Search results for: Thai Speech Therapy
471 The Effectiveness of Metaphor Therapy on Depression among Female Students
Authors: Marzieh Talebzadeh Shoushtari
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The present study aimed to determine the effectiveness of Metaphor therapy on depression among female students. The sample included 60 female students with depression symptoms selected by simple sampling and randomly divided into two equal groups (experimental and control groups). Beck Depression Inventory was used to measure the variables. This was an experimental study with a pre-test/post-test design with control group. Eight metaphor therapy sessions were held for the experimental group. A post-test was administered to both groups. Data were analyzed using multivariate analysis of covariance (MANCOVA). Results showed that the Metaphor therapy decreased depression in the experimental group compared to the control group.Keywords: Metaphor therapy, depression, female, students.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 2722470 Orchestra/Percussion Classification Algorithm for United Speech Audio Coding System
Authors: Yueming Wang, Rendong Ying, Sumxin Jiang, Peilin Liu
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Unified Speech Audio Coding (USAC), the latest MPEG standardization for unified speech and audio coding, uses a speech/audio classification algorithm to distinguish speech and audio segments of the input signal. The quality of the recovered audio can be increased by well-designed orchestra/percussion classification and subsequent processing. However, owing to the shortcoming of the system, introducing an orchestra/percussion classification and modifying subsequent processing can enormously increase the quality of the recovered audio. This paper proposes an orchestra/percussion classification algorithm for the USAC system which only extracts 3 scales of Mel-Frequency Cepstral Coefficients (MFCCs) rather than traditional 13 scales of MFCCs and use Iterative Dichotomiser 3 (ID3) Decision Tree rather than other complex learning method, thus the proposed algorithm has lower computing complexity than most existing algorithms. Considering that frequent changing of attributes may lead to quality loss of the recovered audio signal, this paper also design a modified subsequent process to help the whole classification system reach an accurate rate as high as 97% which is comparable to classical 99%.
Keywords: ID3 Decision Tree, MFCC, Orchestra/Percussion Classification, USAC
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 1658469 A High Quality Speech Coder at 600 bps
Authors: Yong Zhang, Ruimin Hu
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This paper presents a vocoder to obtain high quality synthetic speech at 600 bps. To reduce the bit rate, the algorithm is based on a sinusoidally excited linear prediction model which extracts few coding parameters, and three consecutive frames are grouped into a superframe and jointly vector quantization is used to obtain high coding efficiency. The inter-frame redundancy is exploited with distinct quantization schemes for different unvoiced/voiced frame combinations in the superframe. Experimental results show that the quality of the proposed coder is better than that of 2.4kbps LPC10e and achieves approximately the same as that of 2.4kbps MELP and with high robustness.
Keywords: Speech coding, Vector quantization, linear predicition, Mixed sinusoidal excitation
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 2173468 An ICA Algorithm for Separation of Convolutive Mixture of Speech Signals
Authors: Rajkishore Prasad, Hiroshi Saruwatari, Kiyohiro Shikano
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This paper describes Independent Component Analysis (ICA) based fixed-point algorithm for the blind separation of the convolutive mixture of speech, picked-up by a linear microphone array. The proposed algorithm extracts independent sources by non- Gaussianizing the Time-Frequency Series of Speech (TFSS) in a deflationary way. The degree of non-Gaussianization is measured by negentropy. The relative performances of algorithm under random initialization and Null beamformer (NBF) based initialization are studied. It has been found that an NBF based initial value gives speedy convergence as well as better separation performance
Keywords: Blind signal separation, independent component analysis, negentropy, convolutive mixture.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 1757467 An Advanced Method for Speech Recognition
Authors: Meysam Mohamad pour, Fardad Farokhi
