WorldWideScience

Sample records for machine coded speech

  1. Speech coding

    Energy Technology Data Exchange (ETDEWEB)

    Ravishankar, C., Hughes Network Systems, Germantown, MD

    1998-05-08

    Speech is the predominant means of communication between human beings and since the invention of the telephone by Alexander Graham Bell in 1876, speech services have remained to be the core service in almost all telecommunication systems. Original analog methods of telephony had the disadvantage of speech signal getting corrupted by noise, cross-talk and distortion Long haul transmissions which use repeaters to compensate for the loss in signal strength on transmission links also increase the associated noise and distortion. On the other hand digital transmission is relatively immune to noise, cross-talk and distortion primarily because of the capability to faithfully regenerate digital signal at each repeater purely based on a binary decision. Hence end-to-end performance of the digital link essentially becomes independent of the length and operating frequency bands of the link Hence from a transmission point of view digital transmission has been the preferred approach due to its higher immunity to noise. The need to carry digital speech became extremely important from a service provision point of view as well. Modem requirements have introduced the need for robust, flexible and secure services that can carry a multitude of signal types (such as voice, data and video) without a fundamental change in infrastructure. Such a requirement could not have been easily met without the advent of digital transmission systems, thereby requiring speech to be coded digitally. The term Speech Coding is often referred to techniques that represent or code speech signals either directly as a waveform or as a set of parameters by analyzing the speech signal. In either case, the codes are transmitted to the distant end where speech is reconstructed or synthesized using the received set of codes. A more generic term that is applicable to these techniques that is often interchangeably used with speech coding is the term voice coding. This term is more generic in the sense that the

  2. Principles of speech coding

    CERN Document Server

    Ogunfunmi, Tokunbo

    2010-01-01

    It is becoming increasingly apparent that all forms of communication-including voice-will be transmitted through packet-switched networks based on the Internet Protocol (IP). Therefore, the design of modern devices that rely on speech interfaces, such as cell phones and PDAs, requires a complete and up-to-date understanding of the basics of speech coding. Outlines key signal processing algorithms used to mitigate impairments to speech quality in VoIP networksOffering a detailed yet easily accessible introduction to the field, Principles of Speech Coding provides an in-depth examination of the

  3. An analysis of machine translation and speech synthesis in speech-to-speech translation system

    OpenAIRE

    Hashimoto, K.; Yamagishi, J.; Byrne, W.; King, S.; Tokuda, K.

    2011-01-01

    This paper provides an analysis of the impacts of machine translation and speech synthesis on speech-to-speech translation systems. The speech-to-speech translation system consists of three components: speech recognition, machine translation and speech synthesis. Many techniques for integration of speech recognition and machine translation have been proposed. However, speech synthesis has not yet been considered. Therefore, in this paper, we focus on machine translation and speech synthesis, ...

  4. Speech and audio processing for coding, enhancement and recognition

    CERN Document Server

    Togneri, Roberto; Narasimha, Madihally

    2015-01-01

    This book describes the basic principles underlying the generation, coding, transmission and enhancement of speech and audio signals, including advanced statistical and machine learning techniques for speech and speaker recognition with an overview of the key innovations in these areas. Key research undertaken in speech coding, speech enhancement, speech recognition, emotion recognition and speaker diarization are also presented, along with recent advances and new paradigms in these areas. ·         Offers readers a single-source reference on the significant applications of speech and audio processing to speech coding, speech enhancement and speech/speaker recognition. Enables readers involved in algorithm development and implementation issues for speech coding to understand the historical development and future challenges in speech coding research; ·         Discusses speech coding methods yielding bit-streams that are multi-rate and scalable for Voice-over-IP (VoIP) Networks; ·     �...

  5. Speech coding code- excited linear prediction

    CERN Document Server

    Bäckström, Tom

    2017-01-01

    This book provides scientific understanding of the most central techniques used in speech coding both for advanced students as well as professionals with a background in speech audio and or digital signal processing. It provides a clear connection between the whys hows and whats thus enabling a clear view of the necessity purpose and solutions provided by various tools as well as their strengths and weaknesses in each respect Equivalently this book sheds light on the following perspectives for each technology presented Objective What do we want to achieve and especially why is this goal important Resource Information What information is available and how can it be useful and Resource Platform What kind of platforms are we working with and what are their capabilities restrictions This includes computational memory and acoustic properties and the transmission capacity of devices used. The book goes on to address Solutions Which solutions have been proposed and how can they be used to reach the stated goals and ...

