nodes for K-means clustering as discussed in (Jin, et al) [5].<br />
<br />
It's not exactly clear what method 3 (TFIDF-LDA) is doing, how is it performing TF-IDF on the topics? Also it seems like the preprocessing step only keeps 10/20/30 top words? This seems like an extremely low number especially in comparison with the LDA which has 10/20/30 topics - what is the reason for so strongly limiting the number of words? It would also be interesting to see if both key words and topics are necessary - an ablation study showing the significance of both would be interesting.<br />
<br />
It is better if the paper has an example with some topics on some research papers. Also it is better if we can visualize the distance between each research paper and the topic names<br />
<br />
I am interested in the first step of the general framework, which is the Paper Crawling step. Many conferences actually require the authors to indicate several key words that best describe a paper. For example, a database paper may have keywords such as "large-scale database management", "information retrieval", and "relational table mining". So in addition to crawling text from abstract, it may be more effective to crawl these keywords directly. Not only does this require less time, these keywords may also lead to better performance than the nouns extracted from the abstract section. I am also slightly concerned about the claim made in the paper that "Our methodologies can be applied to text outside of research papers". Research papers are usually carefully revised and well-structured. Extending the algorithm described in the paper to any kind of free-text could be difficult in practice.<br />
<br />
It would be better if the author could provide some application or example of the research algorithm in the real world. It would be helpful for the readers to understand the algorithm.<br />
<br />
The summary clearly goes through the model framework well, starting from data-preprocessing, prediction, and testing. It can be enhanced by applying this model to other similar use-cases and how well the prediction goes.<br />
<br />
It will be better if their is a comparison on the BM25 algorithm v.s. TF-IDF, which is usually get compared in IR papers<br />
<br />
=References=<br />
<br />
Blei DM, el. (2003). Latent Dirichlet allocation. J Mach Learn Res 3:993–1022<br />
<br />
Gil, JM, Kim, SW. (2019). Research paper classification systems based on TF-IDF and LDA schemes. ''Human-centric Computing and Information Sciences'', 9, 30. https://doi.org/10.1186/s13673-019-0192-7<br />
<br />
Liu, S. (2019, January 11). Dirichlet distribution Motivating LDA. Retrieved November 2020, from https://towardsdatascience.com/dirichlet-distribution-a82ab942a879<br />
<br />
Serrano, L. (Director). (2020, March 18). Latent Dirichlet Allocation (Part 1 of 2) [Video file]. Retrieved 2020, from https://www.youtube.com/watch?v=T05t-SqKArY<br />
<br />
Jin, Cui, Yu. (2016). A New Parallelization Method for K-means. https://arxiv.org/ftp/arxiv/papers/1608/1608.06347.pdf</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=Research_Papers_Classification_System&diff=49341Research Papers Classification System2020-12-06T07:52:12Z<p>Iaoellme: /* HDFS */</p>
<hr />
<div>= Presented by =<br />
Jill Wang, Junyi (Jay) Yang, Yu Min (Chris) Wu, Chun Kit (Calvin) Li<br />
<br />
= Introduction =<br />
With the increasing advance of computer science and information technology, there is an increasingly overwhelming number of papers that have been published. Because of the mass number of papers, it has become incredibly hard to find and categorize papers. This paper introduces a paper classification system that utilizes the Term Frequency-Inverse Document Frequency (TF-IDF), Latent Dirichlet Allocation (LDA), and K-means clustering. The most important technology the system used to process big data is the Hadoop Distributed File System (HDFS). The system can handle quantitatively complex research paper classification problems efficiently and accurately.<br />
<br />
===General Framework===<br />
<br />
The paper classification system classifies research papers based on the abstracts given that the core of most papers is presented in the abstracts. <br />
<br />
[[ File:Systemflow.png |right|image on right| 400px]]<br />
<ol><li>Paper Crawling <br />
<p>Collects abstracts from research papers published during a given period</p></li><br />
<li>Preprocessing<br />
<p> <ol style="list-style-type:lower-alpha"><li>Removes stop words in the papers crawled, in which only nouns are extracted from the papers</li><br />
<li>generates a keyword dictionary, keeping only the top-N keywords with the highest frequencies</li> </ol><br />
</p></li> <br />
<li>Topic Modelling<br />
<p> Use the LDA to group the keywords into topics</p><br />
</li><br />
<li>Paper Length Calculation<br />
<p> Calculates the total number of occurrences of words to prevent an unbalanced TF values caused by the various length of abstracts using the map-reduce algorithm</p><br />
</li><br />
<li>Word Frequency Calculation<br />
<p> Calculates the Term Frequency (TF) values which represent the frequency of keywords in a research paper</p><br />
</li><br />
<li>Document Frequency Calculation<br />
<p> Calculates the Document Frequency (DF) values which represents the frequency of keywords in a collection of research papers. The higher the DF value, the lower the importance of a keyword.</p><br />
</li><br />
<li>TF-IDF calculation<br />
<p> Calculates the inverse of the DF which represents the importance of a keyword.</p><br />
</li><br />
<li>Paper Classification<br />
<p> Classify papers by topics using the K-means clustering algorithm.</p><br />
</li><br />
</ol><br />
<br />
<br />
===Technologies===<br />
<br />
The HDFS with a Hadoop cluster composed of one master node, one sub-node, and four data nodes is what is used to process the massive paper data. Hadoop-2.6.5 version in Java is what is used to perform the TF-IDF calculation. Spark MLlib is what is used to perform the LDA. The Scikit-learn library is what is used to perform the K-means clustering.<br />
<br />
===HDFS===<br />
<br />
Hadoop Distributed File System (HDFS) was used to process big data in this system. HFDS has been shown to process big data rapidly and stably with high scalability which makes it a perfect choice for this problem. What Hadoop does is to break a big collection of data into different partitions and pass each partition to one individual processor. Each processor will only have information about the partition of data it has received.<br />
<br />
'''In this summary, we are going to focus on introducing the main algorithms of what this system uses, namely LDA, TF-IDF, and K-Means.'''<br />
<br />
=Data Preprocessing=<br />
===Crawling of Abstract Data===<br />
<br />
Under the assumption that audiences tend to first read the abstract of a paper to gain an overall understanding of the material, it is reasonable to assume the abstract section includes “core words” that can be used to effectively classify a paper's subject.<br />
<br />
An abstract is crawled to have its stop words removed. Stop words are words that are usually ignored by search engines, such as “the”, “a”, and etc. Afterward, nouns are extracted, as a more condensed representation for efficient analysis.<br />
<br />
This is managed on HDFS. The TF-IDF value of each paper is calculated through map-reduce, an easy-to-use programming model and implementation for processing and generating large data sets. The user must specify (i) a map procedure, that filters and sorts the input data to produce a set of intermediate key/value pairs and (ii) a reduce function, which performs a summary operation on the intermediate values with the same key and returns a smaller set of output key/value pairs. The MapReduce interface enables this process by grouping the intermediate values with the same key and passing them as input to the reduce function. For example, one could count the number of times various words appear in a large number of documents by setting your map procedure to count the number of occurrences of each word in a single document, and your reduce function to sum all counts of a given word [[https://dl.acm.org/doi/pdf/10.1145/1327452.1327492?casa_token=_Zg_DWxQzKEAAAAA:EHII0CaP36_ojGMT8huqTGLNMSEc-CKzZAoXBxSXe6pr2WB0DCQvEKa30CFQW0NSbB2-CVo8GcBcJAg 1]].<br />
<br />
===Managing Paper Data===<br />
<br />
To construct an effective keyword dictionary using abstract data and keywords data in all of the crawled papers, the authors categorized keywords with similar meanings using a single representative keyword. The approach is called stemming, which is common in cleaning data, where words are reduced to their word stem. An example is "running" and "ran" would be reduced to "run". 1394 keyword categories are extracted, which is still too much to compute. Hence, only the top 30 keyword categories are used.<br />
<br />
<div align="center">[[File:table_1_kswf.JPG|700px]]</div><br />
<br />
=Topic Modeling Using LDA=<br />
<br />
Latent Dirichlet allocation (LDA) is a generative probabilistic model that views documents as random mixtures over latent topics. Each topic is a distribution over words, and the goal is to extract these topics from documents.<br />
<br />
LDA estimates the topic-word distribution <math>P\left(t | z\right)</math> (probability of word "z" having topic "t") and the document-topic distribution <math>P\left(z | d\right)</math> (probability of finding word "z" within a given document "d") using Dirichlet priors for the distributions with a fixed number of topics. For each document, obtain a feature vector:<br />
<br />
\[F = \left( P\left(z_1 | d\right), P\left(z_2 | d\right), \cdots, P\left(z_k | d\right) \right)\]<br />
<br />
In the paper, authors extract topics from preprocessed paper to generate three kinds of topic sets, each with 10, 20, and 30 topics respectively. The following is a table of the 10 topic sets of highest frequency keywords.<br />
<br />
<div align="center">[[File:table_2_tswtebls.JPG|700px]]</div><br />
<br />
<br />
===LDA Intuition===<br />
<br />
LDA uses the Dirichlet priors of the Dirichlet distribution, which allows the algorithm to model a probability distribution ''over prior probability distributions of words and topics''. The following picture illustrates 2-simplex Dirichlet distributions with different alpha values, one for each corner of the triangles. <br />
<br />
<div align="center">[[File:dirichlet_dist.png|700px]]</div><br />
<br />
Simplex is a generalization of the notion of a triangle in k-1 dimension where k is the number of classes. For example, if you wish to classify essays into three groups, English, History and Math then the simplex would be a 2 dimension triangle, if you add philosophy as one of your potential class, then we would need a tetrahedron in 3 deminsion. In Dirichlet distribution, each parameter will be represented by a corner in simplex, so adding additional parameters implies increasing the dimensions of simplex. As illustrated, when alphas are smaller than 1 the distribution is dense at the corners. When the alphas are greater than 1 the distribution is dense at the centers.<br />
<br />
The following illustration shows an example LDA with 3 topics, 4 words and 7 documents.<br />
<br />
<div align="center">[[File:LDA_example.png|800px]]</div><br />
<br />
In the left diagram, there are three topics, hence it is a 2-simplex. In the right diagram there are four words, hence it is a 3-simplex. LDA essentially adjusts parameters in Dirichlet distributions and multinomial distributions (represented by the points), such that, in the left diagram, all the yellow points representing documents and, in the right diagram, all the points representing topics, are as close to a corner as possible. In other words, LDA finds topics for documents and also finds words for topics. At the end topic-word distribution <math>P\left(t | z\right)</math> and the document-topic distribution <math>P\left(z | d\right)</math> are produced.<br />
<br />
=Term Frequency Inverse Document Frequency (TF-IDF) Calculation=<br />
<br />
TF-IDF is widely used to evaluate the importance of a set of words in the fields of information retrieval and text mining. It is a combination of term frequency (TF) and inverse document frequency (IDF). The idea behind this combination is<br />
* It evaluates the importance of a word within a document<br />
* It evaluates the importance of the word among the collection of all documents<br />
<br />
The inverse of the document frequency accounts for the fact that term frequency will naturally increase as document frequency increases. Thus IDF is needed to counteract a word's TF to give an accurate representation of a word's importance.<br />
<br />
The TF-IDF formula has the following form:<br />
<br />
\[TF-IDF_{i,j} = TF_{i,j} \times IDF_{i}\]<br />
<br />
where i stands for the <math>i^{th}</math> word and j stands for the <math>j^{th}</math> document.<br />
<br />
===Term Frequency (TF)===<br />
<br />
TF evaluates the percentage of a given word in a document. Thus, TF value indicates the importance of a word. The TF has a positive relation with the importance.<br />
<br />
In this paper, we only calculate TF for words in the keyword dictionary obtained. For a given keyword i, <math>TF_{i,j}</math> is the number of times word i appears in document j divided by the total number of words in document j.<br />
<br />
The formula for TF has the following form:<br />
<br />
\[TF_{i,j} = \frac{n_{i,j} }{\sum_k n_{k,j} }\]<br />
<br />
where i stands for the <math>i^{th}</math> word, j stands for the <math>j^{th}</math> document, <math>n_{i,j}</math> stands for the number of times words <math>t_i</math> appear in document <math>d_j</math> and <math>\sum_k n_{k,j} </math> stands for total number of occurence of words in document <math>d_j</math>.<br />
<br />
Note that the denominator is the total number of words remaining in document j after crawling.<br />
<br />
===Document Frequency (DF)===<br />
<br />
DF evaluates the percentage of documents that contain a given word over the entire collection of documents. Thus, the higher DF value is, the less important the word is.<br />
<br />
<math>DF_{i}</math> is the number of documents in the collection with word i divided by the total number of documents in the collection. The formula for DF has the following form:<br />
<br />
\[DF_{i} = \frac{|d_k \in D: n_{i,k} > 0|}{|D|}\]<br />
<br />
where <math>n_{i,k}</math> is the number of times word i appears in document k, |D| is the total number of documents in the collection.<br />
<br />
Since DF and the importance of the word have an inverse relation, we use inverse document frequency (IDF) instead of DF.<br />
<br />
===Inverse Document Frequency (IDF)===<br />
<br />
In this paper, IDF is calculated in a log scale. Since we will receive a large number of documents, i.e, we will have a large |D|<br />
<br />
The formula for IDF has the following form:<br />
<br />
\[IDF_{i} = log\left(\frac{|D|}{|\{d_k \in D: n_{i,k} > 0\}|}\right)\]<br />
<br />
As mentioned before, we will use HDFS. The actual formula applied is:<br />
<br />
\[IDF_{i} = log\left(\frac{|D|+1}{|\{d_k \in D: n_{i,k} > 0\}|+1}\right)\]<br />
<br />
The inverse document frequency gives a measure of how rare a certain term is in a given document corpus.<br />
<br />
=Paper Classification Using K-means Clustering=<br />
<br />
The K-means clustering is an unsupervised classification algorithm that groups similar data into the same class. It is an efficient and simple method that can be applied to different types of data attributes. It is also flexible enough to handle various kinds of noise and outliers.<br />
<br><br />
<br />
Given a set of <math>d</math> by <math>n</math> dataset <math>\mathbf{X} = \left[ \mathbf{x}_1 \cdots \mathbf{x}_n \right]</math>, the algorithm will assign each <math>\mathbf{x}_j</math> into <math>k</math> different clusters based on the characteristics of <math>\mathbf{x}_j</math> itself.<br />
<br><br />
<br />
Moreover, when assigning data into a cluster, the algorithm will also try to minimise the distances between the data and the centre of the cluster which the data belongs to. That is, k-means clustering will minimize the sum of square error:<br />
<br />
\begin{align*}<br />
min \sum_{i=1}^{k} \sum_{j \in C_i} ||x_j - \mu_i||^2<br />
\end{align*}<br />
<br />
where<br />
<ul><br />
<li><math>k</math>: the number of clusters</li><br />
<li><math>C_i</math>: the <math>i^th</math> cluster</li><br />
<li><math>x_j</math>: the <math>j^th</math> data in the <math>C_i</math></li><br />
<li><math>mu_i</math>: the centroid of <math>C_i</math></li><br />
<li><math>||x_j - \mu_i||^2</math>: the Euclidean distance between <math>x_j</math> and <math>\mu_i</math></li><br />
</ul><br />
<br><br />
<br />
K-means Clustering algorithm, an unsupervised algorithm, is chosen because of its advantages to deal with different types of attributes, to run with minimal requirement of domain knowledge, to deal with noise and outliers, to realize clusters with similarities. <br />
<br />
<br />
Since the goal for this paper is to classify research papers and group papers with similar topics based on keywords, the paper uses the K-means clustering algorithm. The algorithm first computes the cluster centroid for each group of papers with a specific topic. Then, it will assign a paper into a cluster based on the Euclidean distance between the cluster centroid and the paper’s TF-IDF value.<br />
<br><br />
<br />
However, different values of <math>k</math> (the number of clusters) will return different clustering results. Therefore, it is important to define the number of clusters before clustering. For example, in this paper, the authors choose to use the Elbow scheme to determine the value of <math>k</math>. The Elbow scheme is a somewhat subjective way of choosing an optimal <math>k</math> that involves plotting the average of the squared distances from the cluster centers of the respective clusters (distortion) as a function of <math>k</math> and choosing a <math>k</math> at which point the decrease in distortion is outweighed by the increase in complexity. Also, to measure the performance of clustering, the authors decide to use the Silhouette scheme. The results of clustering are validated if the Silhouette scheme returns a value greater than <math>0.5</math>.<br />
<br />
=System Testing Results=<br />
<br />
In this paper, the dataset has 3264 research papers from the Future Generation Computer System (FGCS) journal between 1984 and 2017. For constructing keyword dictionaries for each paper, the authors have introduced three methods as shown below:<br />
<br />
<div align="center">[[File:table_3_tmtckd.JPG|700px]]</div><br />
<br />
<br />
Then, the authors use the Elbow scheme to define the number of clusters for each method with different numbers of keywords before running the K-means clustering algorithm. The results are shown below:<br />
<br />
<div align="center">[[File:table_4_nocobes.JPG|700px]]</div><br />
<br />
According to Table 4, there is a positive correlation between the number of keywords and the number of clusters. In addition, method 3 combines the advantages for both method 1 and method 2; thus, method 3 requires the least clusters in total. On the other hand, the wrong keywords might be presented in papers; hence, it might not be possible to group papers with similar subjects correctly by using method 1 and so method 1 needs the most number of clusters in total.<br />
<br />
<br />
Next, the Silhouette scheme had been used for measuring the performance for clustering. The average of the Silhouette values for each method with different numbers of keywords are shown below:<br />
<br />
<div align="center">[[File:table_5_asv.JPG|700px]]</div><br />
<br />
Since the clustering is validated if the Silhouette’s value is greater than 0.5, for methods with 10 and 30 keywords, the K-means clustering algorithm produces good results.<br />
<br />
<br />
To evaluate the accuracy of the classification system in this paper, the authors use the F-Score. The authors execute 5 times of experiment and use 500 randomly selected research papers for each trial. The following histogram shows the average value of F-Score for the three methods and different numbers of keywords:<br />
<br />
<div align="center">[[File:fig_16_fsvotm.JPG|700px]]</div><br />
<br />
Note that “TFIDF” means method 1, “LDA” means method 2, and “TFIDF-LDA” means method 3. The number 10, 20, and 30 after each method is the number of keywords the method has used.<br />
According to the histogram above, method 3 has the highest F-Score values than the other two methods with different numbers of keywords. Therefore, the classification system is most accurate when using method 3 as it combines the advantages for both method 1 and method 2.<br />
<br />
=Conclusion=<br />
<br />
This paper introduces a classification system that classifies research papers into different topics by using TF-IDF and LDA scheme with K-means clustering algorithm. The experimental results showed that the proposed system can classify the papers with similar subjects according to the keywords extracted from the abstracts of papers. This system allows users to search the papers they want quickly and with the most productivity.<br />
<br />
Furthermore, this classification system might be also used in different types of texts (e.g. documents, tweets, etc.) instead of only classifying research papers.<br />
<br />
=Critique=<br />
<br />
In this paper, DF values are calculated within each partition. This results that for each partition, DF value for a given word will vary and may have an inconsistent result for different partition methods. As mentioned above, there might be a divide by zero problem since some partitions do not have documents containing a given word, but this can be solved by introducing a dummy document as the authors did. Another method that might be better at solving inconsistent results and the divide by zero problems is to have all partitions to communicate with their DF value. Then pass the merged DF value to all partitions to do the final IDF and TF-IDF value. Having all partitions to communicate with the DF value will guarantee a consistent DF value across all partitions and helps avoid a divide by zero problem as words in the keyword dictionary must appear in some documents in the whole collection.<br />
<br />
This paper treated the words in the different parts of a document equivalently, it might perform better if it gives different weights to the same word in different parts. For example, if a word appears in the title of the document, it usually shows it's a main topic of this document so we can put more weight on it to categorize.<br />
<br />
When discussing the potential processing advantages of this classification system for other types of text samples, has the effect of processing mixed samples (text and image or text and video) taken into consideration? IF not, in terms of text classification only, does it have an overwhelming advantage over traditional classification models?<br />
<br />
The preprocessing should also include <math>n</math>-gram tokenization for topic modelling because some topics are inherently two words, such as machine learning where if it is seen separately, it implies different topics.<br />
<br />
This system is very compute-intensive due to the large volumes of dictionaries that can be generated by processing large volumes of data. It would be nice to see how much data HDFS had to process and similarly how much time was saved by using Hadoop for data processing as opposed to centralized approach.<br />
<br />
This system can be improved further in terms of computation times by utilizing other big data framework MapReduce, that can also use HDFS, by parallelizing their computation across multiple nodes for K-means clustering as discussed in (Jin, et al) [5].<br />
<br />
It's not exactly clear what method 3 (TFIDF-LDA) is doing, how is it performing TF-IDF on the topics? Also it seems like the preprocessing step only keeps 10/20/30 top words? This seems like an extremely low number especially in comparison with the LDA which has 10/20/30 topics - what is the reason for so strongly limiting the number of words? It would also be interesting to see if both key words and topics are necessary - an ablation study showing the significance of both would be interesting.<br />
<br />
It is better if the paper has an example with some topics on some research papers. Also it is better if we can visualize the distance between each research paper and the topic names<br />
<br />
I am interested in the first step of the general framework, which is the Paper Crawling step. Many conferences actually require the authors to indicate several key words that best describe a paper. For example, a database paper may have keywords such as "large-scale database management", "information retrieval", and "relational table mining". So in addition to crawling text from abstract, it may be more effective to crawl these keywords directly. Not only does this require less time, these keywords may also lead to better performance than the nouns extracted from the abstract section. I am also slightly concerned about the claim made in the paper that "Our methodologies can be applied to text outside of research papers". Research papers are usually carefully revised and well-structured. Extending the algorithm described in the paper to any kind of free-text could be difficult in practice.<br />
<br />
It would be better if the author could provide some application or example of the research algorithm in the real world. It would be helpful for the readers to understand the algorithm.<br />
<br />
The summary clearly goes through the model framework well, starting from data-preprocessing, prediction, and testing. It can be enhanced by applying this model to other similar use-cases and how well the prediction goes.<br />
<br />
It will be better if their is a comparison on the BM25 algorithm v.s. TF-IDF, which is usually get compared in IR papers<br />
<br />
=References=<br />
<br />
Blei DM, el. (2003). Latent Dirichlet allocation. J Mach Learn Res 3:993–1022<br />
<br />
Gil, JM, Kim, SW. (2019). Research paper classification systems based on TF-IDF and LDA schemes. ''Human-centric Computing and Information Sciences'', 9, 30. https://doi.org/10.1186/s13673-019-0192-7<br />
<br />
Liu, S. (2019, January 11). Dirichlet distribution Motivating LDA. Retrieved November 2020, from https://towardsdatascience.com/dirichlet-distribution-a82ab942a879<br />
<br />
Serrano, L. (Director). (2020, March 18). Latent Dirichlet Allocation (Part 1 of 2) [Video file]. Retrieved 2020, from https://www.youtube.com/watch?v=T05t-SqKArY<br />
<br />
Jin, Cui, Yu. (2016). A New Parallelization Method for K-means. https://arxiv.org/ftp/arxiv/papers/1608/1608.06347.pdf</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=Improving_neural_networks_by_preventing_co-adaption_of_feature_detectors&diff=49339Improving neural networks by preventing co-adaption of feature detectors2020-12-06T07:29:11Z<p>Iaoellme: /* Improvement Intro */</p>
<hr />
<div>== Presented by ==<br />
Stan Lee, Seokho Lim, Kyle Jung, Dae Hyun Kim<br />
<br />
= Improvement Intro =<br />
'''Drop Out Model'''<br />
<br />
In this paper, Hinton et al. introduce a novel way to improve neural networks’ performance, particularly in the case that a large feedforward neural network is trained on a small training set, which causes poor performance and leads to “overfitting” problem. This problem can be reduced by randomly omitting half of the feature detectors on each training case. In fact, By omitting neurons in hidden layers with a probability of 0.5, each hidden unit is prevented from relying on other hidden units being present during training. Hence there are fewer co-adaptations among them on the training data. Called “dropout,” this process is also an efficient alternative to train many separate networks and average their predictions on the test set. <br />
<br />
The intuition for dropout is that if neurons are randomly dropped during training, they can no longer rely on their neighbours, thus allowing each neuron to become more robust. Another interpretation is that dropout is similar to training an ensemble of models, since each epoch with randomly dropped neurons can be viewed as its own model. <br />
<br />
They used the standard, stochastic gradient descent algorithm and separated training data into mini-batches. An upper bound was set on the L2 norm of incoming weight vector for each hidden neuron, which was normalized if its size exceeds the bound. They found that using a constraint, instead of a penalty, forced the model to do a more thorough search of the weight-space, when coupled with the very large learning rate that decays during training.<br />
<br />
'''Mean Network'''<br />
<br />
Their dropout models included all of the hidden neurons, and their outgoing weights were halved to account for the chances of omission. This is called a 'Mean Network'. This is similar to taking the geometric mean of the probability distribution predicted by all <math>2^N</math> networks. Due to this cumulative addition, the correct answers will have higher log probability than an individual dropout network, which also leads to a lower squared error of the network. <br />
<br />
<br />
The models were shown to result in lower test error rates on several datasets: MNIST; TIMIT; Reuters Corpus Volume; CIFAR-10; and ImageNet.<br />
<br />
= MNIST =<br />
The MNIST dataset contains 70,000 digit images of size 28 x 28. To see the impact of dropout, they used 4 different neural networks (784-800-800-10, 784-1200-1200-10, 784-2000-2000-10, 784-1200-1200-1200-10), with the same dropout rates, 50%, for hidden neurons and 20% for visible neurons. Stochastic gradient descent was used with mini-batches of size 100 and a cross-entropy objective function as the loss function. Weights were updated after each minibatch, and training was done for 3000 epochs. An exponentially decaying learning rate <math>\epsilon</math> was used, with the initial value set as 10.0, and it was multiplied by the decaying factor <math>f</math> = 0.998 at the end of each epoch. At each hidden layer, the incoming weight vector for each hidden neuron was set an upper bound of its length, <math>l</math>, and they found from cross-validation that the results were the best when <math>l</math> = 15. Initial weights values were pooled from a normal distribution with mean 0 and standard deviation of 0.01. To update weights, an additional variable, ''p'', called momentum, was used to accelerate learning. The initial value of <math>p</math> was 0.5, and it increased linearly to the final value 0.99 during the first 500 epochs, remaining unchanged after. Also, when updating weights, the learning rate was multiplied by <math>1 – p</math>. <math>L</math> denotes the gradient of loss function.<br />
<br />
[[File:weights_mnist2.png|center|400px]]<br />
<br />
The best published result for a standard feedforward neural network was 160 errors. This was reduced to about 130 errors with 0.5 dropout and different L2 constraints for each hidden unit input weight. By omitting a random 20% of the input pixels in addition to the aforementioned changes, the number of errors was further reduced to 110. The following figure visualizes the result.<br />
[[File:mnist_figure.png|center|500px]]<br />
A publicly available pre-trained deep belief net resulted in 118 errors, and it was reduced to 92 errors when the model was fine-tuned with dropout. Another publicly available model was a deep Boltzmann machine, and it resulted in 103, 97, 94, 93 and 88 when the model was fine-tuned using standard backpropagation and was unrolled. They were reduced to 83, 79, 78, 78, and 77 when the model was fine-tuned with dropout – the mean of 79 errors was a record for models that do not use prior knowledge or enhanced training sets.<br />
<br />
= TIMIT = <br />
<br />
TIMIT dataset includes voice samples of 630 American English speakers varying across 8 different dialects. It is often used to evaluate the performance of automatic speech recognition systems. Using Kaldi, the dataset was pre-processed to extract input features in the form of log filter bank responses.<br />
<br />
=== Pre-training and Training ===<br />
<br />
For pretraining, they pretrained their neural network with a deep belief network and the first layer was built using Restricted Boltzmann Machine (RBM). Initializing visible biases with zero, weights were sampled from random numbers that followed normal distribution <math>N(0, 0.01)</math>. Each visible neuron’s variance was set to 1.0 and remained unchanged.<br />
<br />
Minimizing Contrastive Divergence (CD) was used to facilitate learning. Since momentum is used to speed up learning, it was initially set to 0.5 and increased linearly to 0.9 over 20 epochs. The average gradient had 0.001 of a learning rate which was then multiplied by <math>(1-momentum)</math> and L2 weight decay was set to 0.001. After setting up the hyperparameters, the model was done training after 100 epochs. Binary RBMs were used for training all subsequent layers with a learning rate of 0.01. Then, <math>p</math> was set as the mean activation of a neuron in the data set and the visible bias of each neuron was initialized to <math>log(p/(1 − p))</math>. Training each layer with 50 epochs, all remaining hyper-parameters were the same as those for the Gaussian RBM.<br />
<br />
=== Dropout tuning ===<br />
<br />
The initial weights were set in a neural network from the pretrained RBMs. To finetune the network with dropout-backpropagation, momentum was initially set to 0.5 and increased linearly up to 0.9 over 10 epochs. The model had a small constant learning rate of 1.0 and it was used to apply to the average gradient on a minibatch. The model also retained all other hyperparameters the same as the model from MNIST dropout finetuning. The model required approximately 200 epochs to converge. For comparison purpose, they also finetuned the same network with standard backpropagation with a learning rate of 0.1 with the same hyperparameters.<br />
<br />
=== Classification Test and Performance ===<br />
<br />
A Neural network was constructed to output the classification error rate on the test set of TIMIT dataset. They have built the neural network with four fully-connected hidden layers with 4000 neurons per layer. The output layer distinguishes distinct classes from 185 softmax output neurons that are merged into 39 classes. After constructing the neural network, 21 adjacent frames with an advance of 10ms per frame was given as an input.<br />
<br />
Comparing the performance of dropout with standard backpropagation on several network architectures and input representations, dropout consistently achieved lower error and cross-entropy. Results showed that it significantly controls overfitting, making the method robust to choices of network architecture. It also allowed much larger nets to be trained and removed the need for early stopping. Thus, neural network architectures with dropout are not very sensitive to the choice of learning rate and momentum.<br />
<br />
= Reuters Corpus Volume =<br />
Reuters Corpus Volume I archives 804,414 news documents that belong to 103 topics. Under four major themes - corporate/industrial, economics, government/social, and markets – they belonged to 63 classes. After removing 11 classes with no data and one class with insufficient data, they are left with 50 classes and 402,738 documents. The documents were divided into training and test sets equally and randomly, with each document representing the 2000 most frequent words in the dataset, excluding stopwords.<br />
<br />
They trained two neural networks, with size 2000-2000-1000-50, one using dropout and backpropagation, and the other using standard backpropagation. The training hyperparameters are the same as that in MNIST, but training was done for 500 epochs.<br />
<br />
In the following figure, we see the significant improvements by the model with dropout in the test set error. On the right side, we see that learning with dropout also proceeds smoother. <br />
<br />
[[File:reuters_figure.png|700px|center]]<br />
<br />
= CNN =<br />
<br />
Feed-forward neural networks consist of several layers of neurons where each neuron in a layer applies a linear filter to the input image data and is passed on to the neurons in the next layer. When calculating the neuron’s output, scalar bias a.k.a weights is applied to the filter with nonlinear activation function as parameters of the network that are learned by training data. [[File:cnnbigpicture.jpeg|thumb|upright=2|center|alt=text|Figure: Overview of Convolutional Neural Network]] There are several differences between Convolutional Neural networks and ordinary neural networks. The figure above gives a visual representation of a Convolutional Neural Network. First, CNN’s neurons are organized topographically into a bank and laid out on a 2D grid, so it reflects the organization of dimensions of the input data. Secondly, neurons in CNN apply filters which are local, and which are centered at the neuron’s location in the topographic organization. Meaning that useful metrics or clues to identify the object in an input image which can be found by examining local neighborhoods of the image. Next, all neurons in a bank apply the same filter at different locations in the input image. When looking at the image example, green is an input to one neuron bank, yellow is filter bank, and pink is the output of one neuron bank (convolved feature). A bank of neurons in a CNN applies a convolution operation, aka filters, to its input where a single layer in a CNN typically has multiple banks of neurons, each performing a convolution with a different filter. The resulting neuron banks become distinct input channels into the next layer. The whole process reduces the net’s representational capacity, but also reduces the capacity to overfit.<br />
[[File:bankofneurons.gif|thumb|upright=3|center|alt=text|Figure: Bank of neurons]]<br />
<br />
=== Pooling ===<br />
<br />
Pooling layer summarizes the activities of local patches of neurons in the convolutional layer by subsampling the output of a convolutional layer. Pooling is useful for extracting dominant features, to decrease the computational power required to process the data through dimensionality reduction. The procedure of pooling goes on like this; output from convolutional layers is divided into sections called pooling units and they are laid out topographically, connected to a local neighborhood of other pooling units from the same convolutional output. Then, each pooling unit is computed with some function which could be maximum and average. Maximum pooling returns the maximum value from the section of the image covered by the pooling unit while average pooling returns the average of all the values inside the pooling unit (see example). In result, there are fewer total pooling units than convolutional unit outputs from the previous layer, this is due to larger spacing between pixels on pooling layers. Using the max-pooling function reduces the effect of outliers and improves generalization. Other than that, overlapping pooling makes this spacing between pixels smaller than the size of the neighborhood that the pooling units summarize (This spacing is usually referred as the stride between pooling units). With this variant, pooling layer can produce a coarse coding of the outputs which helps generalization. <br />
[[File:maxandavgpooling.jpeg|thumb|upright=2|center|alt=text|Figure: Max pooling and Average pooling]]<br />
<br />
=== Local Response Normalization === <br />
<br />
This network includes local response normalization layers which are implemented in lateral form and used on neurons with unbounded activations and permits the detection of high-frequency features with a big neuron response. This regularizer encourages competition among neurons belonging to different banks. Normalization is done by dividing the activity of a neuron in bank <math>i</math> at position <math>(x,y)</math> by the equation:<br />
[[File:local response norm.png|upright=2|center|]] where the sum runs over <math>N</math> ‘adjacent’ banks of neurons at the same position as in the topographic organization of neuron bank. The constants, <math>N</math>, <math>alpha</math> and <math>betas</math> are hyper-parameters whose values are determined using a validation set. This technique is replaced by better techniques such as the combination of dropout and regularization methods (<math>L1</math> and <math>L2</math>)<br />
<br />
=== Neuron nonlinearities ===<br />
<br />
All of the neurons for this model use the max-with-zero nonlinearity where output within a neuron is computed as <math> a^{i}_{x,y} = max(0, z^i_{x,y})</math> where <math> z^i_{x,y} </math> is the total input to the neuron. The reason they use nonlinearity is because it has several advantages over traditional saturating neuron models, such as significant reduction in training time required to reach a certain error rate. Another advantage is that nonlinearity reduces the need for contrast-normalization and data pre-processing since neurons do not saturate- meaning activities simply scale up little by little with usually large input values. For this model’s only pre-processing step, they subtract the mean activity from each pixel and the result is a centered data.<br />
<br />
=== Objective function ===<br />
<br />
The objective function of their network maximizes the multinomial logistic regression objective which is the same as minimizing the average cross-entropy across training cases between the true label and the model’s predicted label.<br />
<br />
=== Weight Initialization === <br />
<br />
It’s important to note that if a neuron always receives a negative value during training, it will not learn because its output is uniformly zero under the max-with-zero nonlinearity. Hence, the weights in their model were sampled from a zero-mean normal distribution with a high enough variance. High variance in weights will set a certain number of neurons with positive values for learning to happen, and in practice, it’s necessary to try out several candidates for variances until a working initialization is found. In their experiment, setting a positive constant, or 1, as biases of the neurons in the hidden layers was helpful in finding it.<br />
<br />
=== Training ===<br />
<br />
In this model, a batch size of 128 samples and momentum of 0.9, we train our model using stochastic gradient descent. The update rule for weight <math>w</math> is $$ v_{i+1} = 0.9v_i + \epsilon <\frac{dE}{dw_i}> i$$ $$w_{i+1} = w_i + v_{i+1} $$ where <math>i</math> is the iteration index, <math>v</math> is a momentum variable, <math>\epsilon</math> is the learning rate and <math>\frac{dE}{dw}</math> is the average over the <math>i</math>th batch of the derivative of the objective with respect to <math>w_i</math>. The whole training process on CIFAR-10 takes roughly 90 minutes and ImageNet takes 4 days with dropout and two days without.<br />
<br />
=== Learning ===<br />
To determine the learning rate for the network, it is a must to start with an equal learning rate for each layer which produces the largest reduction in the objective function with power of ten. Usually, it is in the order of <math>10^{-2}</math> or <math>10^{-3}</math>. In this case, they reduce the learning rate twice by a factor of ten before termination of training.<br />
<br />
= CIFAR-10 =<br />
<br />
=== CIFAR-10 Dataset ===<br />
<br />
Removing incorrect labels, The CIFAR-10 dataset is a subset of the Tiny Images dataset with 10 classes. It contains 5000 training images and 1000 testing images for each class. The dataset has 32 x 32 color images searched from the web and the images are labeled with the noun used to search the image.<br />
<br />
[[File:CIFAR-10.png|thumb|upright=2|center|alt=text|Figure 4: CIFAR-10 Sample Dataset]]<br />
<br />
=== Models for CIFAR-10 ===<br />
<br />
Two models, one with dropout and one without dropout, were built to test the performance of dropout on CIFAR-10. All models have CNN with three convolutional layers each with a pooling layer. All of the pooling payers use a stride=2 and summarize a 3*3 neighborhood. The max-pooling method is performed by the pooling layer which follows the first convolutional layer, and the average-pooling method is performed by remaining 2 pooling layers. The first and second pooling layers with <math>N = 9, α = 0.001</math>, and <math>β = 0.75</math> are followed by response normalization layers. A ten-unit softmax layer, which is used to output a probability distribution over class labels, is connected with the upper-most pooling layer. Using filter size of 5×5, all convolutional layers have 64 filter banks.<br />
<br />
Additional changes were made with the model with dropout. The model with dropout enables us to use more parameters because dropout forces a strong regularization on the network. Thus, a fourth weight layer is added to take the input from the previous pooling layer. This fourth weight layer is locally connected, but not convolutional, and contains 16 banks of filters of size 3 × 3 with 50% dropout. Lastly, the softmax layer takes its input from this fourth weight layer.<br />
<br />
Thus, with a neural network with 3 convolutional hidden layers with 3 max-pooling layers, the classification error achieved 16.6% to beat 18.5% from the best published error rate without using transformed data. The model with one additional locally-connected layer and dropout at the last hidden layer produced the error rate of 15.6%.<br />
<br />
= ImageNet =<br />
<br />
===ImageNet Dataset===<br />
<br />
ImageNet is a dataset of millions of high-resolution images, and they are labeled among 1000 different categories. The data were collected from the web and manually labeled using MTerk tool, which is a crowd-sourcing tool provided by Amazon.<br />
Because this dataset has millions of labeled images in thousands of categories, it is very difficult to have perfect accuracy on this dataset even for humans because the ImageNet images may contain multiple objects and there are a large number of object classes. ImageNet and CIFAR-10 are very similar, but the scale of ImageNet is about 20 times bigger (1,300,000 vs 60,000). The size of ImageNet is about 1.3 million training images, 50,000 validation images, and 150,000 testing images. They used resized images of 256 x 256 pixels for their experiments.<br />
<br />
'''An ambiguous example to classify:'''<br />
<br />
[[File:imagenet1.png|200px|center]]<br />
<br />
When this paper was written, the best score on this dataset was the error rate of 45.7% by High-dimensional signature compression for large-scale image classification (J. Sanchez, F. Perronnin, CVPR11 (2011)). The authors of this paper could achieve a comparable performance of 48.6% error rate using a single neural network with five convolutional hidden layers with a max-pooling layer in between, followed by two globally connected layers and a final 1000-way softmax layer. When applying 50% dropout to the 6th layer, the error rate was brought down to 42.4%.<br />
<br />
'''ImageNet Dataset:'''<br />
<br />
[[File:imagenet2.png|400px|center]]<br />
<br />
===Models for ImageNet===<br />
<br />
They mostly focused on the model with dropout because the one without dropout had a similar approach, but there was a serious issue with overfitting. They used a convolutional neural network trained by 224×224 patches randomly extracted from the 256 × 256 images. This could reduce the network’s capacity to overfit the training data and helped generalization as a form of data augmentation. The method of averaging the prediction of the net on ten 224 × 224 patches of the 256 × 256 input image was used for testing their model patched at the center, four corners, and their horizontal reflections. To maximize the performance on the validation set, this complicated network architecture was used and it was found that dropout was very effective. Also, it was demonstrated that using non-convolutional higher layers with the number of parameters worked well with dropout, but it had a negative impact to the performance without dropout.<br />
<br />
The network contains seven weight layers. The first five are convolutional, and the last two are globally-connected. Max-pooling layers follow the layer number 1,2, and 5. And then, the output of the last globally-connected layer was fed to a 1000-way softmax output layers. Using this architecture, the authors achieved the error rate of 48.6%. When applying 50% dropout to the 6th layer, the error rate was brought down to 42.4%.<br />
<br />
<br />
[[File:modelh2.png|700px|center]] <br />
<br />
[[File:layer2.png|600px|center]]<br />
<br />
Like the previous datasets, such as the MNIST, TIMIT, Reuters, and CIFAR-10, we also see a significant improvement for the ImageNet dataset. Including complicated architectures like this one, introducing dropout generalizes models better and gives lower test error rates.<br />
<br />
= Conclusion =<br />
<br />
The authors have shown a consistent improvement by the models trained with dropout in classifying objects in the following datasets: MNIST; TIMIT; Reuters Corpus Volume I; CIFAR-10; and ImageNet.<br />
<br />
The authors comment on a theory that sexual reproduction limits biological function to a small number of coadapted genes. The idea is that a given organism is unlikely to receive many coordinated genes from a parent, so will likely die if it relies on many genes to perform a given task. This limits the number of genes required to perform a function, which is like a built-in evolutionary dropout.<br />
<br />
