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Improper sitting positions are known as the primary reason for back pain and the emergence of musculoskeletal disorders (MSDs) among individuals who spend prolonged time working with computer screens, keyboards, and mice. At the same time, it is well understood that automated technological tools can play an important role in the process of unhealthy habit alteration, so plenty of research efforts are focused on research and technology development (RTD) activities that aim to provide support for the prevention of back pain or the development of MSDs. Here, we report on creating a new resource in support of RTD activities aiming at the automated detection of improper sitting positions. It consists of multimodal multisensory recordings of 100 persons, made with a video recorder, camera, and wrist-attached sensors that capture physiological signals (PPG, EDA, skin temperature), as well as motion sensors (three-axis accelerometer). Our multimodal multisensory dataset (MMD-MSD) opens new opportunities for modeling the body stance (sitting posture and movements), physiological state (stress level, attention, emotional arousal and valence), and performance (success rate on the Stroop test) of people working with a computer. Finally, we demonstrate two use cases: improper neck posture detection from pictures, and task-specific cognitive load detection from physiological signals. [ABSTRACT FROM AUTHOR]

Machine learning tools are widely used in support of bioacoustics studies, and there are numerous publications on the applicability of convolutional neural networks (CNNs) to the automated presenceabsence detection of species. However, the relation between the merit of acoustic background modelling and the recognition performance needs to be better understood. In this study, we investigated the influence of acoustic background substance on the performance of the acoustic detector of the White-lored Spinetail (Synallaxis albilora). Two detector designs were evaluated: the 152- layer ResNet with transfer learning and a purposely created CNN. We experimented with acoustic background representations trained with season-specific (dry, wet, and all-season) data and without explicit modelling to evaluate its influence on the detection performance. The detector permits monitoring of the diel behaviour and breeding time of White-lored Spinetail solely based on the changes in the vocal activity patterns. We report an advantageous performance when background modelling is used, precisely when trained with all-season data. The highest classification accuracy (84.5%) was observed for the purposely created CNN model. Our findings contribute to an improved understanding of the importance of acoustic background modelling, which is essential for increasing the performance of CNN-based species detectors. [ABSTRACT FROM AUTHOR]

We propose a novel distributed method for non-convex optimization problems with coupling equality and inequality constraints. This method transforms the optimization problem into a specific form to allow distributed implementation of modified gradient descent and Newton's methods so that they operate as if they were distributed. We demonstrate that for the proposed distributed method: (i) communications are significantly less time-consuming than oracle calls, (ii) its convergence rate is equivalent to the convergence of Newton's method concerning oracle calls, and (iii) for the cases when oracle calls are more expensive than communication between agents, the transition from a centralized to a distributed paradigm does not significantly affect computational time. The proposed method is applicable when the objective function is twice differentiable and constraints are differentiable, which holds for a wide range of machine learning methods and optimization setups. [ABSTRACT FROM AUTHOR]

Computer-assisted species recognition facilitates the analysis of relevant biological information in continuous audio recordings. In the present study, we assess the suitability of this approach for determining distinct life-cycle phases of the Southern Lapwing Vanellus chilensis lampronotus based on adult vocal activity. For this purpose we use passive 14-min and 30-min soundscape recordings (n = 33 201) collected in 24/7 mode between November 2012 and October 2013 in Brazil’s Pantanal wetlands. Time-stamped detections of V. chilensis call events (n = 62 292) were obtained with a species-specific sound recognizer. We demonstrate that the breeding season fell in a three-month period from mid-May to early August 2013, between the end of the flood cycle and the height of the dry season. Several phases of the lapwing’s life history were identified with presumed error margins of a few days: pre-breeding, territory establishment and egg-laying, incubation, hatching, parental defense of chicks, and post-breeding. Diurnal time budgets confirm high acoustic activity levels during midday hours in June and July, indicative of adults defending young. By August, activity patterns had reverted to nonbreeding mode, with peaks around dawn and dusk and low call frequency during midday heat. We assess the current technological limitations of the V. chilensis recognizer through a comprehensive performance assessment and scrutinize the usefulness of automated acoustic recognizers in studies on the distribution pattern, ecology, life history, and conservation status of sound-producing animal species. [ABSTRACT FROM AUTHOR]