  6. Machine speech and speaking about machines

    Energy Technology Data Exchange (ETDEWEB)

    Nye, A. [Univ. of Wisconsin, Whitewater, WI (United States)

    1996-12-31

    Current philosophy of language prides itself on scientific status. It boasts of being no longer contaminated with queer mental entities or idealist essences. It theorizes language as programmable variants of formal semantic systems, reimaginable either as the properly epiphenomenal machine functions of computer science or the properly material neural networks of physiology. Whether or not such models properly capture the physical workings of a living human brain is a question that scientists will have to answer. I, as a philosopher, come at the problem from another direction. Does contemporary philosophical semantics, in its dominant truth-theoretic and related versions, capture actual living human thought as it is experienced, or does it instead reflect, regardless of (perhaps dubious) scientific credentials, pathology of thought, a pathology with a disturbing social history.

  7. Sparsity in Linear Predictive Coding of Speech

    DEFF Research Database (Denmark)

    Giacobello, Daniele

    of the effectiveness of their application in audio processing. The second part of the thesis deals with introducing sparsity directly in the linear prediction analysis-by-synthesis (LPAS) speech coding paradigm. We first propose a novel near-optimal method to look for a sparse approximate excitation using a compressed...... one with direct applications to coding but also consistent with the speech production model of voiced speech, where the excitation of the all-pole filter can be modeled as an impulse train, i.e., a sparse sequence. Introducing sparsity in the LP framework will also bring to de- velop the concept...... sensing formulation. Furthermore, we define a novel re-estimation procedure to adapt the predictor coefficients to the given sparse excitation, balancing the two representations in the context of speech coding. Finally, the advantages of the compact parametric representation of a segment of speech, given...

  8. Man machine interface based on speech recognition

    International Nuclear Information System (INIS)

    Jorge, Carlos A.F.; Aghina, Mauricio A.C.; Mol, Antonio C.A.; Pereira, Claudio M.N.A.

    2007-01-01

    This work reports the development of a Man Machine Interface based on speech recognition. The system must recognize spoken commands, and execute the desired tasks, without manual interventions of operators. The range of applications goes from the execution of commands in an industrial plant's control room, to navigation and interaction in virtual environments. Results are reported for isolated word recognition, the isolated words corresponding to the spoken commands. For the pre-processing stage, relevant parameters are extracted from the speech signals, using the cepstral analysis technique, that are used for isolated word recognition, and corresponds to the inputs of an artificial neural network, that performs recognition tasks. (author)

  9. Ultra low bit-rate speech coding

    CERN Document Server

    Ramasubramanian, V

    2015-01-01

    "Ultra Low Bit-Rate Speech Coding" focuses on the specialized topic of speech coding at very low bit-rates of 1 Kbits/sec and less, particularly at the lower ends of this range, down to 100 bps. The authors set forth the fundamental results and trends that form the basis for such ultra low bit-rates to be viable and provide a comprehensive overview of various techniques and systems in literature to date, with particular attention to their work in the paradigm of unit-selection based segment quantization. The book is for research students, academic faculty and researchers, and industry practitioners in the areas of speech processing and speech coding.

  10. INTEGRATING MACHINE TRANSLATION AND SPEECH SYNTHESIS COMPONENT FOR ENGLISH TO DRAVIDIAN LANGUAGE SPEECH TO SPEECH TRANSLATION SYSTEM

    Directory of Open Access Journals (Sweden)

    J. SANGEETHA

    2015-02-01

    Full Text Available This paper provides an interface between the machine translation and speech synthesis system for converting English speech to Tamil text in English to Tamil speech to speech translation system. The speech translation system consists of three modules: automatic speech recognition, machine translation and text to speech synthesis. Many procedures for incorporation of speech recognition and machine translation have been projected. Still speech synthesis system has not yet been measured. In this paper, we focus on integration of machine translation and speech synthesis, and report a subjective evaluation to investigate the impact of speech synthesis, machine translation and the integration of machine translation and speech synthesis components. Here we implement a hybrid machine translation (combination of rule based and statistical machine translation and concatenative syllable based speech synthesis technique. In order to retain the naturalness and intelligibility of synthesized speech Auto Associative Neural Network (AANN prosody prediction is used in this work. The results of this system investigation demonstrate that the naturalness and intelligibility of the synthesized speech are strongly influenced by the fluency and correctness of the translated text.

  11. Speech coding, reconstruction and recognition using acoustics and electromagnetic waves

    International Nuclear Information System (INIS)

    Holzrichter, J.F.; Ng, L.C.