= Critiques =<br />
It is a very brilliant idea to dropout half of the neurons to reduce co-adaptations. It is mentioned that for fully connected layers, dropout in all hidden layers works better than dropout in only one hidden layer. There is another paper Dropout: A Simple Way to Prevent Neural Networks from<br />
Overfitting[https://www.cs.toronto.edu/~hinton/absps/JMLRdropout.pdf] gives a more detailed explanation.<br />
<br />
It will be interesting to see how this paper could be used to prevent overfitting of LSTMs.<br />
<br />
This paper focused more on CV tasks, it will be interesting to have some discussion on NLP tasks<br />
<br />
Firstly, it is a very interested topic of classification by "dropout" CNN method(omitting neurons in hidden layers). If the author can briefly explain the advantages of this method in processing image data in theory, it will be easier for readers to understand. Also, how to deal with overfitting issue would be valuable.<br />
<br />
The authors mention that they tried various dropout probabilities and that the majority of them improved the model's generalization performance, but that more extreme probabilities tended to be worse which is why a dropout rate of 50% was used in the paper. The authors further develop this point to mention that the method can be improved by adapting individual dropout probabilities of each hidden or input unit using validation tests. This would be an interesting area to further develop and explore, as using a hardcoded 50% dropout for all layers might not be the optimal choice for all CNN applications. It would have been interesting to see the results of their investigations of differing dropout rates.<br />
<br />
The authors don't explain that during training, at each layer that we apply dropout, the values must be scaled by 1/p where p is dropout rate - this way the expected value of the layers is the same in both train and test time. They may have considered another solution for this discrepancy at the time (it is an old paper) but it doesn't seem like any solution was presented here. <br />
<br />
Despite the advantages of using dropout to prevent overfitting and reducing errors in testing, the authors did not discuss much about the effects on the length of training time. In another [https://www.cs.toronto.edu/~hinton/absps/JMLRdropout.pdf paper] published a few years later by the same authors, there was more discussion about this. It appears that dropout increases training time by 2-3 times compared to a standard NN with the same architecture, which is a drawback that might be worth mentioning.<br />
<br />
Dropout layers prevent overfitting by randomly dropout a fraction of the neurons specified in each layer. In fact, the neurons to be dropped out in each layer are randomly selected. Therefore, it might be the case that some important features in the dropout layer are discarded, which leads to a sudden drop in performance. Although this barely happens, and CNN with dropout rates roughly 50% in each layer will lead to generally good performance, some future improvements are still possible if we are able to select dropout neurons cleverly.<br />
<br />
The article does a good job of analyzing the benefit of using the standard dropout method, but I think it would be beneficial to take a look at other dropout variants. For example, the paper may have benefited at looking at DropConnect which was introduced y L. Wan et al and is similar to dropout layers but it does not apply a dropout directly o the neurons but on the weights and the bias linking the neurons. Others that they also could have looked at were Standout, Pooling Drop and MaxDrop. Comparing various dropout methods I think would greatly add to the paper.<br />
<br />
The author analyzed the dropout method for addressing overfitting problems. The key idea is to randomly drop units from the neural network during training. This prevents units from co-adapting too much. In addition, it also fastens the speed of training models since there are fewer neurons, which is a good idea. <br />
<br />
Random dropping was indeed quite effective in the MNIST fashion classification challenge, however it may pose a question if the problem has very few features to begin with.<br />
<br />
The authors mentioned that they used Momentum to speed up the training but didn't show the alternative and the speed of the alternative. This [https://link.springer.com/article/10.1007/s11042-019-08453-9 paper]conducts an empirical study of Dropout vs Batch Normalization as well as compares different optimizers (like SGD which uses momentum) for each technique. It is found that optimizers with momentum out perform adaptive optimizers but at a cost of significantly longer training times.<br />
<br />
== Other Work ==<br />
<br />
In modern training, dropout is not advised for convolutional neural networks because it does not have the effect, interpretation, impact on spatial feature maps as dense features. This is because features in CNNs are spatially correlated. There is an interesting paper on DropBlock [2], a dropout method which drops entire contiguous regions of features, which has been shown to be much more effective for CNNs.<br />
<br />
== Reference ==<br />
[1] N. Srivastave, "Dropout: a simple way to prevent neural networks from overfitting", The Journal of Machine Learning Research, Jan 2014.<br />
<br />
[2] Ghiasi, Golnaz and Lin, Tsung-Yi and Le, Quoc V. "DropBlock: A regularization method for convolutional networks". NeurIPS, 2018.</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=Improving_neural_networks_by_preventing_co-adaption_of_feature_detectors&diff=49338Improving neural networks by preventing co-adaption of feature detectors2020-12-06T07:24:12Z<p>Iaoellme: /* Conclusion */</p>
<hr />
<div>== Presented by ==<br />
Stan Lee, Seokho Lim, Kyle Jung, Dae Hyun Kim<br />
<br />
= Improvement Intro =<br />
'''Drop Out Model'''<br />
<br />
In this paper, Hinton et al. introduce a novel way to improve neural networks’ performance, particularly in the case that a large feedforward neural network is trained on a small training set, which causes poor performance and leads to “overfitting” problem. This problem can be reduced by randomly omitting half of the feature detectors on each training case. In fact, By omitting neurons in hidden layers with a probability of 0.5, each hidden unit is prevented from relying on other hidden units being present during training. Hence there are fewer co-adaptations among them on the training data. Called “dropout,” this process is also an efficient alternative to train many separate networks and average their predictions on the test set. <br />
<br />
The intuition for dropout is that if neurons are randomly dropped during training, they can no longer rely on their neighbours, thus allowing each neutron to become more robust. Another interpretation is that dropout is similar to training an ensemble of models, since each epoch with randomly dropped neurons can be viewed as its own model. <br />
<br />
They used the standard, stochastic gradient descent algorithm and separated training data into mini-batches. An upper bound was set on the L2 norm of incoming weight vector for each hidden neuron, which was normalized if its size exceeds the bound. They found that using a constraint, instead of a penalty, forced the model to do a more thorough search of the weight-space, when coupled with the very large learning rate that decays during training.<br />
<br />
'''Mean Network'''<br />
<br />
Their dropout models included all of the hidden neurons, and their outgoing weights were halved to account for the chances of omission. This is called an 'Mean Network'. This is similar to taking the geometric mean of the probability distribution predicted by all <math>2^N</math> networks. Due to this cumulative addition, the correct answers will have higher log probability than an individual dropout network, which also lead to a lower square error of the network. <br />
<br />
<br />
The models were shown to result in lower test error rates on several datasets: MNIST; TIMIT; Reuters Corpus Volume; CIFAR-10; and ImageNet.<br />
<br />
= MNIST =<br />
The MNIST dataset contains 70,000 digit images of size 28 x 28. To see the impact of dropout, they used 4 different neural networks (784-800-800-10, 784-1200-1200-10, 784-2000-2000-10, 784-1200-1200-1200-10), with the same dropout rates, 50%, for hidden neurons and 20% for visible neurons. Stochastic gradient descent was used with mini-batches of size 100 and a cross-entropy objective function as the loss function. Weights were updated after each minibatch, and training was done for 3000 epochs. An exponentially decaying learning rate <math>\epsilon</math> was used, with the initial value set as 10.0, and it was multiplied by the decaying factor <math>f</math> = 0.998 at the end of each epoch. At each hidden layer, the incoming weight vector for each hidden neuron was set an upper bound of its length, <math>l</math>, and they found from cross-validation that the results were the best when <math>l</math> = 15. Initial weights values were pooled from a normal distribution with mean 0 and standard deviation of 0.01. To update weights, an additional variable, ''p'', called momentum, was used to accelerate learning. The initial value of <math>p</math> was 0.5, and it increased linearly to the final value 0.99 during the first 500 epochs, remaining unchanged after. Also, when updating weights, the learning rate was multiplied by <math>1 – p</math>. <math>L</math> denotes the gradient of loss function.<br />
<br />
[[File:weights_mnist2.png|center|400px]]<br />
<br />
The best published result for a standard feedforward neural network was 160 errors. This was reduced to about 130 errors with 0.5 dropout and different L2 constraints for each hidden unit input weight. By omitting a random 20% of the input pixels in addition to the aforementioned changes, the number of errors was further reduced to 110. The following figure visualizes the result.<br />
[[File:mnist_figure.png|center|500px]]<br />
A publicly available pre-trained deep belief net resulted in 118 errors, and it was reduced to 92 errors when the model was fine-tuned with dropout. Another publicly available model was a deep Boltzmann machine, and it resulted in 103, 97, 94, 93 and 88 when the model was fine-tuned using standard backpropagation and was unrolled. They were reduced to 83, 79, 78, 78, and 77 when the model was fine-tuned with dropout – the mean of 79 errors was a record for models that do not use prior knowledge or enhanced training sets.<br />
<br />
= TIMIT = <br />
<br />
TIMIT dataset includes voice samples of 630 American English speakers varying across 8 different dialects. It is often used to evaluate the performance of automatic speech recognition systems. Using Kaldi, the dataset was pre-processed to extract input features in the form of log filter bank responses.<br />
<br />
=== Pre-training and Training ===<br />
<br />
For pretraining, they pretrained their neural network with a deep belief network and the first layer was built using Restricted Boltzmann Machine (RBM). Initializing visible biases with zero, weights were sampled from random numbers that followed normal distribution <math>N(0, 0.01)</math>. Each visible neuron’s variance was set to 1.0 and remained unchanged.<br />
<br />
Minimizing Contrastive Divergence (CD) was used to facilitate learning. Since momentum is used to speed up learning, it was initially set to 0.5 and increased linearly to 0.9 over 20 epochs. The average gradient had 0.001 of a learning rate which was then multiplied by <math>(1-momentum)</math> and L2 weight decay was set to 0.001. After setting up the hyperparameters, the model was done training after 100 epochs. Binary RBMs were used for training all subsequent layers with a learning rate of 0.01. Then, <math>p</math> was set as the mean activation of a neuron in the data set and the visible bias of each neuron was initialized to <math>log(p/(1 − p))</math>. Training each layer with 50 epochs, all remaining hyper-parameters were the same as those for the Gaussian RBM.<br />
<br />
=== Dropout tuning ===<br />
<br />
The initial weights were set in a neural network from the pretrained RBMs. To finetune the network with dropout-backpropagation, momentum was initially set to 0.5 and increased linearly up to 0.9 over 10 epochs. The model had a small constant learning rate of 1.0 and it was used to apply to the average gradient on a minibatch. The model also retained all other hyperparameters the same as the model from MNIST dropout finetuning. The model required approximately 200 epochs to converge. For comparison purpose, they also finetuned the same network with standard backpropagation with a learning rate of 0.1 with the same hyperparameters.<br />
<br />
=== Classification Test and Performance ===<br />
<br />
A Neural network was constructed to output the classification error rate on the test set of TIMIT dataset. They have built the neural network with four fully-connected hidden layers with 4000 neurons per layer. The output layer distinguishes distinct classes from 185 softmax output neurons that are merged into 39 classes. After constructing the neural network, 21 adjacent frames with an advance of 10ms per frame was given as an input.<br />
<br />
Comparing the performance of dropout with standard backpropagation on several network architectures and input representations, dropout consistently achieved lower error and cross-entropy. Results showed that it significantly controls overfitting, making the method robust to choices of network architecture. It also allowed much larger nets to be trained and removed the need for early stopping. Thus, neural network architectures with dropout are not very sensitive to the choice of learning rate and momentum.<br />
<br />
= Reuters Corpus Volume =<br />
Reuters Corpus Volume I archives 804,414 news documents that belong to 103 topics. Under four major themes - corporate/industrial, economics, government/social, and markets – they belonged to 63 classes. After removing 11 classes with no data and one class with insufficient data, they are left with 50 classes and 402,738 documents. The documents were divided into training and test sets equally and randomly, with each document representing the 2000 most frequent words in the dataset, excluding stopwords.<br />
<br />
They trained two neural networks, with size 2000-2000-1000-50, one using dropout and backpropagation, and the other using standard backpropagation. The training hyperparameters are the same as that in MNIST, but training was done for 500 epochs.<br />
<br />
In the following figure, we see the significant improvements by the model with dropout in the test set error. On the right side, we see that learning with dropout also proceeds smoother. <br />
<br />
[[File:reuters_figure.png|700px|center]]<br />
<br />
= CNN =<br />
<br />
Feed-forward neural networks consist of several layers of neurons where each neuron in a layer applies a linear filter to the input image data and is passed on to the neurons in the next layer. When calculating the neuron’s output, scalar bias a.k.a weights is applied to the filter with nonlinear activation function as parameters of the network that are learned by training data. [[File:cnnbigpicture.jpeg|thumb|upright=2|center|alt=text|Figure: Overview of Convolutional Neural Network]] There are several differences between Convolutional Neural networks and ordinary neural networks. The figure above gives a visual representation of a Convolutional Neural Network. First, CNN’s neurons are organized topographically into a bank and laid out on a 2D grid, so it reflects the organization of dimensions of the input data. Secondly, neurons in CNN apply filters which are local, and which are centered at the neuron’s location in the topographic organization. Meaning that useful metrics or clues to identify the object in an input image which can be found by examining local neighborhoods of the image. Next, all neurons in a bank apply the same filter at different locations in the input image. When looking at the image example, green is an input to one neuron bank, yellow is filter bank, and pink is the output of one neuron bank (convolved feature). A bank of neurons in a CNN applies a convolution operation, aka filters, to its input where a single layer in a CNN typically has multiple banks of neurons, each performing a convolution with a different filter. The resulting neuron banks become distinct input channels into the next layer. The whole process reduces the net’s representational capacity, but also reduces the capacity to overfit.<br />
[[File:bankofneurons.gif|thumb|upright=3|center|alt=text|Figure: Bank of neurons]]<br />
<br />
=== Pooling ===<br />
<br />
Pooling layer summarizes the activities of local patches of neurons in the convolutional layer by subsampling the output of a convolutional layer. Pooling is useful for extracting dominant features, to decrease the computational power required to process the data through dimensionality reduction. The procedure of pooling goes on like this; output from convolutional layers is divided into sections called pooling units and they are laid out topographically, connected to a local neighborhood of other pooling units from the same convolutional output. Then, each pooling unit is computed with some function which could be maximum and average. Maximum pooling returns the maximum value from the section of the image covered by the pooling unit while average pooling returns the average of all the values inside the pooling unit (see example). In result, there are fewer total pooling units than convolutional unit outputs from the previous layer, this is due to larger spacing between pixels on pooling layers. Using the max-pooling function reduces the effect of outliers and improves generalization. Other than that, overlapping pooling makes this spacing between pixels smaller than the size of the neighborhood that the pooling units summarize (This spacing is usually referred as the stride between pooling units). With this variant, pooling layer can produce a coarse coding of the outputs which helps generalization. <br />
[[File:maxandavgpooling.jpeg|thumb|upright=2|center|alt=text|Figure: Max pooling and Average pooling]]<br />
<br />
=== Local Response Normalization === <br />
<br />
This network includes local response normalization layers which are implemented in lateral form and used on neurons with unbounded activations and permits the detection of high-frequency features with a big neuron response. This regularizer encourages competition among neurons belonging to different banks. Normalization is done by dividing the activity of a neuron in bank <math>i</math> at position <math>(x,y)</math> by the equation:<br />
[[File:local response norm.png|upright=2|center|]] where the sum runs over <math>N</math> ‘adjacent’ banks of neurons at the same position as in the topographic organization of neuron bank. The constants, <math>N</math>, <math>alpha</math> and <math>betas</math> are hyper-parameters whose values are determined using a validation set. This technique is replaced by better techniques such as the combination of dropout and regularization methods (<math>L1</math> and <math>L2</math>)<br />
<br />
=== Neuron nonlinearities ===<br />
<br />
All of the neurons for this model use the max-with-zero nonlinearity where output within a neuron is computed as <math> a^{i}_{x,y} = max(0, z^i_{x,y})</math> where <math> z^i_{x,y} </math> is the total input to the neuron. The reason they use nonlinearity is because it has several advantages over traditional saturating neuron models, such as significant reduction in training time required to reach a certain error rate. Another advantage is that nonlinearity reduces the need for contrast-normalization and data pre-processing since neurons do not saturate- meaning activities simply scale up little by little with usually large input values. For this model’s only pre-processing step, they subtract the mean activity from each pixel and the result is a centered data.<br />
<br />
=== Objective function ===<br />
<br />
The objective function of their network maximizes the multinomial logistic regression objective which is the same as minimizing the average cross-entropy across training cases between the true label and the model’s predicted label.<br />
<br />
=== Weight Initialization === <br />
<br />
It’s important to note that if a neuron always receives a negative value during training, it will not learn because its output is uniformly zero under the max-with-zero nonlinearity. Hence, the weights in their model were sampled from a zero-mean normal distribution with a high enough variance. High variance in weights will set a certain number of neurons with positive values for learning to happen, and in practice, it’s necessary to try out several candidates for variances until a working initialization is found. In their experiment, setting a positive constant, or 1, as biases of the neurons in the hidden layers was helpful in finding it.<br />
<br />
=== Training ===<br />
<br />
In this model, a batch size of 128 samples and momentum of 0.9, we train our model using stochastic gradient descent. The update rule for weight <math>w</math> is $$ v_{i+1} = 0.9v_i + \epsilon <\frac{dE}{dw_i}> i$$ $$w_{i+1} = w_i + v_{i+1} $$ where <math>i</math> is the iteration index, <math>v</math> is a momentum variable, <math>\epsilon</math> is the learning rate and <math>\frac{dE}{dw}</math> is the average over the <math>i</math>th batch of the derivative of the objective with respect to <math>w_i</math>. The whole training process on CIFAR-10 takes roughly 90 minutes and ImageNet takes 4 days with dropout and two days without.<br />
<br />
=== Learning ===<br />
To determine the learning rate for the network, it is a must to start with an equal learning rate for each layer which produces the largest reduction in the objective function with power of ten. Usually, it is in the order of <math>10^{-2}</math> or <math>10^{-3}</math>. In this case, they reduce the learning rate twice by a factor of ten before termination of training.<br />
<br />
= CIFAR-10 =<br />
<br />
=== CIFAR-10 Dataset ===<br />
<br />
Removing incorrect labels, The CIFAR-10 dataset is a subset of the Tiny Images dataset with 10 classes. It contains 5000 training images and 1000 testing images for each class. The dataset has 32 x 32 color images searched from the web and the images are labeled with the noun used to search the image.<br />
<br />
[[File:CIFAR-10.png|thumb|upright=2|center|alt=text|Figure 4: CIFAR-10 Sample Dataset]]<br />
<br />
=== Models for CIFAR-10 ===<br />
<br />
Two models, one with dropout and one without dropout, were built to test the performance of dropout on CIFAR-10. All models have CNN with three convolutional layers each with a pooling layer. All of the pooling payers use a stride=2 and summarize a 3*3 neighborhood. The max-pooling method is performed by the pooling layer which follows the first convolutional layer, and the average-pooling method is performed by remaining 2 pooling layers. The first and second pooling layers with <math>N = 9, α = 0.001</math>, and <math>β = 0.75</math> are followed by response normalization layers. A ten-unit softmax layer, which is used to output a probability distribution over class labels, is connected with the upper-most pooling layer. Using filter size of 5×5, all convolutional layers have 64 filter banks.<br />
<br />
Additional changes were made with the model with dropout. The model with dropout enables us to use more parameters because dropout forces a strong regularization on the network. Thus, a fourth weight layer is added to take the input from the previous pooling layer. This fourth weight layer is locally connected, but not convolutional, and contains 16 banks of filters of size 3 × 3 with 50% dropout. Lastly, the softmax layer takes its input from this fourth weight layer.<br />
<br />
Thus, with a neural network with 3 convolutional hidden layers with 3 max-pooling layers, the classification error achieved 16.6% to beat 18.5% from the best published error rate without using transformed data. The model with one additional locally-connected layer and dropout at the last hidden layer produced the error rate of 15.6%.<br />
<br />
= ImageNet =<br />
<br />
===ImageNet Dataset===<br />
<br />
ImageNet is a dataset of millions of high-resolution images, and they are labeled among 1000 different categories. The data were collected from the web and manually labeled using MTerk tool, which is a crowd-sourcing tool provided by Amazon.<br />
Because this dataset has millions of labeled images in thousands of categories, it is very difficult to have perfect accuracy on this dataset even for humans because the ImageNet images may contain multiple objects and there are a large number of object classes. ImageNet and CIFAR-10 are very similar, but the scale of ImageNet is about 20 times bigger (1,300,000 vs 60,000). The size of ImageNet is about 1.3 million training images, 50,000 validation images, and 150,000 testing images. They used resized images of 256 x 256 pixels for their experiments.<br />
<br />
'''An ambiguous example to classify:'''<br />
<br />
[[File:imagenet1.png|200px|center]]<br />
<br />
When this paper was written, the best score on this dataset was the error rate of 45.7% by High-dimensional signature compression for large-scale image classification (J. Sanchez, F. Perronnin, CVPR11 (2011)). The authors of this paper could achieve a comparable performance of 48.6% error rate using a single neural network with five convolutional hidden layers with a max-pooling layer in between, followed by two globally connected layers and a final 1000-way softmax layer. When applying 50% dropout to the 6th layer, the error rate was brought down to 42.4%.<br />
<br />
'''ImageNet Dataset:'''<br />
<br />
[[File:imagenet2.png|400px|center]]<br />
<br />
===Models for ImageNet===<br />
<br />
They mostly focused on the model with dropout because the one without dropout had a similar approach, but there was a serious issue with overfitting. They used a convolutional neural network trained by 224×224 patches randomly extracted from the 256 × 256 images. This could reduce the network’s capacity to overfit the training data and helped generalization as a form of data augmentation. The method of averaging the prediction of the net on ten 224 × 224 patches of the 256 × 256 input image was used for testing their model patched at the center, four corners, and their horizontal reflections. To maximize the performance on the validation set, this complicated network architecture was used and it was found that dropout was very effective. Also, it was demonstrated that using non-convolutional higher layers with the number of parameters worked well with dropout, but it had a negative impact to the performance without dropout.<br />
<br />
The network contains seven weight layers. The first five are convolutional, and the last two are globally-connected. Max-pooling layers follow the layer number 1,2, and 5. And then, the output of the last globally-connected layer was fed to a 1000-way softmax output layers. Using this architecture, the authors achieved the error rate of 48.6%. When applying 50% dropout to the 6th layer, the error rate was brought down to 42.4%.<br />
<br />
<br />
[[File:modelh2.png|700px|center]] <br />
<br />
[[File:layer2.png|600px|center]]<br />
<br />
Like the previous datasets, such as the MNIST, TIMIT, Reuters, and CIFAR-10, we also see a significant improvement for the ImageNet dataset. Including complicated architectures like this one, introducing dropout generalizes models better and gives lower test error rates.<br />
<br />
= Conclusion =<br />
<br />
The authors have shown a consistent improvement by the models trained with dropout in classifying objects in the following datasets: MNIST; TIMIT; Reuters Corpus Volume I; CIFAR-10; and ImageNet.<br />
<br />
The authors comment on a theory that sexual reproduction limits biological function to a small number of coadapted genes. The idea is that a given organism is unlikely to receive many coordinated genes from a parent, so will likely die if it relies on many genes to perform a given task. This limits the number of genes required to perform a function, which is like a built-in evolutionary dropout.<br />
<br />
= Critiques =<br />
It is a very brilliant idea to dropout half of the neurons to reduce co-adaptations. It is mentioned that for fully connected layers, dropout in all hidden layers works better than dropout in only one hidden layer. There is another paper Dropout: A Simple Way to Prevent Neural Networks from<br />
Overfitting[https://www.cs.toronto.edu/~hinton/absps/JMLRdropout.pdf] gives a more detailed explanation.<br />
<br />
It will be interesting to see how this paper could be used to prevent overfitting of LSTMs.<br />
<br />
This paper focused more on CV tasks, it will be interesting to have some discussion on NLP tasks<br />
<br />
Firstly, it is a very interested topic of classification by "dropout" CNN method(omitting neurons in hidden layers). If the author can briefly explain the advantages of this method in processing image data in theory, it will be easier for readers to understand. Also, how to deal with overfitting issue would be valuable.<br />
<br />
The authors mention that they tried various dropout probabilities and that the majority of them improved the model's generalization performance, but that more extreme probabilities tended to be worse which is why a dropout rate of 50% was used in the paper. The authors further develop this point to mention that the method can be improved by adapting individual dropout probabilities of each hidden or input unit using validation tests. This would be an interesting area to further develop and explore, as using a hardcoded 50% dropout for all layers might not be the optimal choice for all CNN applications. It would have been interesting to see the results of their investigations of differing dropout rates.<br />
<br />
The authors don't explain that during training, at each layer that we apply dropout, the values must be scaled by 1/p where p is dropout rate - this way the expected value of the layers is the same in both train and test time. They may have considered another solution for this discrepancy at the time (it is an old paper) but it doesn't seem like any solution was presented here. <br />
<br />
Despite the advantages of using dropout to prevent overfitting and reducing errors in testing, the authors did not discuss much about the effects on the length of training time. In another [https://www.cs.toronto.edu/~hinton/absps/JMLRdropout.pdf paper] published a few years later by the same authors, there was more discussion about this. It appears that dropout increases training time by 2-3 times compared to a standard NN with the same architecture, which is a drawback that might be worth mentioning.<br />
<br />
Dropout layers prevent overfitting by randomly dropout a fraction of the neurons specified in each layer. In fact, the neurons to be dropped out in each layer are randomly selected. Therefore, it might be the case that some important features in the dropout layer are discarded, which leads to a sudden drop in performance. Although this barely happens, and CNN with dropout rates roughly 50% in each layer will lead to generally good performance, some future improvements are still possible if we are able to select dropout neurons cleverly.<br />
<br />
The article does a good job of analyzing the benefit of using the standard dropout method, but I think it would be beneficial to take a look at other dropout variants. For example, the paper may have benefited at looking at DropConnect which was introduced y L. Wan et al and is similar to dropout layers but it does not apply a dropout directly o the neurons but on the weights and the bias linking the neurons. Others that they also could have looked at were Standout, Pooling Drop and MaxDrop. Comparing various dropout methods I think would greatly add to the paper.<br />
<br />
The author analyzed the dropout method for addressing overfitting problems. The key idea is to randomly drop units from the neural network during training. This prevents units from co-adapting too much. In addition, it also fastens the speed of training models since there are fewer neurons, which is a good idea. <br />
<br />
Random dropping was indeed quite effective in the MNIST fashion classification challenge, however it may pose a question if the problem has very few features to begin with.<br />
<br />
The authors mentioned that they used Momentum to speed up the training but didn't show the alternative and the speed of the alternative. This [https://link.springer.com/article/10.1007/s11042-019-08453-9 paper]conducts an empirical study of Dropout vs Batch Normalization as well as compares different optimizers (like SGD which uses momentum) for each technique. It is found that optimizers with momentum out perform adaptive optimizers but at a cost of significantly longer training times.<br />
<br />
== Other Work ==<br />
<br />
In modern training, dropout is not advised for convolutional neural networks because it does not have the effect, interpretation, impact on spatial feature maps as dense features. This is because features in CNNs are spatially correlated. There is an interesting paper on DropBlock [2], a dropout method which drops entire contiguous regions of features, which has been shown to be much more effective for CNNs.<br />
<br />
== Reference ==<br />
[1] N. Srivastave, "Dropout: a simple way to prevent neural networks from overfitting", The Journal of Machine Learning Research, Jan 2014.<br />
<br />
[2] Ghiasi, Golnaz and Lin, Tsung-Yi and Le, Quoc V. "DropBlock: A regularization method for convolutional networks". NeurIPS, 2018.</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=Speech2Face:_Learning_the_Face_Behind_a_Voice&diff=49336Speech2Face: Learning the Face Behind a Voice2020-12-06T07:06:08Z<p>Iaoellme: /* Discussion and Critiques */</p>
<hr />
<div>== Presented by == <br />
Ian Cheung, Russell Parco, Scholar Sun, Jacky Yao, Daniel Zhang<br />
<br />
== Introduction ==<br />
This paper presents a deep neural network architecture called Speech2Face. This architecture utilizes millions of Internet/Youtube videos of people speaking to learn the correlation between a voice and the respective face. The model learns the correlations, allowing it to produce facial reconstruction images that capture specific physical attributes, such as a person's age, gender, or ethnicity, through a self-supervised procedure. Namely, the model utilizes the simultaneous occurrence of faces and speech in videos and does not need to model the attributes explicitly. This model explores what types of facial information could be extracted from speech without the constraints of predefined facial characterizations. Without any prior information or accurate classifiers, the reconstructions revealed correlations between craniofacial features and voice in addition to the correlation between dominant features (gender, age, ethnicity, etc.) and voice. The model is evaluated and numerically quantifies how closely the reconstruction, done by the Speech2Face model, resembles the true face images of the respective speakers.<br />
<br />
== Ethical Considerations ==<br />
<br />
The authors note that due to the potential sensitivity of facial information, they have chosen to explicitly state some ethical considerations. The first of which is privacy. The paper states that the method cannot recover the true identity of the face or produce faces of specific individuals, but rather will show average-looking faces. The paper also addresses that there are potential dataset biases that exist for the voice-face correlations, thus the faces may not accurately represent the intended population. Finally, it acknowledges that the model uses demographic categories that are defined by a commercial face attribute classifier.<br />
<br />
== Previous Work ==<br />
With visual and audio signals being so dominant and accessible in our daily life, there has been huge interest in how visual and audio perceptions interact with each other. Arandjelovic and Zisserman [1] leveraged the existing database of mp4 files to learn a generic audio representation to classify whether a video frame and an audio clip correspond to each other. These learned audio-visual representations have been used in a variety of setting, including cross-modal retrieval, sound source localization and sound source separation. This also paved the path for specifically studying the association between faces and voices of agents in the field of computer vision. In particular, cross-modal signals extracted from faces and voices have been proposed as a binary or multi-task classification task and there have been some promising results. Studies have been able to identify active speakers of a video, separate speech from multiple concurrent sources, predict lip motion from speech, and even learn the emotion of the agents based on their voices. Aytar et al. [6] proposed a student-teacher training procedure in which a well established visual recognition model was used to transfer the knowledge obtained in the visual modality to the sound modality, using unlabeled videos.<br />
<br />
Recently, various methods have been suggested to use various audio signals to reconstruct visual information, where the reconstructed subject is subjected to a priori. Notably, Duarte et al. [2] were able to synthesize the exact face images and expression of an agent from speech using a GAN model. A generative adversarial network (GAN) model is one that uses a generator to produce seemingly possible data for training and a discriminator that identifies if the training data is fabricated by the generator or if it is real [7]. This paper instead hopes to recover the dominant and generic facial structure from a speech.<br />
<br />
== Motivation ==<br />
It seems to be a common trait among humans to imagine what some people look like when we hear their voices before we have seen what they look lke. There is a strong connection between speech and appearance, which is a direct result of the factors that affect speech, including age, gender, and facial bone structure. In addition, other voice-appearance correlations stem from the way in which we talk: language, accent, speed, pronunciations, etc. These properties of speech are often common among many different nationalities and cultures, which can, in turn, translate to common physical features among different voices. Namely, from an input audio segment of a person speaking, the method would reconstruct an image of the person’s face in a canonical form (frontal-facing, neutral expression). The goal was to study to what extent people can infer how someone else looks from the way they talk. Rather than predicting a recognizable image of the exact face, the authors are more interested in capturing the dominant facial features.<br />
<br />
== Model Architecture == <br />
<br />
'''Speech2Face model and training pipeline'''<br />
<br />
[[File:ModelFramework.jpg|center]]<br />
<br />
<div style="text-align:center;"> Figure 1. '''Speech2Face model and training pipeline''' </div><br />
<br />
<br />
<br />
The Speech2Face Model used to achieve the desired result consists of two parts - a voice encoder which takes in a spectrogram of speech as input and outputs low dimensional face features, and a face decoder which takes in face features as input and outputs a normalized image of a face (neutral expression, looking forward). Figure 1 gives a visual representation of the pipeline of the entire model, from video input to a recognizable face. The combination of the voice encoder and face decoder results are combined to form an image. The variability in facial expressions, head positions and lighting conditions of the face images creates a challenge to both the design and training of the Speech2Face model. It needs a model to figure out many irrelevant variations in the data, and to implicitly extract important internal representations of faces. To avoid this problem the model is trained to first regress to a low dimensional intermediate representation of the face. <br />
<br />
'''Face Decoder''' <br />
The face decoder itself was taken from previous work The VGG-Face model by Cole et al [3] (a face recognition model that is pretrained on a largescale face database [5] is used to extract a 4069-D face feature from the penultimate layer of the network.) and will not be explored in great detail here, but in essence the facenet model is combined with a single multilayer perceptron layer, the result of which is passed through a convolutional neural network to determine the texture of the image, and a multilayer perception to determine the landmark locations. The face decoder kept the VGG-Face model's dimension and weights. The weights were also trained separately and remained fixed during the voice encoder training. <br />
<br />
'''Voice Encoder Architecture''' <br />
<br />
[[File:VoiceEncoderArch.JPG|center]]<br />
<br />
<div style="text-align:center;"> Table 1: '''Voice encoder architecture''' </div><br />
<br />
<br />
<br />
The voice encoder itself is a convolutional neural network, which transforms the input spectrogram into pseudo face features. The exact architecture is given in Table 1. The model alternates between convolution, ReLU, batch normalization layers, and layers of max-pooling. In each max-pooling layer, pooling is only done along the temporal dimension of the data. This is to ensure that the frequency, an important factor in determining vocal characteristics such as tone, is preserved. In the final pooling layer, an average pooling is applied along the temporal dimension. This allows the model to aggregate information over time and allows the model to be used for input speeches of varying lengths. Two fully connected layers at the end are used to return a 4096-dimensional facial feature output.<br />
<br />
'''Training'''<br />
<br />
The AVSpeech dataset, a large-scale audio-visual dataset is used for the training. AVSpeech dataset is comprised of millions of video segments from Youtube with over 100,000 different people. The training data is composed of educational videos and does not provide an accurate representation of the global population, which will clearly affect the model. Also note that facial features that are irrelevant to speech, like hair color, may be predicted by the model. From each video, a 224x224 pixels image of the face was passed through the face decoder to compute a facial feature vector. Combined with a spectrogram of the audio, a training and test set of 1.7 and 0.15 million entries respectively were constructed.<br />
<br />
The voice encoder is trained in a self-supervised manner. A frame that contains the face is extracted from each video and then inputted to the VGG-Face model to extract the feature vector <math>v_f</math>, the 4096-dimensional facial feature vector given by the face decoder on a single frame from the input video. This provides the supervision signal for the voice-encoder. The feature <math>v_s</math>, the 4096 dimensional facial feature vector from the voice encoder, is trained to predict <math>v_f</math>.<br />
<br />
In order to train this model, a proper loss function must be defined. The L1 norm of the difference between <math>v_s</math> and <math>v_f</math>, given by <math>||v_f - v_s||_1</math>, may seem like a suitable loss function, but in actuality results in unstable results and long training times. Figure 2, below, shows the difference in predicted facial features given by <math>||v_f - v_s||_1</math> and the following loss. Based on the work of Castrejon et al. [4], a loss function is used which penalizes the differences in the last layer of the VGG-Face model <math>f_{VGG}</math>: <math> \mathbb{R}^{4096} \to \mathbb{R}^{2622}</math> and the first layer of face decoder <math>f_{dec}</math> : <math> \mathbb{R}^{4096} \to \mathbb{R}^{1000}</math>. The final loss function is given by: $$L_{total} = ||f_{dec}(v_f) - f_{dec}(v_s)|| + \lambda_1||\frac{v_f}{||v_f||} - \frac{v_s}{||v_s||}||^2_2 + \lambda_2 L_{distill}(f_{VGG}(v_f), f_{VGG}(v_s))$$<br />
This loss penalizes on both the normalized Euclidean distance between the 2 facial feature vectors and the knowledge distillation loss, which is given by: $$L_{distill}(a,b) = -\sum_ip_{(i)}(a)\text{log}p_{(i)}(b)$$ $$p_{(i)}(a) = \frac{\text{exp}(a_i/T)}{\sum_j \text{exp}(a_j/T)}$$ Knowledge distillation is used as an alternative to Cross-Entropy. By recommendation of Cole et al [3], <math> T = 2 </math> was used to ensure a smooth activation. <math>\lambda_1 = 0.025</math> and <math>\lambda_2 = 200</math> were chosen so that magnitude of the gradient of each term with respect to <math>v_s</math> are of similar scale at the <math>1000^{th}</math> iteration.<br />
<br />
<center><br />
[[File:L1vsTotalLoss.png | 700px]]<br />
</center><br />
<br />
<div style="text-align:center;"> Figure 2: '''Qualitative results on the AVSpeech test set''' </div><br />
<br />
== Results ==<br />
<br />
'''Confusion Matrix and Dataset statistics'''<br />
<br />
<center><br />
[[File:Confusionmatrix.png| 600px]]<br />
</center><br />
<br />
<div style="text-align:center;"> Figure 3. '''Facial attribute evaluation''' </div><br />
<br />
<br />
<br />
In order to determine the similarity between the generated images and the ground truth, a commercial service known as Face++ which classifies faces for distinct attributes (such as gender, ethnicity, etc) was used. Figure 3 gives a confusion matrix based on gender, ethnicity, and age. By examining these matrices, it is seen that the Speech2Face model performs very well on gender, only misclassifying 6% of the time. Similarly, the model performs fairly well on ethnicities, especially with white or Asian faces. Although the model performs worse on black and Indian faces, that can be attributed to the vastly unbalanced data, where 50% of the data represented a white face, and 80% represented a white or Asian face. <br />
<br />
'''Feature Similarity'''<br />
<br />
<center><br />
[[File:FeatSim.JPG]]<br />
</center><br />
<br />
<div style="text-align:center;"> Table 2. '''Feature similarity''' </div><br />
<br />
<br />
<br />
Another examination of the result is the similarity of features predicted by the Speech2Face model. The cosine, L1, and L2 distance between the facial feature vector produced by the model and the true facial feature vector from the face decoder were computed, and presented, above, in Table 2. A comparison of facial similarity was also done based on the length of audio input. From the table, it is evident that the 6-second audio produced a lower cosine, L1, and L2 distance, resulting in a facial feature vector that is closer to the ground truth. <br />
<br />
'''S2F -> Face retrieval performance'''<br />
<br />
<center><br />
[[File: Retrieval.JPG]]<br />
</center><br />
<br />
<div style="text-align:center;"> Table 3. '''S2F -> Face retrieval performance''' </div><br />
<br />
<br />
<br />
The performance of the model was also examined on how well it could produce the original image. The R@K metric, also known as retrieval performance by recall at K, measures the probability that the K closest images to the model output includes the correct image of the speaker's face. A higher R@K score indicates better performance. From Table 3, above, we see that both the 3-second and 6-second audio showed significant improvement over random chance, with the 6-second audio performing slightly better.<br />
<br />
'''Additional Observations''' <br />
<br />
Ablation studies were carried out to test the effect of audio duration and batch normalization. It was found that the duration of input audio during the training stage had little effect on convergence speed (comparing 3 and 6-second speech segments), while in the test stage longer input speech yields improvement in reconstruction quality. With respect to batch normalization (BN), it was found that without BN reconstructed faces would converge to an average face, while the inclusion of BN led to results which contained much richer facial features.<br />
<br />
== Conclusion ==<br />
The report presented a novel study of face reconstruction from audio recordings of a person speaking. The model was demonstrated to be able to predict plausible face reconstructions with similar facial features to real images of the person speaking. The problem was addressed by learning to align the feature space of speech to that of a pretrained face decoder. The model was trained on millions of videos of people speaking from YouTube. The model was then evaluated by comparing the reconstructed faces with a commercial facial detection service. The authors believe that facial reconstruction allows a more comprehensive view of voice-face correlation compared to predicting individual features, which may lead to new research opportunities and applications.<br />
<br />
== Discussion and Critiques ==<br />
<br />
There is evidence that the results of the model may be heavily influenced by external factors:<br />
<br />
1. Their method of sampling random YouTube videos resulted in an unbalanced sample in terms of ethnicity. Over half of the samples were white. We also saw a large bias in the model's prediction of ethnicity towards white. The bias in the results shows that the model may be overfitting the training data and puts into question what the performance of the model would be when trained and tested on a balanced dataset. Figure (11) highlights this shortcoming: The same man heard speaking in either English or Chinese was predicted to have a "white" appearance or an "asian" appearance respectively.<br />
<br />
