Causal discovery in time-series data is critical for analyzing dynamic systems across neuroscience, economics, and biomedical signal processing. Traditional methods, such as Vector Auto-regression (VAR) and constraint-based approaches, struggle with high-dimensional dependencies, nonlinear relationships, and non-stationary dynamics. Deep learning-based models, including cMLP, cLSTM, and VAE-based approaches, aim to address these challenges but suffer from instability, over-pruning, and reliance on sparsity constraints. While cMLP provides lag-specific causal inference, its accuracy is limited, and other methods fail to explicitly capture lag-wise dependencies. This paper introduces DVAE-GC, a structured deep learning framework integrating dynamic variational inference with lag-structured recurrent MLPs (lsrMLP) to explicitly model time-lagged causal dependencies. Unlike prior methods that infer causality via weight sparsity, DVAE-GC progressively refines causal estimation, leveraging a bidirectional recurrent encoder and structured decoder. Additionally, Noise Invalidation Soft Thresholding (NIST) eliminates spurious connections, enhancing interpretability and robustness. Empirically, DVAE-GC outperforms the best baseline (CUTS) on VAR(9) by +18.3 absolute F1 points averaged over multiple noise levels, and on NetSim fMRI-20 by +8.1 absolute F1 points averaged over sequence multiple lengths; in simulated atrial rotor detection, it improves Rotational Activity Estimation Precision (RAEP) by +22.4 % over the best alternative (VAR). These are absolute-point gains, and also precision, recall, and false discovery rate (FDR) has been reported. Although evaluated in biomedical simulations, DVAE-GC applies broadly to time-series domains, including neuroscience, climate science, and financial modeling.

Background and Context: While ample research has examined undergraduate students’ participation in Computer Science I (CS1), far less attention has been paid to Computer Science II (CS2) outcomes. Inspired by self-efficacy, Object-Oriented Programming (OOP) and CS2, we expand our understanding of the traditional computer programming curriculum sequence in CS curricula guidelines (CS1, CS2, and data structures and algorithms).

Chronic stress risks an individual's overall well-being. Chronic stress is associated with allostatic load, the body's wear-and-tear due to prolonged heightened physiological and psychological states. Increased allostatic load among workers increases their risk of injuries and the likelihood of diseases and illnesses. An allostatic load model could explain the basis of a stress response. Stress research in affective computing uses wearable devices, data processing algorithms, and machine learning methods to create models that could benefit from an allostatic load model of stress. We emphasize the need for the allostatic load model in affective computing to create disease and illness prediction models. Predictive models could enhance safeguards in the workplace by helping to create proactive mitigation strategies against chronic stress. First, we briefly introduce allostasis’ physiological and psychological basis. Next, we reviewed stress studies within affective computing that may benefit from an allostatic load model of stress. We focused our review on studies conducted in dynamic settings, such as the workplace, and those incorporating typical stress study elements in affective computing. We conclude our review by identifying gaps between affective computing and neuroscientific stress studies and provide recommendations for adopting the allostatic load model of stress.

Speech, music, and environmental sounds are the main forms of audio signals that are widely studied. There is a certain amount of texture present in every sound, and our human auditory system is not efficient in recognizing and classifying these audio textures present within the sounds. E.g. we are not able to distinguish between two sounds of a fire crackling or two sounds of water-falling. Hence, there is a need for a representation that could model these audio textures. These textures are also present in the audio signals that changes if pathological and pathomorphological conditions are present. To capture and analyze these audio textures, the audio signal is generally transformed into an intermediate time-frequency (t-f) representation such as spectrograms, Mel-spectrograms, and more. But recent studies have shown that joint time-frequency scattering transform is more suitable for classification problems than the standard time-frequency representations because of its inherent property of invariance and invertibility. In this paper, we have investigated the capacity of joint time-frequency scattergrams to capture the audio textures by analyzing the audio structures of biomedical sounds such as COVID-19 cough and breath sounds, pathological speech in children and adults, and infant cry sounds. Accuracy rates up to 96.40% for COVID-19 sounds, 95.50% for pathological speech in children, 94.10% for pathological speech in adults, and 97.30% for infant cry sounds have been achieved with 10-fold cross-validation. The proposed model of using scattergram provides an alternate and an efficient way of representing biomedical audio signals for machine learning applications.