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In this paper in consideration of each available techniques deficiencies for speech recognition, an advanced method is presented that-s able to classify speech signals with the high accuracy (98%) at the minimum time. In the presented method, first, the recorded signal is preprocessed that this section includes denoising with Mels Frequency Cepstral Analysis and feature extraction using discrete wavelet transform (DWT) coefficients; Then these features are fed to Multilayer Perceptron (MLP) network for classification. Finally, after training of neural network effective features are selected with UTA algorithm.Keywords: Multilayer perceptron (MLP) neural network, Discrete Wavelet Transform (DWT) , Mels Scale Frequency Filter , UTA algorithm.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 2335466 The Effect of Different Compression Schemes on Speech Signals
Authors: Jalal Karam, Raed Saad
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This paper studies the effect of different compression constraints and schemes presented in a new and flexible paradigm to achieve high compression ratios and acceptable signal to noise ratios of Arabic speech signals. Compression parameters are computed for variable frame sizes of a level 5 to 7 Discrete Wavelet Transform (DWT) representation of the signals for different analyzing mother wavelet functions. Results are obtained and compared for Global threshold and level dependent threshold techniques. The results obtained also include comparisons with Signal to Noise Ratios, Peak Signal to Noise Ratios and Normalized Root Mean Square Error.Keywords: Speech Compression, Wavelets.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 1716465 Bidirectional Dynamic Time Warping Algorithm for the Recognition of Isolated Words Impacted by Transient Noise Pulses
Authors: G. Tamulevičius, A. Serackis, T. Sledevič, D. Navakauskas
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We consider the biggest challenge in speech recognition – noise reduction. Traditionally detected transient noise pulses are removed with the corrupted speech using pulse models. In this paper we propose to cope with the problem directly in Dynamic Time Warping domain. Bidirectional Dynamic Time Warping algorithm for the recognition of isolated words impacted by transient noise pulses is proposed. It uses simple transient noise pulse detector, employs bidirectional computation of dynamic time warping and directly manipulates with warping results. Experimental investigation with several alternative solutions confirms effectiveness of the proposed algorithm in the reduction of impact of noise on recognition process – 3.9% increase of the noisy speech recognition is achieved.
Keywords: Transient noise pulses, noise reduction, dynamic time warping, speech recognition.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 1920464 Performance, Carcass Yield, Hematological Parameters, and Feather Pecking Damage of Thai Indigenous Chickens Raised Indoors or with Outdoor Access
Authors: W. Molee, P. Puttaraksa, S. Pitakwong, S. Khempaka
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An experiment was conducted to determine the effect of the rearing system on growth performance, carcass yield, hematological parameters, and feather pecking damage of Thai indigenous chickens. Three hundred and sixty 1-d-old chicks were randomly assigned to 2 treatments: indoor treatment and outdoor access treatment. In the indoor treatment, the chickens were housed in floor pens (5 birds/m2). In the outdoor access treatment, the chickens were housed in a similar indoor house; in addition, they also had an outdoor grass paddock (1 bird/m2). All birds were provided with same diet and were raised for 16 wk of age. The results showed that growth performance and carcass yield were not different among treatment (P>0.05). Outdoor access had no effect on hematological parameters (P>0.05). However, the feather pecking damage of the chickens in the outdoor access treatment was lower than that of the chickens in the indoor treatment (P<0.05).Keywords: Hematology, performance, rearing system, Thai indigenous chickens
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 1825463 Marketing Mix for Tourism in the Chonburi Province
Authors: Pisit Potjanajaruwit
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The objectives of the study were to determine the marketing mix factors that influencing tourist’s destination decision making for cultural tourism in the Chonburi province. Both quantitative and qualitative data were applied in this study. The samples of 400 cases for quantitative analysis were tourists (both Thai and foreign) who were interested in cultural tourism in the Chonburi province, and traveled to cultural sites in Chonburi and 14 representatives from provincial tourism committee of Chonburi and local tourism experts. Statistics utilized in this research included frequency, percentage, mean, standard deviation, and multiple regression analysis. The study found that Thai and foreign tourists are influenced by different important marketing mix factors. The important factors for Thai respondents were physical evidence, price, people, and place at high importance level. For foreign respondents, physical evidence, price, people, and process were high importance level, whereas, product, place and promotion were moderate importance level.