2. The model was shown to infer different face features based on language. This puts into question how heavily the model depends on the spoken language. The paper mentioned the quality of face reconstruction may be affected by uncommon languages, where English is the most popular language on Youtube(training set). Testing a more controlled sample where all speech recording was of the same language may help address this concern to determine the model's reliance on spoken language.<br />
<br />
3. The evaluation of the result is also highly dependent on the Face++ classifiers. Since they compare the age, gender, and ethnicity by running the Face++ classifiers on the original images and the reconstructions to evaluate their model, the model that they create can only be as good as the one they are using to evaluate it. Therefore, any limitations of the Face++ classifier may become a limitation of Speech2Face and may result in a compounding effect on the miss-classification rate.<br />
<br />
4. Figure 4.b shows the AVSpeech dataset statistics. However, it doesn't show the statistics about speakers' ethnicity and the language of the video. If we train the model with a more comprehensive dataset that includes enough Asian/Indian English speakers and native language speakers will this increase the accuracy?<br />
<br />
5. One concern about the source of the training data, i.e. the Youtube videos, is that resolution varies a lot since the videos are randomly selected. That may be the reason why the proposed model performs badly on some certain features. For example, it is hard to tell the age when the resolution is bad because the wrinkles on the face are neglected.<br />
<br />
6. The topic of this project is very interesting, but I highly doubt this model will be practical in real-world problems. Because there are many factors to affect a person's sound in a real-world environment. Sounds such as phone clock, TV, car horn and so on. These sounds will decrease the accuracy of the predicted result of the model.<br />
<br />
7. A lot of information can be obtained from someone's voice, this can potentially be useful for detective work and crime scene investigation. In our world of increasing surveillance, public voice recording is quite common and we can reconstruct images of potential suspects based on their voice. In order for this to be achieved, the model has to be thoroughly trained and tested to avoid false positives as it could have a highly destructive outcome for a falsely convicted suspect.<br />
<br />
8. This is a very interesting topic, and this summary has a good structure for readers. Since this model uses Youtube to train model, but I think one problem is that most of the YouTubers are adult, and many additional reasons make this dataset highly unbalanced. What is more, some people may have a baby voice, this also could affect the performance of the model. But overall, this is a meaningful topic, it might help police to locate the suspects. So it might be interesting to apply this to the police.<br />
<br />
9. In addition, it seems very unlikely that any results coming from this model would ever be held in regard even remotely close to being admissible in court to identify a person of interest until the results are improved and the model can be shown to work in real-world applications. Otherwise, there seems to be very little use for such technology and it could have negative impacts on people if they were to be depicted in an unflattering way by the model based on their voice.<br />
<br />
10. Using voice as a factor of constructing the face is a good idea, but it seems like the data they have will have lots of noise and bias. The voice of a video might not come from the person in the video. There are so many YouTubers adjusting their voices before uploading their video and it's really hard to know whether they adjust their voice. Also, most YouTubers are adults so the model cannot have enough training samples about teenagers and kids.<br />
<br />
11. It would be interesting to see how the performance changes with different face encoding sizes (instead of just 4096-D) and also difference face models (encoder/decoders) to see if better performance can be achieved. Also given that the dataset used was unbalanced, was the dataset used to train the face model the same dataset? or was a different dataset used (the model was pretrained). This could affect the performance of the model as well.<br />
<br />
12. The audio input is transformed into a spectrogram before being used for training. They use STFT with a Hann window of 25 mm, a hop length of 10 ms, and 512 FFT frequency bands. They cite this method from a paper that focuses on speech separation, not speech classification. So, it would be interesting to see if there is a better way to do STFT, possibly with different hyperparameters (eg. different windowing, different number of bands), or if another type of transform (eg. wavelet transform) would have better results.<br />
<br />
13. A easy way to get somewhat balanced data is to duplicate the data that are fewer.<br />
<br />
14. This problem is interesting but is hard to generalize. This algorithm didn't account for other genders and mixed-race. In addition, the face recognition software Face++ introduces bias which can carry forward to Speech2Face algorithm. Face recognition algorithms are known to have higher error rates classifying darker-skinned individuals. Thus, it'll be tough to apply it to real-life scenarios like identifying suspects.<br />
<br />
15. This experiment raises a lot of ethical complications when it comes to possible applications in the real world. Even if this model was highly accurate, the implications of being able to discern a person's racial ethnicity, skin tone, etc. based solely on there voice could play in to inherent biases in the application user and this may end up being an issue that needs to be combatted in future research in this area. Another possible issue is that many people will change their intonation or vocal features based on the context (I'll likely have a different voice pattern in a job interview in terms of projection, intonation, etc. than if I was casually chatting/mumbling with a friend while playing video games for example).<br />
<br />
16. Overall a very interesting topic. I want to talk about the technical challenged raised by using the AVSSpeech dataset for training. The paper acknowledges that the AVSSpeech is unbalanced, and 80% of the data are white and Asians. It also says in the results section that "Our model does not perform on other races due to the imbalance in data". There does not seem to be any effort made in balancing the data. I think that there are definitely some data processing techniques that can be used (filtering, data augmentation, etc) to address the class imbalance problem. Not seeing any of these in the paper is a bit disappointing. Another issue I have noticed is that the model aims to predict an average-looking face from certain gender/racial group from voice input, due to ethical considerations. If we cannot reveal the identify of a person, why don't we predict the gender and race directly? Giving an average-looking face does not seem to be the most helpful.<br />
<br />
17. Very interesting research paper to be studied and the main objective was also interesting. This research leads to open question which can be applied to another application such as predicting person's face using voice and can be used in more advanced way. The only risk is how the data is obtained from YouTube where data is not consistent.<br />
<br />
18. The essay uses millions of natural videos of people speaking to find the correlation between face and voice. Since face and voice are commonly used as the identity of a person, there are many possible research opportunities and applications about improving voice and face unlock.<br />
<br />
19. It would be better to have a future work section to discuss the current shortage and explore the possible improvement and applications in the future.<br />
<br />
20. While the idea behind Speech2Face is interesting, ethnic profiling is a huge concern and it can further lead to racial discrimination, racism etc. Developers must put more care and thought into applying Speech2Face in tech before deploying the products.<br />
<br />
21. It would be helpful if the author could explore the different applications of this project in real life. Speech2face can be helpful during criminal investigation and essentially in scenarios when someone's picture is missing and only voice is available. It would also be helpful if the author could state the importance and need of such kind project in the society.<br />
<br />
22. The authors mention that they use the AVSpeech dataset for both training and testing but do not talk about how they split the data. It is possible that the same speakers were used in the training and testing data and so the model is able to recreate a face simply by matching the observed face to the observed audio. This would explain the striking example images shown in the paper.<br />
<br />
== References ==<br />
[1] R. Arandjelovic and A. Zisserman. Look, listen and learn. In<br />
IEEE International Conference on Computer Vision (ICCV),<br />
2017.<br />
<br />
[2] A. Duarte, F. Roldan, M. Tubau, J. Escur, S. Pascual, A. Salvador, E. Mohedano, K. McGuinness, J. Torres, and X. Giroi-Nieto. Wav2Pix: speech-conditioned face generation using generative adversarial networks. In IEEE International<br />
Conference on Acoustics, Speech and Signal Processing<br />
(ICASSP), 2019.<br />
<br />
[3] F. Cole, D. Belanger, D. Krishnan, A. Sarna, I. Mosseri, and W. T. Freeman. Synthesizing normalized faces from facial identity features. In IEEE Conference on Computer Vision and Pattern Recognition (CVPR), 2017.<br />
<br />
[4] L. Castrejon, Y. Aytar, C. Vondrick, H. Pirsiavash, and A. Torralba. Learning aligned cross-modal representations from weakly aligned data. In IEEE Conference on Computer Vision and Pattern Recognition (CVPR), 2016.<br />
<br />
[5] O. M. Parkhi, A. Vedaldi, and A. Zisserman. Deep face recognition. In British Machine Vision Conference (BMVC), 2015.<br />
<br />
[7] “Overview of GAN Structure | Generative Adversarial Networks,” ''Google Developers'', 24-May-2019. [Online]. Available: https://developers.google.com/machine-learning/gan/gan_structure. [Accessed: 02-Dec-2020].</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=Music_Recommender_System_Based_using_CRNN&diff=49334Music Recommender System Based using CRNN2020-12-06T06:50:30Z<p>Iaoellme: /* Critiques/ Insights: */</p>
<hr />
<div>==Introduction and Objective:==<br />
<br />
In the digital era of music streaming, companies, such as Spotify and Pandora, are faced with the following challenge: can they provide users with relevant and personalized music recommendations amidst the ever-growing abundance of music and user data?<br />
<br />
The objective of this paper is to implement a personalized music recommender system that takes user listening history as input and continually finds new music that captures individual user preferences.<br />
<br />
This paper argues that a music recommendation system should vary from the general recommendation system used in practice since it should combine music feature recognition and audio processing technologies to extract music features, and combine them with data on user preferences.<br />
<br />
The authors of this paper took a content-based music approach to build the recommendation system - specifically, comparing the similarity of features based on the audio signal.<br />
<br />
The following two-method approach for building the recommendation system was followed:<br />
#Make recommendations including genre information extracted from classification algorithms.<br />
#Make recommendations without genre information.<br />
<br />
The authors used convolutional recurrent neural networks (CRNN), which is a combination of convolutional neural networks (CNN) and recurrent neural network(RNN), as their main classification model.<br />
<br />
==Methods and Techniques:==<br />
Generally, a music recommender can be divided into three main parts: (i) users, (ii) items, and (iii) user-item matching algorithms. Firstly, a model for a users music taste is generated based on their profiles. Secondly, item profiling based on editorial, cultural, and acoustic metadata is exploited to increase listener satisfaction. Thirdly, a matching algorithm is employed to recommend personalized music to the listener. Two main approaches are currently available;<br />
<br />
(1) Collaborative filtering<br />
<br />
It is based on users' historical listening data and depends on user ratings. Nearest neighbour is the standard method used for collaborative filtering and can be broken into two classes of methods: (i) user-based neighbourhood methods and (ii) item-based neighbourhood methods. <br />
<br />
User-based neighbourhood methods calculate the similarity between the target user and other users, and selects the k most similar. A weighted average of the most similar users' song ratings is then computed to predict how the target user would rate those songs. Songs that have a high predicted rating are then recommended to the user. In contrast, methods that use item-based neighbourhoods calculate similarities between songs that the target user has rated well and songs they have not listened to in order to recommend songs.<br />
<br />
That being said, collaborative filtering faces many challenges. For example, given that each user sees only a small portion of all music libraries, sparsity and scalability become an issue. However, this can be dealt with using matrix factorization. A more difficult challenge to overcome is the fact that users often don't rate songs when they are listening to music. <br />
<br />
(2) Content-based filtering<br />
<br />
Content based recommendation systems base their recommendations on the similarity of an items features and features that the user has enjoyed. It has two-steps; (i) Extract audio content features and (ii) predict user preferences.<br />
<br />
In this work, the authors take a content-based approach, as they compare the similarity of audio signal features to make recommendations. To classify music, the original music’s audio signal is converted into a spectrogram image. Using the image and the Short Time Fourier Transform (STFT), we convert the data into the Mel scale which is used in the CNN and CRNN models. <br />
=== Mel Scale: === <br />
The scale of pitches that are heard by listeners, which translates to equal pitch increments.<br />
<br />
[[File:Mel.png|frame|none|Mel Scale on Spectrogram]]<br />
<br />
=== Short Time Fourier Transform (STFT): ===<br />
The transformation that determines the sinusoidal frequency of the audio, with a Hanning smoothing function. In the continuous case this is written as: <math>\mathbf{STFT}\{x(t)\}(\tau,\omega) \equiv X(\tau, \omega) = \int_{-\infty}^{\infty} x(t) w(t-\tau) e^{-i \omega t} \, d t </math><br />
<br />
where: <math>w(\tau)</math> is the Hanning smoothing function. The STFT is applied over a specified window length at a certain time allowing the frequency to represented for that given window rather than the entire signal as a typical Fourier Transform would.<br />
<br />
=== Convolutional Neural Network (CNN): ===<br />
A Convolutional Neural Network is a Neural Network that uses convolution in place of matrix multiplication for some layer calculations. By training the data, weights for inputs are updated to find the most significant data relevant to classification. These convolutional layers gather small groups of data with kernels and try to find patterns that can help find features in the overall data. The features are then used for classification. Padding is another technique used to extend the pixels on the edge of the original image to allow the kernel to more accurately capture the borderline pixels. Padding is also used if one wishes the convolved output image to have a certain size. The image on the left represents the mathematical expression of a convolution operation, while the right image demonstrates an application of a kernel on the data.<br />
<br />
[[File:Convolution.png|thumb|400px|left|Convolution Operation]]<br />
[[File:PaddingKernels.png|thumb|400px|center|Example of Padding (white 0s) and Kernels (blue square)]]<br />
<br />
=== Convolutional Recurrent Neural Network (CRNN): === <br />
The CRNN is similar to the architecture of a CNN, but with the addition of a GRU, which is a Recurrent Neural Network (RNN). An RNN is used to treat sequential data, by reusing the activation function of previous nodes to update the output. A Gated Recurrent Unit (GRU) is used to store more long-term memory and will help train the early hidden layers. GRUs can be thought of as LSTMs but with a forget gate, and has fewer parameters than an LSTM. These gates are used to determine how much information from the past should be passed along onto the future. They are originally aimed to prevent the vanishing gradient problem, since deeper networks will result in smaller and smaller gradients at each layer. The GRU can choose to copy over all the information in the past, thus eliminating the risk of vanishing gradients.<br />
<br />
[[File:GRU441.png|thumb|400px|left|Gated Recurrent Unit (GRU)]]<br />
[[File:Recurrent441.png|thumb|400px|center|Diagram of General Recurrent Neural Network]]<br />
<br />
==Data Screening:==<br />
<br />
The authors of this paper used a publicly available music dataset made up of 25,000 30-second songs from the Free Music Archives which contains 16 different genres. The data is cleaned up by removing low audio quality songs, wrongly labelled genres and those that have multiple genres. To ensure a balanced dataset, only 1000 songs each from the genres of classical, electronic, folk, hip-hop, instrumental, jazz and rock were used in the final model. <br />
<br />
[[File:Data441.png|thumb|200px|none|Data sorted by music genre]]<br />
<br />
==Implementation:==<br />
<br />
=== Modeling Neural Networks ===<br />
<br />
As noted previously, both CNNs and CRNNs were used to model the data. The advantage of CRNNs is that they are able to model time sequence patterns in addition to frequency features from the spectrogram, allowing for greater identification of important features. Furthermore, feature vectors produced before the classification stage could be used to improve accuracy. <br />
<br />
In implementing the neural networks, the Mel-spectrogram data was split up into training, validation, and test sets at a ratio of 8:1:1 respectively and labelled via one-hot encoding. This made it possible for the categorical data to be labelled correctly for binary classification. As opposed to classical stochastic gradient descent, the authors opted to use binary classier and ADAM optimization to update weights in the training phase, and parameters of <math>\alpha = 0.001, \beta_1 = 0.9, \beta_2 = 0.999</math>. Binary cross-entropy was used as the loss function. <br />
Input spectrogram image are 96x1366. In both the CNN and CRNN models, the data was trained over 100 epochs and batch size of 50 (limited computing power) with a binary cross-entropy loss function. Notable model specific details are below:<br />
<br />
'''CNN'''<br />
* Five convolutional layers with 3x3 kernel, stride 1, padding, batch normalization, and ReLU activation<br />
* Max pooling layers <br />
* The sigmoid function was used as the output layer<br />
<br />
'''CRNN'''<br />
* Four convolutional layers with 3x3 kernel (which construct a 2D temporal pattern - two layers of RNNs with Gated Recurrent Units), stride 1, padding, batch normalization, ReLU activation, and dropout rate 0.1<br />
* Feature maps are N x1x15 (N = number of features maps, 68 feature maps in this case) is used for RNNs.<br />
* 4 Max pooling layers for four convolutional layers with kernel ((2x2)-(3x3)-(4x4)-(4x4)) and same stride<br />
* The sigmoid function was used as the output layer<br />
<br />
The CNN and CRNN architecture is also given in the charts below.<br />
<br />
[[File:CNN441.png|thumb|800px|none|Implementation of CNN Model]]<br />
[[File:CRNN441.png|thumb|800px|none|Implementation of CRNN Model]]<br />
<br />
=== Music Recommendation System ===<br />
<br />
The recommendation system is computed by the cosine similarity of the extraction features from the neural network. Each genre will have a song act as a centre point for each class. The final inputs of the trained neural networks will be the feature variables. The featured variables will be used in the cosine similarity to find the best recommendations. <br />
<br />
The values are between [-1,1], where larger values are songs that have similar features. When the user inputs five songs, those songs become the new inputs in the neural networks and the features are used by the cosine similarity with other music. The largest five cosine similarities are used as recommendations.<br />
[[File:Cosine441.png|frame|100px|none|Cosine Similarity]]<br />
<br />
== Evaluation Metrics ==<br />
=== Precision: ===<br />
* The proportion of True Positives with respect to the '''predicted''' positive cases (true positives and false positives)<br />
* For example, out of all the songs that the classifier '''predicted''' as Classical, how many are actually Classical?<br />
* Describes the rate at which the classifier predicts the true genre of songs among those predicted to be of that certain genre<br />
<br />
=== Recall: ===<br />
* The proportion of True Positives with respect to the '''actual''' positive cases (true positives and false negatives)<br />
* For example, out of all the songs that are '''actually''' Classical, how many are correctly predicted to be Classical?<br />
* Describes the rate at which the classifier predicts the true genre of songs among the correct instances of that genre<br />
<br />
=== F1-Score: ===<br />
An accuracy metric that combines the classifier’s precision and recall scores by taking the harmonic mean between the two metrics:<br />
<br />
[[File:F1441.png|frame|100px|none|F1-Score]]<br />
<br />
=== Receiver operating characteristics (ROC): ===<br />
* A graphical metric that is used to assess a classification model at different classification thresholds <br />
* In the case of a classification threshold of 0.5, this means that if <math>P(Y = k | X = x) > 0.5</math> then we classify this instance as class k<br />
* Plots the true positive rate versus false positive rate as the classification threshold is varied<br />
<br />
[[File:ROCGraph.jpg|thumb|400px|none|ROC Graph. Comparison of True Positive Rate and False Positive Rate]]<br />
<br />
=== Area Under the Curve (AUC) ===<br />
AUC is the area under the ROC in doing so, the ROC provides an aggregate measure across all possible classification thresholds.<br />
<br />
In the context of the paper: When scoring all songs as <math>Prob(Classical | X=x)</math>, it is the probability that the model ranks a random Classical song at a higher probability than a random non-Classical song.<br />
<br />
[[File:AUCGraph.jpg|thumb|400px|none|Area under the ROC curve.]]<br />
<br />
== Results ==<br />
=== Accuracy Metrics ===<br />
The table below is the accuracy metrics with the classification threshold of 0.5.<br />
<br />
[[File:TruePositiveChart.jpg|thumb|none|True Positive / False Positive Chart]]<br />
On average, CRNN outperforms CNN in true positive and false positive cases. In addition, it is very apparent that false positives are much more frequent for songs in the Instrumental genre, perhaps indicating that more pre-processing needs to be done for songs in this genre or that it should be excluded from analysis completely given how most music has instrumental components.<br />
<br />
<br />
[[File:F1Chart441.jpg|thumb|400px|none|F1 Chart]]<br />
On average, CRNN outperforms CNN in F1-score. <br />
<br />
<br />
[[File:AUCChart.jpg|thumb|400px|none|AUC Chart]]<br />
On average, CRNN also outperforms CNN in AUC metric.<br />
<br />
<br />
CRNN models that consider the frequency features and time sequence patterns of songs have a better classification performance through metrics such as F1 score and AUC when comparing to CNN classifier.<br />
<br />
=== Evaluation of Music Recommendation System: ===<br />
<br />
* A listening experiment was performed with 30 participants to access user responses to given music recommendations.<br />
* Participants choose 5 pieces of music they enjoyed and the recommender system generated 5 new recommendations. The participants then evaluated the music recommendation by recording whether the song was liked or disliked.<br />
* The recommendation system takes two approaches to the recommendation:<br />
** Method one uses only the value of cosine similarity.<br />
** Method two uses the value of cosine similarity and information on music genre.<br />
*Perform test of significance of differences in average user likes between the two methods using a t-statistic:<br />
[[File:H0441.png|frame|100px|none|Hypothesis test between method 1 and method 2]]<br />
<br />
Comparing the two methods, <math> H_0: u_1 - u_2 = 0</math>, we have <math> t_{stat} = -4.743 < -2.037 </math>, which demonstrates that the increase in average user likes with the addition of music genre information is statistically significant.<br />
<br />
== Conclusion: ==<br />
<br />
The two two main conclusions obtained from this paper:<br />
<br />
- The music genre should be a key feature to increase the predictive capabilities of the music recommendation system.<br />
<br />
- To extract the song genre from a song’s audio signals and get overall better performance, CRNN’s are superior to CNN’s as they consider frequency in features and time sequence patterns of audio signals. <br />
<br />
According to the paper, the authors suggested adding other music features like tempo gram for capturing local tempo as a way to improve the accuracy of the recommender system.<br />
<br />
== Critiques/ Insights: ==<br />
# The authors fail to give reference to the performance of current recommendation algorithms used in the industry; my critique would be for the authors to bench-mark their novel approach with other recommendation algorithms such as collaborative filtering to see if there is a lift in predictive capabilities.<br />
# The listening experiment used to evaluate the recommendation system only includes songs that are outputted by the model. Users may be biased if they believe all songs have come from a recommendation system. To remove bias, we suggest having 15 songs where 5 songs are recommended and 10 songs are set. With this in the user’s mind, it may remove some bias in response and give more accurate predictive capabilities. <br />
# They could go into more details about how CRNN makes it perform better than CNN, in terms of attributes of each network.<br />
# The methodology introduced in this paper is probably also suitable for movie recommendations. As music is presented as spectrograms (images) in a time sequence, and it is very similar to a movie. <br />
# The way of evaluation is a very interesting approach. Since it's usually not easy to evaluate the testing result when it's subjective. By listing all these evaluations' performance, the result would be more comprehensive. A practice that might reduce bias is by coming back to the participants after a couple of days and asking whether they liked the music that was recommended. Often times music "grows" on people and their opinion of a new song may change after some time has passed. <br />
# The paper lacks the comparison between the proposed algorithm and the music recommendation algorithms being used now. It will be clearer to show the superiority of this algorithm.<br />
# The GAN neural network has been proposed to enhance the performance of the neural network, so an improved result may appear after considering using GAN.<br />
# The limitation of CNN and CRNN could be that they are only able to process the spectrograms with single labels rather than multiple labels. This is far from enough for the music recommender systems in today's music industry since the edges between various genres are blurred.<br />
# Is it possible for CNN and CRNN to identify different songs? The model would be harder to train, based on my experience, the efficiency of CNN in R is not very high, which can be improved for future work.<br />
# according to the author, the recommender system is done by calculating the cosine similarity of extraction features from one music to another music. Is possible to represent it by Euclidean distance or p-norm distances?<br />
# In real-life application, most of the music software will have the ability to recommend music to the listener and ask do they like the music that was recommended. It would be a nice application by involving some new information from the listener.<br />
# This paper is very similar to another [https://link.springer.com/chapter/10.1007/978-3-319-46131-1_29 paper], written by Bruce Fewerda and Markus Schedl. Both papers are suggesting methods of building music recommendation systems. However, this paper recommends music based on genre, but the paper written by Fewerda and Schedl suggests a personality-based user modeling for music recommender systems.<br />
# Actual music listeners do not listen to one genre of music, and in fact listening to the same track or the same genre would be somewhat unusual. Could this method be used to make recommendations not on genre, but based on other categories? (Such as the theme of the lyrics, the pitch of the singer, or the date published). Would this model be able to differentiate between tracks of varying "lyric vocabulation difficulty"? Or would NLP algorithms be needed to consider lyrics?<br />
# This model can be applied to many other fields such as recommending the news in the news app, recommending things to buy in the amazon, recommending videos to watch in YOUTUBE and so on based on the user information.<br />
# Looks like for the most genres, CRNN outperforms CNN, but CNN did do better on a few genres (like Jazz), so it might be better to mix them together or might use CNN for some genres and CRNN for the rest.<br />
# Cosine similarity is used to find songs with similar patterns as the input ones from users. That is, feature variables are extracted from the trained neural network model before the classification layer, and used as the basis to find similar songs. One potential problem of this approach is that if the neural network classifies an input song incorrectly, the extracted feature vector will not be a good representation of the input song. Thus, a song that is in fact really similar to the input song may have a small cosine similarity value, i.e. not be recommended. In conclusion, if the first classification is wrong, future inferences based on that is going to make it deviate further from the true answer. A possible future improvement will be how to offset this inference error.<br />
# In the tables when comparing performance and accuracies of the CNN and CRNN models on different genres of music, the researchers claimed that CRNN had superior performance to CNN models. This seemed intuitive, especially in the cases when the differences in accuracies were large. However, maybe the researchers should consider including some hypothesis testing statistics in such tables, which would support such claims in a more rigorous manner.<br />
# A music recommender system that doesn't use the song's meta data such as artist and genre and rather tries to classify genre itself seems unproductive. I also believe that the specific artist matters much more than the genre since within a genre you have many different styles. It just seems like the authors hamstring their recommender system by excluding other relevant data.<br />
# The genres that are posed in the paper are very broad and may not be specific enough to distinguish a listeners actual tastes (ie, I like rock and roll, but not punk rock, which could both be in the "rock" category). It would be interesting to run similar experiments with more concrete and specific genres to study the possibility of improving accuracy in the model.<br />
# This summary is well organized with detailed explanation to the music recommendation algorithm. However, since the data used in this paper is cleaned to buffer the efficiency of the recommendation, there should be a section evaluating the impact of noise on the performance this algorithm and how to minimize the impact.<br />
# This method will be better if the user choose some certain music genres that they like while doing the sign-up process. This is similar to recommending articles on twitter.<br />
# I have some feedback for the "Evaluation of Music Recommendation System" section. Firstly, there can be a brief mention of the participants' background information. Secondly, the summary mentions that "participants choose 5 pieces of music they enjoyed". Are they free to choose any music they like, or are they choosing from a pool of selections? What are the lengths of these music pieces? Lastly, method one and method two are compared against each other. It's intuitive that method two will outperform method one, since method two makes use of both cosine similarity and information on music genre, whereas method one only makes use of cosine similarity. Thus, saying method two outperforms method one is not necessarily surprising. I would like to see more explanation on why these methods are chosen, and why comparing them directly is considered to be fair.<br />
# It would be better to have more comparison with other existing music recommender system.<br />
# In the Collecting Music Data section, the author has indicated that for maintaining the balance of data for each genre that they are choosing to omit some genres and a portion of the dataset. However, how this was done was not explained explicitly which can be a concern for results replication. It would be better to describe the steps and measures taken to ensure the actions taken by the teams are reproducible. <br />
# For cleaning data, for training purposes, the team is choosing to omit the ones with lower music quality. While this is a sound option, it can be adjusted that the ratings for the music are deducted to adjust the balance. This could be important since a poor music quality could mean either equipment failure or corrupt server storage or it was a recording of a live performance that often does not have a perfect studio quality yet it would be loved by many real-life users. This omission is not entirely justified and feels like a deliberate adjustment for later results.<br />
# It would be more convincing if the author could provide more comparison between CRNN and CNN.<br />
# How is the result used to recommend songs within genres? It looks like it only predicts what genre the user likes to listen and recommends one of the songs from that genre. How can this recommender system be used to recommend songs within the same genre?<br />
# This [https://arxiv.org/pdf/2006.15795.pdf paper] implements CRNN differently; the CNN and RNN are separate and their resulting matrices and combined later. Would using this version of the CRNN potentially improve the accuracy?<br />
# This kind of approach can be used in implementing other recommender systems for like movies, articles, news, websites etc. It would be helpful if the author could explain and generalize the implementation on other forms of recommender systems.<br />
# The accuracy of the genre classifier seemed really low, considering how distinct the genres sound to humans. The authors recommend adding features to the data but these could likely be extracted from the audio signal. Extra preprocessing would likely go a long way to improve the accuracy.<br />
<br />
== References: ==<br />
Nilashi, M., et.al. ''Collaborative Filtering Recommender Systems''. Research Journal of Applied Sciences, Engineering and Technology 5(16):4168-4182, 2013.<br />
Adiyansjah, Alexander A S Gunawan, Derwin Suhartono, Music Recommender System Based on Genre using Convolutional Recurrent Neural Networks, Procedia Computer Science, https://doi.org/10.1016/j.procs.2019.08.146.</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=Deep_Learning_for_Cardiologist-level_Myocardial_Infarction_Detection_in_Electrocardiograms&diff=49333Deep Learning for Cardiologist-level Myocardial Infarction Detection in Electrocardiograms2020-12-06T06:37:18Z<p>Iaoellme: /* Results */</p>
<hr />
<div><br />
== Presented by ==<br />
<br />
Zihui (Betty) Qin, Wenqi (Maggie) Zhao, Muyuan Yang, Amartya (Marty) Mukherjee<br />
<br />
== Introduction ==<br />
<br />
This paper presents ConvNetQuake, an approach on detecting heart disease from ECG signals by fine-tuning the deep learning neural network. For context, ConvNetQuake is a convolutional neural network, used by Perol, Gharbi, and Denolle [4], for Earthquake detection and location from a single waveform. A deep learning approach was used due to the model's ability to be trained using multiple GPUs and terabyte-sized datasets. This, in turn, creates a model that is robust against noise. The purpose of this paper is to provide detailed analyses of the contributions of the ECG leads on identifying heart disease, to show the use of multiple channels in ConvNetQuake enhances prediction accuracy, and to show that feature engineering is not necessary for any of the training, validation, or testing processes. In this area, the combination of data fusion and machine learning techniques exhibits great promise to healthcare innovation, and the analyses in this paper help further this realization. The benefits of translating knowledge between deep learning and its real-world applications in health are also illustrated.<br />
<br />
== Previous Work and Motivation ==<br />
<br />
The database used in previous works is the Physikalisch-Technische Bundesanstalt (PTB) database, which consists of ECG records. Previous papers used techniques, such as CNN, SVM, K-nearest neighbors, naïve Bayes classification, and ANN. From these instances, the paper observes several shortcomings in the previous papers. The first being the issue that most papers use feature selection on the raw ECG data before training the model. Dabanloo and Attarodi [2] used various techniques such as ANN, K-nearest neighbors, and Naïve Bayes. However, they extracted two features, the T-wave integral and the total integral, to aid in localizing and detecting heart disease. Sharma and Sunkaria [3] used SVM and K-nearest neighbors as their classifier, but extracted various features using stationary wavelet transforms to decompose the ECG signal into sub-bands. The second issue is that papers that do not use feature selection would arbitrarily pick ECG leads for classification without rationale. For example, Liu et al. [1] used a deep CNN that uses 3 seconds of ECG signal from lead II at a time as input. The decision for using lead II compared to the other leads was not explained. <br />
<br />
The issue with feature selection is that it can be time-consuming and impractical with large volumes of data. The second issue with the arbitrary selection of leads is that it does not offer insight into why the lead was chosen and the contributions of each lead in the identification of heart disease. Thus, this paper addresses these two issues through implementing a deep learning model that does not rely on feature selection of ECG data and to quantify the contributions of each ECG and Frank lead in identifying heart disease.<br />
<br />
== Model Architecture ==<br />
<br />
The dataset, which was used to train, validate, and test the neural network models, consists of 549 ECG records taken from 290 unique patients. Each ECG record has a mean length of over 100 seconds.<br />
<br />
This Deep Neural Network model was created by modifying the ConvNetQuake model by adding 1D batch normalization layers; this addition helps to combat overfitting. A second modification that was made was to introduce the use of label smoothing, which can help by discouraging the model from making overconfident predictions. Label smoothing refers to the method of relaxing the confidence on the model's prediction labels. The authors' experiments demonstrated that both of these modifications helped to increase model accuracy. <br />
<br />
During the training stage, a 10-second long two-channel input was fed into the neural network. In order to ensure that the two channels were weighted equally, both channels were normalized. Besides, time invariance was incorporated by selecting the 10-second long segment randomly from the entire signal. <br />
<br />
The input layer is a 10-second long ECG signal. There are 8 hidden layers in this model, each of which consists of a 1D convolution layer with the ReLu activation function followed by a batch normalization layer. The output layer is a one-dimensional layer that uses the Sigmoid activation function.<br />
<br />
This model is trained by using batches of size 10. The learning rate is <math>10^{-4}</math>. The ADAM optimizer is used. The ADAM (adaptive moment estimation) optimizer is a stochastic gradient optimization method that uses adaptive learning rates for the parameters used in the estimating the gradient's first and second moments [5]. In training the model, the dataset is split into a train set, validation set, and test set with ratios 80-10-10.<br />
<br />
During the training process, the model was trained from scratch numerous times to avoid inserting unintended variation into the model by randomly initializing weights.<br />
<br />
The following images gives a visual representation of the model.<br />
<br />
[[File:architecture.png | thumb | center | 1000px | Model Architecture (Gupta et al., 2019)]]<br />
<br />
==Results== <br />
<br />
The paper first uses quantification of accuracies for single channels with 20-fold cross-validation, resulting in the highest individual accuracies: v5, v6, vx, vz, and ii. The researchers further investigated the accuracies for pairs of the top 5 highest individual channels using 20-fold cross-validation. They arrived at the conclusion that the highest pairs accuracies to feed into a neural network are lead v6 and lead vz. They then use 100-fold cross validation on v6 and vz pair of channels, compare outliers based on top 20, top 50 and total 100 performing models, finding that standard deviation is non-trivial and there are few models performed very poorly. <br />
<br />
Next, they discussed 2 factors affecting model performance evaluation: 1） Random train-val-test split might have effects on the performance of the model, but it can be improved by access with a larger data set and further discussion; and 2） random initialization of the weights of the neural network shows little effects on the performance of the model performance evaluation, because of showing high average results with a fixed train-val-test split. <br />
<br />
Comparing with other models in the other 12 papers, the model in this article has the highest accuracy, specificity, and precision. The dataset contained 549 records from 290 unique patients. In order to ensure that the model did not overfit specific patient profiles, they performed a patient-wise split, where all records associated with a given patient are either in test data or train data (but not both). They tested the 290 fold patient-wise split, resulting in the same highest accuracy of the pair v6 and vz same as record-wise split. The second best pair was ii and vz, which also contains the vz channel. Combining the two best pair channels into v6, vz, vii ultimately gave the best results over 10 trials which has an average of 97.83% in patient-wise split. Even though the patient-wise split might result in lower accuracy evaluation, however, it still maintains a very high average.<br />
<br />
==Conclusion & Discussion== <br />
<br />
The paper introduced a new architecture for heart condition classification based on raw ECG signals using multiple leads. It outperformed the state-of-art model by a large margin of 1 percent. This study finds that out of the 15 ECG channels(12 conventional ECG leads and 3 Frank Leads), channel v6, vz, and ii contain the most meaningful information for detecting myocardial infarction. Also, recent advances in machine learning can be leveraged to produce a model capable of classifying myocardial infraction with a cardiologist-level success rate. To further improve the performance of the models, access to a larger labeled data set is needed. The PTB database is small. It is difficult to test the true robustness of the model with a relatively small test set. If a larger data set can be found to help correctly identify other heart conditions beyond myocardial infraction, the research group plans to share the deep learning models and develop an open-source, computationally efficient app that can be readily used by cardiologists.<br />
<br />
A detailed analysis of the relative importance of each of the 15 ECG channels indicates that deep learning can identify myocardial infraction by processing only ten seconds of raw ECG data from the v6, vz, and ii leads and reaches a cardiologist-level success rate. Deep learning algorithms may be readily used as commodity software. The neural network model that was originally designed to identify earthquakes may be re-designed and tuned to identify myocardial infarction. Feature engineering of ECG data is not required to identify myocardial infraction in the PTB database. This model only required ten seconds of raw ECG data to identify this heart condition with cardiologist-level performance. Access to a larger database should be provided to deep learning researchers so they can work on detecting different types of heart conditions. Deep learning researchers and the cardiology community can work together to develop deep learning algorithms that provide trustworthy, real-time information regarding heart conditions with minimal computational resources.<br />
<br />
Fourier Transform (such as FFT) can be helpful when dealing with ECG signals. It transforms signals from the time domain to the frequency domain, which means some hidden features in frequency may be discovered.<br />
<br />
A limitation specified by the authors is the lack of labeled data. The use of a small dataset such as PTB makes it difficult to determine the robustness of the model due to the small size of the test set. Given a larger dataset, the model could be tested to see if it generalizes to identify heart conditions other than myocardial infarction.<br />
<br />
==Critiques==<br />
- The lack of large, labelled data sets is often a common problem in most applied deep learning studies. Since the PTB database is as small as you describe it to be, the robustness of the model which may be hard to gauge. There are very likely various other physical factors that may play a role in the study which the deep neural network may not be able to adjust for as well, since health data can be somewhat subjective at times and/or may be somewhat inaccurate, especially if machines are used to measurement. This might mean error was propagated forward in the study.<br />
<br />
- Additionally, there is a risk of confirmation bias, which may occur when a model is self-training, especially given the fact that the training set is small.<br />
<br />
- I feel that the results of deep learning models in medical settings where the consequences of misclassification can be severe should be evaluated by assigning weights to classification. In case if the misclassification can lead to severe consequences, then the network should be trained in such a way that it errs towards safety. For example, in case if heart disease, the consequences will be very high if the system says that there is no heart disease when in fact there is. So, the evaluation metric must be selected carefully.<br />
<br />
- This is a useful and meaningful application topic in machine learning. Using Deep Learning to detect heart disease can be very helpful if it is difficult to detect disease by looking at ECG by humans eys. This model also useful for doing statistics, such as calculating the percentage of people get heart disease. But I think the doctor should not 100% trust the result from the model, it is almost impossible to get 100% accuracy from a model. So, I think double-checking by human eyes is necessary if the result is weird. What is more, I think it will be interesting to discuss more applications in mediccal by using this method, such as detecting the Brainwave diagram to predict a person's mood and to diagnose mental diseases.<br />
<br />
- Compared to the dataset for other topics such as object recognition, the PTB database is pretty small with only 549 ECG records. And these are highly unbiased (Table 1) with 4 records for myocarditis and 148 for myocardial infarction. Medical datasets can only be labeled by specialists. This is why these datasets are related small. It would be great if there will be a larger, more comprehensive dataset.<br />
<br />
- Only results using 20-fold cross validation were presented. It should be shown that the results could be reproduced using a more common number of folds like 5 or 10<br />
<br />
- There are potential issues with the inclusion of Frank leads. From a practitioner standpoint, ECGs taken with Frank leads are less common. This could prevent the use of this technique. Additionally, Frank leads are expressible as a linear combinations of the 12 traditional leads. The authors are not adding any fundamentally new information by including them and their inclusion could be viewed as a form of feature selection (going against the authors' original intentions).<br />
<br />
- It will better if we can see how the model in this paper outperformed those methods that used feature selections. The details of the results are not enough.<br />