Traditional human-observer-based biological surveys are expensive. Therefore most biodiversity studies are implemented only periodically, for short periods, and predominantly during daytime and under favorable weather conditions. Automated data acquisition and analysis can overcome these shortcomings and facilitate continuous monitoring. Here we report on the development of an automated acoustic recognizer for Southern Lapwing Vanellus chilensis lampronotus vocalizations, a first for this species. The recognizer is a species-specific information retrieval agent, which searches throughout long audio recordings in order to detect and timestamp call events of the target species. The recognizer relies on a log-likelihood ratio estimator, based on a Gaussian Mixture Model–Universal Background Model (GMM–UBM), complemented with purposely-developed temporal post-processing that incorporates domain knowledge about the structure of V. chilensis vocalizations. Validation experiments with real-field recordings of complex soundscapes indicate that the recognizer is sensitive enough to register V. chilensis call events with sound levels down to −30 dB and recognition accuracy of up to 85.6%, at zero false positive rates. The recognizer is considered a valuable tool for computer-assisted analysis of hourly and daily acoustic activity of V. chilensis over extended periods of time, as it offers an indispensable support to long-term monitoring studies and conservation efforts in the Pantanal region. 1 INAU Project 3.14 “Monitoring Bioindicators and Migratory Birds in the Pantanal. Applied Acoustomics – a Tool for Bio-sustainability Assessment”, INAU Lab. 3 – Biodiversity and Ecological Processes (INAU: www.inau.org.br ; 2011–2014) – Program CsF ( www.cienciasemfronteiras.gov.br/web/csf ). 1 [ABSTRACT FROM AUTHOR]

Abstract: We propose a two-stage phone duration modelling scheme, which can be applied for the improvement of prosody modelling in speech synthesis systems. This scheme builds on a number of independent feature constructors (FCs) employed in the first stage, and a phone duration model (PDM) which operates on an extended feature vector in the second stage. The feature vector, which acts as input to the first stage, consists of numerical and non-numerical linguistic features extracted from text. The extended feature vector is obtained by appending the phone duration predictions estimated by the FCs to the initial feature vector. Experiments on the American-English KED TIMIT and on the Modern Greek WCL-1 databases validated the advantage of the proposed two-stage scheme, improving prediction accuracy over the best individual predictor, and over a two-stage scheme which just fuses the first-stage outputs. Specifically, when compared to the best individual predictor, a relative reduction in the mean absolute error and the root mean square error of 3.9% and 3.9% on the KED TIMIT, and of 4.8% and 4.6% on the WCL-1 database, respectively, is observed. [Copyright &y& Elsevier]

Abstract: Based on the observation that dissimilar speech enhancement algorithms perform differently for different types of interference and noise conditions, we propose a context-adaptive speech pre-processing scheme, which performs adaptive selection of the most advantageous speech enhancement algorithm for each condition. The selection process is based on an unsupervised clustering of the acoustic feature space and a subsequent mapping function that identifies the most appropriate speech enhancement channel for each audio input, corresponding to unknown environmental conditions. Experiments performed on the MoveOn motorcycle speech and noise database validate the practical value of the proposed scheme for speech enhancement and demonstrate a significant improvement in terms of speech recognition accuracy, when compared to the one of the best performing individual speech enhancement algorithm. This is expressed as accuracy gain of 3.3% in terms of word recognition rate. The advance offered in the present work reaches beyond the specifics of the present application, and can be beneficial to spoken interfaces operating in fast-varying noise environments. [Copyright &y& Elsevier]

In the present work we address the problem of phonetic segmentation of emotional speech. Investigating various traditional and recent HMM-based methods for speech segmentation, which we elaborated for the specifics of emotional speech segmentation, we demonstrate that the HMM-based method with hybrid embedded-isolated training offers advantageous segmentation accuracy, when compared to other HMM-based models used so far. The increased precision of the segmentation is a consequence of the iterative training process employed in the hybrid-training method, which refines the model parameters and the estimated phonetic boundaries taking advantage of the estimations made at previous iterations. Furthermore, we demonstrate the benefits of using purposely-built models for each target category of emotional speech, when compared to the case of one common model built solely from neutral speech. This advantage, in terms of segmentation accuracy, justifies the effort for creating and employing the purposely-built segmentation models per emotion category, since it significantly improves the overall segmentation accuracy. [ABSTRACT FROM AUTHOR]

In the present work, we investigate the performance of a number of traditional and recent speech enhancement algorithms in the adverse non-stationary conditions, which are distinctive for motorcycles on the move. The performance of these algorithms is ranked in terms of the improvement they contribute to the speech recognition accuracy, when compared to the baseline performance, i.e. without speech enhancement. The experiments on the MoveOn motorcycle speech and noise database indicated that there is no equivalence between the ranking of algorithms based on the human perception of speech quality and the speech recognition performance. The Multi-band spectral subtraction method was observed to lead to the highest speech recognition performance. [ABSTRACT FROM AUTHOR]