Keywords: Chonburi Province, Decision Making for cultural tourism, Marketing Mixed.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 3581462 Unit Selection Algorithm Using Bi-grams Model For Corpus-Based Speech Synthesis
Authors: Mohamed Ali KAMMOUN, Ahmed Ben HAMIDA
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In this paper, we present a novel statistical approach to corpus-based speech synthesis. Classically, phonetic information is defined and considered as acoustic reference to be respected. In this way, many studies were elaborated for acoustical unit classification. This type of classification allows separating units according to their symbolic characteristics. Indeed, target cost and concatenation cost were classically defined for unit selection. In Corpus-Based Speech Synthesis System, when using large text corpora, cost functions were limited to a juxtaposition of symbolic criteria and the acoustic information of units is not exploited in the definition of the target cost. In this manuscript, we token in our consideration the unit phonetic information corresponding to acoustic information. This would be realized by defining a probabilistic linguistic Bi-grams model basically used for unit selection. The selected units would be extracted from the English TIMIT corpora.Keywords: Unit selection, Corpus-based Speech Synthesis, Bigram model
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 1421461 Puff Noise Detection and Cancellation for Robust Speech Recognition
Authors: Sangjun Park, Jungpyo Hong, Byung-Ok Kang, Yun-keun Lee, Minsoo Hahn
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In this paper, an algorithm for detecting and attenuating puff noises frequently generated under the mobile environment is proposed. As a baseline system, puff detection system is designed based on Gaussian Mixture Model (GMM), and 39th Mel Frequency Cepstral Coefficient (MFCC) is extracted as feature parameters. To improve the detection performance, effective acoustic features for puff detection are proposed. In addition, detected puff intervals are attenuated by high-pass filtering. The speech recognition rate was measured for evaluation and confusion matrix and ROC curve are used to confirm the validity of the proposed system.Keywords: Gaussian mixture model, puff detection and cancellation, speech enhancement.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 2211460 Human Capital Development for ASEAN Community
Authors: Chutikarn Sriwiboon
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The main purpose of this research paper was to study the requirements for human capital development in order to be ready for ASEAN Community. Thai education institutions are encountering a challenging course of change to be effective members of ASEAN Economic Community (AEC) in 2015. It was vital that everyone and every organization participate in the process of becoming part of the ASEAN community, a pluralistic society. Thai universities will be required to partake in the human capital development in a variety of fields. In order to assist the whole nation to enhance potential development, there was a need to collaborate with other ASEAN leading universities to do researches to ameliorate the qualifications and capabilities of university management, administers, professors, and staffs.Keywords: ASEAN, Education, Human capital development.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 2719459 Analysis of Linguistic Disfluencies in Bilingual Children’s Discourse
Authors: Sheena Christabel Pravin, M. Palanivelan
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Speech disfluencies are common in spontaneous speech. The primary purpose of this study was to distinguish linguistic disfluencies from stuttering disfluencies in bilingual Tamil–English (TE) speaking children. The secondary purpose was to determine whether their disfluencies are mediated by native language dominance and/or on an early onset of developmental stuttering at childhood. A detailed study was carried out to identify the prosodic and acoustic features that uniquely represent the disfluent regions of speech. This paper focuses on statistical modeling of repetitions, prolongations, pauses and interjections in the speech corpus encompassing bilingual spontaneous utterances from school going children – English and Tamil. Two classifiers including Hidden Markov Models (HMM) and the Multilayer Perceptron (MLP), which is a class of feed-forward artificial neural network, were compared in the classification of disfluencies. The results of the classifiers document the patterns of disfluency in spontaneous speech samples of school-aged children to distinguish between Children Who Stutter (CWS) and Children with Language Impairment CLI). The ability of the models in classifying the disfluencies was measured in terms of F-measure, Recall, and Precision.