<br />
- A new extended dataset for PTB dubbed [https://www.nature.com/articles/s41597-020-0495-6 PTB-XL], has 21837 records. Using this dataset could yield a more accurate result, since the original PTB's small dataset posed limitations on the deep learning model.<br />
<br />
- The paper mentions that it has better results, but by how much? what accuracy did the methods you compared to have? Also, what methods did you compare to? (Authors mentioned feature engineering methods but this is vague) Also how much were the labels smoothed? (i.e. 1 -> 0.99 or 1-> 0.95 for example) How much of a difference did the label smoothing make?<br />
<br />
- It is nice to see that the authors also considered training and testing the model on data via a patient-wise split, which gives more insights towards the cases when a patient has multiple records of diagnosis. Obviously and similar to what other critiques suggested, using a patient-wise split might disadvantage from the lack of training data, given that there are only 290 unique patients in the PTB database. Also, acquiring prior knowledge from professionals about correlations, such as causal relationships, between different diagnoses might be helpful for improving the model.<br />
<br />
- As mentioned above, the dataset is comparably small in the context of machine learning. While on the other hand, each record has a length of roughly 100 seconds, which is significantly large as a single input. Therefore, it might be helpful to apply data augmentation algorithms during data preprocessing sections so that there will be a more reasonable dataset than what we currently have so far, which has a high chance of being biased or overfitted.<br />
<br />
- There are several points from the Model Architecture section that can be improved. It mentions that both 1d batch normalization layers and label smoothing are used to improve the accuracy of the models, based on empirical experiment results. Yet, there's no breakdown of how each of these two method improves the accuracy. So it's left unclear whether each method is significant on its own, or the model simultaneously requires both methods in order to achieve improved accuracy. Some more data can be provided about this. It's mentioned that "models are trained from scratch numerous times." How many times is numerous times? Can we get the exact number? Training time about the models should also be provided. This is because if these models take a long time to train, then training them from scratch every time may cause issues with respect to runtime.<br />
<br />
- The authors should have indicated how much the accuracy has been improved by what method. It is a little unclear that how can we define "better results". Also, this paper could be more clear if they included the details about the Model Architecture such as how it was performed and how long was the training time for the model.<br />
<br />
- The summary is lacking several components such as explanation of model, data-preprocessing, result visualization and such. It is hard to understand how the result improved since there is no comparison. Information about dataset is unclear too, it is not explained well what they are and how they are populated.<br />
<br />
- The authors didn't specify how many epochs the model ran for. A common practice when dealing with small datasets is to run more epochs at the risk of overfitting. However the use of batch normalization (and perhaps the introduction of Dropout layers) aid in preventing the model to overfitting the data or affirming the bias of the dataset so more epochs may have improved performance in this case.<br />
<br />
- It is difficult to justify the effectiveness of deep learning for detecting myocardial infarction in EKG due to the lack of information available on the deep learning structure. Meanwhile, false negatives and false positives must be as close to 0 as possible, therefore the authors should test their algorithm on a variety of datasets before determining if deep learning is effective.<br />
<br />
- The authors do not motivate the use of ConvNetQuake as their baseline model for deep transfer learning. There are likely several other model candidates that perform similar signal processing related tasks such as CNN models for gravitational wave detection.<br />
<br />
== References ==<br />
<br />
[1] Na Liu et al. "A Simple and Effective Method for Detecting Myocardial Infarction Based on Deep Convolutional Neural Network". In: Journal of Medical Imaging and Health Informatics (Sept. 2018). doi: 10.1166/jmihi.2018.2463.<br />
<br />
[2] Naser Safdarian, N.J. Dabanloo, and Gholamreza Attarodi. "A New Pattern Recognition Method for Detection and Localization of Myocardial Infarction Using T-Wave Integral and Total Integral as Extracted Features from One Cycle of ECG Signal". In: J. Biomedical Science and Engineering (Aug. 2014). doi: http://dx.doi.org/10.4236/jbise.2014.710081.<br />
<br />
[3] L.D. Sharma and R.K. Sunkaria. "Inferior myocardial infarction detection using stationary wavelet transform and machine learning approach." In: Signal, Image and Video Processing (July 2017). doi: https://doi.org/10.1007/s11760-017-1146-z.<br />
<br />
[4] Perol Thibaut, Gharbi Michaël, and Denolle Marin. "Convolutional neural network for earthquake detection and location". In: Science Advances (Feb. 2018). doi: 10.1126/sciadv.1700578<br />
<br />
[5] Kingma, D. and Ba, J., 2015. Adam: A Method for Stochastic Optimization. In: International Conference for Learning Representations. [online] San Diego: 3rd International Conference for Learning Representations, p.1. Available at: <https://arxiv.org/pdf/1412.6980.pdf> [Accessed 3 December 2020].</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=Deep_Learning_for_Cardiologist-level_Myocardial_Infarction_Detection_in_Electrocardiograms&diff=49332Deep Learning for Cardiologist-level Myocardial Infarction Detection in Electrocardiograms2020-12-06T06:33:25Z<p>Iaoellme: /* Results */</p>
<hr />
<div><br />
== Presented by ==<br />
<br />
Zihui (Betty) Qin, Wenqi (Maggie) Zhao, Muyuan Yang, Amartya (Marty) Mukherjee<br />
<br />
== Introduction ==<br />
<br />
This paper presents ConvNetQuake, an approach on detecting heart disease from ECG signals by fine-tuning the deep learning neural network. For context, ConvNetQuake is a convolutional neural network, used by Perol, Gharbi, and Denolle [4], for Earthquake detection and location from a single waveform. A deep learning approach was used due to the model's ability to be trained using multiple GPUs and terabyte-sized datasets. This, in turn, creates a model that is robust against noise. The purpose of this paper is to provide detailed analyses of the contributions of the ECG leads on identifying heart disease, to show the use of multiple channels in ConvNetQuake enhances prediction accuracy, and to show that feature engineering is not necessary for any of the training, validation, or testing processes. In this area, the combination of data fusion and machine learning techniques exhibits great promise to healthcare innovation, and the analyses in this paper help further this realization. The benefits of translating knowledge between deep learning and its real-world applications in health are also illustrated.<br />
<br />
== Previous Work and Motivation ==<br />
<br />
The database used in previous works is the Physikalisch-Technische Bundesanstalt (PTB) database, which consists of ECG records. Previous papers used techniques, such as CNN, SVM, K-nearest neighbors, naïve Bayes classification, and ANN. From these instances, the paper observes several shortcomings in the previous papers. The first being the issue that most papers use feature selection on the raw ECG data before training the model. Dabanloo and Attarodi [2] used various techniques such as ANN, K-nearest neighbors, and Naïve Bayes. However, they extracted two features, the T-wave integral and the total integral, to aid in localizing and detecting heart disease. Sharma and Sunkaria [3] used SVM and K-nearest neighbors as their classifier, but extracted various features using stationary wavelet transforms to decompose the ECG signal into sub-bands. The second issue is that papers that do not use feature selection would arbitrarily pick ECG leads for classification without rationale. For example, Liu et al. [1] used a deep CNN that uses 3 seconds of ECG signal from lead II at a time as input. The decision for using lead II compared to the other leads was not explained. <br />
<br />
The issue with feature selection is that it can be time-consuming and impractical with large volumes of data. The second issue with the arbitrary selection of leads is that it does not offer insight into why the lead was chosen and the contributions of each lead in the identification of heart disease. Thus, this paper addresses these two issues through implementing a deep learning model that does not rely on feature selection of ECG data and to quantify the contributions of each ECG and Frank lead in identifying heart disease.<br />
<br />
== Model Architecture ==<br />
<br />
The dataset, which was used to train, validate, and test the neural network models, consists of 549 ECG records taken from 290 unique patients. Each ECG record has a mean length of over 100 seconds.<br />
<br />
This Deep Neural Network model was created by modifying the ConvNetQuake model by adding 1D batch normalization layers; this addition helps to combat overfitting. A second modification that was made was to introduce the use of label smoothing, which can help by discouraging the model from making overconfident predictions. Label smoothing refers to the method of relaxing the confidence on the model's prediction labels. The authors' experiments demonstrated that both of these modifications helped to increase model accuracy. <br />
<br />
During the training stage, a 10-second long two-channel input was fed into the neural network. In order to ensure that the two channels were weighted equally, both channels were normalized. Besides, time invariance was incorporated by selecting the 10-second long segment randomly from the entire signal. <br />
<br />
The input layer is a 10-second long ECG signal. There are 8 hidden layers in this model, each of which consists of a 1D convolution layer with the ReLu activation function followed by a batch normalization layer. The output layer is a one-dimensional layer that uses the Sigmoid activation function.<br />
<br />
This model is trained by using batches of size 10. The learning rate is <math>10^{-4}</math>. The ADAM optimizer is used. The ADAM (adaptive moment estimation) optimizer is a stochastic gradient optimization method that uses adaptive learning rates for the parameters used in the estimating the gradient's first and second moments [5]. In training the model, the dataset is split into a train set, validation set, and test set with ratios 80-10-10.<br />
<br />
During the training process, the model was trained from scratch numerous times to avoid inserting unintended variation into the model by randomly initializing weights.<br />
<br />
The following images gives a visual representation of the model.<br />
<br />
[[File:architecture.png | thumb | center | 1000px | Model Architecture (Gupta et al., 2019)]]<br />
<br />
==Results== <br />
<br />
The paper first uses quantification of accuracies for single channels with 20-fold cross-validation, resulting in the highest individual accuracies: v5, v6, vx, vz, and ii. The researchers further investigated the accuracies for pairs of the top 5 highest individual channels using 20-fold cross-validation. They arrived at the conclusion that the highest pairs accuracies to feed into a neural network are lead v6 and lead vz. They then use 100-fold cross validation on v6 and vz pair of channels, compare outliers based on top 20, top 50 and total 100 performing models, finding that standard deviation is non-trivial and there are few models performed very poorly. <br />
<br />
Next, they discussed 2 factors affecting model performance evaluation: 1） Random train-val-test split might have effects on the performance of the model, but it can be improved by access with a larger data set and further discussion; and 2） random initialization of the weights of the neural network shows little effects on the performance of the model performance evaluation, because of showing high average results with a fixed train-val-test split. <br />
<br />
Comparing with other models in the other 12 papers, the model in this article has the highest accuracy, specificity, and precision. With concerns of patients' records affecting the training accuracy, they used 290 fold patient-wise split, resulting in the same highest accuracy of the pair v6 and vz same as record-wise split. The second best pair was ii and vz, which also contains the vz channel. Combining the two best pair channels into v6, vz, vii ultimately gave the best results over 10 trials which has an average of 97.83% in patient-wise split. Even though the patient-wise split might result in lower accuracy evaluation, however, it still maintains a very high average.<br />
<br />
==Conclusion & Discussion== <br />
<br />
The paper introduced a new architecture for heart condition classification based on raw ECG signals using multiple leads. It outperformed the state-of-art model by a large margin of 1 percent. This study finds that out of the 15 ECG channels(12 conventional ECG leads and 3 Frank Leads), channel v6, vz, and ii contain the most meaningful information for detecting myocardial infarction. Also, recent advances in machine learning can be leveraged to produce a model capable of classifying myocardial infraction with a cardiologist-level success rate. To further improve the performance of the models, access to a larger labeled data set is needed. The PTB database is small. It is difficult to test the true robustness of the model with a relatively small test set. If a larger data set can be found to help correctly identify other heart conditions beyond myocardial infraction, the research group plans to share the deep learning models and develop an open-source, computationally efficient app that can be readily used by cardiologists.<br />
<br />
A detailed analysis of the relative importance of each of the 15 ECG channels indicates that deep learning can identify myocardial infraction by processing only ten seconds of raw ECG data from the v6, vz, and ii leads and reaches a cardiologist-level success rate. Deep learning algorithms may be readily used as commodity software. The neural network model that was originally designed to identify earthquakes may be re-designed and tuned to identify myocardial infarction. Feature engineering of ECG data is not required to identify myocardial infraction in the PTB database. This model only required ten seconds of raw ECG data to identify this heart condition with cardiologist-level performance. Access to a larger database should be provided to deep learning researchers so they can work on detecting different types of heart conditions. Deep learning researchers and the cardiology community can work together to develop deep learning algorithms that provide trustworthy, real-time information regarding heart conditions with minimal computational resources.<br />
<br />
Fourier Transform (such as FFT) can be helpful when dealing with ECG signals. It transforms signals from the time domain to the frequency domain, which means some hidden features in frequency may be discovered.<br />
<br />
A limitation specified by the authors is the lack of labeled data. The use of a small dataset such as PTB makes it difficult to determine the robustness of the model due to the small size of the test set. Given a larger dataset, the model could be tested to see if it generalizes to identify heart conditions other than myocardial infarction.<br />
<br />
==Critiques==<br />
- The lack of large, labelled data sets is often a common problem in most applied deep learning studies. Since the PTB database is as small as you describe it to be, the robustness of the model which may be hard to gauge. There are very likely various other physical factors that may play a role in the study which the deep neural network may not be able to adjust for as well, since health data can be somewhat subjective at times and/or may be somewhat inaccurate, especially if machines are used to measurement. This might mean error was propagated forward in the study.<br />
<br />
- Additionally, there is a risk of confirmation bias, which may occur when a model is self-training, especially given the fact that the training set is small.<br />
<br />
- I feel that the results of deep learning models in medical settings where the consequences of misclassification can be severe should be evaluated by assigning weights to classification. In case if the misclassification can lead to severe consequences, then the network should be trained in such a way that it errs towards safety. For example, in case if heart disease, the consequences will be very high if the system says that there is no heart disease when in fact there is. So, the evaluation metric must be selected carefully.<br />
<br />
- This is a useful and meaningful application topic in machine learning. Using Deep Learning to detect heart disease can be very helpful if it is difficult to detect disease by looking at ECG by humans eys. This model also useful for doing statistics, such as calculating the percentage of people get heart disease. But I think the doctor should not 100% trust the result from the model, it is almost impossible to get 100% accuracy from a model. So, I think double-checking by human eyes is necessary if the result is weird. What is more, I think it will be interesting to discuss more applications in mediccal by using this method, such as detecting the Brainwave diagram to predict a person's mood and to diagnose mental diseases.<br />
<br />
- Compared to the dataset for other topics such as object recognition, the PTB database is pretty small with only 549 ECG records. And these are highly unbiased (Table 1) with 4 records for myocarditis and 148 for myocardial infarction. Medical datasets can only be labeled by specialists. This is why these datasets are related small. It would be great if there will be a larger, more comprehensive dataset.<br />
<br />
- Only results using 20-fold cross validation were presented. It should be shown that the results could be reproduced using a more common number of folds like 5 or 10<br />
<br />
- There are potential issues with the inclusion of Frank leads. From a practitioner standpoint, ECGs taken with Frank leads are less common. This could prevent the use of this technique. Additionally, Frank leads are expressible as a linear combinations of the 12 traditional leads. The authors are not adding any fundamentally new information by including them and their inclusion could be viewed as a form of feature selection (going against the authors' original intentions).<br />
<br />
- It will better if we can see how the model in this paper outperformed those methods that used feature selections. The details of the results are not enough.<br />
<br />
- A new extended dataset for PTB dubbed [https://www.nature.com/articles/s41597-020-0495-6 PTB-XL], has 21837 records. Using this dataset could yield a more accurate result, since the original PTB's small dataset posed limitations on the deep learning model.<br />
<br />
- The paper mentions that it has better results, but by how much? what accuracy did the methods you compared to have? Also, what methods did you compare to? (Authors mentioned feature engineering methods but this is vague) Also how much were the labels smoothed? (i.e. 1 -> 0.99 or 1-> 0.95 for example) How much of a difference did the label smoothing make?<br />
<br />
- It is nice to see that the authors also considered training and testing the model on data via a patient-wise split, which gives more insights towards the cases when a patient has multiple records of diagnosis. Obviously and similar to what other critiques suggested, using a patient-wise split might disadvantage from the lack of training data, given that there are only 290 unique patients in the PTB database. Also, acquiring prior knowledge from professionals about correlations, such as causal relationships, between different diagnoses might be helpful for improving the model.<br />
<br />
- As mentioned above, the dataset is comparably small in the context of machine learning. While on the other hand, each record has a length of roughly 100 seconds, which is significantly large as a single input. Therefore, it might be helpful to apply data augmentation algorithms during data preprocessing sections so that there will be a more reasonable dataset than what we currently have so far, which has a high chance of being biased or overfitted.<br />
<br />
- There are several points from the Model Architecture section that can be improved. It mentions that both 1d batch normalization layers and label smoothing are used to improve the accuracy of the models, based on empirical experiment results. Yet, there's no breakdown of how each of these two method improves the accuracy. So it's left unclear whether each method is significant on its own, or the model simultaneously requires both methods in order to achieve improved accuracy. Some more data can be provided about this. It's mentioned that "models are trained from scratch numerous times." How many times is numerous times? Can we get the exact number? Training time about the models should also be provided. This is because if these models take a long time to train, then training them from scratch every time may cause issues with respect to runtime.<br />
<br />
- The authors should have indicated how much the accuracy has been improved by what method. It is a little unclear that how can we define "better results". Also, this paper could be more clear if they included the details about the Model Architecture such as how it was performed and how long was the training time for the model.<br />
<br />
- The summary is lacking several components such as explanation of model, data-preprocessing, result visualization and such. It is hard to understand how the result improved since there is no comparison. Information about dataset is unclear too, it is not explained well what they are and how they are populated.<br />
<br />
- The authors didn't specify how many epochs the model ran for. A common practice when dealing with small datasets is to run more epochs at the risk of overfitting. However the use of batch normalization (and perhaps the introduction of Dropout layers) aid in preventing the model to overfitting the data or affirming the bias of the dataset so more epochs may have improved performance in this case.<br />
<br />
- It is difficult to justify the effectiveness of deep learning for detecting myocardial infarction in EKG due to the lack of information available on the deep learning structure. Meanwhile, false negatives and false positives must be as close to 0 as possible, therefore the authors should test their algorithm on a variety of datasets before determining if deep learning is effective.<br />
<br />
- The authors do not motivate the use of ConvNetQuake as their baseline model for deep transfer learning. There are likely several other model candidates that perform similar signal processing related tasks such as CNN models for gravitational wave detection.<br />
<br />
== References ==<br />
<br />
[1] Na Liu et al. "A Simple and Effective Method for Detecting Myocardial Infarction Based on Deep Convolutional Neural Network". In: Journal of Medical Imaging and Health Informatics (Sept. 2018). doi: 10.1166/jmihi.2018.2463.<br />
<br />
[2] Naser Safdarian, N.J. Dabanloo, and Gholamreza Attarodi. "A New Pattern Recognition Method for Detection and Localization of Myocardial Infarction Using T-Wave Integral and Total Integral as Extracted Features from One Cycle of ECG Signal". In: J. Biomedical Science and Engineering (Aug. 2014). doi: http://dx.doi.org/10.4236/jbise.2014.710081.<br />
<br />
[3] L.D. Sharma and R.K. Sunkaria. "Inferior myocardial infarction detection using stationary wavelet transform and machine learning approach." In: Signal, Image and Video Processing (July 2017). doi: https://doi.org/10.1007/s11760-017-1146-z.<br />
<br />
[4] Perol Thibaut, Gharbi Michaël, and Denolle Marin. "Convolutional neural network for earthquake detection and location". In: Science Advances (Feb. 2018). doi: 10.1126/sciadv.1700578<br />
<br />
[5] Kingma, D. and Ba, J., 2015. Adam: A Method for Stochastic Optimization. In: International Conference for Learning Representations. [online] San Diego: 3rd International Conference for Learning Representations, p.1. Available at: <https://arxiv.org/pdf/1412.6980.pdf> [Accessed 3 December 2020].</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=User:J46hou&diff=48309User:J46hou2020-11-30T04:55:54Z<p>Iaoellme: /* Introduction */</p>
<hr />
<div>DROCC: Deep Robust One-Class Classification<br />
== Presented by == <br />
Jinjiang Lian, Yisheng Zhu, Jiawen Hou, Mingzhe Huang<br />
== Introduction ==<br />
In this paper, the “one-class” classification, whose goal is to obtain accurate discriminators for a special class, has been studied. Popular uses of this technique include anomaly detection, which is widely used to detect unusual patterns in data. Anomaly detection is a well-studied area of research that aims to learn a model that accurately describes "normality". It has many applications, such as risk assessment for security purposes in many fields, health, and medical risk. However, the conventional approach of modeling with typical data using a simple function falls short when it comes to complex domains such as vision or speech. Another case where this would be useful is when recognizing a “wake-word” while waking up AI systems such as Alexa. <br />
<br />
Deep learning based on anomaly detection methods attempts to learn features automatically but has some limitations. One approach is based on extending the classical data modeling techniques over the learned representations, but in this case, all the points may be mapped to a single point, making the layer look "perfect". The second approach is based on learning the salient geometric structure of data and training the discriminator to predict the applied transformation. The result could be considered anomalous if the discriminator fails to predict the transformation accurately.<br />
<br />
Thus, in this paper, a new approach called Deep Robust One-Class Classification (DROCC) was presented to solve the above concerns. DROCC is based on the assumption that the points from the class of interest lie on a well-sampled, locally linear low-dimensional manifold. More specifically, we are presenting DROCC-LF which is an outlier-exposure style extension of DROCC. This extension combines the DROCC's anomaly detection loss with standard classification loss over the negative data and exploits the negative examples to learn a Mahalanobis distance.<br />
<br />
== Previous Work ==<br />
Traditional approaches for one-class problems include one-class SVM (Scholkopf et al., 1999) and Isolation Forest (Liu et al., 2008)[9]. One drawback of these approaches is that they involve careful feature engineering when applied to structured domains like images. The current state of the art methodologies to tackle these kinds of problems are: <br />
<br />
1. Approach based on prediction transformations (Golan & El-Yaniv, 2018; Hendricks et al.,2019a) [1]. This work is based on learning the salient geometric structure of typical data by applying specific transformations to the input data and training the discriminator to predict the applied transformation. This approach has some shortcomings in the sense that it depends heavily on an appropriate domain-specific set of transformations that are in general hard to obtain. <br />
<br />
2. Approach of minimizing a classical one-class loss on the learned final layer representations such as DeepSVDD. (Ruff et al.,2018)[2] This such work has proposed some heuristics to mitigate issues like setting the bias to zero but it is often insufficient in practice. This method suffers from the fundamental drawback of representation collapse, where the learned transformation might map all the points to a single point (like the origin), leading to a degenerate solution and poor discrimination between normal points and the anomalous points.<br />
<br />
3. Approach based on balancing unbalanced training datasets using methods such as SMOTE to synthetically create outlier data to train models on.<br />
<br />
== Motivation ==<br />
Anomaly detection is a well-studied problem with a large body of research (Aggarwal, 2016; Chandola et al., 2009) [3]. The goal is to identify the outliers: points which are not following a typical distribution. The following image provides a visual representation of an outlier/anomaly. <br />
[[File:abnormal.jpeg | thumb | center | 1000px | Abnormal Data (Data Driven Investor, 2020)]]<br />
Classical approaches for anomaly detection are based on modeling the typical data using simple functions over the low-dimensional subspace or a tree-structured partition of the input space to detect anomalies (Schölkopf et al., 1999; Liu et al., 2008; Lakhina et al., 2004) [4], such as constructing a minimum-enclosing ball around the typical data points (Tax & Duin, 2004) [5]. They broadly fall into three categories: AD via generative modeling, Deep Once Class SVM, Transformations based methods, and Side-information based AD. While these techniques are well-suited when the input is featured appropriately, they struggle on complex domains like vision and speech, where hand-designing features are difficult.<br />
<br />
'''AD via Generative Modeling:''' involves deep autoencoders and GAN based methods and have been deeply studied. But, this method solves a much harder problem than required and reconstructs the entire input during the decoding step.<br />
<br />
'''Deep One-Class SVM:''' Deep SVDD attempts to learn a neural network which maps data into a hypersphere. Mappings which fall within the hypersphere are considered "normal". It was the first method to introduce deep one-class classification for the purpose of anomaly detection, but is impeded by representation collapse.<br />
<br />
'''Transformations based methods:''' Are more recent methods that are based on self-supervised training. The training process of these methods applies transformations to the regular points and training the classifier to identify the transformations used. The model relies on the assumption that a point is normal iff the transformations applied to the point can be identified. Some proposed transformations are as simple as rotations and flips, or can be handcrafted and much more complicated. The various transformations that have been proposed are heavily domain dependent and are hard to design.<br />
<br />
'''Side-information based AD:''' incorporate labelled anomalous data or out-of-distribution samples. DROCC makes no assumptions regarding access to side-information.<br />
<br />
Another related problem is the one-class classification under limited negatives (OCLN). In this case, only a few negative samples are available. The goal is to find a classifier that would not misfire close negatives so that the false positive rate will be low. <br />
<br />
DROCC is robust to representation collapse by involving a discriminative component that is general and empirically accurate on most standard domains like tabular, time-series and vision without requiring any additional side information. DROCC is motivated by the key observation that generally, the typical data lies on a low-dimensional manifold, which is well-sampled in the training data. This is believed to be true even in complex domains such as vision, speech, and natural language (Pless & Souvenir, 2009). [6]<br />
<br />
== Model Explanation ==<br />
[[File:drocc_f1.jpg | center]]<br />
<div align="center">'''Figure 1'''</div><br />
<br />
(a): A normal data manifold with red dots representing generated anomalous points in Ni(r). <br />
<br />
(b): Decision boundary learned by DROCC when applied to the data from (a). Blue represents points classified as normal and red points are classified as abnormal. We observe from here that DROCC is able to capture the manifold accurately; whereas the classical methods, OC-SVM and DeepSVDD perform poorly as they both try to learn a minimum enclosing ball for the whole set of positive data points. <br />
<br />
(c), (d): First two dimensions of the decision boundary of DROCC and DROCC–LF, when applied to noisy data (Section 5.2). DROCC–LF is nearly optimal while DROCC’s decision boundary is inaccurate. Yellow color sine wave depicts the train data.<br />
<br />
== DROCC ==<br />
The model is based on the assumption that the true data lies on a manifold. As manifolds resemble Euclidean space locally, our discriminative component is based on classifying a point as anomalous if it is outside the union of small L2 norm balls around the training typical points (See Figure 1a, 1b for an illustration). Importantly, the above definition allows us to synthetically generate anomalous points, and we adaptively generate the most effective anomalous points while training via a gradient ascent phase reminiscent of adversarial training. In other words, DROCC has a gradient ascent phase to adaptively add anomalous points to our training set and a gradient descent phase to minimize the classification loss by learning a representation and a classifier on top of the representations to separate typical points from the generated anomalous points. In this way, DROCC automatically learns an appropriate representation (like DeepSVDD) but is robust to a representation collapse as mapping all points to the same value would lead to poor discrimination between normal points and the generated anomalous points.<br />
<br />
The algorithm that was used to train the model is laid out below in pseudocode.<br />
<center><br />
[[File:DROCCtrain.png]]<br />
</center><br />
<br />
For a DNN <math>f_\theta: \mathbb{R}^d \to \mathbb{R}</math> that is parameterized by a set of parameters <math>\theta</math>, DROCC estimates <math>\theta^{dr} = \min_\theta\ell^{dr}(\theta)</math> where <br />
$$\ell^{dr}(\theta) = \lambda\|\theta\|^2 + \sum_{i=1}^n[\ell(f_\theta(x_i),1)+\mu\max_{\tilde{x}_i \in N_i(r)}\ell(f_\theta(\tilde{x}_i),-1)]$$<br />
Here, <math>N_i(r) = \{\|\tilde{x}_i-x_i\|_2\leq\gamma\cdot r; r \leq \|\tilde{x}_i - x_j\|, \forall j=1,2,...n\}</math> contains all the points that are at least distance <math>r</math> from the training points. The <math>\gamma \geq 1</math> is a regularization term, and <math>\ell:\mathbb{R} \times \mathbb{R} \to \mathbb{R}</math> is a loss function. The <math>x_i</math> are normal points that should be classified as positive and the <math>\tilde{x}_i</math> are anomalous points that should be classified as negative. This formulation is a saddle point problem.<br />
<br />
== DROCC-LF ==<br />
To especially tackle problems such as anomaly detection and outlier exposure (Hendrycks et al., 2019a) [7], DROCC–LF, an outlier-exposure style extension of DROCC was proposed. Intuitively, DROCC–LF combines DROCC’s anomaly detection loss (that is over only the positive data points) with standard classification loss over the negative data. In addition, DROCC–LF exploits the negative examples to learn a Mahalanobis distance to compare points over the manifold instead of using the standard Euclidean distance, which can be inaccurate for high-dimensional data with relatively fewer samples. (See Figure 1c, 1d for illustration)<br />
<br />
== Popular Dataset Benchmark Result ==<br />
<br />
[[File:drocc_auc.jpg | center]]<br />
<div align="center">'''Figure 2: AUC result'''</div><br />
<br />
The CIFAR-10 dataset consists of 60000 32x32 color images in 10 classes, with 6000 images per class. There are 50000 training images and 10000 test images. The dataset is divided into five training batches and one test batch, each with 10000 images. The test batch contains exactly 1000 randomly selected images from each class. The training batches contain the remaining images in random order, but some training batches may contain more images from one class than another. Between them, the training batches contain exactly 5000 images from each class. The average AUC (with standard deviation) for one-vs-all anomaly detection on CIFAR-10 is shown in table 1. DROCC outperforms baselines on most classes, with gains as high as 20%, and notably, nearest neighbors (NN) beats all the baselines on 2 classes.<br />
<br />
[[File:drocc_f1score.jpg | center]]<br />
<div align="center">'''Figure 3: F1-Score'''</div><br />
<br />
Figure 3 shows F1-Score (with standard deviation) for one-vs-all anomaly detection on Thyroid, Arrhythmia, and Abalone datasets from the UCI Machine Learning Repository. DROCC outperforms the baselines on all three datasets by a minimum of 0.07 which is about an 11.5% performance increase.<br />
Results on One-class Classification with Limited Negatives (OCLN): <br />
[[File:ocln.jpg | center]]<br />
<div align="center">'''Figure 4: Sample positives, negatives and close negatives for MNIST digit 0 vs 1 experiment (OCLN).'''</div><br />
MNIST 0 vs. 1 Classification: <br />
We consider an experimental setup on the MNIST dataset, where the training data consists of Digit 0, the normal class, and Digit 1 as the anomaly. During the evaluation, in addition to samples from training distribution, we also have half zeros, which act as challenging OOD points (close negatives). These half zeros are generated by randomly masking 50% of the pixels (Figure 2). BCE performs poorly, with a recall of 54% only at a fixed FPR of 3%. DROCC–OE gives a recall value of 98:16% outperforming DeepSAD by a margin of 7%, which gives a recall value of 90:91%. DROCC–LF provides further improvement with a recall of 99:4% at 3% FPR. <br />
<br />
[[File:ocln_2.jpg | center]]<br />
<div align="center">'''Figure 5: OCLN on Audio Commands.'''</div><br />
Wake word Detection: <br />
Finally, we evaluate DROCC–LF on the practical problem of wake word detection with low FPR against arbitrary OOD negatives. To this end, we identify a keyword, say “Marvin” from the audio commands dataset (Warden, 2018) [8] as the positive class, and the remaining 34 keywords are labeled as the negative class. For training, we sample points uniformly at random from the above-mentioned dataset. However, for evaluation, we sample positives from the train distribution, but negatives contain a few challenging OOD points as well. Sampling challenging negatives itself is a hard task and is the key motivating reason for studying the problem. So, we manually list close-by keywords to Marvin such as Mar, Vin, Marvelous, etc. We then generate audio snippets for these keywords via a speech synthesis tool 2 with a variety of accents.<br />
Figure 5 shows that for 3% and 5% FPR settings, DROCC–LF is significantly more accurate than the baselines. For example, with FPR=3%, DROCC–LF is 10% more accurate than the baselines. We repeated the same experiment with the keyword: Seven, and observed a similar trend. In summary, DROCC–LF is able to generalize well against negatives that are “close” to the true positives even when such negatives were not supplied with the training data.<br />
<br />
== Conclusion and Future Work ==<br />
We introduced DROCC method for deep anomaly detection. It models normal data points using a low-dimensional sub-manifold inside the feature space, and the anomalous points are characterized via their Euclidean distance from the sub-manifold. Based on this intuition, DROCC’s optimization is formulated as a saddle point problem which is solved via a standard gradient descent-ascent algorithm. We then extended DROCC to OCLN problem where the goal is to generalize well against arbitrary negatives, assuming the positive class is well sampled and a small number of negative points are also available. Both the methods perform significantly better than strong baselines, in their respective problem settings. <br />
<br />
For computational efficiency, we simplified the projection set of both methods which can perhaps slow down the convergence of the two methods. Designing optimization algorithms that can work with the stricter set is an exciting research direction. Further, we would also like to rigorously analyze DROCC, assuming enough samples from a low-curvature manifold. Finally, as OCLN is an exciting problem that routinely comes up in a variety of real-world applications, we would like to apply DROCC–LF to a few high impact scenarios. Possible applications of this work are financial fraud detection, medical anomalies, or key words in audio processing.<br />
<br />
The results of this study showed that DROCC is comparatively better for anomaly detection across many different areas, such as tabular data, images, audio, and time series, when compared to existing state-of-the-art techniques.<br />
<br />
== References ==<br />
[1]: Golan, I. and El-Yaniv, R. Deep anomaly detection using geometric transformations. In Advances in Neural Information Processing Systems (NeurIPS), 2018.<br />
<br />
[2]: Ruff, L., Vandermeulen, R., Goernitz, N., Deecke, L., Siddiqui, S. A., Binder, A., M¨uller, E., and Kloft, M. Deep one-class classification. In International Conference on Machine Learning (ICML), 2018.<br />
<br />
[3]: Aggarwal, C. C. Outlier Analysis. Springer Publishing Company, Incorporated, 2nd edition, 2016. ISBN 3319475770.<br />
<br />
[4]: Sch¨olkopf, B., Williamson, R., Smola, A., Shawe-Taylor, J., and Platt, J. Support vector method for novelty detection. In Proceedings of the 12th International Conference on Neural Information Processing Systems, 1999.<br />
<br />
[5]: Tax, D. M. and Duin, R. P. Support vector data description. Machine Learning, 54(1), 2004.<br />
<br />
[6]: Pless, R. and Souvenir, R. A survey of manifold learning for images. IPSJ Transactions on Computer Vision and Applications, 1, 2009.<br />
<br />
[7]: Hendrycks, D., Mazeika, M., and Dietterich, T. Deep anomaly detection with outlier exposure. In International Conference on Learning Representations (ICLR), 2019a.<br />
<br />
[8]: Warden, P. Speech commands: A dataset for limited vocabulary speech recognition, 2018. URL https: //arxiv.org/abs/1804.03209.<br />
<br />
[9]: Liu, F. T., Ting, K. M., and Zhou, Z.-H. Isolation forest. In Proceedings of the 2008 Eighth IEEE International Conference on Data Mining, 2008.<br />
<br />
== Critiques/Insights ==<br />
<br />
1. It would be interesting to see this implemented in self-driving cars, for instance, to detect unusual road conditions.<br />
<br />
2. Figure 1 shows a good representation on how this model works. However, how can we know that this model is not prone to overfitting? There are many situations where there are valid points that lie outside of the line, especially new data that the model has never see before. An explanation on how this is avoided would be good.<br />
<br />
3.In the introduction part, it should first explain what is "one class", and then make a detailed application. Moreover, special definition words are used in many places in the text. No detailed explanation was given. In the end, the future application fields of DROCC and the research direction of the group can be explained.<br />
<br />
4. It will also be interesting to see if one change from using <math>\ell_{2}</math> Euclidean distance to other distances. When the low-dimensional manifold is highly non-linear, using the local linear distance to characterize anomalous points might fail.<br />
<br />
5. This is a nice summary and the authors introduce clearly on the performance of DROCC. It is nice to use Alexa as an example to catch readers' attention. I think it will be nice to include the algorithm of the DROCC or the architecture of DROCC in this summary to help us know the whole view of this method. Maybe it will be interesting to apply DROCC in biomedical studies? since one-class classification is often used in biomedical studies.<br />
<br />
6. The training method resembles adversarial learning with gradient ascent, however, there is no evaluation of this method on adversarial examples. This is quite unusual considering the paper proposed a method for robust one-class classification, and can be a security threat in real life in critical applications.<br />
<br />
7. The underlying idea behind OCLN is very similar to how neural networks are implemented in recommender systems and trained over positive/negative triplet models. In that case as well, due to the nature of implicit and explicit feedback, positive data tends to dominate the system. It would be interesting to see if insights from that area could be used to further boost the model presented in this paper.<br />
<br />
8. The paper shows the performance of DROCC being evaluated for time series data. It is interesting to see high AUC scores for DROCC against baselines like nearest neighbours and REBMs.Because detecting abnormal data in time series datasets is not common to practice.<br />
<br />
9. Figure1 presented results on a simple 2-D sine wave dataset to visualize the kind of classifiers learnt by DROCC. And the 1a is the positive data lies on a 1-D manifold. We can see from 1b that DROCC is able to capture the manifold accurately.<br />
<br />
10. In the MNIST 0 vs. 1 Classification dataset, why is 1 the only digit that is considered an anomoly? Couldn't all of the non-0 digits be left in the dataset to serve as "anomolies"?<br />
<br />
11. For future work the authors suggest considering DROCC for a low curvature manifold but do not motivate the benefits of such a direction.<br />
<br />
12. One of the problems is that in this model we might need to map all the points to one point to make the layer looks "perfect". However, this might not be a good choice since each point is distinct and if we map them together to one point, then this point cannot tell everything. If authors can specify more details on this it would be better.<br />
<br />
13. This project introduced DROCC for “one-class” classification. It will be interesting if such kind of classification can be compared with any other classification such as binary classification, etc. If “one-class” classification would be more speedy than the others.<br />
<br />
14. The dimensions and feature values must be so different across datasets in different domains. I would love to see how this algorithm is performing so well applied on different domains as it is mentioned that it could be used on datasets including images, audio, time-series, etc.<br />
<br />
15. It would be interesting to show the performance of DROCC against popular models used for outlier prediction such as PCA, EVA, etc. Perhaps show their accuracy scores so we can better compare.<br />
<br />
16. It would be greater if an visualization of how much performance DROCC improved compare to traditional binary classifier like SVM, isolation Forest.</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=User:J46hou&diff=48305User:J46hou2020-11-30T04:52:20Z<p>Iaoellme: /* Conclusion and Future Work */</p>
<hr />
<div>DROCC: Deep Robust One-Class Classification<br />
== Presented by == <br />
Jinjiang Lian, Yisheng Zhu, Jiawen Hou, Mingzhe Huang<br />
== Introduction ==<br />
In this paper, the “one-class” classification, whose goal is to obtain accurate discriminators for a special class, has been studied. Popular uses of this technique include anomaly detection, which is widely used for anomaly detection. Anomaly detection is a well-studied area of research that aims to learn a model that accurately describes "normality". It has many applications, such as risk assessment for security purposes in many fields, health, and medical risk. However, the conventional approach of modeling with typical data using a simple function falls short when it comes to complex domains such as vision or speech. Another case where this would be useful is when recognizing “wake-word” while waking up AI systems such as Alexa. <br />
<br />
Deep learning based on anomaly detection methods attempts to learn features automatically but has some limitations. One approach is based on extending the classical data modeling techniques over the learned representations, but in this case, all the points may be mapped to a single point, making the layer look "perfect". The second approach is based on learning the salient geometric structure of data and training the discriminator to predict the applied transformation. The result could be considered anomalous if the discriminator fails to predict the transformation accurately.<br />
<br />
Thus, in this paper, a new approach called Deep Robust One-Class Classification (DROCC) was presented to solve the above concerns. DROCC is based on the assumption that the points from the class of interest lie on a well-sampled, locally linear low-dimensional manifold. More specifically, we are presenting DROCC-LF which is an outlier-exposure style extension of DROCC. This extension combines the DROCC's anomaly detection loss with standard classification loss over the negative data and exploits the negative examples to learn a Mahalanobis distance.<br />
<br />
== Previous Work ==<br />
Traditional approaches for one-class problems include one-class SVM (Scholkopf et al., 1999) and Isolation Forest (Liu et al., 2008)[9]. One drawback of these approaches is that they involve careful feature engineering when applied to structured domains like images. The current state of the art methodologies to tackle these kinds of problems are: <br />
<br />