    1998-01-01

    The use of EM radiation in conjunction with simultaneously recorded acoustic speech information enables a complete mathematical coding of acoustic speech. The methods include the forming of a feature vector for each pitch period of voiced speech and the forming of feature vectors for each time frame of unvoiced, as well as for combined voiced and unvoiced speech. The methods include how to deconvolve the speech excitation function from the acoustic speech output to describe the transfer function each time frame. The formation of feature vectors defining all acoustic speech units over well defined time frames can be used for purposes of speech coding, speech compression, speaker identification, language-of-speech identification, speech recognition, speech synthesis, speech translation, speech telephony, and speech teaching. 35 figs

  12. Speech coding, reconstruction and recognition using acoustics and electromagnetic waves

    Science.gov (United States)

    Holzrichter, John F.; Ng, Lawrence C.

    1998-01-01

    The use of EM radiation in conjunction with simultaneously recorded acoustic speech information enables a complete mathematical coding of acoustic speech. The methods include the forming of a feature vector for each pitch period of voiced speech and the forming of feature vectors for each time frame of unvoiced, as well as for combined voiced and unvoiced speech. The methods include how to deconvolve the speech excitation function from the acoustic speech output to describe the transfer function each time frame. The formation of feature vectors defining all acoustic speech units over well defined time frames can be used for purposes of speech coding, speech compression, speaker identification, language-of-speech identification, speech recognition, speech synthesis, speech translation, speech telephony, and speech teaching.

  13. Compressed Domain Packet Loss Concealment of Sinusoidally Coded Speech

    DEFF Research Database (Denmark)

    Rødbro, Christoffer A.; Christensen, Mads Græsbøll; Andersen, Søren Vang

    2003-01-01

    We consider the problem of packet loss concealment for voice over IP (VoIP). The speech signal is compressed at the transmitter using a sinusoidal coding scheme working at 8 kbit/s. At the receiver, packet loss concealment is carried out working directly on the quantized sinusoidal parameters......, based on time-scaling of the packets surrounding the missing ones. Subjective listening tests show promising results indicating the potential of sinusoidal speech coding for VoIP....

  14. Ocean circulation code on machine connection

    International Nuclear Information System (INIS)

    Vitart, F.

    1993-01-01

    This work is part of a development of a global climate model based on a coupling between an ocean model and an atmosphere model. The objective was to develop this global model on a massively parallel machine (CM2). The author presents the OPA7 code (equations, boundary conditions, equation system resolution) and parallelization on the CM2 machine. CM2 data structure is briefly evoked, and two tests are reported (on a flat bottom basin, and a topography with eight islands). The author then gives an overview of studies aimed at improving the ocean circulation code: use of a new state equation, use of a formulation of surface pressure, use of a new mesh. He reports the study of the use of multi-block domains on CM2 through advection tests, and two-block tests

  15. Reversible machine code and its abstract processor architecture

    DEFF Research Database (Denmark)

    Axelsen, Holger Bock; Glück, Robert; Yokoyama, Tetsuo

    2007-01-01

    A reversible abstract machine architecture and its reversible machine code are presented and formalized. For machine code to be reversible, both the underlying control logic and each instruction must be reversible. A general class of machine instruction sets was proven to be reversible, building...

  16. Optimal interference code based on machine learning

    Science.gov (United States)

    Qian, Ye; Chen, Qian; Hu, Xiaobo; Cao, Ercong; Qian, Weixian; Gu, Guohua

    2016-10-01

    In this paper, we analyze the characteristics of pseudo-random code, by the case of m sequence. Depending on the description of coding theory, we introduce the jamming methods. We simulate the interference effect or probability model by the means of MATLAB to consolidate. In accordance with the length of decoding time the adversary spends, we find out the optimal formula and optimal coefficients based on machine learning, then we get the new optimal interference code. First, when it comes to the phase of recognition, this study judges the effect of interference by the way of simulating the length of time over the decoding period of laser seeker. Then, we use laser active deception jamming simulate interference process in the tracking phase in the next block. In this study we choose the method of laser active deception jamming. In order to improve the performance of the interference, this paper simulates the model by MATLAB software. We find out the least number of pulse intervals which must be received, then we can make the conclusion that the precise interval number of the laser pointer for m sequence encoding. In order to find the shortest space, we make the choice of the greatest common divisor method. Then, combining with the coding regularity that has been found before, we restore pulse interval of pseudo-random code, which has been already received. Finally, we can control the time period of laser interference, get the optimal interference code, and also increase the probability of interference as well.