Abstract: In the present work we study the appropriateness of a number of linear and non-linear regression methods, employed on the task of speech segmentation, for combining multiple phonetic boundary predictions which are obtained through various segmentation engines. The proposed fusion schemes are independent of the implementation of the individual segmentation engines as well as from their number. In order to illustrate the practical significance of the proposed approach, we employ 112 speech segmentation engines based on hidden Markov models (HMMs), which differ in the setup of the HMMs and in the speech parameterization techniques they employ. Specifically we relied on sixteen different HMMs setups and on seven speech parameterization techniques, four of which are recent and their performance on the speech segmentation task have not been evaluated yet. In the evaluation experiments we contrast the performance of the proposed fusion schemes for phonetic boundary predictions against some recently reported methods. Throughout this comparison, on the established for the phonetic segmentation task TIMIT database, we demonstrate that the support vector regression scheme is capable of achieving more accurate predictions, when compared to other fusion schemes reported so far. [Copyright &y& Elsevier]

In the present contribution we propose an integral training procedure for the Locally Recurrent Probabilistic Neural Networks (LR PNNs). Specifically, the adjustment of the smoothing factor "sigma" in the pattern layer of the LR PNN and the training of the recurrent layer weights are integrated in an automatic process that iteratively estimates all adjustable parameters of the LR PNN from the available training data. Furthermore, in contrast to the original LR PNN, whose recurrent layer was trained to provide optimum separation among the classes on the training dataset, while striving to keep a balance between the learning rates for all classes, here the training strategy is oriented towards optimizing the overall classification accuracy, straightforwardly. More precisely, the new training strategy directly targets at maximizing the posterior probabilities for the target class and minimizing the posterior probabilities estimated for the non-target classes. The new fitness function requires fewer computations for each evaluation, and therefore the overall computational demands for training the recurrent layer weights are reduced. The performance of the integrated training procedure is illustrated on three different speech processing tasks: emotion recognition, speaker identification and speaker verification. [ABSTRACT FROM AUTHOR]

An extension of the well-known probabilistic neural network (PNN) to generalized locally recurrent PNN (GLR PNN) is introduced. The GLR PNN is derived from the original PNN by incorporating a fully connected recurrent layer between the pattern and output layers. This extension renders GLR PNN sensitive to the context in which events occur, and therefore, capable of identifying temporal and spatial correlations. In the present work, this capability is exploited to improve the speaker verification performance. A fast three-step method for training GLR PNNs is proposed. The first two steps are identical to the training of original PNNs, while the third step is based on the differential evolution (DE) optimization method. [Copyright &y& Elsevier]

The aim of our investigations was to define the influence of reserpine on iron absorption in rats. Iron absorption was determined in vivo at the 24th hour and in situ in a tied-off intestinal segment. Five-days treatment with reserpine (1mg/kg b.wt) led to suppressed bone marrow ertyhropoiesis, manifested a considerable decrease in reticulocyte count in a percentage of (sup)59Fe-incorporation into newly formed erythrocytes. Increased plasma iron and decreased plasma ferritin levels were found. Iron absorption was found to be significantly increased, in spite of the suppressed erythropoiesis. The major conlusions of the experiments are, that reserpine interferes with the regulation of iron metabolism and erythropoiesis. [ABSTRACT FROM AUTHOR]

Acoustic monitoring provides the opportunity to study ecological processes that are difficult to assess with traditional surveys. Elachistocleis matogrosso is an anuran species, described in 2010, for which limited biological information is available. This study investigated the calling activity of the species in the north-eastern portion of the Pantanal, Brazil, a wetland area with marked seasonality between the dry and wet seasons. The calling activity of E. matogrosso was monitored using automated digital recorders in combination with automated signal recognition software over two different annual cycles. The species was vocally active only during the wet season (October – April), with a peak in November-December during the 2013–2014 annual cycle and in February-March during the 2015–2016 annual cycle. The peak calling activity occurred at dusk. This species has nocturnal habits and an explosive breeding activity. The detection of the species was intermittent, which suggests that environmental predictors or site-specific conditions might play an important role in species detection. Moreover, this intermittent occupancy indicated that surveys that employ traditional field techniques would likely fail to detect this species. We describe an effective protocol for detecting E. matogrosso with acoustic monitoring, which requires recording during 20 days in February from 17:01 to 05:00. Our procedure would be easy to adapt to other anuran species, and it could be used for investigating new localities and assessing population changes over time. [ABSTRACT FROM AUTHOR]