Keywords: Bilingual, children who stutter, children with language impairment, Hidden Markov Models, multi-layer perceptron, linguistic disfluencies, stuttering disfluencies.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 1009458 Applications of Support Vector Machines on Smart Phone Systems for Emotional Speech Recognition
Authors: Wernhuar Tarng, Yuan-Yuan Chen, Chien-Lung Li, Kun-Rong Hsie, Mingteh Chen
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An emotional speech recognition system for the applications on smart phones was proposed in this study to combine with 3G mobile communications and social networks to provide users and their groups with more interaction and care. This study developed a mechanism using the support vector machines (SVM) to recognize the emotions of speech such as happiness, anger, sadness and normal. The mechanism uses a hierarchical classifier to adjust the weights of acoustic features and divides various parameters into the categories of energy and frequency for training. In this study, 28 commonly used acoustic features including pitch and volume were proposed for training. In addition, a time-frequency parameter obtained by continuous wavelet transforms was also used to identify the accent and intonation in a sentence during the recognition process. The Berlin Database of Emotional Speech was used by dividing the speech into male and female data sets for training. According to the experimental results, the accuracies of male and female test sets were increased by 4.6% and 5.2% respectively after using the time-frequency parameter for classifying happy and angry emotions. For the classification of all emotions, the average accuracy, including male and female data, was 63.5% for the test set and 90.9% for the whole data set.Keywords: Smart phones, emotional speech recognition, socialnetworks, support vector machines, time-frequency parameter, Mel-scale frequency cepstral coefficients (MFCC).
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 1828457 Marketing Strategy Analysis of Thai Asia Pacific Brewery Company
Authors: Sinee Sankrusme
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The study was a case study analysis about Thai Asia Pacific Brewery Company. The purpose was to analyze the company’s marketing objective, marketing strategy at company level, and marketing mix before liquor liberalization in 2000. Methods used in this study were qualitative and descriptive research approach which demonstrated the following results of the study demonstrated as follows: (1) Marketing objective was to increase market share of Heineken and Amtel, (2) the company’s marketing strategies were brand building strategy and distribution strategy. Additionally, the company also conducted marketing mix strategy as follows. Product strategy: The company added more beer brands namely Amstel and Tiger to provide additional choice to consumers, product and marketing research, and product development. Price strategy: the company had taken the following into consideration: cost, competitor, market, economic situation and tax. Promotion strategy: the company conducted sales promotion and advertising. Distribution strategy: the company extended channels its channels of distribution into food shops, pubs and various entertainment places. This strategy benefited interested persons and people who were engaged in the beer business.
Keywords: Marketing Strategy, Beer, Thai Asia Pacific Brewery Company.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 6215456 Applications of Conic Optimization and Quadratic Programming in the Investigation of Index Arbitrage in the Thai Derivatives and Equity Markets
Authors: Satjaporn Tungsong, Gun Srijuntongsiri
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This research seeks to investigate the frequency and profitability of index arbitrage opportunities involving the SET50 futures, SET50 component stocks, and the ThaiDEX SET50 ETF (ticker symbol: TDEX). In particular, the frequency and profit of arbitrage are measured in the following three arbitrage tests: (1) SET50 futures vs. ThaiDEX SET50 ETF, (2) SET50 futures vs. SET50 component stocks, and (3) ThaiDEX SET50 ETF vs. SET50 component stocks are investigated. For tests (2) and (3), the problems involve conic optimization and quadratic programming as subproblems. This research is first to apply conic optimization and quadratic programming techniques in the context of index arbitrage and is first to investigate such index arbitrage in the Thai equity and derivatives markets. Thus, the contribution of this study is twofold. First, its results would help understand the contribution of the derivatives securities to the efficiency of the Thai markets. Second, the methodology employed in this study can be applied to other geographical markets, with minor adjustments.Keywords: Conic optimization, Equity index arbitrage, Executionlags, Quadratic programming, SET50 index futures, ThaiDEX SET50ETF, Transaction costs