  17. Understanding and Writing G & M Code for CNC Machines

    Science.gov (United States)

    Loveland, Thomas

    2012-01-01

    In modern CAD and CAM manufacturing companies, engineers design parts for machines and consumable goods. Many of these parts are cut on CNC machines. Whether using a CNC lathe, milling machine, or router, the ideas and designs of engineers must be translated into a machine-readable form called G & M Code that can be used to cut parts to precise…

  18. A portable virtual machine target for proof-carrying code

    DEFF Research Database (Denmark)

    Franz, Michael; Chandra, Deepak; Gal, Andreas

    2005-01-01

    Virtual Machines (VMs) and Proof-Carrying Code (PCC) are two techniques that have been used independently to provide safety for (mobile) code. Existing virtual machines, such as the Java VM, have several drawbacks: First, the effort required for safety verification is considerable. Second and mor...... simultaneously providing efficient justin-time compilation and target-machine independence. In particular, our approach reduces the complexity of the required proofs, resulting in fewer proof obligations that need to be discharged at the target machine....

  19. Continuous speech recognition with sparse coding

    CSIR Research Space (South Africa)

    Smit, WJ

    2009-04-01

    Full Text Available generative model. The spike train is classified by making use of a spike train model and dynamic programming. It is computationally expensive to find a sparse code. We use an iterative subset selection algorithm with quadratic programming for this process...

  20. Shared acoustic codes underlie emotional communication in music and speech-Evidence from deep transfer learning.

    Directory of Open Access Journals (Sweden)

    Eduardo Coutinho

    Full Text Available Music and speech exhibit striking similarities in the communication of emotions in the acoustic domain, in such a way that the communication of specific emotions is achieved, at least to a certain extent, by means of shared acoustic patterns. From an Affective Sciences points of view, determining the degree of overlap between both domains is fundamental to understand the shared mechanisms underlying such phenomenon. From a Machine learning perspective, the overlap between acoustic codes for emotional expression in music and speech opens new possibilities to enlarge the amount of data available to develop music and speech emotion recognition systems. In this article, we investigate time-continuous predictions of emotion (Arousal and Valence in music and speech, and the Transfer Learning between these domains. We establish a comparative framework including intra- (i.e., models trained and tested on the same modality, either music or speech and cross-domain experiments (i.e., models trained in one modality and tested on the other. In the cross-domain context, we evaluated two strategies-the direct transfer between domains, and the contribution of Transfer Learning techniques (feature-representation-transfer based on Denoising Auto Encoders for reducing the gap in the feature space distributions. Our results demonstrate an excellent cross-domain generalisation performance with and without feature representation transfer in both directions. In the case of music, cross-domain approaches outperformed intra-domain models for Valence estimation, whereas for Speech intra-domain models achieve the best performance. This is the first demonstration of shared acoustic codes for emotional expression in music and speech in the time-continuous domain.

  1. Shared acoustic codes underlie emotional communication in music and speech-Evidence from deep transfer learning.

    Science.gov (United States)

    Coutinho, Eduardo; Schuller, Björn

    2017-01-01

    Music and speech exhibit striking similarities in the communication of emotions in the acoustic domain, in such a way that the communication of specific emotions is achieved, at least to a certain extent, by means of shared acoustic patterns. From an Affective Sciences points of view, determining the degree of overlap between both domains is fundamental to understand the shared mechanisms underlying such phenomenon. From a Machine learning perspective, the overlap between acoustic codes for emotional expression in music and speech opens new possibilities to enlarge the amount of data available to develop music and speech emotion recognition systems. In this article, we investigate time-continuous predictions of emotion (Arousal and Valence) in music and speech, and the Transfer Learning between these domains. We establish a comparative framework including intra- (i.e., models trained and tested on the same modality, either music or speech) and cross-domain experiments (i.e., models trained in one modality and tested on the other). In the cross-domain context, we evaluated two strategies-the direct transfer between domains, and the contribution of Transfer Learning techniques (feature-representation-transfer based on Denoising Auto Encoders) for reducing the gap in the feature space distributions. Our results demonstrate an excellent cross-domain generalisation performance with and without feature representation transfer in both directions. In the case of music, cross-domain approaches outperformed intra-domain models for Valence estimation, whereas for Speech intra-domain models achieve the best performance. This is the first demonstration of shared acoustic codes for emotional expression in music and speech in the time-continuous domain.

  2. Tools for signal compression applications to speech and audio coding

    CERN Document Server

    Moreau, Nicolas

    2013-01-01

    This book presents tools and algorithms required to compress/uncompress signals such as speech and music. These algorithms are largely used in mobile phones, DVD players, HDTV sets, etc. In a first rather theoretical part, this book presents the standard tools used in compression systems: scalar and vector quantization, predictive quantization, transform quantization, entropy coding. In particular we show the consistency between these different tools. The second part explains how these tools are used in the latest speech and audio coders. The third part gives Matlab programs simulating t

  3. A Support Vector Machine Approach to Dutch Part-of-Speech Tagging

    NARCIS (Netherlands)

    Poel, Mannes; Stegeman, L.; op den Akker, Hendrikus J.A.; Berthold, M.R.; Shawe-Taylor, J.; Lavrac, N.