A major challenge in the automated acoustic recognition of bird species is the audio segmentation, which aims to select portions of audio that contain meaningful sound events and eliminates segments that contain predominantly background noise or sound events of other origin. Here we report on the development of an audio parameterization method with integrated robust frame selection that makes use of morphological filtering applied on the spectrogram seen as an image. The morphological filtering allows to exclude from further processing certain audio events, which otherwise could cause misclassification errors. The Mel Frequency Cepstral Coefficients (MFCCs) computed for the selected audio frames offer a good representation of the spectral information for dominant vocalizations because the morphological filtering eliminates short bursts of noise and suppresses weak competing signals. Experimental validation of the proposed method on the identification of 40 bird species from Brazil demonstrated superior accuracy and faster operation than three traditional and recent approaches. This is expressed as reduction of the relative error rate by 3.4% and the overall operational time by 7.5% when compared to the second best result. The improved frame selection robustness, precision, and operational speed facilitate applications like multi-species identification of real-field recordings. [ABSTRACT FROM AUTHOR]

Audio event recognition methods based on the Hidden Markov Model / Gaussian Mixture Model (HMM/GMM) often depend on a large number of mixture components or multi-stage models that require significant computational and memory resources during their operation. A widely used approach for coping with complexity is employing an acoustic activity detector, which selects for further processing only those portions of the audio that are considered promising. As a result, the audio feature extraction and the subsequent pattern recognition stages will process only a subset of the original audio stream, helping to reduce the misclassification rates while lowering the computational demands. In the present work we propose a method for bird acoustic activity detection, based on morphological filtering of the spectrogram seen as an image. The practical significance of the proposed method is validated on the automated acoustic recognition of Southern Lapwing Vanellus chilensis , a common Neotropical bird species. Compared with other methods of acoustic activity detection it demonstrates advantageous performance. [ABSTRACT FROM AUTHOR]

The MoveOn speech and noise database was purposely designed and implemented in support of research on spoken dialogue interaction in a motorcycle environment. The distinctiveness of the MoveOn database results from the requirements of the application domain-an information support and operational command and control system for the two-wheel police force-and also from the specifics of the adverse open-air acoustic environment. In this article, we first outline the target application, motivating the database design and purpose, and then report on the implementation details. The main challenges related to the choice of equipment, the organization of recording sessions, and some difficulties that were experienced during this effort, are discussed. We offer a detailed account of the database statistics, the suggested data splits in subsets, and discuss results from automatic speech recognition experiments which illustrate the degree of complexity of the operational environment. [ABSTRACT FROM AUTHOR]

Abstract: In the present work, we propose a scheme for the fusion of different phone duration models, operating in parallel. Specifically, the predictions from a group of dissimilar and independent to each other individual duration models are fed to a machine learning algorithm, which reconciles and fuses the outputs of the individual models, yielding more precise phone duration predictions. The performance of the individual duration models and of the proposed fusion scheme is evaluated on the American-English KED TIMIT and on the Greek WCL-1 databases. On both databases, the SVR-based individual model demonstrates the lowest error rate. When compared to the second-best individual algorithm, a relative reduction of the mean absolute error (MAE) and the root mean square error (RMSE) by 5.5% and 3.7% on KED TIMIT, and 6.8% and 3.7% on WCL-1 is achieved. At the fusion stage, we evaluate the performance of 12 fusion techniques. The proposed fusion scheme, when implemented with SVR-based fusion, contributes to the improvement of the phone duration prediction accuracy over the one of the best individual model, by 1.9% and 2.0% in terms of relative reduction of the MAE and RMSE on KED TIMIT, and by 2.6% and 1.8% on the WCL-1 database. [ABSTRACT FROM AUTHOR]