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 1558455 Voice Disorders Identification Using Hybrid Approach: Wavelet Analysis and Multilayer Neural Networks
Authors: L. Salhi, M. Talbi, A. Cherif
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This paper presents a new strategy of identification and classification of pathological voices using the hybrid method based on wavelet transform and neural networks. After speech acquisition from a patient, the speech signal is analysed in order to extract the acoustic parameters such as the pitch, the formants, Jitter, and shimmer. Obtained results will be compared to those normal and standard values thanks to a programmable database. Sounds are collected from normal people and patients, and then classified into two different categories. Speech data base is consists of several pathological and normal voices collected from the national hospital “Rabta-Tunis". Speech processing algorithm is conducted in a supervised mode for discrimination of normal and pathology voices and then for classification between neural and vocal pathologies (Parkinson, Alzheimer, laryngeal, dyslexia...). Several simulation results will be presented in function of the disease and will be compared with the clinical diagnosis in order to have an objective evaluation of the developed tool.Keywords: Formants, Neural Networks, Pathological Voices, Pitch, Wavelet Transform.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 2827454 Preliminary Study of the Phonological Development in Three- and Four-Year-Old Bulgarian Children
Authors: Tsvetomira Braynova, Miglena Simonska
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The article presents the results of a research of phonological processes in three- and four-year-old children. A test, created for the purpose of the study, was developed and conducted among 120 children. The study included three areas of research - at the level of words (96 words), at the level of sentence repetition (10 sentences) and at the level of generating own speech from a picture (15 pictures). The test also gives us additional information about the articulation errors of the assessed children. The main purpose of the research is to analyze all phonological processes that occur at this age in Bulgarian children and to identify which are typical and atypical for this age. The results show that the most common phonology errors that children make are: sound substitution, elision of sound, metathesis of sound, elision of syllable, elision of consonants clustered in a syllable. Measuring the correlation between average length of repeated speech and average length of generated speech, the analysis does not prove that the more words a child can repeat in part “repeated speech”, the more words they can be expected to generate in part “generating sentence”. The results of this study show that the task of naming a word provides sufficient and representative information to assess the child's phonology.
Keywords: Articulation, phonology, speech, language development.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 353453 Continuous Feature Adaptation for Non-Native Speech Recognition
Authors: Y. Deng, X. Li, C. Kwan, B. Raj, R. Stern
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The current speech interfaces in many military applications may be adequate for native speakers. However, the recognition rate drops quite a lot for non-native speakers (people with foreign accents). This is mainly because the nonnative speakers have large temporal and intra-phoneme variations when they pronounce the same words. This problem is also complicated by the presence of large environmental noise such as tank noise, helicopter noise, etc. In this paper, we proposed a novel continuous acoustic feature adaptation algorithm for on-line accent and environmental adaptation. Implemented by incremental singular value decomposition (SVD), the algorithm captures local acoustic variation and runs in real-time. This feature-based adaptation method is then integrated with conventional model-based maximum likelihood linear regression (MLLR) algorithm. Extensive experiments have been performed on the NATO non-native speech corpus with baseline acoustic model trained on native American English. The proposed feature-based adaptation algorithm improved the average recognition accuracy by 15%, while the MLLR model based adaptation achieved 11% improvement. The corresponding word error rate (WER) reduction was 25.8% and 2.73%, as compared to that without adaptation. The combined adaptation achieved overall recognition accuracy improvement of 29.5%, and WER reduction of 31.8%, as compared to that without adaptation. Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 3199452 Simulation of Climate Variability for Assessing Impacts on Yield and Genetic Change of Thai Soybean
Authors: Kanita Thanacharoenchanaphas, Orose Rugchati