    Part-of-Speech tagging, the assignment of Parts-of-Speech to the words in a given context of use, is a basic technique in many systems that handle natural languages. This paper describes a method for supervised training of a Part-of-Speech tagger using a committee of Support Vector Machines on a

  4. Machine function based control code algebras

    NARCIS (Netherlands)

    Bergstra, J.A.

    Machine functions have been introduced by Earley and Sturgis in [6] in order to provide a mathematical foundation of the use of the T-diagrams proposed by Bratman in [5]. Machine functions describe the operation of a machine at a very abstract level. A theory of hardware and software based on

  5. Relating dynamic brain states to dynamic machine states: Human and machine solutions to the speech recognition problem.

    Directory of Open Access Journals (Sweden)

    Cai Wingfield

    2017-09-01

    Full Text Available There is widespread interest in the relationship between the neurobiological systems supporting human cognition and emerging computational systems capable of emulating these capacities. Human speech comprehension, poorly understood as a neurobiological process, is an important case in point. Automatic Speech Recognition (ASR systems with near-human levels of performance are now available, which provide a computationally explicit solution for the recognition of words in continuous speech. This research aims to bridge the gap between speech recognition processes in humans and machines, using novel multivariate techniques to compare incremental 'machine states', generated as the ASR analysis progresses over time, to the incremental 'brain states', measured using combined electro- and magneto-encephalography (EMEG, generated as the same inputs are heard by human listeners. This direct comparison of dynamic human and machine internal states, as they respond to the same incrementally delivered sensory input, revealed a significant correspondence between neural response patterns in human superior temporal cortex and the structural properties of ASR-derived phonetic models. Spatially coherent patches in human temporal cortex responded selectively to individual phonetic features defined on the basis of machine-extracted regularities in the speech to lexicon mapping process. These results demonstrate the feasibility of relating human and ASR solutions to the problem of speech recognition, and suggest the potential for further studies relating complex neural computations in human speech comprehension to the rapidly evolving ASR systems that address the same problem domain.

  6. Memory of AMR coded speech distorted by packet loss

    OpenAIRE

    Nykänen, Arne; Lindegren, David; Wruck, Louisa; Ljung, Robert; Odelius, Johan; Möller, Sebastian

    2014-01-01

    Previous studies have shown that free recall of spoken word lists is impaired if the speech is presented in background noise, even if the signal-to-noise ratio is kept at a level allowing full word identification. The objective of this study was to examine recall rates for word lists presented in noise and word lists coded by an AMR (Adaptive Multi Rate) telephone codec distorted by packet loss. Twenty subjects performed a word recall test. Word lists consisting of ten words were played to th...

  7. Fifty years of progress in speech coding standards

    Science.gov (United States)

    Cox, Richard

    2004-10-01

    Over the past 50 years, speech coding has taken root worldwide. Early applications were for the military and transmission for telephone networks. The military gave equal priority to intelligibility and low bit rate. The telephone network gave priority to high quality and low delay. These illustrate three of the four areas in which requirements must be set for any speech coder application: bit rate, quality, delay, and complexity. While the military could afford relatively expensive terminal equipment for secure communications, the telephone network needed low cost for massive deployment in switches and transmission equipment worldwide. Today speech coders are at the heart of the wireless phones and telephone answering systems we use every day. In addition to the technology and technical invention that has occurred, standards make it possible for all these different systems to interoperate. The primary areas of standardization are the public switched telephone network, wireless telephony, and secure telephony for government and military applications. With the advent of IP telephony there are additional standardization efforts and challenges. In this talk the progress in all areas is reviewed as well as a reflection on Jim Flanagan's impact on this field during the past half century.

  8. Complete permutation Gray code implemented by finite state machine

    Directory of Open Access Journals (Sweden)

    Li Peng

    2014-09-01

    Full Text Available An enumerating method of complete permutation array is proposed. The list of n! permutations based on Gray code defined over finite symbol set Z(n = {1, 2, …, n} is implemented by finite state machine, named as n-RPGCF. An RPGCF can be used to search permutation code and provide improved lower bounds on the maximum cardinality of a permutation code in some cases.