Abstract: Aiming at robust spoken dialogue interaction in motorcycle environment, we investigate various configurations for a speech front-end, which consists of speech pre-processing, speech enhancement and speech recognition components. These components are implemented as agents in the Olympus/RavenClaw framework, which is the core of a multimodal dialogue interaction interface of a wearable solution for information support of the motorcycle police force on the move. In the present effort, aiming at optimizing the speech recognition performance, different experimental setups are considered for the speech front-end. The practical value of various speech enhancement techniques is assessed and, after analysis of their performances, a collaborative scheme is proposed. In this collaborative scheme independent speech enhancement channels operate in parallel on a common input and their outputs are fed to the multithread speech recognition component. The outcome of the speech recognition process is post-processed by an appropriate fusion technique, which contributes for a more accurate interpretation of the input. Investigating various fusion algorithms, we identified the Adaboost.M1 algorithm as the one performing best. Utilizing the fusion collaborative scheme based on the Adaboost.M1 algorithm, significant improvement of the overall speech recognition performance was achieved. This is expressed in terms of word recognition rate and correctly recognized words, as accuracy gain of 8.0% and 5.48%, respectively, when compared to the performance of the best speech enhancement channel, alone. The advance offered in the present work reaches beyond the specifics of the present application, and can be beneficial to spoken interfaces operating in non-stationary noise environments. [Copyright &y& Elsevier]

This work presents an integrated system that uses speech as a natural input modality to provide user-friendly access to information and entertainment devices installed in a real home environment The practical limitations introduced by the on-line nature of the application as well as the implementation challenges and solutions are analyzed. The focus of the present study is on the implementation on the front-end signal pre-processing block that consist of an array of 8 microphones connected to a multi-channel soundcard and a tandem of workstations performing all signal pre-processing tasks, such as, acquisition, filtering, and beamforming. Evaluation of the beamformer's performance in realistic home environment with controllable noise sources is provided. Furthermore, speech and speaker recognition results using the front-end that was deployed are presented. [ABSTRACT FROM AUTHOR]

This study evaluates the relationship between shape and size of tegmen, harp, mirror, and spectral range of calling song frequency of a Neotropical cricket subpopulation (Lerneca inalata beripocone. In addition, we compare intraspecific morphological divergence and calling song properties between individuals from different sites of the Pantanal of Poconé, Mato Grosso, Brazil. Regression analysis showed that the dominant and maximum calling song frequencies were negatively correlated with tegmen size, i.e. frequencies are either lower or higher depending on the corresponding size variation in resonance structures of the forewings. Canonical variable analysis demonstrated marked intraspecific differences in morphometric characters between localities of a L. inalata subpopulation c. 35 km apart (SESC-Pantanal Advanced Research Base and Pouso Alegre Farm, Mato Grosso, Brazil). Lerneca inalata beripocone at SESC had larger forewings than conspecifics from Pouso Alegre Farm. These morphological variations of wing properties related to reproductive behaviours were interpreted as fitness parameters, likely shaped by restricted gene flow during temporal habitat isolation episodes. Such isolation patterns occur in the Pantanal wetlands for several months during the annual hydrological cycle. [ABSTRACT FROM AUTHOR]

Abstract: We describe a novel design, implementation and evaluation of a speech interface, as part of a platform for the development of serious games. The speech interface consists of the speech recognition component and the emotion recognition from speech component. The speech interface relies on a platform designed and implemented to support the development of serious games, which supports cognitive-based treatment of patients with mental disorders. The implementation of the speech interface is based on the Olympus/RavenClaw framework. This framework has been extended for the needs of the specific serious games and the respective application domain, by integrating new components, such as emotion recognition from speech. The evaluation of the speech interface utilized purposely collected domain-specific dataset. The speech recognition experiments show that emotional speech moderately affects the performance of the speech interface. Furthermore, the emotion detectors demonstrated satisfying performance for the emotion states of interest, Anger and Boredom, and contributed towards successful modelling of the patient’s emotion status. The performance achieved for speech recognition and for the detection of the emotional states of interest was satisfactory. Recent evaluation of the serious games showed that the patients started to show new coping styles with negative emotions in normal stress life situations. [Copyright &y& Elsevier]

Previous review studies have suggested that computer games can serve as an alternative or additional form of treatment in several areas (schizophrenia, asthma or motor rehabilitation). Although several naturalistic studies have been conducted showing the usefulness of serious video games in the treatment of some abnormal behaviours, there is a lack of serious games specially designed for treating mental disorders. The purpose of our project was to develop and evaluate a serious video game designed to remediate attitudinal, behavioural and emotional processes of patients with impulse-related disorders. The video game was created and developed within the European research project PlayMancer. It aims to prove potential capacity to change underlying attitudinal, behavioural and emotional processes of patients with impulse-related disorders. New interaction modes were provided by newly developed components, such as emotion recognition from speech, face and physiological reactions, while specific impulsive reactions were elicited. The video game uses biofeedback for helping patients to learn relaxation skills, acquire better self-control strategies and develop new emotional regulation strategies. In this article, we present a description of the video game used, rationale, user requirements, usability and preliminary data, in several mental disorders. [ABSTRACT FROM AUTHOR]