In this work, we consider distance-based clustering of partial lexicographic preference trees (PLP-trees), intuitive and compact graphical representations of user preferences over multi-valued attributes. To compute distances between PLP-trees, we propose a polynomial time algorithm that computes Kendall's Tau distance directly from the trees and show its efficacy compared to the brute-force algorithm. To this end, we implement several clustering methods (i.e., spectral clustering, affinity propagation, and agglomerative nesting) augmented by our distance algorithm, experiment with clustering of up to 10,000 PLP-trees, and show the effectiveness of the clustering methods and visualizations of their results.

Several experiments on the effects of pair programming versus solo programming in the context of education have been reported in the research literature. We present a meta-analysis of these studies that accounted for 18 manuscripts with 28 independent effect sizes in the domains of programming assignments, exams, passing rates, and affective measures. In total, our sample accounts for N = 3,308 students either using pair programming as a treatment variable or using traditional solo programming in the context of a computing course. Our findings suggest positive results in favor of pair programming in three of four domains with exception to affective measures. We provide a comprehensive review of our results and discuss our findings.

Archives of human rights violations reports, by virtue of their poor metadata, basis in natural language, and scale, obscure fine grain analyses of violation event patterns. Cross-document coreference of victim or perpetrator occurrences from across a corpus is challenging, particularly when those mentions relate to different events. These challenges are emblematic of the transition from small scale to big data analysis in the humanities. This paper discusses these issues and proposes a framework to address these challenges so as to explore narrative construction and the formation of collective memory. Though our framework is based on processing human rights violation reports, it can be readily extended to support other big data problems in the humanities.

Research in signal processing shows that a variety of transforms have been introduced to map the data from the original space into the feature space, in order to efficiently analyze a signal. These techniques differ in their basis functions, that is used for projecting the signal into a higher dimensional space. One of the widely used schemes for quasi-stationary and non-stationary signals is the time-frequency (TF) transforms, characterized by specific kernel functions. This work introduces a novel class of Ramanujan Fourier Transform (RFT) based TF transform functions, constituted by Ramanujan sums (RS) basis. The proposed special class of transforms offer high immunity to noise interference, since the computation is carried out only on co-resonant components, during analysis of signals. Further, we also provide a 2-D formulation of the RFT function. Experimental validation using synthetic examples, indicates that this technique shows potential for obtaining relatively sparse TF-equivalent representation and can be optimized for characterization of certain real-life signals.

The design of large and complex enterprise integration solutions is a difficult task. It can require solutions that are unique because of constraints from the current set of legacy applications. Design knowledge for enterprise integration solutions is, therefore, difficult to articulate and reuse. In particular, the nature and form of knowledge for conceptual design of integration solutions is difficult to pin down. In this paper, we investigate whether design knowledge for enterprise integration in the form of patterns can be reused to develop systems integration solutions, and whether such reuse leads to more effective design outcomes. The research follows design science guidelines in which we describe a research artifact, and evaluate it to assess whether it meets the intended goals. The results indicate that approaches to facilitate reuse of conceptual design knowledge are feasible in the domain of enterprise integration, and that such reuse does, in fact, lead to more effective design solutions.

Audio feature extraction plays an important role in analyzing and characterizing audio content. Auditory scene analysis, content-based retrieval, indexing, and fingerprinting of audio are few of the applications that require efficient feature extraction. The key to extract strong features that characterize the complex nature of audio signals is to identify their discriminatory subspaces. In this paper, we propose an audio feature extraction and a multigroup classification scheme that focuses on identifying discriminatory time-frequency subspaces using the local discriminant bases (LDB) technique. Two dissimilarity measures were used in the process of selecting the LDB nodes and extracting features from them. The extracted features were then fed to a linear discriminant analysis-based classifier for a three-level hierarchical classification of audio signals into ten classes. In the first level, the audio signals were grouped into artificial and natural sounds. Each of the first level groups were subdivided to form the second level groups viz. instrumental, automobile, human, and nonhuman sounds. The third level was formed by subdividing the four groups of the second level into the final ten groups (drums, flute, piano, aircraft, helicopter, male, female, animals, birds and insects). A database of 213 audio signals were used in this study and an average classification accuracy of 83% for the first level (113 artificial and 100 natural sounds), 92% for the second level (73 instrumental and 40 automobile sounds; 40 human and 60 nonhuman sounds), and 89% for the third level (27 drums, 15 flute, and 31 piano sounds; 23 aircraft and 17 helicopter sounds; 20 male and 20 female speech; 20 animals, 20 birds and 20 insects sounds) were achieved. In addition to the above, a separate classification was also performed combining the LDB features with the mel-frequency cepstral coefficients. The average classification accuracies achieved using the combined features were 91% for the first level, 99% for the second level, and 95% for the third level