  9. Anthropomorphic Coding of Speech and Audio: A Model Inversion Approach

    Directory of Open Access Journals (Sweden)

    W. Bastiaan Kleijn

    2005-06-01

    Full Text Available Auditory modeling is a well-established methodology that provides insight into human perception and that facilitates the extraction of signal features that are most relevant to the listener. The aim of this paper is to provide a tutorial on perceptual speech and audio coding using an invertible auditory model. In this approach, the audio signal is converted into an auditory representation using an invertible auditory model. The auditory representation is quantized and coded. Upon decoding, it is then transformed back into the acoustic domain. This transformation converts a complex distortion criterion into a simple one, thus facilitating quantization with low complexity. We briefly review past work on auditory models and describe in more detail the components of our invertible model and its inversion procedure, that is, the method to reconstruct the signal from the output of the auditory model. We summarize attempts to use the auditory representation for low-bit-rate coding. Our approach also allows the exploitation of the inherent redundancy of the human auditory system for the purpose of multiple description (joint source-channel coding.

  10. 3D equilibrium codes for mirror machines

    International Nuclear Information System (INIS)

    Kaiser, T.B.

    1983-01-01

    The codes developed for cumputing three-dimensional guiding center equilibria for quadrupole tandem mirrors are discussed. TEBASCO (Tandem equilibrium and ballooning stability code) is a code developed at LLNL that uses a further expansion of the paraxial equilibrium equation in powers of β (plasma pressure/magnetic pressure). It has been used to guide the design of the TMX-U and MFTF-B experiments at Livermore. Its principal weakness is its perturbative nature, which renders its validity for high-β calculation open to question. In order to compute high-β equilibria, the reduced MHD technique that has been proven useful for determining toroidal equilibria was adapted to the tandem mirror geometry. In this approach, the paraxial expansion of the MHD equations yields a set of coupled nonlinear equations of motion valid for arbitrary β, that are solved as an initial-value problem. Two particular formulations have been implemented in computer codes developed at NYU/Kyoto U and LLNL. They differ primarily in the type of grid, the location of the lateral boundary and the damping techniques employed, and in the method of calculating pressure-balance equilibrium. Discussions on these codes are presented in this paper. (Kato, T.)

  11. Speech rhythms and multiplexed oscillatory sensory coding in the human brain.

    Directory of Open Access Journals (Sweden)

    Joachim Gross

    2013-12-01

    Full Text Available Cortical oscillations are likely candidates for segmentation and coding of continuous speech. Here, we monitored continuous speech processing with magnetoencephalography (MEG to unravel the principles of speech segmentation and coding. We demonstrate that speech entrains the phase of low-frequency (delta, theta and the amplitude of high-frequency (gamma oscillations in the auditory cortex. Phase entrainment is stronger in the right and amplitude entrainment is stronger in the left auditory cortex. Furthermore, edges in the speech envelope phase reset auditory cortex oscillations thereby enhancing their entrainment to speech. This mechanism adapts to the changing physical features of the speech envelope and enables efficient, stimulus-specific speech sampling. Finally, we show that within the auditory cortex, coupling between delta, theta, and gamma oscillations increases following speech edges. Importantly, all couplings (i.e., brain-speech and also within the cortex attenuate for backward-presented speech, suggesting top-down control. We conclude that segmentation and coding of speech relies on a nested hierarchy of entrained cortical oscillations.

  12. Speech Rhythms and Multiplexed Oscillatory Sensory Coding in the Human Brain

    Science.gov (United States)

    Gross, Joachim; Hoogenboom, Nienke; Thut, Gregor; Schyns, Philippe; Panzeri, Stefano; Belin, Pascal; Garrod, Simon

    2013-01-01

    Cortical oscillations are likely candidates for segmentation and coding of continuous speech. Here, we monitored continuous speech processing with magnetoencephalography (MEG) to unravel the principles of speech segmentation and coding. We demonstrate that speech entrains the phase of low-frequency (delta, theta) and the amplitude of high-frequency (gamma) oscillations in the auditory cortex. Phase entrainment is stronger in the right and amplitude entrainment is stronger in the left auditory cortex. Furthermore, edges in the speech envelope phase reset auditory cortex oscillations thereby enhancing their entrainment to speech. This mechanism adapts to the changing physical features of the speech envelope and enables efficient, stimulus-specific speech sampling. Finally, we show that within the auditory cortex, coupling between delta, theta, and gamma oscillations increases following speech edges. Importantly, all couplings (i.e., brain-speech and also within the cortex) attenuate for backward-presented speech, suggesting top-down control. We conclude that segmentation and coding of speech relies on a nested hierarchy of entrained cortical oscillations. PMID:24391472

  13. The Cortical Organization of Speech Processing: Feedback Control and Predictive Coding the Context of a Dual-Stream Model

    Science.gov (United States)

    Hickok, Gregory

    2012-01-01

    Speech recognition is an active process that involves some form of predictive coding. This statement is relatively uncontroversial. What is less clear is the source of the prediction. The dual-stream model of speech processing suggests that there are two possible sources of predictive coding in speech perception: the motor speech system and the…