1. Approach based on prediction transformations (Golan & El-Yaniv, 2018; Hendricks et al.,2019a) [1]. This work is based on learning the salient geometric structure of typical data by applying specific transformations to the input data and training the discriminator to predict the applied transformation. This approach has some shortcomings in the sense that it depends heavily on an appropriate domain-specific set of transformations that are in general hard to obtain. <br />
<br />
2. Approach of minimizing a classical one-class loss on the learned final layer representations such as DeepSVDD. (Ruff et al.,2018)[2] This such work has proposed some heuristics to mitigate issues like setting the bias to zero but it is often insufficient in practice. This method suffers from the fundamental drawback of representation collapse, where the learned transformation might map all the points to a single point (like the origin), leading to a degenerate solution and poor discrimination between normal points and the anomalous points.<br />
<br />
3. Approach based on balancing unbalanced training datasets using methods such as SMOTE to synthetically create outlier data to train models on.<br />
<br />
== Motivation ==<br />
Anomaly detection is a well-studied problem with a large body of research (Aggarwal, 2016; Chandola et al., 2009) [3]. The goal is to identify the outliers: points which are not following a typical distribution. The following image provides a visual representation of an outlier/anomaly. <br />
[[File:abnormal.jpeg | thumb | center | 1000px | Abnormal Data (Data Driven Investor, 2020)]]<br />
Classical approaches for anomaly detection are based on modeling the typical data using simple functions over the low-dimensional subspace or a tree-structured partition of the input space to detect anomalies (Schölkopf et al., 1999; Liu et al., 2008; Lakhina et al., 2004) [4], such as constructing a minimum-enclosing ball around the typical data points (Tax & Duin, 2004) [5]. They broadly fall into three categories: AD via generative modeling, Deep Once Class SVM, Transformations based methods, and Side-information based AD. While these techniques are well-suited when the input is featured appropriately, they struggle on complex domains like vision and speech, where hand-designing features are difficult.<br />
<br />
'''AD via Generative Modeling:''' involves deep autoencoders and GAN based methods and have been deeply studied. But, this method solves a much harder problem than required and reconstructs the entire input during the decoding step.<br />
<br />
'''Deep One-Class SVM:''' Deep SVDD attempts to learn a neural network which maps data into a hypersphere. Mappings which fall within the hypersphere are considered "normal". It was the first method to introduce deep one-class classification for the purpose of anomaly detection, but is impeded by representation collapse.<br />
<br />
'''Transformations based methods:''' Are more recent methods that are based on self-supervised training. The training process of these methods applies transformations to the regular points and training the classifier to identify the transformations used. The model relies on the assumption that a point is normal iff the transformations applied to the point can be identified. Some proposed transformations are as simple as rotations and flips, or can be handcrafted and much more complicated. The various transformations that have been proposed are heavily domain dependent and are hard to design.<br />
<br />
'''Side-information based AD:''' incorporate labelled anomalous data or out-of-distribution samples. DROCC makes no assumptions regarding access to side-information.<br />
<br />
Another related problem is the one-class classification under limited negatives (OCLN). In this case, only a few negative samples are available. The goal is to find a classifier that would not misfire close negatives so that the false positive rate will be low. <br />
<br />
DROCC is robust to representation collapse by involving a discriminative component that is general and empirically accurate on most standard domains like tabular, time-series and vision without requiring any additional side information. DROCC is motivated by the key observation that generally, the typical data lies on a low-dimensional manifold, which is well-sampled in the training data. This is believed to be true even in complex domains such as vision, speech, and natural language (Pless & Souvenir, 2009). [6]<br />
<br />
== Model Explanation ==<br />
[[File:drocc_f1.jpg | center]]<br />
<div align="center">'''Figure 1'''</div><br />
<br />
(a): A normal data manifold with red dots representing generated anomalous points in Ni(r). <br />
<br />
(b): Decision boundary learned by DROCC when applied to the data from (a). Blue represents points classified as normal and red points are classified as abnormal. We observe from here that DROCC is able to capture the manifold accurately; whereas the classical methods, OC-SVM and DeepSVDD perform poorly as they both try to learn a minimum enclosing ball for the whole set of positive data points. <br />
<br />
(c), (d): First two dimensions of the decision boundary of DROCC and DROCC–LF, when applied to noisy data (Section 5.2). DROCC–LF is nearly optimal while DROCC’s decision boundary is inaccurate. Yellow color sine wave depicts the train data.<br />
<br />
== DROCC ==<br />
The model is based on the assumption that the true data lies on a manifold. As manifolds resemble Euclidean space locally, our discriminative component is based on classifying a point as anomalous if it is outside the union of small L2 norm balls around the training typical points (See Figure 1a, 1b for an illustration). Importantly, the above definition allows us to synthetically generate anomalous points, and we adaptively generate the most effective anomalous points while training via a gradient ascent phase reminiscent of adversarial training. In other words, DROCC has a gradient ascent phase to adaptively add anomalous points to our training set and a gradient descent phase to minimize the classification loss by learning a representation and a classifier on top of the representations to separate typical points from the generated anomalous points. In this way, DROCC automatically learns an appropriate representation (like DeepSVDD) but is robust to a representation collapse as mapping all points to the same value would lead to poor discrimination between normal points and the generated anomalous points.<br />
<br />
The algorithm that was used to train the model is laid out below in pseudocode.<br />
<center><br />
[[File:DROCCtrain.png]]<br />
</center><br />
<br />
For a DNN <math>f_\theta: \mathbb{R}^d \to \mathbb{R}</math> that is parameterized by a set of parameters <math>\theta</math>, DROCC estimates <math>\theta^{dr} = \min_\theta\ell^{dr}(\theta)</math> where <br />
$$\ell^{dr}(\theta) = \lambda\|\theta\|^2 + \sum_{i=1}^n[\ell(f_\theta(x_i),1)+\mu\max_{\tilde{x}_i \in N_i(r)}\ell(f_\theta(\tilde{x}_i),-1)]$$<br />
Here, <math>N_i(r) = \{\|\tilde{x}_i-x_i\|_2\leq\gamma\cdot r; r \leq \|\tilde{x}_i - x_j\|, \forall j=1,2,...n\}</math> contains all the points that are at least distance <math>r</math> from the training points. The <math>\gamma \geq 1</math> is a regularization term, and <math>\ell:\mathbb{R} \times \mathbb{R} \to \mathbb{R}</math> is a loss function. The <math>x_i</math> are normal points that should be classified as positive and the <math>\tilde{x}_i</math> are anomalous points that should be classified as negative. This formulation is a saddle point problem.<br />
<br />
== DROCC-LF ==<br />
To especially tackle problems such as anomaly detection and outlier exposure (Hendrycks et al., 2019a) [7], DROCC–LF, an outlier-exposure style extension of DROCC was proposed. Intuitively, DROCC–LF combines DROCC’s anomaly detection loss (that is over only the positive data points) with standard classification loss over the negative data. In addition, DROCC–LF exploits the negative examples to learn a Mahalanobis distance to compare points over the manifold instead of using the standard Euclidean distance, which can be inaccurate for high-dimensional data with relatively fewer samples. (See Figure 1c, 1d for illustration)<br />
<br />
== Popular Dataset Benchmark Result ==<br />
<br />
[[File:drocc_auc.jpg | center]]<br />
<div align="center">'''Figure 2: AUC result'''</div><br />
<br />
The CIFAR-10 dataset consists of 60000 32x32 color images in 10 classes, with 6000 images per class. There are 50000 training images and 10000 test images. The dataset is divided into five training batches and one test batch, each with 10000 images. The test batch contains exactly 1000 randomly selected images from each class. The training batches contain the remaining images in random order, but some training batches may contain more images from one class than another. Between them, the training batches contain exactly 5000 images from each class. The average AUC (with standard deviation) for one-vs-all anomaly detection on CIFAR-10 is shown in table 1. DROCC outperforms baselines on most classes, with gains as high as 20%, and notably, nearest neighbors (NN) beats all the baselines on 2 classes.<br />
<br />
[[File:drocc_f1score.jpg | center]]<br />
<div align="center">'''Figure 3: F1-Score'''</div><br />
<br />
Figure 3 shows F1-Score (with standard deviation) for one-vs-all anomaly detection on Thyroid, Arrhythmia, and Abalone datasets from the UCI Machine Learning Repository. DROCC outperforms the baselines on all three datasets by a minimum of 0.07 which is about an 11.5% performance increase.<br />
Results on One-class Classification with Limited Negatives (OCLN): <br />
[[File:ocln.jpg | center]]<br />
<div align="center">'''Figure 4: Sample positives, negatives and close negatives for MNIST digit 0 vs 1 experiment (OCLN).'''</div><br />
MNIST 0 vs. 1 Classification: <br />
We consider an experimental setup on the MNIST dataset, where the training data consists of Digit 0, the normal class, and Digit 1 as the anomaly. During the evaluation, in addition to samples from training distribution, we also have half zeros, which act as challenging OOD points (close negatives). These half zeros are generated by randomly masking 50% of the pixels (Figure 2). BCE performs poorly, with a recall of 54% only at a fixed FPR of 3%. DROCC–OE gives a recall value of 98:16% outperforming DeepSAD by a margin of 7%, which gives a recall value of 90:91%. DROCC–LF provides further improvement with a recall of 99:4% at 3% FPR. <br />
<br />
[[File:ocln_2.jpg | center]]<br />
<div align="center">'''Figure 5: OCLN on Audio Commands.'''</div><br />
Wake word Detection: <br />
Finally, we evaluate DROCC–LF on the practical problem of wake word detection with low FPR against arbitrary OOD negatives. To this end, we identify a keyword, say “Marvin” from the audio commands dataset (Warden, 2018) [8] as the positive class, and the remaining 34 keywords are labeled as the negative class. For training, we sample points uniformly at random from the above-mentioned dataset. However, for evaluation, we sample positives from the train distribution, but negatives contain a few challenging OOD points as well. Sampling challenging negatives itself is a hard task and is the key motivating reason for studying the problem. So, we manually list close-by keywords to Marvin such as Mar, Vin, Marvelous, etc. We then generate audio snippets for these keywords via a speech synthesis tool 2 with a variety of accents.<br />
Figure 5 shows that for 3% and 5% FPR settings, DROCC–LF is significantly more accurate than the baselines. For example, with FPR=3%, DROCC–LF is 10% more accurate than the baselines. We repeated the same experiment with the keyword: Seven, and observed a similar trend. In summary, DROCC–LF is able to generalize well against negatives that are “close” to the true positives even when such negatives were not supplied with the training data.<br />
<br />
== Conclusion and Future Work ==<br />
We introduced DROCC method for deep anomaly detection. It models normal data points using a low-dimensional sub-manifold inside the feature space, and the anomalous points are characterized via their Euclidean distance from the sub-manifold. Based on this intuition, DROCC’s optimization is formulated as a saddle point problem which is solved via a standard gradient descent-ascent algorithm. We then extended DROCC to OCLN problem where the goal is to generalize well against arbitrary negatives, assuming the positive class is well sampled and a small number of negative points are also available. Both the methods perform significantly better than strong baselines, in their respective problem settings. <br />
<br />
For computational efficiency, we simplified the projection set of both methods which can perhaps slow down the convergence of the two methods. Designing optimization algorithms that can work with the stricter set is an exciting research direction. Further, we would also like to rigorously analyze DROCC, assuming enough samples from a low-curvature manifold. Finally, as OCLN is an exciting problem that routinely comes up in a variety of real-world applications, we would like to apply DROCC–LF to a few high impact scenarios. Possible applications of this work are financial fraud detection, medical anomalies, or key words in audio processing.<br />
<br />
The results of this study showed that DROCC is comparatively better for anomaly detection across many different areas, such as tabular data, images, audio, and time series, when compared to existing state-of-the-art techniques.<br />
<br />
== References ==<br />
[1]: Golan, I. and El-Yaniv, R. Deep anomaly detection using geometric transformations. In Advances in Neural Information Processing Systems (NeurIPS), 2018.<br />
<br />
[2]: Ruff, L., Vandermeulen, R., Goernitz, N., Deecke, L., Siddiqui, S. A., Binder, A., M¨uller, E., and Kloft, M. Deep one-class classification. In International Conference on Machine Learning (ICML), 2018.<br />
<br />
[3]: Aggarwal, C. C. Outlier Analysis. Springer Publishing Company, Incorporated, 2nd edition, 2016. ISBN 3319475770.<br />
<br />
[4]: Sch¨olkopf, B., Williamson, R., Smola, A., Shawe-Taylor, J., and Platt, J. Support vector method for novelty detection. In Proceedings of the 12th International Conference on Neural Information Processing Systems, 1999.<br />
<br />
[5]: Tax, D. M. and Duin, R. P. Support vector data description. Machine Learning, 54(1), 2004.<br />
<br />
[6]: Pless, R. and Souvenir, R. A survey of manifold learning for images. IPSJ Transactions on Computer Vision and Applications, 1, 2009.<br />
<br />
[7]: Hendrycks, D., Mazeika, M., and Dietterich, T. Deep anomaly detection with outlier exposure. In International Conference on Learning Representations (ICLR), 2019a.<br />
<br />
[8]: Warden, P. Speech commands: A dataset for limited vocabulary speech recognition, 2018. URL https: //arxiv.org/abs/1804.03209.<br />
<br />
[9]: Liu, F. T., Ting, K. M., and Zhou, Z.-H. Isolation forest. In Proceedings of the 2008 Eighth IEEE International Conference on Data Mining, 2008.<br />
<br />
== Critiques/Insights ==<br />
<br />
1. It would be interesting to see this implemented in self-driving cars, for instance, to detect unusual road conditions.<br />
<br />
2. Figure 1 shows a good representation on how this model works. However, how can we know that this model is not prone to overfitting? There are many situations where there are valid points that lie outside of the line, especially new data that the model has never see before. An explanation on how this is avoided would be good.<br />
<br />
3.In the introduction part, it should first explain what is "one class", and then make a detailed application. Moreover, special definition words are used in many places in the text. No detailed explanation was given. In the end, the future application fields of DROCC and the research direction of the group can be explained.<br />
<br />
4. It will also be interesting to see if one change from using <math>\ell_{2}</math> Euclidean distance to other distances. When the low-dimensional manifold is highly non-linear, using the local linear distance to characterize anomalous points might fail.<br />
<br />
5. This is a nice summary and the authors introduce clearly on the performance of DROCC. It is nice to use Alexa as an example to catch readers' attention. I think it will be nice to include the algorithm of the DROCC or the architecture of DROCC in this summary to help us know the whole view of this method. Maybe it will be interesting to apply DROCC in biomedical studies? since one-class classification is often used in biomedical studies.<br />
<br />
6. The training method resembles adversarial learning with gradient ascent, however, there is no evaluation of this method on adversarial examples. This is quite unusual considering the paper proposed a method for robust one-class classification, and can be a security threat in real life in critical applications.<br />
<br />
7. The underlying idea behind OCLN is very similar to how neural networks are implemented in recommender systems and trained over positive/negative triplet models. In that case as well, due to the nature of implicit and explicit feedback, positive data tends to dominate the system. It would be interesting to see if insights from that area could be used to further boost the model presented in this paper.<br />
<br />
8. The paper shows the performance of DROCC being evaluated for time series data. It is interesting to see high AUC scores for DROCC against baselines like nearest neighbours and REBMs.Because detecting abnormal data in time series datasets is not common to practice.<br />
<br />
9. Figure1 presented results on a simple 2-D sine wave dataset to visualize the kind of classifiers learnt by DROCC. And the 1a is the positive data lies on a 1-D manifold. We can see from 1b that DROCC is able to capture the manifold accurately.<br />
<br />
10. In the MNIST 0 vs. 1 Classification dataset, why is 1 the only digit that is considered an anomoly? Couldn't all of the non-0 digits be left in the dataset to serve as "anomolies"?<br />
<br />
11. For future work the authors suggest considering DROCC for a low curvature manifold but do not motivate the benefits of such a direction.<br />
<br />
12. One of the problems is that in this model we might need to map all the points to one point to make the layer looks "perfect". However, this might not be a good choice since each point is distinct and if we map them together to one point, then this point cannot tell everything. If authors can specify more details on this it would be better.<br />
<br />
13. This project introduced DROCC for “one-class” classification. It will be interesting if such kind of classification can be compared with any other classification such as binary classification, etc. If “one-class” classification would be more speedy than the others.<br />
<br />
14. The dimensions and feature values must be so different across datasets in different domains. I would love to see how this algorithm is performing so well applied on different domains as it is mentioned that it could be used on datasets including images, audio, time-series, etc.<br />
<br />
15. It would be interesting to show the performance of DROCC against popular models used for outlier prediction such as PCA, EVA, etc. Perhaps show their accuracy scores so we can better compare.<br />
<br />
16. It would be greater if an visualization of how much performance DROCC improved compare to traditional binary classifier like SVM, isolation Forest.</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=Point-of-Interest_Recommendation:_Exploiting_Self-Attentive_Autoencoders_with_Neighbor-Aware_Influence&diff=48290Point-of-Interest Recommendation: Exploiting Self-Attentive Autoencoders with Neighbor-Aware Influence2020-11-30T04:33:48Z<p>Iaoellme: /* Introduction */</p>
<hr />
<div>== Presented by == <br />
Guanting(Tony) Pan, Zaiwei(Zara) Zhang, Haocheng(Beren) Chang<br />
<br />
== Introduction == <br />
With the development of mobile devices and location-acquisition technologies, accessing real-time location information is becoming easier and more efficient. Precisely because of this development, Location-based Social Networks (LBSNs) like Yelp and Foursquare have become an important part of human life. People can share their experiences in locations, such as restaurants and parks, on the Internet. These locations can be seen as a Point-of-Interest (POI) in software such as Maps on our phone. These large amounts of user-POI interaction data can be used to provide a service, which is called personalized POI recommendation, to give recommendations to users of a location they might be interested in. These large amounts of data can be used to train a model through Machine Learning methods(i.e., Classification, Clustering, etc.) to predict a POI that users might be interested in. The POI recommendation system still faces some challenging issues: (1) the difficulty of modeling complex user-POI interactions from sparse implicit feedback; (2) the difficulty of incorporating geographic background information. In order to meet these challenges, this paper will introduce a novel autoencoder-based model to learn non-linear user-POI relations, which is called SAE-NAD. SAE stands for the self-attentive encoder, while NAD stands for the neighbor-aware decoder. An autoencoder is an unsupervised learning technique that we implement in the neural network model for representation learning, meaning that our neural network will contain a "bottleneck" layer that produces a compressed knowledge representation of the original input. This method will utilize machine learning knowledge that we learned in this course.<br />
<br />
== Previous Work == <br />
<br />
In the previous works, the method is just equally treating users checked in POIs. The drawback of equally treating users checked in POIs is that valuable information about the similarity between users is not utilized, thus reducing the power of such recommenders. However, the SAE adaptively differentiates the user preference degrees in multiple aspects.<br />
<br />
Previous methods mainly used a process called collaborative filtering which can be divided into memory based methods and model based methods. Memory based methods predict a users preference based on a weighted average of similar users or POIs. Model based methods use user-POI data to build a model for generating recommendations. Both methods typically model user preferences linearly, which may be an oversimplification.<br />
<br />
There are some other personalized POI recommendation methods that can be used. Some famous software (e.g., Netflix) uses model-based methods that are built on matrix factorization (MF). For example, ranked based Geographical Factorization Method in [1] adopted weighted regularized MF to serve people on POI. Machine learning is popular in this area. POI recommendation is an important topic in the domain of recommender systems [4]. This paper also described related work in Personalized location recommendation and attention mechanism in the recommendation.<br />
<br />
== Motivation == <br />
This paper reviews encoder and decoder. A single hidden-layer autoencoder is an unsupervised neural network, which consists of two parts: an encoder and a decoder. The encoder has one activation function that maps the input data to the latent space. The decoder also has one activation function mapping the representations in the latent space to the reconstruction space. And here is the formula:<br />
<br />
[[File: formula.png|center]](Note: a is the activation function)<br />
<br />
The proposed method uses a two-layer neural network to compute the score matrix in the architecture of the SAE. The NAD adopts the RBF kernel to make checked-in POIs exert more influence on nearby unvisited POIs. To train this model, Network training is required.<br />
<br />
This paper will use the datasets in the real world, which are from Gowalla[2], Foursquare [3], and Yelp[3]. These datasets would be used to train by using the method introduced in this paper and compare the performance of SAE-NAD with other POI recommendation methods. Three groups of methods are used to compare with the proposed method, which are traditional MF methods for implicit feedback, Classical POI recommendation methods, and Deep learning-based methods. Specifically, the Deep learning-based methods contain a DeepAE which is a three-hidden-layer autoencoder with a weighted loss function, we can connect this to the material in this course.<br />
<br />
Autoencoders (AE), due to their ability to represent complex data, have become very useful in recommendation systems. The primary reason it is used is because with deep neural network and with non-linear activation function it effectively captures the non-linear and non-trivial relationships between users and POI's.<br />
<br />
== Methodology == <br />
<br />
=== Notations ===<br />
<br />
Here are the notations used in this paper. It will be helpful when trying to understand the structure and equations in the algorithm.<br />
[[File:notations.JPG|500px|x300px|center]]<br />
<br />
=== Structure ===<br />
<br />
The structure of the network in this paper includes a self-attentive encoder as the input layer(yellow), and a neighbor-aware decoder as the output layer(green).<br />
<br />
[[File:1.JPG|1200px|x600px]]<br />
<br />
=== Self-Attentive Encoder ===<br />
<br />
The self-attentive encoder is the input layer. It transfers the preference vector <math>x_u</math> to hidden representation <math>A_u</math> using weight matrix <math>W^1</math> and the activation function <math>softmax</math> and <math>tanh</math>. The 0's and 1's in <math>x_u</math> indicates whether the user has been to a certain POI. The weight matrix <math>W_a</math> assigns different weights on various features of POIs. <math>A_u</math> is the importance score matrix, where each column is a POI, and each row is the importance level of the <math>n</math> checked in POIs in one aspect.<br />
<br />
[[File:encoder.JPG|center]]<br />
<br />
<math>\mathbf{Z}_u^{(1)} = \mathbf{A}_u \cdot (\mathbf{W}^{(1)}[L_u])^\top</math> is the multiplication of the importance score matrix and the POI embeddings, and represents the user from <math>d_a</math> aspects. <math>\mathbf{Z}_u^{(1)}</math> is a <math>d_a \times H_1</math> matrix. The aggregation layer then combines multiple aspects that represent the user into one, so that it fits into the Neighbor-Aware decoder: <math>\mathbf{z}_u^{(1)} = a_t(\mathbf{Z}_u^{(1)\top}\mathbf{w}_t + \mathbf{b}_t)</math>.<br />
<br />
=== Neighbor-Aware Decoder ===<br />
<br />
POI recommendation uses the geographical clustering phenomenon, which increases the weight of the unvisited POIs that surround the visited POIs. Also, an aggregation layer is added to the network to aggregate users’ representations from different aspects into one aspect. This means that a person who has visited a location is very likely to return to this location again in the future, so the user is recommended POIs surrounding this area. An example would be someone who has been to the UW plaza and bought Lazeez, is very likely to return to the plaza, therefore the person is recommended to try Mr. Panino's Beijing House.<br />
<br />
[[File:decoder.JPG|center]]<br />
<br />
=== Objective Function ===<br />
<br />
By minimizing the objective function, the partial derivatives with respect to all the parameters can be computed by gradient descent with backpropagation. After that, the training is complete.<br />
<br />
[[File:objective_function.JPG|center]]<br />
<br />
<br />
== Comparative analysis ==<br />
<br />
=== Metrics introduction ===<br />
To obtain a comprehensive evaluation of the effectiveness of the model, the authors performed a thorough comparison between the proposed model and the existing major POI recommendation methods. These methods can be further broken down into three categories: traditional matrix factorization methods for implicit feedback, classical POI recommendation methods, and deep learning-based methods. Here, three key evaluation metrics were introduced as Precison@k, Recall@k, and MAP@k. Through comparing all models on three datasets using the above metrics, it is concluded that the proposed model achieved the best performance.<br />
<br />
To better understand the comparison results, it is critical to understand the meanings behind each evaluation metric. Suppose the proposed model generated k recommended POIs for the user. The first metric, Precison@k, measures the percentage of the recommended POIs which the user has visited. Recall@k is also associated with the user’s behavior. However, it will measure the percentage of recommended POIs in all POIs which have been visited by the user. Lastly, MAP@k represents the mean average precision at k, where average precision is the average of precision values at all k ranks, where relevant POIs are found. They are formally defined as follows, <br />
$$ Recall@k=\frac{1}{M} \sum_{i=1}^M \frac{S_i(k) \cap T_i}{|T_i|} \\<br />
Precision@k = \frac{1}{M} \sum_{i=1}^M \frac{S_i(k) \cap T_i}{k} \\<br />
MAP@k = \frac{1}{M} \sum_{i=1}^M \frac{\sum_{j=1}^k p(j) \times rel(j)}{|T_i|} \\$$<br />
<br />
where <math>S_i(k) </math> us a set of top- <math>k </math> unvisited locations recommended to user <math> i </math> excluding those locations in the training, and <math>T_i </math> is a set of locations that are visited by user <math> i</math> in the testing. <math> p(j)</math> is the precision of a cut-off rank list from 1 to <math> j</math>, and <math>rel(j) </math> is an indicator function that equals to 1 if the location is visited in the testing, otherwise equals to 0.<br />
<br />
=== Model Comparison ===<br />
Among all models in the comparison group, RankGeoFM, IRNMF, and PACE produced the best results. Nonetheless, these models are still incomparable to our proposed model. The reasons are explained in details as follows:<br />
<br />
Both RankGeoFM and IRNMF incorporate geographical influence into their ranking models, which is significant for generating POI recommendations. However, they are not capable of capturing non-linear interactions between users and POIs. In comparison, the proposed model, while incorporating geographical influence, adopts a deep neural structure which enables it to measure non-linear and complex interactions. As a result, it outperforms the two methods in the comparison group.<br />
<br />
Moreover, compared to PACE, which is a deep learning-based method, the proposed model offers a more precise measurement of geographical influence. Though PACE is able to capture complex interactions, it models the geographical influence by a context graph, which fails to incorporate user reachability into the modeling process. In contrast, the proposed model is able to capture geographical influence directly through its neighbor-aware decoder, which allows it to achieve better performance than the PACE model.<br />
<br />
[[File:model_comparison.JPG|center]]<br />
<br />
== Conclusion ==<br />
In summary, the proposed model, namely SAE-NAD, clearly showed its advantages compared to many state-of-the-art baseline methods. Its self-attentive encoder effectively discriminates user preferences on check-in POIs, and its neighbor-aware decoder measures geographical influence precisely through differentiating user reachability on unvisited POIs. By leveraging these two components together, it can generate recommendations that are highly relevant to its users.<br />
<br />
== Critiques ==<br />
Besides developing the model and conducting a detailed analysis, the authors also did very well in constructing this paper. The paper is well-written and has a highly logical structure. Definitions, notations, and metrics are introduced and explained clearly, which enables readers to follow through the analysis easily. Last but not least, both the abstract and the conclusion of this paper are strong. The abstract concisely reported the objectives and outcomes of the experiment, whereas the conclusion is succinct and precise.<br />
<br />
This idea would have many applications, such as suggesting new restaurants to customers in the food delivery service app. Would the improvement in accuracy outweigh the increased complexity of the model when it comes to use in industry?<br />
<br />
It would be nice if the authours could describe the extensive experiments on the real-world datasets, with different baseline methods and evaluation metrics, to demonstrate the effectiveness of the proposed model. Moreover, show the comparison result in tables vs other methodologies, both in terms of accuracy and time-efficiency. In addition, the drawbacks of this new methodology are unknown to the readers. In other words, how does this compare to the already established recommendation systems found in large scale applications utilize by companies like Netflix? Why use the proposed method over something simpler such as matrix factorization or collaborative filtering? <br />
<br />
It would also be nice if the authors provided some more ablation on the various components of the proposed method. Even after reading some of their experiments, we do not have a clear understanding of how important each component is to the recommendation quality.<br />
<br />
It is recommended that the author present how the encoder and the decoder help in the process by comparing different models with or without encoder and decoder.<br />
<br />
It would have been better if the author included all figures to explain the sensitivity of parameters more elaborately. In the paper he only included plots showing effects on Gowalla and Foursquare datasets but not other datasets.<br />
<br />
Some additional explanations can be inserted in the Objective Function part. From the paper, we can find <math>\lambda</math> is the regularization parameter and <math>W_a</math> and <math>w_t</math> are the learned parameters in the attention layer and aggregation layer.<br />
<br />
It would be more attractive if there is a section to introduce the applications that based on such algorithm in daily life. For instance, which application we use nowadays is based on this algorithm and what are the advantages of it compared to other similar algorithm?<br />
<br />
It would be useful to the readers if the authors briefly described covariates with which they are using to predict POIs. All that is detailed in this paper is that geographical information is used. Taking note of the characteristics that the data set the proposed method deals with is important, especially as phones become capabable of collecting more and more kinds of data. That is to say, the information available for predicting POI's in the future may be different to the information available today, so it is important to describe the data. (The information could even decrease in the future if privacy laws are enacted).<br />
<br />
== References ==<br />
[1] Defu Lian, Cong Zhao, Xing Xie, Guangzhong Sun, Enhong Chen, and Yong Rui. 2014. GeoMF: joint geographical modeling and matrix factorization for point-of-interest recommendation. In KDD. ACM, 831–840.<br />
<br />
[2] Eunjoon Cho, Seth A. Myers, and Jure Leskovec. 2011. Friendship and mobility: user movement in location-based social networks. In KDD. ACM, 1082–1090.<br />
<br />
[3] Yiding Liu, Tuan-Anh Pham, Gao Cong, and Quan Yuan. 2017. An Experimental Evaluation of Point-of-interest Recommendation in Location-based Social Networks. PVLDB 10, 10 (2017), 1010–1021.<br />
<br />
[4] Jie Bao, Yu Zheng, David Wilkie, and Mohamed F. Mokbel. 2015. Recommendations in location-based social networks: a survey. GeoInformatica 19, 3 (2015), 525–565.<br />
<br />
[5] Xiangnan He, Lizi Liao, Hanwang Zhang, Liqiang Nie, Xia Hu, and Tat-Seng Chua. 2017. Neural Collaborative Filtering. In WWW. ACM, 173–182.<br />
<br />
[6] Yifan Hu, Yehuda Koren, and Chris Volinsky. 2008. Collaborative Filtering for Implicit Feedback Datasets. In ICDM. IEEE Computer Society, 263–272.<br />
<br />
[7] Santosh Kabbur, Xia Ning, and George Karypis. 2013. FISM: factored item similarity models for top-N recommender systems. In KDD. ACM, 659–667. [12] Diederik P. Kingma and Jimmy Ba. 2014. Adam: A Method for Stochastic Optimization. CoRR abs/1412.6980 (2014).<br />
<br />
[8] Yong Liu,WeiWei, Aixin Sun, and Chunyan Miao. 2014. Exploiting Geographical<br />
Neighborhood Characteristics for Location Recommendation. In CIKM. ACM,<br />
739–748<br />
<br />
[9] Xutao Li, Gao Cong, Xiaoli Li, Tuan-Anh Nguyen Pham, and Shonali Krishnaswamy.<br />
2015. Rank-GeoFM: A Ranking based Geographical Factorization<br />
Method for Point of Interest Recommendation. In SIGIR. ACM, 433–442.<br />
<br />
[10] Carl Yang, Lanxiao Bai, Chao Zhang, Quan Yuan, and Jiawei Han. 2017. Bridging<br />
Collaborative Filtering and Semi-Supervised Learning: A Neural Approach for<br />
POI Recommendation. In KDD. ACM, 1245–1254.</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=Point-of-Interest_Recommendation:_Exploiting_Self-Attentive_Autoencoders_with_Neighbor-Aware_Influence&diff=48286Point-of-Interest Recommendation: Exploiting Self-Attentive Autoencoders with Neighbor-Aware Influence2020-11-30T04:30:26Z<p>Iaoellme: /* Previous Work */</p>
<hr />
<div>== Presented by == <br />
Guanting(Tony) Pan, Zaiwei(Zara) Zhang, Haocheng(Beren) Chang<br />
<br />
== Introduction == <br />
With the development of mobile devices and location-acquisition technologies, accessing real-time location information is becoming easier and more efficient. Precisely because of this development, Location-based Social Networks (LBSNs) like Yelp and Foursquare have become an important part of human’s life. People can share their experiences in locations, such as restaurants and parks, on the Internet. These locations can be seen as a Point-of-Interest (POI) in software such as Maps on our phone. These large amounts of user-POI interaction data can provide a service, which is called personalized POI recommendation, to give recommendations to users that the location they might be interested in. These large amounts of data can be used to train a model through Machine Learning methods(i.e., Classification, Clustering, etc.) to predict a POI that users might be interested in. The POI recommendation system still faces some challenging issues: (1) the difficulty of modeling complex user-POI interactions from sparse implicit feedback; (2) the difficulty of incorporating geographic background information. In order to meet these challenges, this paper will introduce a novel autoencoder-based model to learn non-linear user-POI relations, which is called SAE-NAD. SAE stands for the self-attentive encoder, while NAD stands for the neighbor-aware decoder. Autoencoder is an unsupervised learning technique that we implement in the neural network model for representation learning, meaning that our neural network will contain a "bottleneck" layer that produces a compressed knowledge representation of the original input. This method will include machine learning knowledge that we learned in this course.<br />
<br />
== Previous Work == <br />
<br />
In the previous works, the method is just equally treating users checked in POIs. The drawback of equally treating users checked in POIs is that valuable information about the similarity between users is not utilized, thus reducing the power of such recommenders. However, the SAE adaptively differentiates the user preference degrees in multiple aspects.<br />
<br />
Previous methods mainly used a process called collaborative filtering which can be divided into memory based methods and model based methods. Memory based methods predict a users preference based on a weighted average of similar users or POIs. Model based methods use user-POI data to build a model for generating recommendations. Both methods typically model user preferences linearly, which may be an oversimplification.<br />
<br />
There are some other personalized POI recommendation methods that can be used. Some famous software (e.g., Netflix) uses model-based methods that are built on matrix factorization (MF). For example, ranked based Geographical Factorization Method in [1] adopted weighted regularized MF to serve people on POI. Machine learning is popular in this area. POI recommendation is an important topic in the domain of recommender systems [4]. This paper also described related work in Personalized location recommendation and attention mechanism in the recommendation.<br />
<br />
== Motivation == <br />
This paper reviews encoder and decoder. A single hidden-layer autoencoder is an unsupervised neural network, which consists of two parts: an encoder and a decoder. The encoder has one activation function that maps the input data to the latent space. The decoder also has one activation function mapping the representations in the latent space to the reconstruction space. And here is the formula:<br />
<br />
[[File: formula.png|center]](Note: a is the activation function)<br />
<br />
The proposed method uses a two-layer neural network to compute the score matrix in the architecture of the SAE. The NAD adopts the RBF kernel to make checked-in POIs exert more influence on nearby unvisited POIs. To train this model, Network training is required.<br />
<br />
This paper will use the datasets in the real world, which are from Gowalla[2], Foursquare [3], and Yelp[3]. These datasets would be used to train by using the method introduced in this paper and compare the performance of SAE-NAD with other POI recommendation methods. Three groups of methods are used to compare with the proposed method, which are traditional MF methods for implicit feedback, Classical POI recommendation methods, and Deep learning-based methods. Specifically, the Deep learning-based methods contain a DeepAE which is a three-hidden-layer autoencoder with a weighted loss function, we can connect this to the material in this course.<br />
<br />
Autoencoders (AE), due to their ability to represent complex data, have become very useful in recommendation systems. The primary reason it is used is because with deep neural network and with non-linear activation function it effectively captures the non-linear and non-trivial relationships between users and POI's.<br />
<br />
== Methodology == <br />
<br />
=== Notations ===<br />
<br />
Here are the notations used in this paper. It will be helpful when trying to understand the structure and equations in the algorithm.<br />
[[File:notations.JPG|500px|x300px|center]]<br />
<br />
=== Structure ===<br />
<br />
The structure of the network in this paper includes a self-attentive encoder as the input layer(yellow), and a neighbor-aware decoder as the output layer(green).<br />
<br />
[[File:1.JPG|1200px|x600px]]<br />
<br />
=== Self-Attentive Encoder ===<br />
<br />
The self-attentive encoder is the input layer. It transfers the preference vector <math>x_u</math> to hidden representation <math>A_u</math> using weight matrix <math>W^1</math> and the activation function <math>softmax</math> and <math>tanh</math>. The 0's and 1's in <math>x_u</math> indicates whether the user has been to a certain POI. The weight matrix <math>W_a</math> assigns different weights on various features of POIs. <math>A_u</math> is the importance score matrix, where each column is a POI, and each row is the importance level of the <math>n</math> checked in POIs in one aspect.<br />
<br />
[[File:encoder.JPG|center]]<br />
<br />
<math>\mathbf{Z}_u^{(1)} = \mathbf{A}_u \cdot (\mathbf{W}^{(1)}[L_u])^\top</math> is the multiplication of the importance score matrix and the POI embeddings, and represents the user from <math>d_a</math> aspects. <math>\mathbf{Z}_u^{(1)}</math> is a <math>d_a \times H_1</math> matrix. The aggregation layer then combines multiple aspects that represent the user into one, so that it fits into the Neighbor-Aware decoder: <math>\mathbf{z}_u^{(1)} = a_t(\mathbf{Z}_u^{(1)\top}\mathbf{w}_t + \mathbf{b}_t)</math>.<br />
<br />
=== Neighbor-Aware Decoder ===<br />
<br />
POI recommendation uses the geographical clustering phenomenon, which increases the weight of the unvisited POIs that surround the visited POIs. Also, an aggregation layer is added to the network to aggregate users’ representations from different aspects into one aspect. This means that a person who has visited a location is very likely to return to this location again in the future, so the user is recommended POIs surrounding this area. An example would be someone who has been to the UW plaza and bought Lazeez, is very likely to return to the plaza, therefore the person is recommended to try Mr. Panino's Beijing House.<br />
<br />
[[File:decoder.JPG|center]]<br />
<br />
=== Objective Function ===<br />
<br />
By minimizing the objective function, the partial derivatives with respect to all the parameters can be computed by gradient descent with backpropagation. After that, the training is complete.<br />
<br />
[[File:objective_function.JPG|center]]<br />
<br />
<br />
== Comparative analysis ==<br />
<br />
=== Metrics introduction ===<br />
To obtain a comprehensive evaluation of the effectiveness of the model, the authors performed a thorough comparison between the proposed model and the existing major POI recommendation methods. These methods can be further broken down into three categories: traditional matrix factorization methods for implicit feedback, classical POI recommendation methods, and deep learning-based methods. Here, three key evaluation metrics were introduced as Precison@k, Recall@k, and MAP@k. Through comparing all models on three datasets using the above metrics, it is concluded that the proposed model achieved the best performance.<br />
<br />
To better understand the comparison results, it is critical to understand the meanings behind each evaluation metric. Suppose the proposed model generated k recommended POIs for the user. The first metric, Precison@k, measures the percentage of the recommended POIs which the user has visited. Recall@k is also associated with the user’s behavior. However, it will measure the percentage of recommended POIs in all POIs which have been visited by the user. Lastly, MAP@k represents the mean average precision at k, where average precision is the average of precision values at all k ranks, where relevant POIs are found. They are formally defined as follows, <br />
$$ Recall@k=\frac{1}{M} \sum_{i=1}^M \frac{S_i(k) \cap T_i}{|T_i|} \\<br />
Precision@k = \frac{1}{M} \sum_{i=1}^M \frac{S_i(k) \cap T_i}{k} \\<br />
MAP@k = \frac{1}{M} \sum_{i=1}^M \frac{\sum_{j=1}^k p(j) \times rel(j)}{|T_i|} \\$$<br />
<br />
where <math>S_i(k) </math> us a set of top- <math>k </math> unvisited locations recommended to user <math> i </math> excluding those locations in the training, and <math>T_i </math> is a set of locations that are visited by user <math> i</math> in the testing. <math> p(j)</math> is the precision of a cut-off rank list from 1 to <math> j</math>, and <math>rel(j) </math> is an indicator function that equals to 1 if the location is visited in the testing, otherwise equals to 0.<br />
<br />
=== Model Comparison ===<br />
Among all models in the comparison group, RankGeoFM, IRNMF, and PACE produced the best results. Nonetheless, these models are still incomparable to our proposed model. The reasons are explained in details as follows:<br />
<br />
Both RankGeoFM and IRNMF incorporate geographical influence into their ranking models, which is significant for generating POI recommendations. However, they are not capable of capturing non-linear interactions between users and POIs. In comparison, the proposed model, while incorporating geographical influence, adopts a deep neural structure which enables it to measure non-linear and complex interactions. As a result, it outperforms the two methods in the comparison group.<br />
<br />
Moreover, compared to PACE, which is a deep learning-based method, the proposed model offers a more precise measurement of geographical influence. Though PACE is able to capture complex interactions, it models the geographical influence by a context graph, which fails to incorporate user reachability into the modeling process. In contrast, the proposed model is able to capture geographical influence directly through its neighbor-aware decoder, which allows it to achieve better performance than the PACE model.<br />
<br />
[[File:model_comparison.JPG|center]]<br />
<br />
== Conclusion ==<br />
In summary, the proposed model, namely SAE-NAD, clearly showed its advantages compared to many state-of-the-art baseline methods. Its self-attentive encoder effectively discriminates user preferences on check-in POIs, and its neighbor-aware decoder measures geographical influence precisely through differentiating user reachability on unvisited POIs. By leveraging these two components together, it can generate recommendations that are highly relevant to its users.<br />
<br />
== Critiques ==<br />
Besides developing the model and conducting a detailed analysis, the authors also did very well in constructing this paper. The paper is well-written and has a highly logical structure. Definitions, notations, and metrics are introduced and explained clearly, which enables readers to follow through the analysis easily. Last but not least, both the abstract and the conclusion of this paper are strong. The abstract concisely reported the objectives and outcomes of the experiment, whereas the conclusion is succinct and precise.<br />
<br />
This idea would have many applications, such as suggesting new restaurants to customers in the food delivery service app. Would the improvement in accuracy outweigh the increased complexity of the model when it comes to use in industry?<br />
<br />