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This study assessed the effects of climate change on Thai soybeans under simulation situations. Our study is focused on temperature variability and effects on growth, yield, and genetic changes in 2 generations of Chiang Mai 60 cultivars. In the experiment, soybeans were exposed to 3 levels of air temperature for 8 h day-1 in an open top chamber for 2 cropping periods. Air temperature levels in each treatment were controlled at 30-33°C (± 2.3) for LT-treatment, 33-36°C ( ± 2.4) for AT-treatment, and 36-40 °C ( ± 3.2) for HT-treatment, respectively. Positive effects of high temperature became obvious at the maturing stage when yield significantly increased in both cropping periods. Results in growth indicated that shoot length at the pre-maturing stage (V3-R3) was more positively affected by high temperature than at the maturing stage. However, the positive effect on growth under high temperature was not found in the 2nd cropping period. Finally, genetic changes were examined in phenotype characteristics by the AFLPs technique. The results showed that the high temperature factor clearly caused genetic change in the soybeans and showed more alteration in the 2nd cropping period.Keywords: simulation, air temperature, variability, Thai soybean, yield , genetic change
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 1668451 Automatic Detection of Syllable Repetition in Read Speech for Objective Assessment of Stuttered Disfluencies
Authors: K. M. Ravikumar, Balakrishna Reddy, R. Rajagopal, H. C. Nagaraj
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Automatic detection of syllable repetition is one of the important parameter in assessing the stuttered speech objectively. The existing method which uses artificial neural network (ANN) requires high levels of agreement as prerequisite before attempting to train and test ANNs to separate fluent and nonfluent. We propose automatic detection method for syllable repetition in read speech for objective assessment of stuttered disfluencies which uses a novel approach and has four stages comprising of segmentation, feature extraction, score matching and decision logic. Feature extraction is implemented using well know Mel frequency Cepstra coefficient (MFCC). Score matching is done using Dynamic Time Warping (DTW) between the syllables. The Decision logic is implemented by Perceptron based on the score given by score matching. Although many methods are available for segmentation, in this paper it is done manually. Here the assessment by human judges on the read speech of 10 adults who stutter are described using corresponding method and the result was 83%.Keywords: Assessment, DTW, MFCC, Objective, Perceptron, Stuttering.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 2782450 A Mixing Matrix Estimation Algorithm for Speech Signals under the Under-Determined Blind Source Separation Model
Authors: Jing Wu, Wei Lv, Yibing Li, Yuanfan You
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The separation of speech signals has become a research hotspot in the field of signal processing in recent years. It has many applications and influences in teleconferencing, hearing aids, speech recognition of machines and so on. The sounds received are usually noisy. The issue of identifying the sounds of interest and obtaining clear sounds in such an environment becomes a problem worth exploring, that is, the problem of blind source separation. This paper focuses on the under-determined blind source separation (UBSS). Sparse component analysis is generally used for the problem of under-determined blind source separation. The method is mainly divided into two parts. Firstly, the clustering algorithm is used to estimate the mixing matrix according to the observed signals. Then the signal is separated based on the known mixing matrix. In this paper, the problem of mixing matrix estimation is studied. This paper proposes an improved algorithm to estimate the mixing matrix for speech signals in the UBSS model. The traditional potential algorithm is not accurate for the mixing matrix estimation, especially for low signal-to noise ratio (SNR).In response to this problem, this paper considers the idea of an improved potential function method to estimate the mixing matrix. The algorithm not only avoids the inuence of insufficient prior information in traditional clustering algorithm, but also improves the estimation accuracy of mixing matrix. This paper takes the mixing of four speech signals into two channels as an example. The results of simulations show that the approach in this paper not only improves the accuracy of estimation, but also applies to any mixing matrix.Keywords: Clustering algorithm, potential function, speech signal, the UBSS model.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 655449 Pilot Trial of Evidence-Based Integrative Group Therapy to Improve Executive Functioning among Adults: Implications for Community Mental Health and Training Clinics