  14. Model-Based Speech Signal Coding Using Optimized Temporal Decomposition for Storage and Broadcasting Applications

    Science.gov (United States)

    Athaudage, Chandranath R. N.; Bradley, Alan B.; Lech, Margaret

    2003-12-01

    A dynamic programming-based optimization strategy for a temporal decomposition (TD) model of speech and its application to low-rate speech coding in storage and broadcasting is presented. In previous work with the spectral stability-based event localizing (SBEL) TD algorithm, the event localization was performed based on a spectral stability criterion. Although this approach gave reasonably good results, there was no assurance on the optimality of the event locations. In the present work, we have optimized the event localizing task using a dynamic programming-based optimization strategy. Simulation results show that an improved TD model accuracy can be achieved. A methodology of incorporating the optimized TD algorithm within the standard MELP speech coder for the efficient compression of speech spectral information is also presented. The performance evaluation results revealed that the proposed speech coding scheme achieves 50%-60% compression of speech spectral information with negligible degradation in the decoded speech quality.

  15. Model-Driven Engineering of Machine Executable Code

    Science.gov (United States)

    Eichberg, Michael; Monperrus, Martin; Kloppenburg, Sven; Mezini, Mira

    Implementing static analyses of machine-level executable code is labor intensive and complex. We show how to leverage model-driven engineering to facilitate the design and implementation of programs doing static analyses. Further, we report on important lessons learned on the benefits and drawbacks while using the following technologies: using the Scala programming language as target of code generation, using XML-Schema to express a metamodel, and using XSLT to implement (a) transformations and (b) a lint like tool. Finally, we report on the use of Prolog for writing model transformations.

  16. Spotlight on Speech Codes 2011: The State of Free Speech on Our Nation's Campuses

    Science.gov (United States)

    Foundation for Individual Rights in Education (NJ1), 2011

    2011-01-01

    Each year, the Foundation for Individual Rights in Education (FIRE) conducts a rigorous survey of restrictions on speech at America's colleges and universities. The survey and accompanying report explore the extent to which schools are meeting their legal and moral obligations to uphold students' and faculty members' rights to freedom of speech,…

  17. Spotlight on Speech Codes 2009: The State of Free Speech on Our Nation's Campuses

    Science.gov (United States)

    Foundation for Individual Rights in Education (NJ1), 2009

    2009-01-01

    Each year, the Foundation for Individual Rights in Education (FIRE) conducts a wide, detailed survey of restrictions on speech at America's colleges and universities. The survey and resulting report explore the extent to which schools are meeting their obligations to uphold students' and faculty members' rights to freedom of speech, freedom of…

  18. Spotlight on Speech Codes 2010: The State of Free Speech on Our Nation's Campuses

    Science.gov (United States)

    Foundation for Individual Rights in Education (NJ1), 2010

    2010-01-01

    Each year, the Foundation for Individual Rights in Education (FIRE) conducts a rigorous survey of restrictions on speech at America's colleges and universities. The survey and resulting report explore the extent to which schools are meeting their legal and moral obligations to uphold students' and faculty members' rights to freedom of speech,…

  19. A wireless brain-machine interface for real-time speech synthesis.

    Directory of Open Access Journals (Sweden)

    Frank H Guenther

    2009-12-01

    Full Text Available Brain-machine interfaces (BMIs involving electrodes implanted into the human cerebral cortex have recently been developed in an attempt to restore function to profoundly paralyzed individuals. Current BMIs for restoring communication can provide important capabilities via a typing process, but unfortunately they are only capable of slow communication rates. In the current study we use a novel approach to speech restoration in which we decode continuous auditory parameters for a real-time speech synthesizer from neuronal activity in motor cortex during attempted speech.Neural signals recorded by a Neurotrophic Electrode implanted in a speech-related region of the left precentral gyrus of a human volunteer suffering from locked-in syndrome, characterized by near-total paralysis with spared cognition, were transmitted wirelessly across the scalp and used to drive a speech synthesizer. A Kalman filter-based decoder translated the neural signals generated during attempted speech into continuous parameters for controlling a synthesizer that provided immediate (within 50 ms auditory feedback of the decoded sound. Accuracy of the volunteer's vowel productions with the synthesizer improved quickly with practice, with a 25% improvement in average hit rate (from 45% to 70% and 46% decrease in average endpoint error from the first to the last block of a three-vowel task.Our results support the feasibility of neural prostheses that may have the potential to provide near-conversational synthetic speech output for individuals with severely impaired speech motor control. They also provide an initial glimpse into the functional properties of neurons in speech motor cortical areas.