Time-frequency (TF) signal decompositions provide us with ample information and extreme flexibility for signal analysis. By applying suitable processing on the TF decomposition parameters, even subtle signal characteristics can be revealed. In many real world applications, identification of these subtle differences make a significant impact in signal analysis. Particularly in classification applications using TF approaches, there may be situations where a localized high discriminative signal structure is diluted due to the presence of other overlapping signal structures. To address this problem we propose a novel approach to construct multiple time-width vs frequency band mappings based on the energy decomposition pattern of the signal. These mapping are then analyzed to locate the highly discriminative features for classification. Initial results with two real world biomedical signal databases: (1) vibroarthrographic (VAG) signals; and (2) pathological speech signals, indicate high potential for the proposed technique.

The ongoing advancements in the multimedia technologies drive the need for efficient classification of the audio signals to make the content-based retrieval process more accurate and much easier from huge databases. The challenge of this task lies in an accurate extraction of signal characteristics so as to derive a strong discriminatory feature suitable for classification. In this paper, a time-frequency (TF) approach for audio classification is proposed. Audio signals are nonstationary in nature and TF approach is the best way to analyze them. The audio signals were decomposed using an adaptive TF decomposition algorithm, and the signal decomposition parameter based on octave (scaling) was used to generate a set of 42 features over three frequency bands within the auditory range. These features were analyzed using linear discriminant functions and classified into six music groups (rock, classical, country, jazz, folk and pop). Overall classification accuracies as high as 97.6 % was achieved by linear discriminant analysis of 170 audio signals.

One of the major challenges in classification problems, based on the signal decomposition approach, is to identify the right basis function and its derivatives that can provide optimal features to distinguish the classes. With the vast amount of available libraries of orthonormal bases, it is hard to select an optimal set of basis functions for a specific dataset. To address this problem, pruning algorithms based on certain selection criteria, are needed. The local discriminant bases (LDB) algorithm is one such algorithm, which efficiently selects a set of significant basis functions from the library of orthonormal bases based on a certain defined dissimilarity measure. The selection of this dissimilarity measure is critical as they indirectly contribute to the performance accuracy of the LDB algorithm. In this paper, we study the impact of the dissimilarity measures on the performance of the LDB algorithm with two classification examples. Two biomedical signal databases used are: 1) vibroarthographic signals (VAG) - 89 signals with 51 normal and 38 abnormal; and 2) pathological speech signals - 100 signals with 50 normal and 50 pathological. Classification accuracies of 76.4% with the VAG database and 96% with the pathological speech database were obtained. This modified method of signal analysis using LDB has shown its powerfulness in analyzing non-stationary signals.

Currently Web services composition problems are addressed using AI planning techniques. The team-based approach, with emphases on the sharing of mental models and proactive collaboration, provides an alternative to current static approaches to Web service composition. The approach provides clear advantages for proactive handling of failures that may be encountered during execution of a complex Web service. The paper proposes a generic framework for dynamic Web-service composition, and extends the CAST architecture to realize the framework.

Acoustic measures of vocal function are routinely used for the assessment of disordered voice, and for monitoring patient's progress over the course of therapy. In current clinical practice, acoustic measures extracted from sustained vowels are used for vocal function characterization. However, the measures derived from continuous speech samples are required for accurate assessment of voice quality. In this paper, a time-frequency approach for pathological voice discrimination has been proposed. The speech signals were decomposed using an adaptive time-frequency transform algorithm, and the signal decomposition parameters such as the octave (scale) maximum, octave mean, energy rate, and length ratio were analyzed using the maximum likelihood method and Jack-knife algorithm for classification. A classification accuracy of 90% was obtained with a database of 40 speech signals (20 normal and 20 pathological cases).

The ongoing advancements in the multimedia technologies drive the need for efficient classification of the audio signals to make the content-based retrieval process more accurate and much easier from huge databases. The challenge of this task lies in an accurate extraction of signal characteristics so as to derive a strong discriminatory feature suitable for retrieval process. A time-frequency approach for audio classification is proposed. The audio signals were decomposed using an adaptive time-frequency decomposition algorithm, and the signal decomposition parameter octave (scale) was used to create patterns based on a similarity measure of the audio signals. These patterns were used to generate templates to classify the audio signals into different categories. Initial studies have yielded a overall correct classification accuracy of 90% with a database of 64 audio segments.