Authors: B. Parchem, M. Watanabe, D. Modrakovic, L. Mathew, A. Franklin, M. Cao, R. E. Broudy
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Objective: Executive functioning (EF) deficits underlie several mental health diagnoses including ADHD, anxiety, and depression. Community mental health clinics face extensive waitlists for services with many referrals involving EF deficits. A pilot trial of a four-week group therapy was developed using key components from Cognitive-Behavioral Therapy (CBT), Dialectical Behavior Therapy (DBT), and mindfulness with an aim to improve EF skills and offer low-fee services. Method: Eight adults (M = 34.5) waiting for services at a community clinic were enrolled in a four-week group therapy at an in-house training clinic for doctoral trainees. Baseline EF, pre-/post-intervention ADHD and distress symptoms, group satisfaction, and curriculum helpfulness were assessed. Results: Downward trends in ADHD and distress symptoms pre/post-intervention were not significant. Favorable responses on group satisfaction and helpfulness suggest clinical utility. Conclusion: Preliminary pilot data from a brief group therapy to improve EF may be an efficacious, acceptable, and feasible intervention for adults waiting for services at community mental health and training clinics where there are high demands and limits to services and staffs.Keywords: Executive functioning, cognitive-behavioral therapy, dialectical behavior therapy, mindfulness, adult group therapy.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 747448 Spectral Entropy Employment in Speech Enhancement based on Wavelet Packet
Authors: Talbi Mourad, Salhi Lotfi, Chérif Adnen
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In this work, we are interested in developing a speech denoising tool by using a discrete wavelet packet transform (DWPT). This speech denoising tool will be employed for applications of recognition, coding and synthesis. For noise reduction, instead of applying the classical thresholding technique, some wavelet packet nodes are set to zero and the others are thresholded. To estimate the non stationary noise level, we employ the spectral entropy. A comparison of our proposed technique to classical denoising methods based on thresholding and spectral subtraction is made in order to evaluate our approach. The experimental implementation uses speech signals corrupted by two sorts of noise, white and Volvo noises. The obtained results from listening tests show that our proposed technique is better than spectral subtraction. The obtained results from SNR computation show the superiority of our technique when compared to the classical thresholding method using the modified hard thresholding function based on u-law algorithm.
Keywords: Enhancement, spectral subtraction, SNR, discrete wavelet packet transform, spectral entropy Histogram
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 1967447 Study of Incineration of Acacia Wood Chips for Biomass Power Plant of the Royal Thai Navy in Sattahip, Chonburi Province, Thailand
Authors: Thanapong Suriyea, Teeratas Pornyungyuen, Koonlaya Kanokjaruvijit
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This research is aimed to find optimal values of parameters of acacia wood chips combustion in a bubbling fluidized bed for electrification within the area of the Royal Thai Navy in Sattahip, Chonburi province, Thailand. The size of wood chips falls in the range of 25 mm in diameter. The bed temperature is set within the range of 800±10 oC with the air flow rate of 2.1-3.1 m/min corresponding to the air-fuel ratio between 0.71 to 1.03. The resulting thermal efficiency is approximately 95% with a thermal output of 474.76 kWth, which produced the electricity 0.131 kW-hr.
Keywords: Acacia Wood Chips, Biomass, Combustion, Fluidized Bed.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 2688446 Bangla Vowel Characterization Based on Analysis by Synthesis
Authors: Syed Akhter Hossain, M. Lutfar Rahman, Farruk Ahmed
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Bangla Vowel characterization determines the spectral properties of Bangla vowels for efficient synthesis as well as recognition of Bangla vowels. In this paper, Bangla vowels in isolated word have been analyzed based on speech production model within the framework of Analysis-by-Synthesis. This has led to the extraction of spectral parameters for the production model in order to produce different Bangla vowel sounds. The real and synthetic spectra are compared and a weighted square error has been computed along with the error in the formant bandwidths for efficient representation of Bangla vowels. The extracted features produced good representation of targeted Bangla vowel. Such a representation also plays essential role in low bit rate speech coding and vocoders.