It would be nice if the authours could describe the extensive experiments on the real-world datasets, with different baseline methods and evaluation metrics, to demonstrate the effectiveness of the proposed model. Moreover, show the comparison result in tables vs other methodologies, both in terms of accuracy and time-efficiency. In addition, the drawbacks of this new methodology are unknown to the readers. In other words, how does this compare to the already established recommendation systems found in large scale applications utilize by companies like Netflix? Why use the proposed method over something simpler such as matrix factorization or collaborative filtering? <br />
<br />
It would also be nice if the authors provided some more ablation on the various components of the proposed method. Even after reading some of their experiments, we do not have a clear understanding of how important each component is to the recommendation quality.<br />
<br />
It is recommended that the author present how the encoder and the decoder help in the process by comparing different models with or without encoder and decoder.<br />
<br />
It would have been better if the author included all figures to explain the sensitivity of parameters more elaborately. In the paper he only included plots showing effects on Gowalla and Foursquare datasets but not other datasets.<br />
<br />
Some additional explanations can be inserted in the Objective Function part. From the paper, we can find <math>\lambda</math> is the regularization parameter and <math>W_a</math> and <math>w_t</math> are the learned parameters in the attention layer and aggregation layer.<br />
<br />
It would be more attractive if there is a section to introduce the applications that based on such algorithm in daily life. For instance, which application we use nowadays is based on this algorithm and what are the advantages of it compared to other similar algorithm?<br />
<br />
It would be useful to the readers if the authors briefly described covariates with which they are using to predict POIs. All that is detailed in this paper is that geographical information is used. Taking note of the characteristics that the data set the proposed method deals with is important, especially as phones become capabable of collecting more and more kinds of data. That is to say, the information available for predicting POI's in the future may be different to the information available today, so it is important to describe the data. (The information could even decrease in the future if privacy laws are enacted).<br />
<br />
== References ==<br />
[1] Defu Lian, Cong Zhao, Xing Xie, Guangzhong Sun, Enhong Chen, and Yong Rui. 2014. GeoMF: joint geographical modeling and matrix factorization for point-of-interest recommendation. In KDD. ACM, 831–840.<br />
<br />
[2] Eunjoon Cho, Seth A. Myers, and Jure Leskovec. 2011. Friendship and mobility: user movement in location-based social networks. In KDD. ACM, 1082–1090.<br />
<br />
[3] Yiding Liu, Tuan-Anh Pham, Gao Cong, and Quan Yuan. 2017. An Experimental Evaluation of Point-of-interest Recommendation in Location-based Social Networks. PVLDB 10, 10 (2017), 1010–1021.<br />
<br />
[4] Jie Bao, Yu Zheng, David Wilkie, and Mohamed F. Mokbel. 2015. Recommendations in location-based social networks: a survey. GeoInformatica 19, 3 (2015), 525–565.<br />
<br />
[5] Xiangnan He, Lizi Liao, Hanwang Zhang, Liqiang Nie, Xia Hu, and Tat-Seng Chua. 2017. Neural Collaborative Filtering. In WWW. ACM, 173–182.<br />
<br />
[6] Yifan Hu, Yehuda Koren, and Chris Volinsky. 2008. Collaborative Filtering for Implicit Feedback Datasets. In ICDM. IEEE Computer Society, 263–272.<br />
<br />
[7] Santosh Kabbur, Xia Ning, and George Karypis. 2013. FISM: factored item similarity models for top-N recommender systems. In KDD. ACM, 659–667. [12] Diederik P. Kingma and Jimmy Ba. 2014. Adam: A Method for Stochastic Optimization. CoRR abs/1412.6980 (2014).<br />
<br />
[8] Yong Liu,WeiWei, Aixin Sun, and Chunyan Miao. 2014. Exploiting Geographical<br />
Neighborhood Characteristics for Location Recommendation. In CIKM. ACM,<br />
739–748<br />
<br />
[9] Xutao Li, Gao Cong, Xiaoli Li, Tuan-Anh Nguyen Pham, and Shonali Krishnaswamy.<br />
2015. Rank-GeoFM: A Ranking based Geographical Factorization<br />
Method for Point of Interest Recommendation. In SIGIR. ACM, 433–442.<br />
<br />
[10] Carl Yang, Lanxiao Bai, Chao Zhang, Quan Yuan, and Jiawei Han. 2017. Bridging<br />
Collaborative Filtering and Semi-Supervised Learning: A Neural Approach for<br />
POI Recommendation. In KDD. ACM, 1245–1254.</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=User:Cvmustat&diff=48265User:Cvmustat2020-11-30T04:11:44Z<p>Iaoellme: /* Interpreting Learned CRNN Weights */</p>
<hr />
<div><br />
== Combine Convolution with Recurrent Networks for Text Classification == <br />
'''Team Members''': Bushra Haque, Hayden Jones, Michael Leung, Cristian Mustatea<br />
<br />
'''Date''': Week of Nov 23 <br />
<br />
== Introduction ==<br />
<br />
<br />
Text classification is the task of assigning a set of predefined categories to natural language texts. It involves learning an embedding layer which allows context-dependent classification. It is a fundamental task in Natural Language Processing (NLP) with various applications such as sentiment analysis, and topic classification. A classic example involving text classification is given a set of News articles, is it possible to classify the genre or subject of each article? Text classification is useful as text data is a rich source of information, but extracting insights from it directly can be difficult and time-consuming as most text data is unstructured.[1] NLP text classification can help automatically structure and analyze text quickly and cost-effectively, allowing for individuals to extract important features from the text easier than before. <br />
<br />
Text classification work mainly focuses on three topics: feature engineering, feature selection, and the use of different types of machine learning algorithms.<br />
:1. Feature engineering, the most widely used feature is the bag of words feature. Some more complex functions are also designed, such as part-of-speech tags, noun phrases, and tree kernels.<br />
:2. Feature selection aims to remove noisy features and improve classification performance. The most common feature selection method is to delete stop words.<br />
:3. Machine learning algorithms usually use classifiers, such as Logistic Regression (LR), Naive Bayes (NB), and Support Vector Machine (SVM).<br />
<br />
In practice, pre-trained word embeddings and deep neural networks are used together for NLP text classification. Word embeddings are used to map the raw text data to an implicit space where the semantic relationships of the words are preserved; words with similar meaning have a similar representation. One can then feed these embeddings into deep neural networks to learn different features of the text. Convolutional neural networks can be used to determine the semantic composition of the text(the meaning), as it treats texts as a 2D matrix by concatenating the embedding of words together. It uses a 1D convolution operator to perform the feature mapping and then conducts a 1D pooling operation over the time domain for obtaining a fixed-length output feature vector, and it can capture both local and position invariant features of the text.[2] Alternatively, Recurrent Neural Networks can be used to determine the contextual meaning of each word in the text (how each word relates to one another) by treating the text as sequential data and then analyzing each word separately. [3] Previous approaches to attempt to combine these two neural networks to incorporate the advantages of both models involve streamlining the two networks, which might decrease their performance. Besides, most methods incorporating a bi-directional Recurrent Neural Network usually concatenate the forward and backward hidden states at each time step, which results in a vector that does not have the interaction information between the forward and backward hidden states.[4] The hidden state in one direction contains only the contextual meaning in that particular direction, however a word's contextual representation, intuitively, is more accurate when collected and viewed from both directions. This paper argues that the failure to observe the meaning of a word in both directions causes the loss of the true meaning of the word, especially for polysemic words (words with more than one meaning) that are context-sensitive.<br />
<br />
== Paper Key Contributions ==<br />
<br />
This paper suggests an enhanced method of text classification by proposing a new way of combining Convolutional and Recurrent Neural Networks (CRNN) involving the addition of a neural tensor layer. The proposed method maintains each network's respective strengths that are normally lost in previous combination methods. The new suggested architecture is called CRNN, which utilizes both a CNN and RNN that run in parallel on the same input sentence. CNN uses weight matrix learning and produces a 2D matrix that shows the importance of each word based on local and position-invariant features. The bidirectional RNN produces a matrix that learns each word's contextual representation; the words' importance about to with concerning the rest of the sentence. A neural tensor layer is introduced on top of the RNN to obtain the fusion of bi-directional contextual information surrounding a particular word. This method combines these two matrix representations and classifies the text, providing the important information of each word for prediction, which helps to explain the results. The model also uses dropout and L2 regularization to prevent overfitting.<br />
<br />
== CRNN Results vs Benchmarks ==<br />
<br />
In order to benchmark the performance of the CRNN model, as well as compare it to other previous efforts, multiple datasets and classification problems were used. All of these datasets are publicly available and can be easily downloaded by any user for testing.<br />
<br />
- '''Movie Reviews:''' a sentiment analysis dataset, with two classes (positive and negative).<br />
<br />
- '''Yelp:''' a sentiment analysis dataset, with five classes. For this test, a subset of 120,000 reviews was randomly chosen from each class for a total of 600,000 reviews.<br />
<br />
- '''AG's News:''' a news categorization dataset, using only the 4 largest classes from the dataset. There are 30000 training samples and 1900 test samples.<br />
<br />
- '''20 Newsgroups:''' a news categorization dataset, again using only 4 large classes (comp, politics, rec, and religion) from the dataset.<br />
<br />
- '''Sogou News:''' a Chinese news categorization dataset, using 10 major classes as a multi-class classification and include 6500 samples randomly from each class.<br />
<br />
- '''Yahoo! Answers:''' a topic classification dataset, with 10 classes and each class contains 140000 training samples and 5000 testing samples.<br />
<br />
For the English language datasets, the initial word representations were created using the publicly available ''word2vec'' [https://code.google.com/p/word2vec/] from Google news. For the Chinese language dataset, ''jieba'' [https://github.com/fxsjy/jieba] was used to segment sentences, and then 50-dimensional word vectors were trained on Chinese ''wikipedia'' using ''word2vec''.<br />
<br />
A number of other models are run against the same data after preprocessing. Some of these models include:<br />
<br />
- '''Self-attentive LSTM:''' an LSTM model with multi-hop attention for sentence embedding.<br />
<br />
- '''RCNN:''' the RCNN's recurrent structure allows for increased depth of capture for contextual information. Less noise is introduced on account of the model's holistic structure (compared to local features).<br />
<br />
The following results are obtained:<br />
<br />
[[File:table of results.png|550px|center]]<br />
<br />
The bold results represent the best performing model for a given dataset. These results show that the CRNN model is the best model for 4 of the 6 datasets, with the Self-attentive LSTM beating the CRNN by 0.03 and 0.12 on the news categorization problems. Considering that the CRNN model has better performance than the Self-attentive LSTM on the other 4 datasets, this suggests that the CRNN model is a better performer overall in the conditions of this benchmark.<br />
<br />
It should be noted that including the neural tensor layer in the CRNN model leads to a significant performance boost compared to the CRNN models without it. The performance boost can be attributed to the fact that the neural tensor layer captures the surrounding contextual information for each word, and brings this information between the forward and backward RNN in a direct method. As seen in the table, this leads to a better classification accuracy across all datasets.<br />
<br />
Another important result was that the CRNN model filter size impacted performance only in the sentiment analysis datasets, as seen in the following table, where the results for the AG's news, Yelp, and Yahoo! Answer datasets are displayed:<br />
<br />
[[File:filter_effects.png|550px|center]]<br />
<br />
== CRNN Model Architecture ==<br />
<br />
The CRNN model is a combination of RNN and CNN. It uses a CNN to compute the importance of each word in the text and utilizes a neural tensor layer to fuse forward and backward hidden states of bi-directional RNN.<br />
<br />
The input of the network is a text, which is a sequence of words. The output of the network is the text representation that is subsequently used as input of a fully-connected layer to obtain the class prediction.<br />
<br />
'''RNN Pipeline:'''<br />
<br />
The goal of the RNN pipeline is to input each word in a text, and retrieve the contextual information surrounding the word and compute the contextual representation of the word itself. This is accomplished by the use of a bi-directional RNN, such that a Neural Tensor Layer (NTL) can combine the results of the RNN to obtain the final output. RNNs are well-suited to NLP tasks because of their ability to sequentially process data such as ordered text.<br />
<br />
A RNN is similar to a feed-forward neural network, but it relies on the use of hidden states. Hidden states are layers in the neural net that produce two outputs: <math> \hat{y}_{t} </math> and <math> h_t </math>. For a time step <math> t </math>, <math> h_t </math> is fed back into the layer to compute <math> \hat{y}_{t+1} </math> and <math> h_{t+1} </math>. <br />
<br />
The pipeline will actually use a variant of RNN called GRU, short for Gated Recurrent Units. This is done to address the vanishing gradient problem which causes the network to struggle to memorize words that came earlier in the sequence. Traditional RNNs are only able to remember the most recent words in a sequence, which may be problematic since words that came at the beginning of the sequence that is important to the classification problem may be forgotten. A GRU attempts to solve this by controlling the flow of information through the network using update and reset gates. <br />
<br />
Let <math>h_{t-1} \in \mathbb{R}^m, x_t \in \mathbb{R}^d </math> be the inputs, and let <math>\mathbf{W}_z, \mathbf{W}_r, \mathbf{W}_h \in \mathbb{R}^{m \times d}, \mathbf{U}_z, \mathbf{U}_r, \mathbf{U}_h \in \mathbb{R}^{m \times m}</math> be trainable weight matrices. Then the following equations describe the update and reset gates:<br />
<br />
<br />
<math><br />
z_t = \sigma(\mathbf{W}_zx_t + \mathbf{U}_zh_{t-1}) \text{update gate} \\<br />
r_t = \sigma(\mathbf{W}_rx_t + \mathbf{U}_rh_{t-1}) \text{reset gate} \\<br />
\tilde{h}_t = \text{tanh}(\mathbf{W}_hx_t + r_t \circ \mathbf{U}_hh_{t-1}) \text{new memory} \\<br />
h_t = (1-z_t)\circ \tilde{h}_t + z_t\circ h_{t-1}<br />
</math><br />
<br />
<br />
Note that <math> \sigma, \text{tanh}, \circ </math> are all element-wise functions. The above equations do the following:<br />
<br />
<ol><br />
<li> <math>h_{t-1}</math> carries information from the previous iteration and <math>x_t</math> is the current input </li><br />
<li> the update gate <math>z_t</math> controls how much past information should be forwarded to the next hidden state </li><br />
<li> the rest gate <math>r_t</math> controls how much past information is forgotten or reset </li><br />
<li> new memory <math>\tilde{h}_t</math> contains the relevant past memory as instructed by <math>r_t</math> and current information from the input <math>x_t</math> </li><br />
<li> then <math>z_t</math> is used to control what is passed on from <math>h_{t-1}</math> and <math>(1-z_t)</math> controls the new memory that is passed on<br />
</ol><br />
<br />
We treat <math>h_0</math> and <math> h_{n+1} </math> as zero vectors in the method. Thus, each <math>h_t</math> can be computed as above to yield results for the bi-directional RNN. The resulting hidden states <math>\overrightarrow{h_t}</math> and <math>\overleftarrow{h_t}</math> contain contextual information around the <math> t</math>-th word in forward and backward directions respectively. Contrary to convention, instead of concatenating these two vectors, it is argued that the word's contextual representation is more precise when the context information from different directions is collected and fused using a neural tensor layer as it permits greater interactions among each element of hidden states. Using these two vectors as input to the neural tensor layer, <math>V^i </math>, we compute a new representation that aggregates meanings from the forward and backward hidden states more accurately as follows:<br />
<br />
<math> <br />
[\hat{h_t}]_i = tanh(\overrightarrow{h_t}V^i\overleftarrow{h_t} + b_i) <br />
</math><br />
<br />
Where <math>V^i \in \mathbb{R}^{m \times m} </math> is the learned tensor layer, and <math> b_i \in \mathbb{R} </math> is the bias.We repeat this <math> m </math> times with different <math>V^i </math> matrices and <math> b_i </math> vectors. Through the neural tensor layer, each element in <math> [\hat{h_t}]_i </math> can be viewed as a different type of intersection between the forward and backward hidden states. In the model, <math> [\hat{h_t}]_i </math> will have the same size as the forward and backward hidden states. We then concatenate the three hidden states vectors to form a new vector that summarizes the context information :<br />
<math><br />
\overleftrightarrow{h_t} = [\overrightarrow{h_t}^T,\overleftarrow{h_t}^T,\hat{h_t}]^T <br />
</math><br />
<br />
We calculate this vector for every word in the text and then stack them all into matrix <math> H </math> with shape <math>n</math>-by-<math>3m</math>.<br />
<br />
<math><br />
H = [\overleftrightarrow{h_1};...\overleftrightarrow{h_n}]<br />
</math><br />
<br />
This <math>H</math> matrix is then forwarded as the results from the Recurrent Neural Network.<br />
<br />
<br />
'''CNN Pipeline:'''<br />
<br />
The goal of the CNN pipeline is to learn the relative importance of words in an input sequence based on different aspects. The process of this CNN pipeline is summarized as the following steps:<br />
<br />
<ol><br />
<li> Given a sequence of words, each word is converted into a word vector using the word2vec algorithm which gives matrix X. <br />
</li><br />
<br />
<li> Word vectors are then convolved through the temporal dimension with filters of various sizes (ie. different K) with learnable weights to capture various numerical K-gram representations. These K-gram representations are stored in matrix C.<br />
</li><br />
<br />
<ul><br />
<li> The convolution makes this process capture local and position-invariant features. Local means the K words are contiguous. Position-invariant means K contiguous words at any position are detected in this case via convolution.<br />
<br />
<li> Temporal dimension example: convolve words from 1 to K, then convolve words 2 to K+1, etc<br />
</li><br />
</ul><br />
<br />
<li> Since not all K-gram representations are equally meaningful, there is a learnable matrix W which takes the linear combination of K-gram representations to more heavily weigh the more important K-gram representations for the classification task.<br />
</li><br />
<br />
<li> Each linear combination of the K-gram representations gives the relative word importance based on the aspect that the linear combination encodes.<br />
</li><br />
<br />
<li> The relative word importance vs aspect gives rise to an interpretable attention matrix A, where each element says the relative importance of a specific word for a specific aspect.<br />
</li><br />
<br />
</ol><br />
<br />
[[File:Group12_Figure1.png |center]]<br />
<br />
<div align="center">Figure 1: The architecture of CRNN.</div><br />
<br />
== Merging RNN & CNN Pipeline Outputs ==<br />
<br />
The results from both the RNN and CNN pipeline can be merged by simply multiplying the output matrices. That is, we compute <math>S=A^TH</math> which has shape <math>z \times 3m</math> and is essentially a linear combination of the hidden states. The concatenated rows of S results in a vector in <math>\mathbb{R}^{3zm}</math> and can be passed to a fully connected Softmax layer to output a vector of probabilities for our classification task. <br />
<br />
To train the model, we make the following decisions:<br />
<ul><br />
<li> Use cross-entropy loss as the loss function (A cross-entropy loss function usually takes in two distributions, a true distribution p and an estimated distribution q, and measures the average number of bits need to identify an event. This calculation is independent of the kind of layers used in the network as well as the kind of activation being implemented.) </li><br />
<li> Perform dropout on random columns in matrix C in the CNN pipeline </li><br />
<li> Perform L2 regularization on all parameters </li><br />
<li> Use stochastic gradient descent with a learning rate of 0.001 </li><br />
</ul><br />
<br />
== Interpreting Learned CRNN Weights ==<br />
<br />
Recall that attention matrix A essentially stores the relative importance of every word in the input sequence for every aspect chosen. Naturally, this means that A is an n-by-z matrix, with n being the number of words in the input sequence and z being the number of aspects considered in the classification task. <br />
<br />
Furthermore, for any specific aspect, words with higher attention values are more important relative to other words in the same input sequence. likewise, for any specific word, aspects with higher attention values prioritize the specific word more than other aspects.<br />
<br />
For example, in this paper, a sentence is sampled from the Movie Reviews dataset, and the transpose of attention matrix A is visualized. Each word represents an element in matrix A, the intensity of red represents the magnitude of an attention value in A, and each sentence is the relative importance of each word for a specific context. In the first row, the words are weighted in terms of a positive aspect, in the last row, the words are weighted in terms of a negative aspect, and in the middle row, the words are weighted in terms of a positive and negative aspect. Notice how the relative importance of words is a function of the aspect.<br />
<br />
[[File:Interpretation example.png|800px|center]]<br />
<br />
From the above sample, it is interesting that the word "but" is viewed as a negative aspect. From a linguistic perspective, the semantic of "but" is incredibly difficult to capture because of the degree of contextual information it needs. In this case, "but" is in the middle of a transition from a negative to a positive so the first row should also have given attention to that word. Also, it seems that the model has learned to give very high attention to the two words directly adjacent to the word of high attention: "is" and "and" beside "powerful", and "an" and "cast" beside "unwieldy".<br />
<br />
The paper also shows that the model can determine important words in the news. The authors take Ag's news dataset and randomly select 10 important words for each class. The table (shown below) contains eye-catching words which fit their classes well.<br />
<br />
[[File:news_words.png|480px|center]]<br />
<br />
== Conclusion & Summary ==<br />
<br />
This paper proposed a new architecture, the Convolutional Recurrent Neural Network, for text classification. The Convolutional Neural Network is used to learn the relative importance of each word from their different aspects and stores this information into a weight matrix. The Recurrent Neural Network learns each word's contextual representation through the combination of the forward and backward context information that is fused using a neural tensor layer and is stored as a matrix. These two matrices are then combined to get the text representation used for classification. Although the specifics of the performed tests are lacking, the experiment's results indicate that the proposed method performed well in comparison to most previous methods. In addition to performing well, the proposed method also provides insight into which words contribute greatly to the classification decision as to the learned matrix from the Convolutional Neural Network stores the relative importance of each word. This information can then be used in other applications or analyses. In the future, one can explore the features extracted from the model and use them to potentially learn new methods such as model space. [5]<br />
<br />
== Critiques ==<br />
<br />
1. In the '''Method''' section of the paper, some explanations used the same notation for multiple different elements of the model. This made the paper harder to follow and understand since they were referring to different elements by identical notation. Additionally, the decision to use sigmoid and hyperbolic tangent functions as nonlinearities for representation learning, is not supported with evidence that these are optimal.<br />
<br />
2. In '''Comparison of Methods''', the authors discuss the range of hyperparameter settings that they search through. While some of the hyperparameters have a large range of search values, three parameters are fixed without much explanation as to why for all experiments, size of the hidden state of GRU, number of layers, and dropout. These parameters have a lot to do with the complexity of the model and this paper could be improved by providing relevant reasoning behind these values, or by providing additional experimental results over different values of these parameters.<br />
<br />
3. In the '''Results''' section of the paper, they tried to show that the classification results from the CRNN model can be better interpreted than other models. In these explanations, the details were lacking and the authors did not adequately demonstrate how their model is better than others.<br />
<br />
4. Finally, in the '''Results''' section again, the paper compares the CRNN model to several models which they did not implement and reproduce results with. This can be seen in the chart of results above, where several models do not have entries in the table for all six datasets. Since the authors used a subset of the datasets, these other models which were not reproduced could have different accuracy scores if they had been tested on the same data as the CRNN model. This difference in training and testing data is not mentioned in the paper, and the conclusion that the CRNN model is better in all cases may not be valid.<br />
<br />
5. Considering the general methodology, the author in the paper chose to fuse CNN with Gated recurrent unit (GRU) with, which is only one version of RNN. However, it has been shown that LSTM generally performs better than GRU, and the author should discuss their choice of using GRU in CRNN instead of LSTM in more detail. Looking at the authors' experimental results, LSTM even outperforms CRNN (with GRU) in some cases, which further motivates the idea of adopting LSTM for RNN to combine with CNN. Whether combining LSTM with CNN will lead to a better performance will of course need further verification, but in principle author should at least address the issue. <br />
<br />
6. This is an interesting method, I would be curious to see if this can be combined or compared with Quasi-Recurrent Neural Networks (https://arxiv.org/abs/1611.01576). In my experience, QRNNs perform similarly to LSTMs while running significantly faster using convolutions with a special temporal pooling. This seems compatible with the neural tensor layer proposed in this paper, which may be combined to yield stronger performance with faster runtimes.<br />
<br />
7. It would be interesting to see how the attention matrix is being constructed and how attention values are being determined in each matrix. For instance, does every different subject have its own attention matrix? If so, how will the situation be handled when the same attention matrix is used in different settings?<br />
<br />
8. The paper shows the CRNN model not performing the best with Ag's news and 20newsgroups. It would be interesting to investigate this in detail and see the difference in the way the data is handled in the model compared to the best performing model(self-attentive LSTM in both datasets).<br />
<br />
9. From the Interpreting Learned CRNN Weights part, the samples are labeled as positive and negative, and their words all have opposite emotional polarities. It can be observed that regardless of whether the polarity of the example is positive or negative, the keyword can be extracted by this method, reflecting that it can capture multiple semantically meaningful components. At the same time it will be very interesting to see if this method is applicable to other specific categories.<br />
<br />
10. The authors of this paper provide 2 examples of what topic classification is, but do not provide any explicit examples of "polysemic words whose meanings are context-sensitive", one of their main critiques of current methods. This is an opportunity to promote the usefulness of their method and engage and inform the reader, simply by listing examples of these words.<br />
<br />
11. In the "Interpreting Learned CRNN Weights" parts, authors gave a figure but did not explain whether this is an example of positive case or negative case. I suggest the author to label the figure. Also, in the paper, we can see there are 2 figures and both of the figures have been labelled as positive or negative but there is only one figure without labelled in the summary.<br />
<br />
12. In the CRNN Results vs Benchmarks, it would be better to provide more details and comparison of other methods other than CRNN. It would be clear and more intuitive for the readers why the author will choose CRNN rather than the others.<br />
<br />
== Comments ==<br />
<br />
- Could this be applied to hieroglyphs to decipher/better understand them?<br />
<br />
- Another application for CRNN might be classifying spoken language.<br />
<br />
-I think it will be better to show more results by using this method. Maybe it will be better to put the result part after the architecture part? Writing a motivation will be better since it will catch readers' "eyes". I think it will be interesting to ask: whether can we apply this to ancient Chinese poetry? Since there are lots of types of ancient Chinese poetry, doing a classification for them will be interesting.<br />
<br />
- In another [https://www.aclweb.org/anthology/W99-0908/ paper] written by Andrew McCallum and Kamal Nigam, they introduce a different method of text classification. Namely, instead of a combination of recurrent and convolutional neural networks, they instead utilized bootstrapping with keywords, Expectation-Maximization algorithm, and shrinkage.<br />
<br />
== References ==<br />
----<br />
<br />
[1] Grimes, Seth. “Unstructured Data and the 80 Percent Rule.” Breakthrough Analysis, 1 Aug. 2008, breakthroughanalysis.com/2008/08/01/unstructured-data-and-the-80-percent-rule/.<br />
<br />
[2] N. Kalchbrenner, E. Grefenstette, and P. Blunsom, “A convolutional neural network for modeling sentences,”<br />
arXiv preprint arXiv:1404.2188, 2014.<br />
<br />
[3] K. Cho, B. V. Merri¨enboer, C. Gulcehre, D. Bahdanau, F. Bougares, H. Schwenk, and Y. Bengio, “Learning<br />
phrase representations using RNN encoder-decoder for statistical machine translation,” arXiv preprint<br />
arXiv:1406.1078, 2014.<br />
<br />
[4] S. Lai, L. Xu, K. Liu, and J. Zhao, “Recurrent convolutional neural networks for text classification,” in Proceedings<br />
of AAAI, 2015, pp. 2267–2273.<br />
<br />
[5] H. Chen, P. Tio, A. Rodan, and X. Yao, “Learning in the model space for cognitive fault diagnosis,” IEEE<br />
Transactions on Neural Networks and Learning Systems, vol. 25, no. 1, pp. 124–136, 2014.</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=File:news_words.png&diff=48259File:news words.png2020-11-30T04:05:27Z<p>Iaoellme: </p>
<hr />
<div></div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=A_universal_SNP_and_small-indel_variant_caller_using_deep_neural_networks&diff=48160A universal SNP and small-indel variant caller using deep neural networks2020-11-30T02:21:33Z<p>Iaoellme: </p>
<hr />
<div>== Background ==<br />
<br />
<br />
Biological functions are determined by genes, and differences in function are determined by mutants, or alleles, of those genes. Determining novel alleles is very important in understanding the genetic variation within a species. For example, most eye colors are determined by different alleles of the gene OCA2. All animals receive one copy of each gene from each of their parents. Mutations of a gene are classified as either homozygous (both copies are the same) or heterozygous (the two copies are different).<br />
<br />
Next generation sequencing is a very popular technique for sequencing, or reading, DNA. Since all genes are encoded as DNA, sequencing is an essential tool for understanding genes. Next generation sequencing works by reading short sections of DNA of length k, called k-mers, and then piecing them together or aligning them to a reference genome. Next generation sequencing is relatively fast and inexpensive, although it can randomly misidentify some nucleotides, introducing errors. However, NGS reading is errorful and arises from a complex error process depending on various factors.<br />
<br />
The process of variant calling is determining novel alleles from sequencing data (typically next generation sequencing data). Some significant alleles only differ from the “standard” version of a gene by only a single base pair, such as the mutation which causes multiple sclerosis. Therefore it is important to be able to accurately call single nucleotide swaps/polymorphisms (SNPs), insertions, and deletions (indels). Calling SNPs and small indels are technically challenging since it requires a program to be able to distinguish between truly novel mutations and errors in the sequencing data.<br />
<br />
Previous approaches usually involve using various statistical techniques, a widely used one is GATK. However, these methods have their weaknesses as some assumptions simply don't hold(i.e. independence assumptions), and it's hard to generalize them to other sequencing technologies.<br />
<br />
This paper aims to solve the problem of calling SNPs and small indels using a convolutional neural net by casting the reads as images and classifying whether or not they contain a mutation. It introduces a variant caller called "DeepVarient", which requires no specialized knowledge, but performs better than previous state-of-art methods.<br />
<br />
== Overview ==<br />
<br />
In Figure 1, the DeepVariant workflow overview is illustrated.<br />
<br />
[[File:figure 111.JPG|Figure 1. In all panels, blue boxes represent data and red boxes are processes]]<br />
<br />
<br />
Initially, the NGS reads aligned to a reference genome are scanned for candidate variants which are different sites from the reference genome. The read and reference data are encoded as an image for each candidate variant site. Then, trained CNN can compute the genotype likelihoods, (heterozygous or homozygous) for each of the candidate variants (figure1, left box). <br />
To train the CNN for image classification purposes, the DeepVariant machinery makes pileup images for a labeled sample with known genotypes. These labeled images and known genotypes are provided to CNN for training, and a stochastic gradient descent algorithm is used to optimize the CNN parameters to maximize genotype prediction accuracy. After the convergence of the model, the final model is frozen to use for calling mutations for other image classification tests (figure1, middle box).<br />
For example, in figure 1 (right box), the reference and read bases are encoded into a pileup image at a candidate variant site. CNN using this encoded image computes the genotype likelihoods for the three diploid genotype states of homozygous reference (hom-ref), heterozygous (het) or homozygous alternate (hom-alt). In this example, a heterozygous variant call is emitted, as the most probable genotype here is “het”.<br />
<br />
== Preprocessing ==<br />
<br />
Before the sequencing reads can be fed into the classifier, they must be preprocessed. There are many pre-processing steps that are necessary for this algorithm. These steps represent the real novelty in this technique, by transforming the data in a way that allows us to use more common neural network architectures for classification. The preprocessing of the data can be broken into three main phases: the realignment of reads, finding candidate variants and creating images of the candidate variants. <br />
<br />
The realignment of the reads phase of the preprocessing is important in order to ensure the sequences can be properly compared to the reference sequences. First, the sequences are aligned to a reference sequence. Reads that align poorly are grouped with other reads around them to build that section, or haplotype, from scratch. If there is strong evidence that the new version of the haplotype fits the reads well, the reads are re-aligned to it. This process updates the CIGAR (Compact Idiosyncratic Gapped Alignment Report) string, a way to represent the alignment of a sequence to a reference, for each read.<br />
<br />
Once the reads are properly aligned, the algorithm then proceeds to find candidate variants, regions in the DNA sequence that may contain variants. It is these candidate variants that will eventually be passed as input to the neural network. To find these, we need to consider each position in the reference sequence independently. Any unusable reads are filtered at this point. This includes reads that are not aligned properly, ones that are marked as duplicates, those that fail vendor quality checks, or whose mapping quality is less than ten. For each site in the genome, we collect all the remaining reads that overlap that site. The corresponding allele aligned to that site is then determined by decoding the CIGAR string, which was updated in the realignment phase, of each read. The alleles are then classified into one of four categories: reference-matching base, reference-mismatching base, insertion with a specific sequence, or deletion with a specific length, and the number of occurrences of each distinct allele across all reads is counted. Read bases are only included as potential alleles if each base in the allele has a quality score of at least 10.<br />
<br />
With candidate variants identified, the last phase of pre-processing is to convert these candidate variants into images representing the data. This allows for the use of well established convolutional neural networks for image classification for this specialized problem. Each colour channel is used to store a different piece of information about a candidate variant. The red channel encodes which base we have (A, G, C, or T), by mapping each base to a particular value. The quality of the read is mapped to the green colour channel. And finally, the blue channel encodes whether or not the reference is on the positive strand of the DNA. Each row of the image represents a read, and each column represents a particular base in that read. The reference strand is repeated for the first five rows of the encoded image, in order to maintain its information after a 5x5 convolution is applied.<br />
With the data preprocessing complete, the images can then be passed into the neural network for classification.<br />
<br />
== Neural Network ==<br />
<br />
The neural network used is a convolutional neural network. Although the full network architecture is not revealed in the paper, there are several details which we can discuss. The architecture of the network is an input layer attached to an adapted Inception v2 ImageNet model with nine partitions. The input layer takes as input the images representing the candidate variants and rescales them to 299x299 pixels. The output layer is a three-class Softmax layer initialized with Gaussian random weights with a standard deviation of 0.001. This final layer is fully connected to the previous layer. The three classes are the homozygous reference (meaning it is not a variant), heterozygous variant, and homozygous variant. The candidate variant is classified into the class with the highest probability. The model is trained using stochastic gradient descent with a weight decay of 0.00004. The training was done in mini-batches, each with 32 images, using a root mean squared (RMS) decay of 0.9. For the multiple sequencing technologies experiments, a single model was trained with a learning rate of 0.0015 and momentum 0.8 for 250,000 update steps. For all other experiments, multiple models were trained, and the one with the highest accuracy on the training set was chosen as the final model. The multiple models stem from using each combination of the possible parameter values for the learning rate (0.00095, 0.001, 0.0015) and momentum (0.8, 0.85, 0.9). These models were trained for 80 hours, or until the training accuracy converged.<br />
<br />
== Results ==<br />
<br />
DeepVariant was trained using data available from the CEPH (Centre d’Etude du Polymorphism Humain) female sample NA12878 and was evaluated on the unseen Ashkenazi male sample NA24385. The results were compared with other most commonly used bioinformatics methods, such as the GATK, FreeBayes22, SAMtools23, 16GT24 and Strelka25 (Table 1). For better comparison, the overall accuracy (F1), recall, precision, and numbers of true positives (TP), false negatives (FN) and false positives (FP) are illustrated over the whole genome.<br />
<br />
[[File:table 11.JPG]]<br />
<br />
DeepVariant showed the highest accuracy and more than 50% fewer errors per genome compared to the next best algorithm. <br />
<br />
They also evaluated the same set of algorithms using the synthetic diploid sample CHM1-CHM1326 (Table 2).<br />
<br />
[[File:Table 333.JPG]]<br />
<br />
Results illustrated that the DeepVariant method outperformed all other algorithms for variant calling (SNP and indel) and showed the highest accuracy in terms of F1, Recall, precision and TP.<br />
<br />
== Conclusion ==<br />
<br />
DeepVariant’s strong performance on human data proves that deep learning is a promising technique for variant calling. Perhaps the most exciting feature of DeepVariant is its simplicity. Unlike other states of the art variant callers, DeepVariant has no knowledge of the sequencing technologies that create the reads, or even the biological processes that introduce mutations. This simplifies the problem of variant calling to preprocessing the reads and training a generic deep learning model. It also suggests that DeepVariant could be significantly improved by tailoring the preprocessing to specific sequencing technologies and/or developing a dedicated CNN architecture for the reads, rather than trying to cast them as images.<br />
<br />
== Critique ==<br />
<br />
The paper presents an interesting method for solving an important problem. Building "images" of reads and running them through a generic image classification CNN seems like a strange approach, and it is interesting that it works well. The biggest issues with the paper are the lack of specific information about how the methods. Some extra information is included in the supplementary material, but there are still some big gaps. In particular:<br />
<br />
1. What is the structure of the neural net? How many layers, and what sizes? The paper for ConvNet which is cited does not have this information. We suspect that this might be a trade secret that Google is protecting.<br />
<br />
2. How is the realignment step implemented? The paper mentions that it uses a "De-Bruijn-graph-based read assembly procedure" to realign reads to a new haplotype. This is a non-standard step in most genomics workflows yet the paper does not describe how they do the realignment or how they build the haplotypes.<br />
<br />
3. How did they settle on the image construction algorithm? The authors provide pseudocode for the construction of pileup images but they do not describe how the decisions for made. For instance, the colour values for different base pairs are not evenly spaced. Also, the image begins with 5 rows of the reference genome.<br />
<br />
One thing we appreciated about the paper was their commentary on future developments. The authors make it very clear that this approach can be improved on and provide specific ideas for next steps.<br />
<br />
Overall, the paper presents an interesting idea with strong results, but lacks detail in some key pieces of the implementation.<br />
<br />
== References ==<br />
<br />
[1] Poplin, R. ''et. al''. A universal SNP and small-indel variant caller using deep neural networks. ''Nature Biotechnology'' '''36''', 983-987 (2018).</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=Neural_Speed_Reading_via_Skim-RNN&diff=47571Neural Speed Reading via Skim-RNN2020-11-29T03:14:23Z<p>Iaoellme: /* Applications */</p>
<hr />
<div>== Group ==<br />
<br />
Mingyan Dai, Jerry Huang, Daniel Jiang<br />
<br />
== Introduction ==<br />
<br />
Recurrent Neural Network (RNN) is a class of artificial neural networks where the connection between nodes form a directed graph along with time series and also has time dynamic behavior. RNN is derived from a feedforward neural network and can use its memory to process variable-length input sequences. This makes it suitable for tasks such as unsegmented, connected handwriting recognition, and speech recognition.<br />
<br />
In Natural Language Processing, recurrent neural networks (RNNs) are a common architecture used to sequentially ‘read’ input tokens and output a distributed representation for each token. By recurrently updating the hidden state of the neural network, an RNN can inherently require the same computational cost across time. However, when it comes to processing input tokens, it is usually the case that some tokens are less important to the overall representation of a piece of text or a query when compared to others. In particular, when considering question answering, many times the neural network will encounter parts of a passage that are irrelevant when it comes to answering a query that is being asked.<br />
<br />
A variant of LSTMs, LSTM-Jump (Yu et al., 2017) was introduced to improve efficiency by skipping multiple tokens at a given step. In contrast, Skim-RNN takes advantage of 'skimming' rather than 'skipping tokens'. The paper demonstrates that skimming achieves higher accuracy compared to skipping tokens, implying that paying attention to unimportant tokens is better than completely ignoring them.<br />
<br />
== Model ==<br />
<br />
In this paper, the authors introduce a model called 'skim-RNN', which takes advantage of ‘skimming’ less important tokens or pieces of text rather than ‘skipping’ them entirely. This models the human ability to skim through passages, or to spend less time reading parts that do not affect the reader’s main objective. While this leads to a loss in the comprehension rate of the text [1], it greatly reduces the amount of time spent reading by not focusing on areas that will not significantly affect efficiency when it comes to the reader's objective.<br />
<br />
'Skim-RNN' works by rapidly determining the significance of each input and spending less time processing unimportant input tokens by using a smaller RNN to update only a fraction of the hidden state. When the decision is to ‘fully read’, that is to not skim the text, Skim-RNN updates the entire hidden state with the default RNN cell. Since the hard decision function (‘skim’ or ‘read’) is non-differentiable, the authors use a gumbel-softmax [2] to estimate the gradient of the function, rather than traditional methods such as REINFORCE (policy gradient)[3]. The switching mechanism between the two RNN cells enables Skim-RNN to reduce the total number of float operations (Flop reduction, or Flop-R). When the skimming rate is high, which often leads to faster inference on CPUs, which makes it very useful for large-scale products and small devices.<br />
<br />
The Skim-RNN has the same input and output interfaces as standard RNNs, so it can be conveniently used to speed up RNNs in existing models. In addition, the speed of Skim-RNN can be dynamically controlled at inference time by adjusting a parameter for the threshold for the ‘skim’ decision.<br />
<br />
=== Related Works ===<br />
<br />
As the popularity of neural networks has grown, significant attention has been given to make them faster and lighter. In particular, relevant work focused on reducing the computational cost of recurrent neural networks has been carried out by several other related works. For example, LSTM-Jump (You et al., 2017) [8] models aim to speed up run times by skipping certain input tokens, as opposed to skimming them. Choi et al. (2017)[9] proposed a model which uses a CNN-based sentence classifier to determine the most relevant sentence(s) to the question and then uses an RNN-based question-answering model. This model focuses on reducing GPU run-times (as opposed to Skim-RNN which focuses on minimizing CPU-time or Flop), and is also focused only on question answering. <br />
<br />
=== Implementation ===<br />
<br />
A Skim-RNN consists of two RNN cells, a default (big) RNN cell of hidden state size <math>d</math> and small RNN cell of hidden state size <math>d'</math>, where <math>d</math> and <math>d'</math> are parameters defined by the user and <math>d' \ll d</math>. This follows the fact that there should be a small RNN cell defined for when text is meant to be skimmed and a larger one for when the text should be processed as normal.<br />
<br />
Each RNN cell will have its own set of weights and bias as well as be any variant of an RNN. There is no requirement on how the RNN itself is structured, rather the core concept is to allow the model to dynamically make a decision as to which cell to use when processing input tokens. Note that skipping text can be incorporated by setting <math>d'</math> to 0, which means that when the input token is deemed irrelevant to a query or classification task, nothing about the information in the token is retained within the model.<br />
<br />
Experimental results suggest that this model is faster than using a single large RNN to process all input tokens, as the smaller RNN requires fewer floating-point operations to process the token. Additionally, higher accuracy and computational efficiency are achieved. <br />
<br />
==== Inference ====<br />
<br />
At each time step <math>t</math>, the Skim-RNN unit takes in an input <math>{\bf x}_t \in \mathbb{R}^d</math> as well as the previous hidden state <math>{\bf h}_{t-1} \in \mathbb{R}^d</math> and outputs the new state <math>{\bf h}_t </math> (although the dimensions of the hidden state and input are the same, this process holds for different sizes as well). In the Skim-RNN, there is a hard decision that needs to be made whether to read or skim the input, although there could be potential to include options for multiple levels of skimming.<br />
<br />
The decision to read or skim is done using a multinomial random variable <math>Q_t</math> over the probability distribution of choices <math>{\bf p}_t</math>, where<br />
<br />
<div class="center" style="width: auto; margin-left: auto; margin-right: auto;"><br />
<math>{\bf p}_t = \text{softmax}(\alpha({\bf x}_t, {\bf h}_{t-1})) = \text{softmax}({\bf W}[{\bf x}_t; {\bf h}_{t-1}]+{\bf b}) \in \mathbb{R}^k</math><br />
</div><br />
<br />
where <math>{\bf W} \in \mathbb{R}^{k \times 2d}</math>, <math>{\bf b} \in \mathbb{R}^{k}</math> are weights to be learned and <math>[{\bf x}_t; {\bf h}_{t-1}] \in \mathbb{R}^{2d}</math> indicates the row concatenation of the two vectors. In this case, <math> \alpha </math> can have any form as long as the complexity of calculating it is less than <math> O(d^2)</math>. Letting <math>{\bf p}^1_t</math> indicate the probability for fully reading and <math>{\bf p}^2_t</math> indicate the probability for skimming the input at time <math> t</math>, it follows that the decision to read or skim can be modelled using a random variable <math> Q_t</math> by sampling from the distribution <math>{\bf p}_t</math> and<br />
<div class="center" style="width: auto; margin-left: auto; margin-right: auto;"><br />
<math>Q_t \sim \text{Multinomial}({\bf p}_t)</math><br />
</div><br />
<br />
Without loss of generality, we can define <math> Q_t = 1</math> to indicate that the input will be read while <math> Q_t = 2</math> indicates that it will be skimmed. Reading requires applying the full RNN on the input as well as the previous hidden state to modify the entire hidden state while skimming only modifies part of the prior hidden state.<br />
<br />
<div class="center" style="width: auto; margin-left: auto; margin-right: auto;"><br />
<math><br />
{\bf h}_t = \begin{cases}<br />
f({\bf x}_t, {\bf h}_{t-1}) & Q_t = 1\\<br />
[f'({\bf x}_t, {\bf h}_{t-1});{\bf h}_{t-1}(d'+1:d)] & Q_t = 2<br />
\end{cases}<br />
</math><br />
</div><br />
<br />
where <math> f </math> is a full RNN with output of dimension <math>d</math> and <math>f'</math> is a smaller RNN with <math>d'</math>-dimensional output. This has advantage that when the model decides to skim, then the computational complexity of that step is only <math>O(d'd)</math>, which is much smaller than <math>O(d^2)</math> due to previously defining <math> d' \ll d</math>.<br />
<br />
==== Training ====<br />
<br />
Since the expected loss/error of the model is a random variable that depends on the sequence of random variables <math> \{Q_t\} </math>, the loss is minimized with respect to the distribution of the variables. Defining the loss to be minimized while conditioning on a particular sequence of decisions<br />
<br />
<div class="center" style="width: auto; margin-left: auto; margin-right: auto;"><br />
<math><br />
L(\theta\vert Q)<br />
</math><br />
</div><br />
where <math>Q=Q_1\dots Q_T</math> is a sequence of decisions of length <math>T</math>, then the expected loss over the distribution of the sequence of decisions is<br />
<br />
<div class="center" style="width: auto; margin-left: auto; margin-right: auto;"><br />
<math><br />
\mathbb{E}[L(\theta)] = \sum_{Q} L(\theta\vert Q)P(Q) = \sum_Q L(\theta\vert Q) \Pi_j {\bf p}_j^{Q_j}<br />
</math><br />
</div><br />
<br />
Since calculating <math>\delta \mathbb{E}_{Q_t}[L(\theta)]</math> directly is rather infeasible, it is possible to approximate the gradients with a gumbel-softmax distribution [2]. Reparameterizing <math> {\bf p}_t</math> as <math> {\bf r}_t</math>, then the back-propagation can flow to <math> {\bf p}_t</math> without being blocked by <math> Q_t</math> and the approximation can arbitrarily approach <math> Q_t</math> by controlling the parameters. The reparameterized distribution is therefore<br />
<br />
<div class="center" style="width: auto; margin-left: auto; margin-right: auto;"><br />
<math><br />
{\bf r}_t^i = \frac{\text{exp}(\log({\bf p}_t^i + {g_t}^i)/\tau)}{\sum_j\text{exp}(\log({\bf p}_t^j + {g_t}^j)/\tau)}<br />
</math><br />
</div><br />
<br />
where <math>{g_t}^i</math> is an independent sample from a <math>\text{Gumbel}(0, 1) = -\log(-\log(\text{Uniform}(0, 1))</math> random variable and <math>\tau</math> is a parameter that represents a temperature. Then it can be rewritten that<br />
<br />
<div class="center" style="width: auto; margin-left: auto; margin-right: auto;"><br />
<math><br />
{\bf h}_t = \sum_i {\bf r}_t^i {\bf \tilde{h}}_t<br />
</math><br />
</div><br />
<br />
where <math>{\bf \tilde{h}}_t</math> is the previous equation for <math>{\bf h}_t</math>. The temperature parameter gradually decreases with time, and <math>{\bf r}_t^i</math> becomes more discrete as it approaches 0.<br />
<br />
A final addition to the model is to encourage skimming when possible. Therefore an extra term related to the negative log probability of skimming and the sequence length. Therefore the final loss function used for the model is denoted by <br />
<br />
<div class="center" style="width: auto; margin-left: auto; margin-right: auto;"><br />
<math><br />
L'(\theta) =L(\theta) + \gamma \cdot\frac{1}{T} \sum_i -\log({\bf \tilde{p}}^i_t)<br />
</math><br />
</div><br />
where <math> \gamma </math> is a parameter used to control the ratio between the main loss function and the negative log probability of skimming.<br />
<br />
== Experiment ==<br />
<br />
The effectiveness of Skim-RNN was measured in terms of accuracy and float operation reduction on four classification tasks and a question-answering task. These tasks were chosen because they do not require one’s full attention to every detail of the text, but rather ask for capturing the high-level information (classification) or focusing on a specific portion (QA) of the text, which a common context for speed reading. The tasks themselves are listed in the table below.<br />
<br />
[[File:Table1SkimRNN.png|center|1000px]]<br />
<br />
=== Classification Tasks ===<br />
<br />
In a language classification task, the input was a sequence of words and the output was the vector of categorical probabilities. Each word is embedded into a <math>d</math>-dimensional vector. We initialize the vector with GloVe [4] to form representations of the words and use those as the inputs for a long short-term memory (LSTM) architecture. A linear transformation on the last hidden state of the LSTM and then a softmax function was applied to obtain the classification probabilities. Adam [5] was used for optimization, with an initial learning rate of 0.0001. For Skim-LSTM, <math>\tau = \max(0.5, exp(−rn))</math> where <math>r = 1e-4</math> and <math>n</math> is the global training step, following [2]. We experiment on different sizes of big LSTM (<math>d \in \{100, 200\}</math>) and small LSTM (<math>d' \in \{5, 10, 20\}</math>) and the ratio between the model loss and the skim loss (<math>\gamma\in \{0.01, 0.02\}</math>) for Skim-LSTM. The batch sizes used were 32 for SST and Rotten Tomatoes, and 128 for others. For all models, early stopping was used when the validation accuracy did not increase for 3000 global steps.<br />
<br />
==== Results ====<br />
<br />
[[File:Table2SkimRNN.png|center|1000px]]<br />
<br />
[[File:Figure2SkimRNN.png|center|1000px]]<br />
<br />
Table 2 shows the accuracy and computational cost of the Skim-RNN model compared with other standard models. It is evident that the Skim-RNN model produces a speed-up on the computational complexity of the task while maintaining a high degree of accuracy. Also, it is interesting to know that the accuracy improvement over LSTM could be due to the increased stability of the hidden state, as the majority of the hidden state is not updated when skimming. Figure 2 meanwhile demonstrates the effect of varying the size of the small hidden state as well as the parameter <math>\gamma</math> on the accuracy and computational cost.<br />
<br />
[[File:Table3SkimRNN.png|center|1000px]]<br />
<br />
Table 3 shows an example of a classification task over a IMDb dataset, where Skim-RNN with <math>d = 200</math>, <math>d' = 10</math>, and <math>\gamma = 0.01</math> correctly classifies it with a high skimming rate (92%). The goal was to classify the review as either positive or negative. The black words are skimmed, and the blue words are fully read. The skimmed words are clearly irrelevant and the model learns to only carefully read the important words, such as ‘liked’, ‘dreadful’, and ‘tiresome’.<br />
<br />
=== Question Answering Task ===<br />
<br />
In the Stanford Question Answering Dataset, the task was to locate the answer span for a given question in a context paragraph. The effectiveness of Skim-RNN for SQuAD was evaluated using two different models: LSTM+Attention and BiDAF [6]. The first model was inspired by most then-present QA systems consisting of multiple LSTM layers and an attention mechanism. This type of model is complex enough to reach reasonable accuracy on the dataset and simple enough to run well-controlled analyses for the Skim-RNN. The second model was an open-source model designed for SQuAD, used primarily to show that Skim-RNN could replace RNN in existing complex systems.<br />
<br />
==== Training ==== <br />
<br />
Adam was used with an initial learning rate of 0.0005. For stable training, the model was pre-trained with a standard LSTM for the first 5k steps, and then fine-tuned with Skim-LSTM.<br />
<br />
==== Results ====<br />
<br />
[[File:Table4SkimRNN.png|center|1000px]]<br />
<br />
Table 4 shows the accuracy (F1 and EM) of LSTM+Attention and Skim-LSTM+Attention models as well as VCRNN [7]. It can be observed from the table that the skimming models achieve higher or similar accuracy scores compared to the non-skimming models while also reducing the computational cost by more than 1.4 times. In addition, decreasing layers (1 layer) or hidden size (<math>d=5</math>) improved the computational cost but significantly decreases the accuracy compared to skimming. The table also shows that replacing LSTM with Skim-LSTM in an existing complex model (BiDAF) stably gives reduced computational cost without losing much accuracy (only 0.2% drop from 77.3% of BiDAF to 77.1% of Sk-BiDAF with <math>\gamma = 0.001</math>).<br />
<br />
An explanation for this trend that was given is that the model is more confident about which tokens are important in the second layer. Second, higher <math>\gamma</math> values lead to a higher skimming rate, which agrees with its intended functionality.<br />
<br />
Figure 4 shows the F1 score of LSTM+Attention model using standard LSTM and Skim LSTM, sorted in ascending order by Flop-R (computational cost). While models tend to perform better with larger computational cost, Skim LSTM (Red) outperforms standard LSTM (Blue) with a comparable computational cost. It can also be seen that the computational cost of Skim-LSTM is more stable across different configurations and computational cost. Moreover, increasing the value of <math>\gamma</math> for Skim-LSTM gradually increases the skipping rate and Flop-R, while it also led to reduced accuracy.<br />
<br />
=== Runtime Benchmark ===<br />
<br />
[[File:Figure6SkimRNN.png|center|1000px]]<br />
<br />
The details of the runtime benchmarks for LSTM and Skim-LSTM, which are used to estimate the speedup of Skim-LSTM-based models in the experiments, are also discussed. A CPU-based benchmark was assumed to be the default benchmark, which has a direct correlation with the number of float operations that can be performed per second. As mentioned previously, the speed-up results in Table 2 (as well as Figure 7) are benchmarked using Python (NumPy), instead of popular frameworks such as TensorFlow or PyTorch.<br />
<br />
Figure 7 shows the relative speed gain of Skim-LSTM compared to standard LSTM with varying hidden state size and skim rate. NumPy was used, with the inferences run on a single thread of CPU. The ratio between the reduction of the number of float operations (Flop-R) of LSTM and Skim-LSTM was plotted, with the ratio acting as a theoretical upper bound of the speed gain on CPUs. From here, it can be noticed that there is a gap between the actual gain and the theoretical gain in speed, with the gap being larger with more overhead of the framework or more parallelization. The gap also decreases as the hidden state size increases because the overhead becomes negligible with very large matrix operations. This indicates that Skim-RNN provides greater benefits for RNNs with larger hidden state size. However, combining Skim-RNN with a CPU-based framework can lead to substantially lower latency than GPUs.<br />
<br />
== Results ==<br />
<br />
The results clearly indicate that the Skim-RNN model provides features that are suitable for general reading tasks, which include classification and question answering. While the tables indicate that minor losses in accuracy occasionally did result when parameters were set at specific values, they were minor and were acceptable given the improvement in runtime.<br />
<br />
An important advantage of Skim-RNN is that the skim rate (and thus computational cost) can be dynamically controlled at inference time by adjusting the threshold for<br />
‘skim’ decision probability <math>{\bf p}^1_t</math>. Figure 5 shows the trade-off between the accuracy and computational cost for two settings, confirming the importance of skimming (<math>d' > 0</math>) compared to skipping (<math>d' = 0</math>).<br />
<br />
Figure 6 shows that the model does not skim when the input seems to be relevant to answering the question, which was as expected by the design of the model. In addition, the LSTM in the second layer skims more than that in the first layer mainly because the second layer is more confident about the importance of each token.<br />
<br />
== Conclusion ==<br />
<br />
A Skim-RNN can offer better latency results on a CPU compared to a standard RNN on a GPU, with lower computational cost, as demonstrated through the results of this study. Future work (as stated by the authors) involves using Skim-RNN for applications that require much higher hidden state size, such as video understanding, and using multiple small RNN cells for varying degrees of skimming. Further, since it has the same input and output interface as a regular RNN, it can replace RNNs in existing applications.<br />
<br />
== Critiques ==<br />
<br />
1. It seems like Skim-RNN is using the not full RNN of processing words that are not important, thus it can increase speed in some very particular circumstances (ie, only small networks). The extra model complexity did slow down the speed while trying to "optimizing" the efficiency and sacrifice part of accuracy while doing so. It is only trying to target a very specific situation (classification/question-answering) and made comparisons only with the baseline LSTM model. It would be definitely more persuasive if the model can compare with some of the state of art neural network models.<br />
<br />
2. This model of Skim-RNN is pretty good to extract binary classification type of text, thus it would be interesting for this to be applied to stock market news analysis. For example, a press release from a company can be analyzed quickly using this model and immediately give the trader a positive or negative summary of the news. Would be beneficial in trading since time and speed is an important factor when executing a trade.<br />
<br />
3. An appropriate application for Skim-RNN could be customer service chatbots as they can analyze a customer's message and skim associated company policies to craft a response. In this circumstance, quickly analyzing text is ideal to not waste customers' time.<br />
<br />
4. This could be applied to news apps to improve readability by highlighting important sections.<br />
<br />
5. This summary describes an interesting and useful model that can save readers time for reading an article. I think it will be interesting that discuss more on training a model by Skim-RNN to highlight the important sections in very long textbooks. As a student, having highlights in the textbook is really helpful to study. But highlight the important parts in a time-consuming work for the author, maybe using Skim-RNN can provide a nice model to do this job. <br />
<br />
6. Besides the good training performance of Skim-RNN, it's good to see the algorithm even performs well simply by training with CPU. It would make it possible to perform the result on lite-platforms.<br />
<br />
== Applications ==<br />
<br />
Recurrent architectures are used in many other applications, such as for processing video. Real-time video processing is an exceedingly demanding and resource-constrained task, particularly in edge settings. It would be interesting to see if this method could be applied to those cases for more efficient inference, such as on drones or self-driving cars. Another possible application is real-time edge processing of game video for sports arenas.<br />
<br />
== References ==<br />
<br />
[1] Patricia Anderson Carpenter Marcel Adam Just. The Psychology of Reading and Language Comprehension. 1987.<br />
<br />
[2] Eric Jang, Shixiang Gu, and Ben Poole. Categorical reparameterization with gumbel-softmax. In ICLR, 2017.<br />
<br />
[3] Ronald J Williams. Simple statistical gradient-following algorithms for connectionist reinforcement learning. Machine learning, 8(3-4):229–256, 1992.<br />
<br />
[4] Jeffrey Pennington, Richard Socher, and Christopher D Manning. Glove: Global vectors for word representation. In EMNLP, 2014.<br />
<br />
[5] Diederik Kingma and Jimmy Ba. Adam: A method for stochastic optimization. In ICLR, 2015.<br />
<br />
[6] Minjoon Seo, Aniruddha Kembhavi, Ali Farhadi, and Hannaneh Hajishirzi. Bidirectional attention flow for machine comprehension. In ICLR, 2017a.<br />
<br />
[7] Yacine Jernite, Edouard Grave, Armand Joulin, and Tomas Mikolov. Variable computation in recurrent neural networks. In ICLR, 2017.<br />
<br />
[8] Adams Wei Yu, Hongrae Lee, and Quoc V Le. Learning to skim text. In ACL, 2017.<br />
<br />
[9] Eunsol Choi, Daniel Hewlett, Alexandre Lacoste, Illia Polosukhin, Jakob Uszkoreit, and Jonathan Berant. Coarse-to-fine question answering for long documents. In ACL, 2017.</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=Graph_Structure_of_Neural_Networks&diff=47569Graph Structure of Neural Networks2020-11-29T03:08:26Z<p>Iaoellme: /* Conclusions */</p>
<hr />
<div>= Presented By =<br />
<br />
Xiaolan Xu, Robin Wen, Yue Weng, Beizhen Chang<br />
<br />
= Introduction =<br />
<br />
A deep neural network is composed of neurons organized into layers and the connections between them. The architecture of a neural network can be captured by its "computational graph", where neurons are represented as nodes, and directed edges link neurons in different layers. This graphical representation demonstrates how the network transmits and transforms information through its input neurons through the hidden layers and ultimately to the output neurons.<br />
<br />
In Neural Network research, it is often important to build a relation between a neural network’s accuracy and its underlying graph structure. A natural choice is to use computational graph representation, but this has many limitations including a lack of generality and disconnection with biology/neuroscience. This disconnection between biology/neuroscience makes knowledge less transferable and interdisciplinary research more difficult.<br />
<br />
Thus, the authors developed a new way of representing a neural network as a graph, called a relational graph. The key insight in the new representation is to focus on message exchange, rather than just on directed data flow. For example, for a fixed-width fully-connected layer, an input channel and output channel pair can be represented as a single node, while an edge in the relational graph can represent the message exchange between the two nodes. Under this formulation, using the appropriate message exchange definition, it can be shown that the relational graph can represent many types of neural network layers.<br />
<br />
WS-flex is a graph generator that allows systematically exploring the design space of neural networks. Neural networks are characterized by the clustering coefficient and average path length of their relational graphs under the insights of neuroscience.<br />
<br />
= Neural Network as Relational Graph =<br />
<br />
The author proposes the concept of relational graph to study the graphical structure of neural network. Each relational graph is based on an undirected graph <math>G =(V; E)</math>, where <math>V =\{v_1,...,v_n\}</math> is the set of all the nodes, and <math>E \subseteq \{(v_i,v_j)|v_i,v_j\in V\}</math> is the set of all edges that connect nodes. Note that for the graph used here, all nodes have self edges, that is <math>(v_i,v_i)\in E</math>. <br />
<br />
To build a relational graph that captures the message exchange between neurons in the network, we associate various mathematical quantities to the graph <math>G</math>. First, a feature quantity <math>x_v</math> is associated with each node. The quantity <math>x_v</math> might be a scalar, vector or tensor depending on different types of neural networks (see the Table at the end of the section). Then a message function <math>f_{uv}(·)</math> is associated with every edge in the graph. A message function specifically takes a node’s feature as the input and then output a message. An aggregation function <math>{\rm AGG}_v(·)</math> then takes a set of messages (the outputs of message function) and outputs the updated node feature. <br />
<br />
A relation graph is a graph <math>G</math> associated with several rounds of message exchange, which transform the feature quantity <math>x_v</math> with the message function <math>f_{uv}(·)</math> and the aggregation function <math>{\rm AGG}_v(·)</math>. At each round of message exchange, each node sends messages to its neighbors and aggregates incoming messages from its neighbors. Each message is transformed at each edge through the message function, then they are aggregated at each node via the aggregation function. Suppose we have already conducted <math>r-1</math> rounds of message exchange, then the <math>r^{th}</math> round of message exchange for a node <math>v</math> can be described as<br />
<br />
<div style="text-align:center;"><math>\mathbf{x}_v^{(r+1)}= {\rm AGG}^{(r)}(\{f_v^{(r)}(\textbf{x}_u^{(r)}), \forall u\in N(v)\})</math></div> <br />
<br />
where <math>\mathbf{x}^{(r+1)}</math> is the feature of the <math>v</math> node in the relational graph after the <math>r^{th}</math> round of update. <math>u,v</math> are nodes in Graph <math>G</math>. <math>N(u)=\{u|(u,v)\in E\}</math> is the set of all the neighbor nodes of <math>u</math> in graph <math>G</math>.<br />
<br />
To further illustrate the above, we use the basic Multilayer Perceptron (MLP) as an example. An MLP consists of layers of neurons, where each neuron performs a weighted sum over scalar inputs and outputs, followed by some non-linearity. Suppose the <math>r^{th}</math> layer of an MLP takes <math>x^{(r)}</math> as input and <math>x^{(r+1)}</math> as output, then a neuron computes <br />
<br />
<div style="text-align:center;"><math>x_i^{(r+1)}= \sigma(\Sigma_jw_{ij}^{(r)}x_j^{(r)})</math>.</div> <br />
<br />
where <math>w_{ij}^{(r)}</math> is the trainable weight and <math>\sigma</math> is the non-linearity function. Let's first consider the special case where the input and output of all the layers <math>x^{(r)}</math>, <math>1 \leq r \leq R </math> have the same feature dimensions <math>d</math>. In this scenario, we can have <math>d</math> nodes in the Graph <math>G</math> with each node representing a neuron in MLP. Each layer of neural network will correspond with a round of message exchange, so there will be <math>R</math> rounds of message exchange in total. The aggregation function here will be the summation with non-linearity transform <math>\sigma(\Sigma)</math>, while the message function is simply the scalar multipication with weight. A fully-connected, fixed-width MLP layer can then be expressed with a complete relational graph, where each node <math>x_v</math> connects to all the other nodes in <math>G</math>, that is neighborhood set <math>N(v) = V</math> for each node <math>v</math>. The figure below shows the correspondence between the complete relation graph with a 5-layer 4-dimension fully-connected MLP.<br />
<br />
<div style="text-align:center;">[[File:fully_connnected_MLP.png]]</div><br />
<br />
In fact, a fixed-width fully-connected MLP is only a special case under a much more general model family, where the message function, aggregation function, and most importantly, the relation graph structure can vary. The different relational graph will represent the different topological structure and information exchange pattern of the network, which is the property that the paper wants to examine. The plot below shows two examples of non-fully connected fixed-width MLP and their corresponding relational graphs. <br />
<br />
<div style="text-align:center;">[[File:otherMLP.png]]</div><br />
<br />
We can generalize the above definitions for fixed-width MLP to Variable-width MLP, Convolutional Neural Network (CNN), and other modern network architecture like Resnet by allowing the node feature quantity <math>\textbf{x}_j^{(r)}</math> to be a vector or tensor respectively. In this case, each node in the relational graph will represent multiple neurons in the network, and the number of neurons contained in each node at each round of message exchange does not need to be the same, which gives us a flexible representation of different neural network architecture. The message function will then change from the simple scalar multiplication to either matrix/tensor multiplication or convolution. And the weight matrix will change to the convolutional filter. The representation of these more complicated networks are described in details in the paper, and the correspondence between different networks and their relational graph properties is summarized in the table below. <br />
<br />
<div style="text-align:center;">[[File:relational_specification.png]]</div><br />
<br />
Overall, relational graphs provide a general representation for neural networks. With proper definitions of node features and message exchange, relational graphs can represent diverse neural architectures, thereby allowing us to study the performance of different graph structures.<br />
<br />
= Exploring and Generating Relational Graphs=<br />
<br />
We will deal with the design and how to explore the space of relational graphs in this section. There are three parts we need to consider:<br />
<br />
(1) '''Graph measures''' that characterize graph structural properties:<br />
<br />
We will use one global graph measure, average path length, and one local graph measure, clustering coefficient in this paper.<br />
To explain clearly, average path length measures the average shortest path distance between any pair of nodes; the clustering coefficient measures the proportion of edges between the nodes within a given node’s neighborhood, divided by the number of edges that could possibly exist between them, averaged over all the nodes.<br />
<br />
(2) '''Graph generators''' that can generate the diverse graph:<br />
<br />
With selected graph measures, we use a graph generator to generate diverse graphs to cover a large span of graph measures. To figure out the limitation of the graph generator and find out the best, we investigate some generators including ER, WS, BA, Harary, Ring, Complete graph and results shows as below:<br />
<br />
<div style="text-align:center;">[[File:3.2 graph generator.png]]</div><br />
<br />
Thus, from the picture, we could obtain the WS-flex graph generator that can generate graphs with a wide coverage of graph measures; notably, WS-flex graphs almost encompass all the graphs generated by classic random generators mentioned above.<br />
<br />
(3) '''Computational Budget''' that we need to control so that the differences in performance of different neural networks are due to their diverse relational graph structures.<br />
<br />
It is important to ensure that all networks have approximately the same complexities so that the differences in performance are due to their relational graph structures when comparing neutral work by their diverse graph.<br />
<br />
We use FLOPS (# of multiply-adds) as the metric. We first compute the FLOPS of our baseline network instantiations (i.e., complete relational graph) and use them as the reference complexity in each experiment. From the description in section 2, a relational graph structure can be instantiated as a neural network with variable width. Therefore, we can adjust the width of a neural network to match the reference complexity without changing the relational graph structures.<br />
<br />
= Experimental Setup =<br />
The author studied the performance of 3942 sampled relational graphs (generated by WS-flex from the last section) of 64 nodes with two experiments: <br />
<br />
(1) CIFAR-10 dataset: 10 classes, 50K training images, and 10K validation images<br />
<br />
Relational Graph: all 3942 sampled relational graphs of 64 nodes<br />
<br />
Studied Network: 5-layer MLP with 512 hidden units<br />
<br />
<br />
(2) ImageNet classification: 1K image classes, 1.28M training images and 50K validation images<br />
<br />
Relational Graph: Due to high computational cost, 52 graphs are uniformly sampled from the 3942 available graphs.<br />
<br />
Studied Network: <br />
*ResNet-34, which only consists of basic blocks of 3×3 convolutions (He et al., 2016)<br />
<br />
*ResNet-34-sep, a variant where we replace all 3×3 dense convolutions in ResNet-34 with 3×3 separable convolutions (Chollet, 2017)<br />
<br />
*ResNet-50, which consists of bottleneck blocks (He et al., 2016) of 1×1, 3×3, 1×1 convolutions<br />
<br />
*EfficientNet-B0 architecture (Tan & Le, 2019)<br />
<br />
*8-layer CNN with 3×3 convolution<br />
<br />
= Results and Discussions =<br />
<br />
The paper summarizes the result of the experiment among multiple different relational graphs through sampling and analyzing and list six important observations during the experiments, These are:<br />
<br />
* There always exists a graph structure that has higher predictive accuracy under Top-1 error compared to the complete graph<br />
<br />
* There is a sweet spot such that the graph structure near the sweet spot usually outperforms the base graph<br />
<br />
* The predictive accuracy under top-1 error can be represented by a smooth function of Average Path Length <math> (L) </math> and Clustering Coefficient <math> (C) </math><br />
<br />
* The Experiments are consistent across multiple datasets and multiple graph structures with similar Average Path Length and Clustering Coefficient.<br />
<br />
* The best graph structure can be identified easily.<br />
<br />
* There are similarities between the best artificial neurons and biological neurons.<br />
<br />
----<br />
<br />
<br />
<br />
[[File:Result2_441_2020Group16.png]]<br />
<br />
$$\text{Figure - Results from Experiments}$$<br />
<br />
== Neural networks performance depends on its structure ==<br />
During the experiment, Top-1 errors for all sampled relational graph among multiple tasks and graph structures are recorded. The parameters of the models are average path length and clustering coefficient. Heat maps were created to illustrate the difference in predictive performance among possible average path length and clustering coefficient. In '''Figure - Results from Experiments (a)(c)(f)''', The darker area represents a smaller top-1 error which indicates the model performs better than the light area.<br />
<br />
Compared to the complete graph which has parameter <math> L = 1 </math> and <math> C = 1 </math>, the best performing relational graph can outperform the complete graph baseline by 1.4% top-1 error for MLP on CIFAR-10, and 0.5% to 1.2% for models on ImageNet. Hence it is an indicator that the predictive performance of the neural networks highly depends on the graph structure, or equivalently that the completed graph does not always have the best performance.<br />
<br />
== Sweet spot where performance is significantly improved ==<br />
It had been recognized that training noises often results in inconsistent predictive results. In the paper, the 3942 graphs in the sample had been grouped into 52 bins, each bin had been colored based on the average performance of graphs that fall into the bin. By taking the average, the training noises had been significantly reduced. Based on the heat map '''Figure - Results from Experiments (f)''', the well-performing graphs tend to cluster into a special spot that the paper called “sweet spot” shown in the red rectangle, the rectangle is approximately included clustering coefficient in the range <math>[0.1,0.7]</math> and average path length within <math>[1.5,3]</math>.<br />
<br />
== Relationship between neural network’s performance and parameters == <br />
When we visualize the heat map, we can see that there is no significant jump of performance that occurred as a small change of clustering coefficient and average path length ('''Figure - Results from Experiments (a)(c)(f)'''). In addition, if one of the variables is fixed in a small range, it is observed that a second-degree polynomial is a good visualization tool for the overall trend ('''Figure - Results from Experiments (b)(d)'''). Therefore, both the clustering coefficient and average path length are highly related to neural network performance by a U-shape. <br />
<br />
== Consistency among many different tasks and datasets ==<br />
They observe that relational graphs with certain graph measures may consistently perform well regardless of how they are instantiated. The paper presents consistency uses two perspectives, one is qualitative consistency and another one is quantitative consistency.<br />
<br />
(1) '''Qualitative Consistency'''<br />
It is observed that the results are consistent from different points of view. Among multiple architecture dataset, it is observed that the clustering coefficient within <math>[0.1,0.7]</math> and average path length within <math>[1.5,3]</math> consistently outperform the baseline complete graph. <br />
<br />
(2) '''Quantitative Consistency'''<br />
Among different dataset with the network that has similar clustering coefficient and average path length, the results are correlated, The paper mentioned that ResNet-34 is much more complex than 5-layer MLP but a fixed set relational graph would perform similarly in both settings, with Pearson correlation of <math>0.658</math>, the p-value for the Null hypothesis is less than <math>10^{-8}</math>.<br />
<br />
== Top architectures can be identified efficiently ==<br />
The computation cost of finding top architectures can be significantly reduced without training the entire data set for a large value of epoch or a relatively large sample. To achieve the top architectures, the number of graphs and training epochs need to be identified. For the number of graphs, a heatmap is a great tool to demonstrate the result. In the 5-layer MLP on CIFAR-10 example, taking a sample of the data around 52 graphs would have a correlation of 0.9, which indicates that fewer samples are needed for a similar analysis in practice. When determining the number of epochs, correlation can help to show the result. In ResNet34 on ImageNet example, the correlation between the variables is already high enough for future computation within 3 epochs. This means that good relational graphs perform well even at the<br />
initial training epochs.<br />
<br />
== Well-performing neural networks have graph structure surprisingly similar to those of real biological neural networks==<br />
The way we define relational graphs and average length in the graph is similar to the way information is exchanged in network science. The biological neural network also has a similar relational graph representation and graph measure with the best-performing relational graph.<br />
<br />
While there is some organizational similarity between a computational neural network and a biological neural network, we should refrain from saying that both these networks share many similarities or are essentially the same with just different substrates. The biological neurons are still quite poorly understood and it may take a while before their mechanisms are better understood.<br />
<br />
= Conclusions=<br />
Our works provide a new viewpoint about combining the fields of graph neural networks (GNNs) and general architecture design by establishing graph structures. In particular, GNNs are instances of general neural architectures where graph structures are seen as input rather than part of the architecture and message functions are shared across all edges of the graph. We found that the graph theories and techniques implemented in other disciplines are able to provide a good reference for understanding and designing the structures and functions of neural networks. This could provide effective help and inspiration for the study of neural networks.<br />
<br />
= Critique =<br />
<br />
1. The experiment is only measuring on a single data set which might not be representative enough. As we can see in the whole paper, the "sweet spot" we talked about might be a special feature for the given data set only which is the CIFAR-10 data set. If we change the data set to another imaging data set like CK+, whether we are going to get a similar result is not shown by the paper. Hence, the result that is being concluded from the paper might not be representative enough. <br />
<br />
2. When we are fitting the model in practice, we will fit the model with more than one epoch. The order of the model fitting should be randomized since we should create more random jumps to avoid staked inside a local minimum. With the same order within each epoch, the data might be grouped by different classes or levels, the model might result in a better performance with certain classes and worse performance with other classes. In this particular example, without randomization of the training data, the conclusion might not be precise enough.<br />
<br />
3. This study shows empirical justification for choosing well-performing models from graphs differing only by average path length and clustering coefficient. An equally important question is whether there is a theoretical justification for why these graph properties may (or may not) contribute to the performance of a general classifier - for example, is there a combination of these properties that is sufficient to recover the universality theorem for MLP's?<br />
<br />
4. It might be worth looking into how to identify the "sweet spot" for different datasets.<br />
<br />
5. What would be considered a "best graph structure " in the discussion and conclusion part? It seems that the intermediate result of getting an accurate result was by binning graphs into smaller bins, what should we do if the graphs can not be binned into significantly smaller bins in order to proceed with the methodologies mentioned in the paper. Both CIFAR - 10 and ImageNet seem too trivial considering the amount of variation and categories in the dataset. What would the generalizability be to other presentations of images?<br />
<br />
6. There is an interesting insight that the idea of the relational graph is kind of similar to applying causal graphs in neuro networks, which is also closer to biology and neuroscience because human beings learning things based on causality. This new approach may lead to higher prediction accuracy but it needs more assumptions, such as correct relations and causalities.<br />
<br />
7. This is an interesting topic that uses the knowledge in graph theory to introduce this new structure of Neural Networks. Using more data sets to discuss this approach might be more interesting, such as the MNIST dataset. We think it is interesting to discuss whether this structure will provide a better performance compare to the "traditional" structure of NN in any type of Neural Networks.</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=Graph_Structure_of_Neural_Networks&diff=47567Graph Structure of Neural Networks2020-11-29T03:00:55Z<p>Iaoellme: /* Results and Discussions */</p>
<hr />
<div>= Presented By =<br />
<br />
Xiaolan Xu, Robin Wen, Yue Weng, Beizhen Chang<br />
<br />
= Introduction =<br />
<br />
A deep neural network is composed of neurons organized into layers and the connections between them. The architecture of a neural network can be captured by its "computational graph", where neurons are represented as nodes, and directed edges link neurons in different layers. This graphical representation demonstrates how the network transmits and transforms information through its input neurons through the hidden layers and ultimately to the output neurons.<br />
<br />
In Neural Network research, it is often important to build a relation between a neural network’s accuracy and its underlying graph structure. A natural choice is to use computational graph representation, but this has many limitations including a lack of generality and disconnection with biology/neuroscience. This disconnection between biology/neuroscience makes knowledge less transferable and interdisciplinary research more difficult.<br />
<br />
Thus, the authors developed a new way of representing a neural network as a graph, called a relational graph. The key insight in the new representation is to focus on message exchange, rather than just on directed data flow. For example, for a fixed-width fully-connected layer, an input channel and output channel pair can be represented as a single node, while an edge in the relational graph can represent the message exchange between the two nodes. Under this formulation, using the appropriate message exchange definition, it can be shown that the relational graph can represent many types of neural network layers.<br />
<br />
WS-flex is a graph generator that allows systematically exploring the design space of neural networks. Neural networks are characterized by the clustering coefficient and average path length of their relational graphs under the insights of neuroscience.<br />
<br />
= Neural Network as Relational Graph =<br />
<br />
The author proposes the concept of relational graph to study the graphical structure of neural network. Each relational graph is based on an undirected graph <math>G =(V; E)</math>, where <math>V =\{v_1,...,v_n\}</math> is the set of all the nodes, and <math>E \subseteq \{(v_i,v_j)|v_i,v_j\in V\}</math> is the set of all edges that connect nodes. Note that for the graph used here, all nodes have self edges, that is <math>(v_i,v_i)\in E</math>. <br />
<br />
To build a relational graph that captures the message exchange between neurons in the network, we associate various mathematical quantities to the graph <math>G</math>. First, a feature quantity <math>x_v</math> is associated with each node. The quantity <math>x_v</math> might be a scalar, vector or tensor depending on different types of neural networks (see the Table at the end of the section). Then a message function <math>f_{uv}(·)</math> is associated with every edge in the graph. A message function specifically takes a node’s feature as the input and then output a message. An aggregation function <math>{\rm AGG}_v(·)</math> then takes a set of messages (the outputs of message function) and outputs the updated node feature. <br />
<br />
A relation graph is a graph <math>G</math> associated with several rounds of message exchange, which transform the feature quantity <math>x_v</math> with the message function <math>f_{uv}(·)</math> and the aggregation function <math>{\rm AGG}_v(·)</math>. At each round of message exchange, each node sends messages to its neighbors and aggregates incoming messages from its neighbors. Each message is transformed at each edge through the message function, then they are aggregated at each node via the aggregation function. Suppose we have already conducted <math>r-1</math> rounds of message exchange, then the <math>r^{th}</math> round of message exchange for a node <math>v</math> can be described as<br />
<br />
<div style="text-align:center;"><math>\mathbf{x}_v^{(r+1)}= {\rm AGG}^{(r)}(\{f_v^{(r)}(\textbf{x}_u^{(r)}), \forall u\in N(v)\})</math></div> <br />
<br />
where <math>\mathbf{x}^{(r+1)}</math> is the feature of the <math>v</math> node in the relational graph after the <math>r^{th}</math> round of update. <math>u,v</math> are nodes in Graph <math>G</math>. <math>N(u)=\{u|(u,v)\in E\}</math> is the set of all the neighbor nodes of <math>u</math> in graph <math>G</math>.<br />
<br />
To further illustrate the above, we use the basic Multilayer Perceptron (MLP) as an example. An MLP consists of layers of neurons, where each neuron performs a weighted sum over scalar inputs and outputs, followed by some non-linearity. Suppose the <math>r^{th}</math> layer of an MLP takes <math>x^{(r)}</math> as input and <math>x^{(r+1)}</math> as output, then a neuron computes <br />
<br />
<div style="text-align:center;"><math>x_i^{(r+1)}= \sigma(\Sigma_jw_{ij}^{(r)}x_j^{(r)})</math>.</div> <br />
<br />
where <math>w_{ij}^{(r)}</math> is the trainable weight and <math>\sigma</math> is the non-linearity function. Let's first consider the special case where the input and output of all the layers <math>x^{(r)}</math>, <math>1 \leq r \leq R </math> have the same feature dimensions <math>d</math>. In this scenario, we can have <math>d</math> nodes in the Graph <math>G</math> with each node representing a neuron in MLP. Each layer of neural network will correspond with a round of message exchange, so there will be <math>R</math> rounds of message exchange in total. The aggregation function here will be the summation with non-linearity transform <math>\sigma(\Sigma)</math>, while the message function is simply the scalar multipication with weight. A fully-connected, fixed-width MLP layer can then be expressed with a complete relational graph, where each node <math>x_v</math> connects to all the other nodes in <math>G</math>, that is neighborhood set <math>N(v) = V</math> for each node <math>v</math>. The figure below shows the correspondence between the complete relation graph with a 5-layer 4-dimension fully-connected MLP.<br />
<br />
<div style="text-align:center;">[[File:fully_connnected_MLP.png]]</div><br />
<br />
In fact, a fixed-width fully-connected MLP is only a special case under a much more general model family, where the message function, aggregation function, and most importantly, the relation graph structure can vary. The different relational graph will represent the different topological structure and information exchange pattern of the network, which is the property that the paper wants to examine. The plot below shows two examples of non-fully connected fixed-width MLP and their corresponding relational graphs. <br />
<br />
<div style="text-align:center;">[[File:otherMLP.png]]</div><br />
<br />
We can generalize the above definitions for fixed-width MLP to Variable-width MLP, Convolutional Neural Network (CNN), and other modern network architecture like Resnet by allowing the node feature quantity <math>\textbf{x}_j^{(r)}</math> to be a vector or tensor respectively. In this case, each node in the relational graph will represent multiple neurons in the network, and the number of neurons contained in each node at each round of message exchange does not need to be the same, which gives us a flexible representation of different neural network architecture. The message function will then change from the simple scalar multiplication to either matrix/tensor multiplication or convolution. And the weight matrix will change to the convolutional filter. The representation of these more complicated networks are described in details in the paper, and the correspondence between different networks and their relational graph properties is summarized in the table below. <br />