Keywords: Speech, vowel, formant, synthesis, spectrum, LPC.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 2353445 Speech Recognition Using Scaly Neural Networks
Authors: Akram M. Othman, May H. Riadh
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This research work is aimed at speech recognition using scaly neural networks. A small vocabulary of 11 words were established first, these words are “word, file, open, print, exit, edit, cut, copy, paste, doc1, doc2". These chosen words involved with executing some computer functions such as opening a file, print certain text document, cutting, copying, pasting, editing and exit. It introduced to the computer then subjected to feature extraction process using LPC (linear prediction coefficients). These features are used as input to an artificial neural network in speaker dependent mode. Half of the words are used for training the artificial neural network and the other half are used for testing the system; those are used for information retrieval. The system components are consist of three parts, speech processing and feature extraction, training and testing by using neural networks and information retrieval. The retrieve process proved to be 79.5-88% successful, which is quite acceptable, considering the variation to surrounding, state of the person, and the microphone type.Keywords: Feature extraction, Liner prediction coefficients, neural network, Speech Recognition, Scaly ANN.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 1718444 Production of Glucose from the Hydrolysis of Cassava Residue using Bacteria Isolates from Thai Higher Termites
Authors: Pitcha Wongskeo, Pramoch Rangsunvigit, Sumaeth Chavadej
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The possibility of using cassava residue containing 49.66% starch, 21.47% cellulose, 12.97% hemicellulose, and 21.86% lignin as a raw material to produce glucose using enzymatic hydrolysis was investigated. In the experiment, each reactor contained the cassava residue, bacteria cells, and production medium. The effects of particles size (40 mesh and 60 mesh) and strains of bacteria (A002 and M015) isolated from Thai higher termites, Microcerotermes sp., on the glucose concentration at 37°C were focused. High performance liquid chromatography (HPLC) with a refractive index detector was used to determine the quantity of glucose. The maximum glucose concentration obtained at 37°C using strain A002 and 60 mesh of the cassava residue was 1.51 g/L at 10 h.Keywords: Hydrolysis, termites, glucose, cassava
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 2436443 Forensic Speaker Verification in Noisy Environmental by Enhancing the Speech Signal Using ICA Approach
Authors: Ahmed Kamil Hasan Al-Ali, Bouchra Senadji, Ganesh Naik
Abstract:
We propose a system to real environmental noise and channel mismatch for forensic speaker verification systems. This method is based on suppressing various types of real environmental noise by using independent component analysis (ICA) algorithm. The enhanced speech signal is applied to mel frequency cepstral coefficients (MFCC) or MFCC feature warping to extract the essential characteristics of the speech signal. Channel effects are reduced using an intermediate vector (i-vector) and probabilistic linear discriminant analysis (PLDA) approach for classification. The proposed algorithm is evaluated by using an Australian forensic voice comparison database, combined with car, street and home noises from QUT-NOISE at a signal to noise ratio (SNR) ranging from -10 dB to 10 dB. Experimental results indicate that the MFCC feature warping-ICA achieves a reduction in equal error rate about (48.22%, 44.66%, and 50.07%) over using MFCC feature warping when the test speech signals are corrupted with random sessions of street, car, and home noises at -10 dB SNR.Keywords: Noisy forensic speaker verification, ICA algorithm, MFCC, MFCC feature warping.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 968442 A Smart-Visio Microphone for Audio-Visual Speech Recognition “Vmike“
Abstract:
The practical implementation of audio-video coupled speech recognition systems is mainly limited by the hardware complexity to integrate two radically different information capturing devices with good temporal synchronisation. In this paper, we propose a solution based on a smart CMOS image sensor in order to simplify the hardware integration difficulties. By using on-chip image processing, this smart sensor can calculate in real time the X/Y projections of the captured image. This on-chip projection reduces considerably the volume of the output data. This data-volume reduction permits a transmission of the condensed visual information via the same audio channel by using a stereophonic input available on most of the standard computation devices such as PC, PDA and mobile phones. A prototype called VMIKE (Visio-Microphone) has been designed and realised by using standard 0.35um CMOS technology. A preliminary experiment gives encouraged results. Its efficiency will be further investigated in a large variety of applications such as biometrics, speech recognition in noisy environments, and vocal control for military or disabled persons, etc.
Keywords: Audio-Visual Speech recognition, CMOS Smartsensor, On-Chip image processing.
Procedia APA BibTeX Chicago EndNote Harvard JSON MLA RIS XML ISO 690 PDF Downloads 1808