<br />
<div style="text-align:center;">[[File:relational_specification.png]]</div><br />
<br />
Overall, relational graphs provide a general representation for neural networks. With proper definitions of node features and message exchange, relational graphs can represent diverse neural architectures, thereby allowing us to study the performance of different graph structures.<br />
<br />
= Exploring and Generating Relational Graphs=<br />
<br />
We will deal with the design and how to explore the space of relational graphs in this section. There are three parts we need to consider:<br />
<br />
(1) '''Graph measures''' that characterize graph structural properties:<br />
<br />
We will use one global graph measure, average path length, and one local graph measure, clustering coefficient in this paper.<br />
To explain clearly, average path length measures the average shortest path distance between any pair of nodes; the clustering coefficient measures the proportion of edges between the nodes within a given node’s neighborhood, divided by the number of edges that could possibly exist between them, averaged over all the nodes.<br />
<br />
(2) '''Graph generators''' that can generate the diverse graph:<br />
<br />
With selected graph measures, we use a graph generator to generate diverse graphs to cover a large span of graph measures. To figure out the limitation of the graph generator and find out the best, we investigate some generators including ER, WS, BA, Harary, Ring, Complete graph and results shows as below:<br />
<br />
<div style="text-align:center;">[[File:3.2 graph generator.png]]</div><br />
<br />
Thus, from the picture, we could obtain the WS-flex graph generator that can generate graphs with a wide coverage of graph measures; notably, WS-flex graphs almost encompass all the graphs generated by classic random generators mentioned above.<br />
<br />
(3) '''Computational Budget''' that we need to control so that the differences in performance of different neural networks are due to their diverse relational graph structures.<br />
<br />
It is important to ensure that all networks have approximately the same complexities so that the differences in performance are due to their relational graph structures when comparing neutral work by their diverse graph.<br />
<br />
We use FLOPS (# of multiply-adds) as the metric. We first compute the FLOPS of our baseline network instantiations (i.e., complete relational graph) and use them as the reference complexity in each experiment. From the description in section 2, a relational graph structure can be instantiated as a neural network with variable width. Therefore, we can adjust the width of a neural network to match the reference complexity without changing the relational graph structures.<br />
<br />
= Experimental Setup =<br />
The author studied the performance of 3942 sampled relational graphs (generated by WS-flex from the last section) of 64 nodes with two experiments: <br />
<br />
(1) CIFAR-10 dataset: 10 classes, 50K training images, and 10K validation images<br />
<br />
Relational Graph: all 3942 sampled relational graphs of 64 nodes<br />
<br />
Studied Network: 5-layer MLP with 512 hidden units<br />
<br />
<br />
(2) ImageNet classification: 1K image classes, 1.28M training images and 50K validation images<br />
<br />
Relational Graph: Due to high computational cost, 52 graphs are uniformly sampled from the 3942 available graphs.<br />
<br />
Studied Network: <br />
*ResNet-34, which only consists of basic blocks of 3×3 convolutions (He et al., 2016)<br />
<br />
*ResNet-34-sep, a variant where we replace all 3×3 dense convolutions in ResNet-34 with 3×3 separable convolutions (Chollet, 2017)<br />
<br />
*ResNet-50, which consists of bottleneck blocks (He et al., 2016) of 1×1, 3×3, 1×1 convolutions<br />
<br />
*EfficientNet-B0 architecture (Tan & Le, 2019)<br />
<br />
*8-layer CNN with 3×3 convolution<br />
<br />
= Results and Discussions =<br />
<br />
The paper summarizes the result of the experiment among multiple different relational graphs through sampling and analyzing and list six important observations during the experiments, These are:<br />
<br />
* There always exists a graph structure that has higher predictive accuracy under Top-1 error compared to the complete graph<br />
<br />
* There is a sweet spot such that the graph structure near the sweet spot usually outperforms the base graph<br />
<br />
* The predictive accuracy under top-1 error can be represented by a smooth function of Average Path Length <math> (L) </math> and Clustering Coefficient <math> (C) </math><br />
<br />
* The Experiments are consistent across multiple datasets and multiple graph structures with similar Average Path Length and Clustering Coefficient.<br />
<br />
* The best graph structure can be identified easily.<br />
<br />
* There are similarities between the best artificial neurons and biological neurons.<br />
<br />
----<br />
<br />
<br />
<br />
[[File:Result2_441_2020Group16.png]]<br />
<br />
$$\text{Figure - Results from Experiments}$$<br />
<br />
== Neural networks performance depends on its structure ==<br />
During the experiment, Top-1 errors for all sampled relational graph among multiple tasks and graph structures are recorded. The parameters of the models are average path length and clustering coefficient. Heat maps were created to illustrate the difference in predictive performance among possible average path length and clustering coefficient. In '''Figure - Results from Experiments (a)(c)(f)''', The darker area represents a smaller top-1 error which indicates the model performs better than the light area.<br />
<br />
Compared to the complete graph which has parameter <math> L = 1 </math> and <math> C = 1 </math>, the best performing relational graph can outperform the complete graph baseline by 1.4% top-1 error for MLP on CIFAR-10, and 0.5% to 1.2% for models on ImageNet. Hence it is an indicator that the predictive performance of the neural networks highly depends on the graph structure, or equivalently that the completed graph does not always have the best performance.<br />
<br />
== Sweet spot where performance is significantly improved ==<br />
It had been recognized that training noises often results in inconsistent predictive results. In the paper, the 3942 graphs in the sample had been grouped into 52 bins, each bin had been colored based on the average performance of graphs that fall into the bin. By taking the average, the training noises had been significantly reduced. Based on the heat map '''Figure - Results from Experiments (f)''', the well-performing graphs tend to cluster into a special spot that the paper called “sweet spot” shown in the red rectangle, the rectangle is approximately included clustering coefficient in the range <math>[0.1,0.7]</math> and average path length within <math>[1.5,3]</math>.<br />
<br />
== Relationship between neural network’s performance and parameters == <br />
When we visualize the heat map, we can see that there is no significant jump of performance that occurred as a small change of clustering coefficient and average path length ('''Figure - Results from Experiments (a)(c)(f)'''). In addition, if one of the variables is fixed in a small range, it is observed that a second-degree polynomial is a good visualization tool for the overall trend ('''Figure - Results from Experiments (b)(d)'''). Therefore, both the clustering coefficient and average path length are highly related to neural network performance by a U-shape. <br />
<br />
== Consistency among many different tasks and datasets ==<br />
They observe that relational graphs with certain graph measures may consistently perform well regardless of how they are instantiated. The paper presents consistency uses two perspectives, one is qualitative consistency and another one is quantitative consistency.<br />
<br />
(1) '''Qualitative Consistency'''<br />
It is observed that the results are consistent from different points of view. Among multiple architecture dataset, it is observed that the clustering coefficient within <math>[0.1,0.7]</math> and average path length within <math>[1.5,3]</math> consistently outperform the baseline complete graph. <br />
<br />
(2) '''Quantitative Consistency'''<br />
Among different dataset with the network that has similar clustering coefficient and average path length, the results are correlated, The paper mentioned that ResNet-34 is much more complex than 5-layer MLP but a fixed set relational graph would perform similarly in both settings, with Pearson correlation of <math>0.658</math>, the p-value for the Null hypothesis is less than <math>10^{-8}</math>.<br />
<br />
== Top architectures can be identified efficiently ==<br />
The computation cost of finding top architectures can be significantly reduced without training the entire data set for a large value of epoch or a relatively large sample. To achieve the top architectures, the number of graphs and training epochs need to be identified. For the number of graphs, a heatmap is a great tool to demonstrate the result. In the 5-layer MLP on CIFAR-10 example, taking a sample of the data around 52 graphs would have a correlation of 0.9, which indicates that fewer samples are needed for a similar analysis in practice. When determining the number of epochs, correlation can help to show the result. In ResNet34 on ImageNet example, the correlation between the variables is already high enough for future computation within 3 epochs. This means that good relational graphs perform well even at the<br />
initial training epochs.<br />
<br />
== Well-performing neural networks have graph structure surprisingly similar to those of real biological neural networks==<br />
The way we define relational graphs and average length in the graph is similar to the way information is exchanged in network science. The biological neural network also has a similar relational graph representation and graph measure with the best-performing relational graph.<br />
<br />
While there is some organizational similarity between a computational neural network and a biological neural network, we should refrain from saying that both these networks share many similarities or are essentially the same with just different substrates. The biological neurons are still quite poorly understood and it may take a while before their mechanisms are better understood.<br />
<br />
= Conclusions=<br />
Our works provide a new viewpoint about combining the fields GNNs and general architecture design bu establishing graph structures. We found that the graph theories and techniques implemented in other disciplines are able to provide a good reference for understanding and designing the structures and functions of neural networks. This could provide effective help and inspiration for the study of neural networks.<br />
<br />
= Critique =<br />
<br />
1. The experiment is only measuring on a single data set which might not be representative enough. As we can see in the whole paper, the "sweet spot" we talked about might be a special feature for the given data set only which is the CIFAR-10 data set. If we change the data set to another imaging data set like CK+, whether we are going to get a similar result is not shown by the paper. Hence, the result that is being concluded from the paper might not be representative enough. <br />
<br />
2. When we are fitting the model in practice, we will fit the model with more than one epoch. The order of the model fitting should be randomized since we should create more random jumps to avoid staked inside a local minimum. With the same order within each epoch, the data might be grouped by different classes or levels, the model might result in a better performance with certain classes and worse performance with other classes. In this particular example, without randomization of the training data, the conclusion might not be precise enough.<br />
<br />
3. This study shows empirical justification for choosing well-performing models from graphs differing only by average path length and clustering coefficient. An equally important question is whether there is a theoretical justification for why these graph properties may (or may not) contribute to the performance of a general classifier - for example, is there a combination of these properties that is sufficient to recover the universality theorem for MLP's?<br />
<br />
4. It might be worth looking into how to identify the "sweet spot" for different datasets.<br />
<br />
5. What would be considered a "best graph structure " in the discussion and conclusion part? It seems that the intermediate result of getting an accurate result was by binning graphs into smaller bins, what should we do if the graphs can not be binned into significantly smaller bins in order to proceed with the methodologies mentioned in the paper. Both CIFAR - 10 and ImageNet seem too trivial considering the amount of variation and categories in the dataset. What would the generalizability be to other presentations of images?<br />
<br />
6. There is an interesting insight that the idea of the relational graph is kind of similar to applying causal graphs in neuro networks, which is also closer to biology and neuroscience because human beings learning things based on causality. This new approach may lead to higher prediction accuracy but it needs more assumptions, such as correct relations and causalities.<br />
<br />
7. This is an interesting topic that uses the knowledge in graph theory to introduce this new structure of Neural Networks. Using more data sets to discuss this approach might be more interesting, such as the MNIST dataset. We think it is interesting to discuss whether this structure will provide a better performance compare to the "traditional" structure of NN in any type of Neural Networks.</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=A_universal_SNP_and_small-indel_variant_caller_using_deep_neural_networks&diff=47333A universal SNP and small-indel variant caller using deep neural networks2020-11-28T18:36:41Z<p>Iaoellme: </p>
<hr />
<div>== Background ==<br />
<br />
<br />
Biological functions are determined by genes, and differences in function are determined by mutants, or alleles, of those genes. Determining novel alleles is very important in understanding the genetic variation within a species. For example, most eye colours are determined by different alleles of the gene OCA2. All animals receive one copy of each gene from each of their parents. Mutations of a gene are classified as either homozygous (both copies are the same) or heterozygous (the two copies are different).<br />
<br />
Next generation sequencing is a very popular technique for sequencing, or reading, DNA. Since all genes are encoded as DNA, sequencing is an essential tool for understanding genes. Next generation sequencing works by reading short sections of DNA of length k, called k-mers, and then piecing them together or aligning them to a reference genome. Next generation sequencing is relatively fast and inexpensive, although can randomly misidentify some nucleotides, introducing errors.<br />
<br />
The process of variant calling is determining novel alleles from sequencing data (typically next generation sequencing data). Some significant alleles only differ from the “standard” version of a gene by only a single base pair, such as the mutation which causes multiple sclerosis. Therefore it is important to be able to accurately call single nucleotide swaps/polymorphisms (SNPs), insertions, and deletions (indels). Calling SNPs and small indels is technically challenging, since it requires a program to be able to distinguish between truly novel mutations and errors in the sequencing data.<br />
<br />
This paper aims to solve the problem of calling SNPs and small indels using a convolutional neural net by casting the reads as images, and classifying whether or not they contain a mutation.<br />
<br />
== Overview ==<br />
<br />
In Figure 1, DeepVariant workflow overview is illustrated.<br />
<br />
Figure 1. In all panels, blue boxes represent data and red boxes are processes.<br />
<br />
Initially, the NGS reads aligned to a reference genome are scanned for candidate variants which are different sites from the reference genome. The read and reference data are encoded as an image for each candidate variant site. Then, trained CNN can compute the genotype likelihoods, (heterozygous or homozygous) for each of the candidate variants (figure1, left box). <br />
To train the CNN for image classification purposes, the DeepVariant machinery makes pileup images for a labeled sample with known genotypes. These labeled images and known genotypes are provided to CNN for training, and stochastic gradient descent algorithm is used to optimize the CNN parameters to maximize genotype prediction accuracy. After the convergence of the model, the final model is frozen to use for calling mutations for other image classification tests (figure1, middle box).<br />
For example, in figure 1 (right box), the reference and read bases are encoded into a pileup image at a candidate variant site. CNN using this encoded image computes the genotype likelihoods for the three diploid genotype states of homozygous reference (hom-ref), heterozygous (het) or homozygous alternate (hom-alt). In this example a heterozygous variant call is emitted, as the most probable genotype here is “het”. <br />
<br />
== Preprocessing ==<br />
<br />
Before the sequencing reads can be fed into the classifier, they must be preprocessed. There are many pre-processing steps that are necessary for this algorithm. These steps represent the real novelty in this technique, by transforming the data in a way that allows us to use more common neural network architectures for classification. The preprocessing of the data can be broken into three main phases: the realignment of reads, finding candidate variants, and creating images of the candidate variants. <br />
<br />
The realignment of the reads phase of the preprocessing is important in order to ensure the sequences can be properly compared to the reference sequences. First, the sequences are aligned to a reference sequence. Reads that align poorly are grouped with other reads around them to build that section, or haplotype, from scratch. If there is strong evidence that the new version of the haplotype fits the reads well, the reads are re-aligned to it. This process updates the CIGAR (Compact Idiosyncratic Gapped Alignment Report) string, a way to represent the alignment of a sequence to a reference, for each read.<br />
<br />
Once the reads are properly aligned, the algorithm then proceeds to finding candidate variants, regions in the DNA sequence that may contain variants. It is these candidate variants that will eventually be passed as input to the neural network. To find these, we need to consider each position in the reference sequence independently. Any unusable reads are filtered at this point. This includes reads that are not aligned properly, ones that are marked as duplicates, those that fail vendor quality checks, or whose mapping quality is less than ten. For each site in the genome, we collect all the remaining reads that overlap that site. The corresponding allele aligned to that site is then determined by decoding the CIGAR string, that was updated in the realignment phase, of each read. The alleles are then classified into one of four categories: reference-matching base, reference-mismatching base, insertion with a specific sequence, or deletion with a specific length, and the number of occurrences of each distinct allele across all reads is counted. Read bases are only included as potential alleles if each base in the allele has a quality score of at least 10.<br />
<br />
With candidate variants identified, the last phase of pre-processing is to convert these candidate variants into images representing the data. This allows for the use of well established convolutional neural networks for image classification for this specialized problem. Each colour channel is used to store a different piece of information about a candidate variant. The red channel encodes which base we have (A, G, C, or T), by mapping each base to a particular value. The quality of the read is mapped to the green colour channel. And finally, the blue channel encodes whether or not the reference is on the positive strand of the DNA. Each row of the image represents a read, and each column represents a particular base in that read. The reference strand is repeated for the first five rows of the encoded image, in order to maintain its information after a 5x5 convolution is applied.<br />
With the data preprocessing complete, the images can then be passed into the neural network for classification.<br />
<br />
== Neural Network ==<br />
<br />
The neural network used is a convolutional neural network. Although the full network architecture is not revealed in the paper, there are several details which we can discuss. The architecture of the network is an input layer attached to an adapted Inception v2 ImageNet model with nine partitions. The input layer takes as input the images representing the candidate variants and rescales them to 299x299 pixels. The output layer is a three class Softmax layer initialized with Gaussian random weights with standard deviation 0.001. This final layer is fully connected to the previous layer. The three classes are: homozygous reference (meaning it is not a variant), heterozygous variant, and homozygous variant. The candidate variant is classified into the class with the highest probability. The model is trained using stochastic gradient descent with a weight decay of 0.00004. The training was done in mini-batches, each with 32 images, using a root mean squared (RMS) decay of 0.9. For the multiple sequencing technologies experiments, a single model was trained with a learning rate of 0.0015 and momentum 0.8 for 250,000 update steps. For all other experiments, multiple models were trained, and the one with the highest accuracy on the training set was chosen as the final model. The multiple models stem from using each combination of the possible parameter values for the learning rate (0.00095, 0.001, 0.0015) and momentum (0.8, 0.85, 0.9). These models were trained for 80 hours, or until the training accuracy converged.<br />
<br />
== Results ==<br />
<br />
DeepVariant was trained using data available from the CEPH (Centre d’Etude du Polymorphisme Humain) female sample NA12878 and was evaluated on the unseen Ashkenazi male sample NA24385. The results were compared with other most commonly used bioinformatics methods, such as the GATK, FreeBayes22, SAMtools23, 16GT24 and Strelka25 (Table 1). For better comparison, the overall accuracy (F1), recall, precision, and numbers of true positives (TP), false negatives (FN) and false positives (FP) are illustrated over the whole genome.<br />
<br />
DeepVariant showed highest accuracy and more than 50% fewer errors per genome compared to the next best algorithm. <br />
<br />
They also evaluated the same set of algorithms using the synthetic diploid sample CHM1-CHM1326 (Table 2). Results illustrated that, DeepVariant method outperformed all other algorithms for variant calling (SNP and indel) and showed the highest accuracy in terms of F1, Recall, precision and TP.<br />
<br />
== Conclusion ==<br />
<br />
DeepVariant’s strong performance on human data proves that deep learning is a promising technique for variant calling. Perhaps the most exciting feature of DeepVariant is its simplicity. Unlike other state of the art variant callers, DeepVariant has no knowledge of the sequencing technologies that create the reads, or even the biological processes that introduce mutations. This simplifies the problem of variant calling to preprocessing the reads and training a generic deep learning model. It also suggests that DeepVariant could be significantly improved by tailoring the preprocessing to specific sequencing technologies and/or developing a dedicated CNN architecture for the reads, rather than trying to cast them as images.</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=stat441F21&diff=46863stat441F212020-11-26T22:21:32Z<p>Iaoellme: /* Paper presentation */</p>
<hr />
<div><br />
<br />
== [[F20-STAT 441/841 CM 763-Proposal| Project Proposal ]] ==<br />
<br />
<!--[https://goo.gl/forms/apurag4dr9kSR76X2 Your feedback on presentations]--><br />
<br />
= Record your contributions here [https://docs.google.com/spreadsheets/d/10CHiJpAylR6kB9QLqN7lZHN79D9YEEW6CDTH27eAhbQ/edit?usp=sharing]=<br />
<br />
Use the following notations:<br />
<br />
P: You have written a summary/critique on the paper.<br />
<br />
T: You had a technical contribution on a paper (excluding the paper that you present).<br />
<br />
E: You had an editorial contribution on a paper (excluding the paper that you present).<br />
<br />
=Paper presentation=<br />
{| class="wikitable"<br />
<br />
{| border="1" cellpadding="3"<br />
|-<br />
|width="60pt"|Date<br />
|width="250pt"|Name <br />
|width="15pt"|Paper number <br />
|width="700pt"|Title<br />
|width="15pt"|Link to the paper<br />
|width="30pt"|Link to the summary<br />
|width="30pt"|Link to the video<br />
|-<br />
|Sep 15 (example)||Ri Wang || ||Sequence to sequence learning with neural networks.||[http://papers.nips.cc/paper/5346-sequence-to-sequence-learning-with-neural-networks.pdf Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Going_Deeper_with_Convolutions Summary] || [https://youtu.be/JWozRg_X-Vg?list=PLehuLRPyt1HzXDemu7K4ETcF0Ld_B5adG&t=539]<br />
|-<br />
|Week of Nov 16 ||Sharman Bharat, Li Dylan,Lu Leonie, Li Mingdao || 1|| Risk prediction in life insurance industry using supervised learning algorithms || [https://rdcu.be/b780J Paper] ||[https://wiki.math.uwaterloo.ca/statwiki/index.php?title=User:Bsharman Summary] ||<br />
[https://www.youtube.com/watch?v=TVLpSFYgF0c&feature=youtu.be]<br />
|-<br />
|Week of Nov 16 || Delaney Smith, Mohammad Assem Mahmoud || 2|| Influenza Forecasting Framework based on Gaussian Processes || [https://proceedings.icml.cc/static/paper_files/icml/2020/1239-Paper.pdf Paper] ||[https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Influenza_Forecasting_Framework_based_on_Gaussian_Processes Summary]|| [https://www.youtube.com/watch?v=HZG9RAHhpXc&feature=youtu.be]<br />
|-<br />
|Week of Nov 16 || Tatianna Krikella, Swaleh Hussain, Grace Tompkins || 3|| Processing of Missing Data by Neural Networks || [http://papers.nips.cc/paper/7537-processing-of-missing-data-by-neural-networks.pdf Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=User:Gtompkin Summary] || [https://learn.uwaterloo.ca/d2l/ext/rp/577051/lti/framedlaunch/6ec1ebea-5547-46a2-9e4f-e3dc9d79fd54]<br />
|-<br />
|Week of Nov 16 ||Jonathan Chow, Nyle Dharani, Ildar Nasirov ||4 ||Streaming Bayesian Inference for Crowdsourced Classification ||[https://papers.nips.cc/paper/9439-streaming-bayesian-inference-for-crowdsourced-classification.pdf Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Streaming_Bayesian_Inference_for_Crowdsourced_Classification Summary] || [https://www.youtube.com/watch?v=UgVRzi9Ubws]<br />
|-<br />
|Week of Nov 16 || Matthew Hall, Johnathan Chalaturnyk || 5|| Neural Ordinary Differential Equations || [https://papers.nips.cc/paper/7892-neural-ordinary-differential-equations.pdf] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Neural_ODEs Summary]||<br />
|-<br />
|Week of Nov 16 || Luwen Chang, Qingyang Yu, Tao Kong, Tianrong Sun || 6|| Adversarial Attacks on Copyright Detection Systems || Paper [https://proceedings.icml.cc/static/paper_files/icml/2020/1894-Paper.pdf] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Adversarial_Attacks_on_Copyright_Detection_Systems Summary] || [https://www.youtube.com/watch?v=bQI9S6bCo8o]<br />
|-<br />
|Week of Nov 16 || Casey De Vera, Solaiman Jawad || 7|| IPBoost – Non-Convex Boosting via Integer Programming || [https://arxiv.org/pdf/2002.04679.pdf Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=IPBoost Summary] || [https://www.youtube.com/watch?v=4DhJDGC4pyI&feature=youtu.be]<br />
|-<br />
|Week of Nov 16 || Yuxin Wang, Evan Peters, Yifan Mou, Sangeeth Kalaichanthiran || 8|| What Game Are We Playing? End-to-end Learning in Normal and Extensive Form Games || [https://arxiv.org/pdf/1805.02777.pdf] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=what_game_are_we_playing Summary] || [https://www.youtube.com/watch?v=9qJoVxo3hnI&feature=youtu.be]<br />
|-<br />
|Week of Nov 16 || Yuchuan Wu || 9|| || || ||<br />
|-<br />
|Week of Nov 16 || Zhou Zeping, Siqi Li, Yuqin Fang, Fu Rao || 10|| A survey of neural network-based cancer prediction models from microarray data || [https://www.sciencedirect.com/science/article/pii/S0933365717305067 Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=User:Y93fang Summary] || [https://youtu.be/B8pPUU8ypO0]<br />
|-<br />
|Week of Nov 23 ||Jinjiang Lian, Jiawen Hou, Yisheng Zhu, Mingzhe Huang || 11|| DROCC: Deep Robust One-Class Classification || [https://proceedings.icml.cc/static/paper_files/icml/2020/6556-Paper.pdf paper] ||[https://wiki.math.uwaterloo.ca/statwiki/index.php?title=User:J46hou Summary] || [https://www.youtube.com/watch?v=tvCEvvy54X8&ab_channel=JJLian]<br />
|-<br />
|Week of Nov 23 || Bushra Haque, Hayden Jones, Michael Leung, Cristian Mustatea || 12|| Combine Convolution with Recurrent Networks for Text Classification || [https://arxiv.org/pdf/2006.15795.pdf Paper] ||[https://wiki.math.uwaterloo.ca/statwiki/index.php?title=User:Cvmustat Summary] || [https://www.youtube.com/watch?v=or5RTxDnZDo]<br />
|-<br />
|Week of Nov 23 || Taohao Wang, Zeren Shen, Zihao Guo, Rui Chen || 13|| Large Scale Landmark Recognition via Deep Metric Learning || [https://arxiv.org/pdf/1908.10192.pdf paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=User:T358wang Summary] || [https://www.youtube.com/watch?v=K9NypDNPLJo Video]<br />
|-<br />
|Week of Nov 23 || Qianlin Song, William Loh, Junyue Bai, Phoebe Choi || 14|| Task Understanding from Confusing Multi-task Data || [https://proceedings.icml.cc/static/paper_files/icml/2020/578-Paper.pdf Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Task_Understanding_from_Confusing_Multi-task_Data Summary] || [https://youtu.be/i_5PQdfuH-Y]<br />
|-<br />
|Week of Nov 23 || Rui Gong, Xuetong Wang, Xinqi Ling, Di Ma || 15|| Semantic Relation Classification via Convolution Neural Network|| [https://www.aclweb.org/anthology/S18-1127.pdf paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Semantic_Relation_Classification——via_Convolution_Neural_Network Summary]|| [https://www.youtube.com/watch?v=m9o3NuMUKkU&ab_channel=DiMa video]<br />
|-<br />
|Week of Nov 23 || Xiaolan Xu, Robin Wen, Yue Weng, Beizhen Chang || 16|| Graph Structure of Neural Networks || [https://proceedings.icml.cc/paper/2020/file/757b505cfd34c64c85ca5b5690ee5293-Paper.pdf Paper] ||[https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Graph_Structure_of_Neural_Networks Summary] || [https://youtu.be/x9eUgEwntcs Video]<br />
|-<br />
|Week of Nov 23 ||Hansa Halim, Sanjana Rajendra Naik, Samka Marfua, Shawrupa Proshasty || 17|| Superhuman AI for multiplayer poker || [https://www.cs.cmu.edu/~noamb/papers/19-Science-Superhuman.pdf Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Superhuman_AI_for_Multiplayer_Poker Summary]|| [https://www.youtube.com/watch?v=kazqcOwbtTI Video]<br />
|-<br />
|Week of Nov 23 ||Guanting Pan, Haocheng Chang, Zaiwei Zhang || 18|| Point-of-Interest Recommendation: Exploiting Self-Attentive Autoencoders with Neighbor-Aware Influence || [https://arxiv.org/pdf/1809.10770.pdf Paper] ||[https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Point-of-Interest_Recommendation:_Exploiting_Self-Attentive_Autoencoders_with_Neighbor-Aware_Influence Summary] || [https://www.youtube.com/watch?v=aAwjaos_Hus Video]<br />
|-<br />
|Week of Nov 23 || Jerry Huang, Daniel Jiang, Minyan Dai || 19|| Neural Speed Reading Via Skim-RNN ||[https://arxiv.org/pdf/1711.02085.pdf?fbclid=IwAR3EeFsKM_b5p9Ox7X9mH-1oI3U3oOKPBy3xUOBN0XvJa7QW2ZeJJ9ypQVo Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Neural_Speed_Reading_via_Skim-RNN Summary]|| [https://youtu.be/vOENmt9jgVE Video]<br />
|-<br />
|Week of Nov 23 ||Ruixian Chin, Yan Kai Tan, Jason Ong, Wen Cheen Chiew || 20|| DivideMix: Learning with Noisy Labels as Semi-supervised Learning || [https://openreview.net/pdf?id=HJgExaVtwr Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=User:Yktan Summary]|| [https://www.youtube.com/watch?v=48xYZXifjS0&ab_channel=SeakraChin]<br />
|-<br />
|Week of Nov 30 || Banno Dion, Battista Joseph, Kahn Solomon || 21|| Music Recommender System Based on Genre using Convolutional Recurrent Neural Networks || [https://www.sciencedirect.com/science/article/pii/S1877050919310646] || ||<br />
|-<br />
|Week of Nov 30 || Sai Arvind Budaraju, Isaac Ellmen, Dorsa Mohammadrezaei, Emilee Carson || 22|| A universal SNP and small-indel variant caller using deep neural networks||[https://www.nature.com/articles/nbt.4235.epdf?author_access_token=q4ZmzqvvcGBqTuKyKgYrQ9RgN0jAjWel9jnR3ZoTv0NuM3saQzpZk8yexjfPUhdFj4zyaA4Yvq0LWBoCYQ4B9vqPuv8e2HHy4vShDgEs8YxI_hLs9ov6Y1f_4fyS7kGZ Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=A_universal_SNP_and_small-indel_variant_caller_using_deep_neural_networks Summary] ||<br />
|-<br />
|Week of Nov 30 || Daniel Fagan, Cooper Brooke, Maya Perelman || 23|| Efficient kNN Classification With Different Number of Nearest Neighbors || [https://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=7898482 Paper] || ||<br />
|-<br />
|Week of Nov 30 || Karam Abuaisha, Evan Li, Jason Pu, Nicholas Vadivelu || 24|| Being Bayesian about Categorical Probability || [https://proceedings.icml.cc/static/paper_files/icml/2020/3560-Paper.pdf Paper] || ||<br />
|-<br />
|Week of Nov 30 || Anas Mahdi Will Thibault Jan Lau Jiwon Yang || 25|| Loss Function Search for Face Recognition<br />
|| [https://proceedings.icml.cc/static/paper_files/icml/2020/245-Paper.pdf] paper || Summary [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Loss_Function_Search_for_Face_Recognition] ||<br />
|-<br />
|Week of Nov 30 ||Zihui (Betty) Qin, Wenqi (Maggie) Zhao, Muyuan Yang, Amartya (Marty) Mukherjee || 26|| Deep Learning for Cardiologist-level Myocardial Infarction Detection in Electrocardiograms || [https://arxiv.org/pdf/1912.07618.pdf?fbclid=IwAR0RwATSn4CiT3qD9LuywYAbJVw8YB3nbex8Kl19OCExIa4jzWaUut3oVB0 Paper] || Summary [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Deep_Learning_for_Cardiologist-level_Myocardial_Infarction_Detection_in_Electrocardiograms&fbclid=IwAR1Tad2DAM7LT6NXXuSYDZtHHBvN0mjZtDdCOiUFFq_XwVcQxG3hU-3XcaE] || [https://www.youtube.com/watch?v=kiYbAmd_3IA]<br />
|-<br />
|Week of Nov 30 || Stan Lee, Seokho Lim, Kyle Jung, Daehyun Kim || 27|| Bag of Tricks for Efficient Text Classification || [https://arxiv.org/pdf/1607.01759.pdf paper] || ||<br />
|-<br />
|Week of Nov 30 || Yawen Wang, Danmeng Cui, ZiJie Jiang, Mingkang Jiang, Haotian Ren, Haris Bin Zahid || 28|| A Brief Survey of Text Mining: Classification, Clustering and Extraction Techniques || [https://arxiv.org/pdf/1707.02919.pdf Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Describtion_of_Text_Mining Summary] ||<br />
|-<br />
|Week of Nov 30 || Qing Guo, XueGuang Ma, James Ni, Yuanxin Wang || 29|| Mask R-CNN || [https://arxiv.org/pdf/1703.06870.pdf Paper] || ||<br />
|-<br />
|Week of Nov 30 || Junyi Yang, Jill Yu Chieh Wang, Yu Min Wu, Calvin Li || 30|| Research paper classifcation systems based on TF‑IDF and LDA schemes || [https://hcis-journal.springeropen.com/articles/10.1186/s13673-019-0192-7?fbclid=IwAR3swO-eFrEbj1BUQfmomJazxxeFR6SPgr6gKayhs38Y7aBG-zX1G3XWYRM Paper] || ||<br />
|-<br />
|Week of Nov 30 || Daniel Zhang, Jacky Yao, Scholar Sun, Russell Parco, Ian Cheung || 31 || Speech2Face: Learning the Face Behind a Voice || [https://arxiv.org/pdf/1905.09773.pdf?utm_source=thenewstack&utm_medium=website&utm_campaign=platform Paper] || ||<br />
|-<br />
|Week of Nov 30 || Siyuan Xia, Jiaxiang Liu, Jiabao Dong, Yipeng Du || 32 || Evaluating Machine Accuracy on ImageNet || [https://proceedings.icml.cc/static/paper_files/icml/2020/6173-Paper.pdf] || ||<br />
|-<br />
|Week of Nov 30 || Mushi Wang, Siyuan Qiu, Yan Yu || 33 || Surround Vehicle Motion Prediction Using LSTM-RNN for Motion Planning of Autonomous Vehicles at Multi-Lane Turn Intersections || [https://ieeexplore.ieee.org/abstract/document/8957421 Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Surround_Vehicle_Motion_Prediction Summary] ||</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=stat441F21&diff=46862stat441F212020-11-26T22:20:41Z<p>Iaoellme: /* Paper presentation */</p>
<hr />
<div><br />
<br />
== [[F20-STAT 441/841 CM 763-Proposal| Project Proposal ]] ==<br />
<br />
<!--[https://goo.gl/forms/apurag4dr9kSR76X2 Your feedback on presentations]--><br />
<br />
= Record your contributions here [https://docs.google.com/spreadsheets/d/10CHiJpAylR6kB9QLqN7lZHN79D9YEEW6CDTH27eAhbQ/edit?usp=sharing]=<br />
<br />
Use the following notations:<br />
<br />
P: You have written a summary/critique on the paper.<br />
<br />
T: You had a technical contribution on a paper (excluding the paper that you present).<br />
<br />
E: You had an editorial contribution on a paper (excluding the paper that you present).<br />
<br />
=Paper presentation=<br />
{| class="wikitable"<br />
<br />
{| border="1" cellpadding="3"<br />
|-<br />
|width="60pt"|Date<br />
|width="250pt"|Name <br />
|width="15pt"|Paper number <br />
|width="700pt"|Title<br />
|width="15pt"|Link to the paper<br />
|width="30pt"|Link to the summary<br />
|width="30pt"|Link to the video<br />
|-<br />
|Sep 15 (example)||Ri Wang || ||Sequence to sequence learning with neural networks.||[http://papers.nips.cc/paper/5346-sequence-to-sequence-learning-with-neural-networks.pdf Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Going_Deeper_with_Convolutions Summary] || [https://youtu.be/JWozRg_X-Vg?list=PLehuLRPyt1HzXDemu7K4ETcF0Ld_B5adG&t=539]<br />
|-<br />
|Week of Nov 16 ||Sharman Bharat, Li Dylan,Lu Leonie, Li Mingdao || 1|| Risk prediction in life insurance industry using supervised learning algorithms || [https://rdcu.be/b780J Paper] ||[https://wiki.math.uwaterloo.ca/statwiki/index.php?title=User:Bsharman Summary] ||<br />
[https://www.youtube.com/watch?v=TVLpSFYgF0c&feature=youtu.be]<br />
|-<br />
|Week of Nov 16 || Delaney Smith, Mohammad Assem Mahmoud || 2|| Influenza Forecasting Framework based on Gaussian Processes || [https://proceedings.icml.cc/static/paper_files/icml/2020/1239-Paper.pdf Paper] ||[https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Influenza_Forecasting_Framework_based_on_Gaussian_Processes Summary]|| [https://www.youtube.com/watch?v=HZG9RAHhpXc&feature=youtu.be]<br />
|-<br />
|Week of Nov 16 || Tatianna Krikella, Swaleh Hussain, Grace Tompkins || 3|| Processing of Missing Data by Neural Networks || [http://papers.nips.cc/paper/7537-processing-of-missing-data-by-neural-networks.pdf Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=User:Gtompkin Summary] || [https://learn.uwaterloo.ca/d2l/ext/rp/577051/lti/framedlaunch/6ec1ebea-5547-46a2-9e4f-e3dc9d79fd54]<br />
|-<br />
|Week of Nov 16 ||Jonathan Chow, Nyle Dharani, Ildar Nasirov ||4 ||Streaming Bayesian Inference for Crowdsourced Classification ||[https://papers.nips.cc/paper/9439-streaming-bayesian-inference-for-crowdsourced-classification.pdf Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Streaming_Bayesian_Inference_for_Crowdsourced_Classification Summary] || [https://www.youtube.com/watch?v=UgVRzi9Ubws]<br />
|-<br />
|Week of Nov 16 || Matthew Hall, Johnathan Chalaturnyk || 5|| Neural Ordinary Differential Equations || [https://papers.nips.cc/paper/7892-neural-ordinary-differential-equations.pdf] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Neural_ODEs Summary]||<br />
|-<br />
|Week of Nov 16 || Luwen Chang, Qingyang Yu, Tao Kong, Tianrong Sun || 6|| Adversarial Attacks on Copyright Detection Systems || Paper [https://proceedings.icml.cc/static/paper_files/icml/2020/1894-Paper.pdf] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Adversarial_Attacks_on_Copyright_Detection_Systems Summary] || [https://www.youtube.com/watch?v=bQI9S6bCo8o]<br />
|-<br />
|Week of Nov 16 || Casey De Vera, Solaiman Jawad || 7|| IPBoost – Non-Convex Boosting via Integer Programming || [https://arxiv.org/pdf/2002.04679.pdf Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=IPBoost Summary] || [https://www.youtube.com/watch?v=4DhJDGC4pyI&feature=youtu.be]<br />
|-<br />
|Week of Nov 16 || Yuxin Wang, Evan Peters, Yifan Mou, Sangeeth Kalaichanthiran || 8|| What Game Are We Playing? End-to-end Learning in Normal and Extensive Form Games || [https://arxiv.org/pdf/1805.02777.pdf] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=what_game_are_we_playing Summary] || [https://www.youtube.com/watch?v=9qJoVxo3hnI&feature=youtu.be]<br />
|-<br />
|Week of Nov 16 || Yuchuan Wu || 9|| || || ||<br />
|-<br />
|Week of Nov 16 || Zhou Zeping, Siqi Li, Yuqin Fang, Fu Rao || 10|| A survey of neural network-based cancer prediction models from microarray data || [https://www.sciencedirect.com/science/article/pii/S0933365717305067 Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=User:Y93fang Summary] || [https://youtu.be/B8pPUU8ypO0]<br />
|-<br />
|Week of Nov 23 ||Jinjiang Lian, Jiawen Hou, Yisheng Zhu, Mingzhe Huang || 11|| DROCC: Deep Robust One-Class Classification || [https://proceedings.icml.cc/static/paper_files/icml/2020/6556-Paper.pdf paper] ||[https://wiki.math.uwaterloo.ca/statwiki/index.php?title=User:J46hou Summary] || [https://www.youtube.com/watch?v=tvCEvvy54X8&ab_channel=JJLian]<br />
|-<br />
|Week of Nov 23 || Bushra Haque, Hayden Jones, Michael Leung, Cristian Mustatea || 12|| Combine Convolution with Recurrent Networks for Text Classification || [https://arxiv.org/pdf/2006.15795.pdf Paper] ||[https://wiki.math.uwaterloo.ca/statwiki/index.php?title=User:Cvmustat Summary] || [https://www.youtube.com/watch?v=or5RTxDnZDo]<br />
|-<br />
|Week of Nov 23 || Taohao Wang, Zeren Shen, Zihao Guo, Rui Chen || 13|| Large Scale Landmark Recognition via Deep Metric Learning || [https://arxiv.org/pdf/1908.10192.pdf paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=User:T358wang Summary] || [https://www.youtube.com/watch?v=K9NypDNPLJo Video]<br />
|-<br />
|Week of Nov 23 || Qianlin Song, William Loh, Junyue Bai, Phoebe Choi || 14|| Task Understanding from Confusing Multi-task Data || [https://proceedings.icml.cc/static/paper_files/icml/2020/578-Paper.pdf Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Task_Understanding_from_Confusing_Multi-task_Data Summary] || [https://youtu.be/i_5PQdfuH-Y]<br />
|-<br />
|Week of Nov 23 || Rui Gong, Xuetong Wang, Xinqi Ling, Di Ma || 15|| Semantic Relation Classification via Convolution Neural Network|| [https://www.aclweb.org/anthology/S18-1127.pdf paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Semantic_Relation_Classification——via_Convolution_Neural_Network Summary]|| [https://www.youtube.com/watch?v=m9o3NuMUKkU&ab_channel=DiMa video]<br />
|-<br />
|Week of Nov 23 || Xiaolan Xu, Robin Wen, Yue Weng, Beizhen Chang || 16|| Graph Structure of Neural Networks || [https://proceedings.icml.cc/paper/2020/file/757b505cfd34c64c85ca5b5690ee5293-Paper.pdf Paper] ||[https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Graph_Structure_of_Neural_Networks Summary] || [https://youtu.be/x9eUgEwntcs Video]<br />
|-<br />
|Week of Nov 23 ||Hansa Halim, Sanjana Rajendra Naik, Samka Marfua, Shawrupa Proshasty || 17|| Superhuman AI for multiplayer poker || [https://www.cs.cmu.edu/~noamb/papers/19-Science-Superhuman.pdf Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Superhuman_AI_for_Multiplayer_Poker Summary]|| [https://www.youtube.com/watch?v=kazqcOwbtTI Video]<br />
|-<br />
|Week of Nov 23 ||Guanting Pan, Haocheng Chang, Zaiwei Zhang || 18|| Point-of-Interest Recommendation: Exploiting Self-Attentive Autoencoders with Neighbor-Aware Influence || [https://arxiv.org/pdf/1809.10770.pdf Paper] ||[https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Point-of-Interest_Recommendation:_Exploiting_Self-Attentive_Autoencoders_with_Neighbor-Aware_Influence Summary] || [https://www.youtube.com/watch?v=aAwjaos_Hus Video]<br />
|-<br />
|Week of Nov 23 || Jerry Huang, Daniel Jiang, Minyan Dai || 19|| Neural Speed Reading Via Skim-RNN ||[https://arxiv.org/pdf/1711.02085.pdf?fbclid=IwAR3EeFsKM_b5p9Ox7X9mH-1oI3U3oOKPBy3xUOBN0XvJa7QW2ZeJJ9ypQVo Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Neural_Speed_Reading_via_Skim-RNN Summary]|| [https://youtu.be/vOENmt9jgVE Video]<br />
|-<br />
|Week of Nov 23 ||Ruixian Chin, Yan Kai Tan, Jason Ong, Wen Cheen Chiew || 20|| DivideMix: Learning with Noisy Labels as Semi-supervised Learning || [https://openreview.net/pdf?id=HJgExaVtwr Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=User:Yktan Summary]|| [https://www.youtube.com/watch?v=48xYZXifjS0&ab_channel=SeakraChin]<br />
|-<br />
|Week of Nov 30 || Banno Dion, Battista Joseph, Kahn Solomon || 21|| Music Recommender System Based on Genre using Convolutional Recurrent Neural Networks || [https://www.sciencedirect.com/science/article/pii/S1877050919310646] || ||<br />
|-<br />
|Week of Nov 30 || Sai Arvind Budaraju, Isaac Ellmen, Dorsa Mohammadrezaei, Emilee Carson || 22|| A universal SNP and small-indel variant caller using deep neural networks||[https://www.nature.com/articles/nbt.4235.epdf?author_access_token=q4ZmzqvvcGBqTuKyKgYrQ9RgN0jAjWel9jnR3ZoTv0NuM3saQzpZk8yexjfPUhdFj4zyaA4Yvq0LWBoCYQ4B9vqPuv8e2HHy4vShDgEs8YxI_hLs9ov6Y1f_4fyS7kGZ Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=A_universal_SNP_and_small-indel_variant_caller_using_deep_neural_networks] ||<br />
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|Week of Nov 30 || Daniel Fagan, Cooper Brooke, Maya Perelman || 23|| Efficient kNN Classification With Different Number of Nearest Neighbors || [https://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=7898482 Paper] || ||<br />
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|Week of Nov 30 || Karam Abuaisha, Evan Li, Jason Pu, Nicholas Vadivelu || 24|| Being Bayesian about Categorical Probability || [https://proceedings.icml.cc/static/paper_files/icml/2020/3560-Paper.pdf Paper] || ||<br />
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|Week of Nov 30 || Anas Mahdi Will Thibault Jan Lau Jiwon Yang || 25|| Loss Function Search for Face Recognition<br />
|| [https://proceedings.icml.cc/static/paper_files/icml/2020/245-Paper.pdf] paper || Summary [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Loss_Function_Search_for_Face_Recognition] ||<br />
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|Week of Nov 30 ||Zihui (Betty) Qin, Wenqi (Maggie) Zhao, Muyuan Yang, Amartya (Marty) Mukherjee || 26|| Deep Learning for Cardiologist-level Myocardial Infarction Detection in Electrocardiograms || [https://arxiv.org/pdf/1912.07618.pdf?fbclid=IwAR0RwATSn4CiT3qD9LuywYAbJVw8YB3nbex8Kl19OCExIa4jzWaUut3oVB0 Paper] || Summary [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Deep_Learning_for_Cardiologist-level_Myocardial_Infarction_Detection_in_Electrocardiograms&fbclid=IwAR1Tad2DAM7LT6NXXuSYDZtHHBvN0mjZtDdCOiUFFq_XwVcQxG3hU-3XcaE] || [https://www.youtube.com/watch?v=kiYbAmd_3IA]<br />
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|Week of Nov 30 || Stan Lee, Seokho Lim, Kyle Jung, Daehyun Kim || 27|| Bag of Tricks for Efficient Text Classification || [https://arxiv.org/pdf/1607.01759.pdf paper] || ||<br />
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|Week of Nov 30 || Yawen Wang, Danmeng Cui, ZiJie Jiang, Mingkang Jiang, Haotian Ren, Haris Bin Zahid || 28|| A Brief Survey of Text Mining: Classification, Clustering and Extraction Techniques || [https://arxiv.org/pdf/1707.02919.pdf Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Describtion_of_Text_Mining Summary] ||<br />
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|Week of Nov 30 || Qing Guo, XueGuang Ma, James Ni, Yuanxin Wang || 29|| Mask R-CNN || [https://arxiv.org/pdf/1703.06870.pdf Paper] || ||<br />
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|Week of Nov 30 || Junyi Yang, Jill Yu Chieh Wang, Yu Min Wu, Calvin Li || 30|| Research paper classifcation systems based on TF‑IDF and LDA schemes || [https://hcis-journal.springeropen.com/articles/10.1186/s13673-019-0192-7?fbclid=IwAR3swO-eFrEbj1BUQfmomJazxxeFR6SPgr6gKayhs38Y7aBG-zX1G3XWYRM Paper] || ||<br />
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|Week of Nov 30 || Daniel Zhang, Jacky Yao, Scholar Sun, Russell Parco, Ian Cheung || 31 || Speech2Face: Learning the Face Behind a Voice || [https://arxiv.org/pdf/1905.09773.pdf?utm_source=thenewstack&utm_medium=website&utm_campaign=platform Paper] || ||<br />
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|Week of Nov 30 || Siyuan Xia, Jiaxiang Liu, Jiabao Dong, Yipeng Du || 32 || Evaluating Machine Accuracy on ImageNet || [https://proceedings.icml.cc/static/paper_files/icml/2020/6173-Paper.pdf] || ||<br />
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|Week of Nov 30 || Mushi Wang, Siyuan Qiu, Yan Yu || 33 || Surround Vehicle Motion Prediction Using LSTM-RNN for Motion Planning of Autonomous Vehicles at Multi-Lane Turn Intersections || [https://ieeexplore.ieee.org/abstract/document/8957421 Paper] || [https://wiki.math.uwaterloo.ca/statwiki/index.php?title=Surround_Vehicle_Motion_Prediction Summary] ||</div>Iaoellmehttp://wiki.math.uwaterloo.ca/statwiki/index.php?title=A_universal_SNP_and_small-indel_variant_caller_using_deep_neural_networks&diff=46861A universal SNP and small-indel variant caller using deep neural networks2020-11-26T22:18:48Z<p>Iaoellme: Created blank page</p>
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