  20. ISOLATED SPEECH RECOGNITION SYSTEM FOR TAMIL LANGUAGE USING STATISTICAL PATTERN MATCHING AND MACHINE LEARNING TECHNIQUES

    Directory of Open Access Journals (Sweden)

    VIMALA C.

    2015-05-01

    Full Text Available In recent years, speech technology has become a vital part of our daily lives. Various techniques have been proposed for developing Automatic Speech Recognition (ASR system and have achieved great success in many applications. Among them, Template Matching techniques like Dynamic Time Warping (DTW, Statistical Pattern Matching techniques such as Hidden Markov Model (HMM and Gaussian Mixture Models (GMM, Machine Learning techniques such as Neural Networks (NN, Support Vector Machine (SVM, and Decision Trees (DT are most popular. The main objective of this paper is to design and develop a speaker-independent isolated speech recognition system for Tamil language using the above speech recognition techniques. The background of ASR system, the steps involved in ASR, merits and demerits of the conventional and machine learning algorithms and the observations made based on the experiments are presented in this paper. For the above developed system, highest word recognition accuracy is achieved with HMM technique. It offered 100% accuracy during training process and 97.92% for testing process.

  1. Compiler design handbook optimizations and machine code generation

    CERN Document Server

    Srikant, YN

    2003-01-01

    The widespread use of object-oriented languages and Internet security concerns are just the beginning. Add embedded systems, multiple memory banks, highly pipelined units operating in parallel, and a host of other advances and it becomes clear that current and future computer architectures pose immense challenges to compiler designers-challenges that already exceed the capabilities of traditional compilation techniques. The Compiler Design Handbook: Optimizations and Machine Code Generation is designed to help you meet those challenges. Written by top researchers and designers from around the

  2. Development of a speech-based dialogue system for report dictation and machine control in the endoscopic laboratory.

    Science.gov (United States)

    Molnar, B; Gergely, J; Toth, G; Pronai, L; Zagoni, T; Papik, K; Tulassay, Z

    2000-01-01

    Reporting and machine control based on speech technology can enhance work efficiency in the gastrointestinal endoscopy laboratory. The status and activation of endoscopy laboratory equipment were described as a multivariate parameter and function system. Speech recognition, text evaluation and action definition engines were installed. Special programs were developed for the grammatical analysis of command sentences, and a rule-based expert system for the definition of machine answers. A speech backup engine provides feedback to the user. Techniques were applied based on the "Hidden Markov" model of discrete word, user-independent speech recognition and on phoneme-based speech synthesis. Speech samples were collected from three male low-tone investigators. The dictation module and machine control modules were incorporated in a personal computer (PC) simulation program. Altogether 100 unidentified patient records were analyzed. The sentences were grouped according to keywords, which indicate the main topics of a gastrointestinal endoscopy report. They were: "endoscope", "esophagus", "cardia", "fundus", "corpus", "antrum", "pylorus", "bulbus", and "postbulbar section", in addition to the major pathological findings: "erosion", "ulceration", and "malignancy". "Biopsy" and "diagnosis" were also included. We implemented wireless speech communication control commands for equipment including an endoscopy unit, video, monitor, printer, and PC. The recognition rate was 95%. Speech technology may soon become an integrated part of our daily routine in the endoscopy laboratory. A central speech and laboratory computer could be the most efficient alternative to having separate speech recognition units in all items of equipment.

  3. Integrating Automatic Speech Recognition and Machine Translation for Better Translation Outputs

    DEFF Research Database (Denmark)

    Liyanapathirana, Jeevanthi

    translations, combining machine translation with computer assisted translation has drawn attention in current research. This combines two prospects: the opportunity of ensuring high quality translation along with a significant performance gain. Automatic Speech Recognition (ASR) is another important area......, which caters important functionalities in language processing and natural language understanding tasks. In this work we integrate automatic speech recognition and machine translation in parallel. We aim to avoid manual typing of possible translations as dictating the translation would take less time...... to the n-best list rescoring, we also use word graphs with the expectation of arriving at a tighter integration of ASR and MT models. Integration methods include constraining ASR models using language and translation models of MT, and vice versa. We currently develop and experiment different methods...

  4. Magnified Neural Envelope Coding Predicts Deficits in Speech Perception in Noise.

    Science.gov (United States)

    Millman, Rebecca E; Mattys, Sven L; Gouws, André D; Prendergast, Garreth

    2017-08-09

    Verbal communication in noisy backgrounds is challenging. Understanding speech in background noise that fluctuates in intensity over time is particularly difficult for hearing-impaired listeners with a sensorineural hearing loss (SNHL). The reduction in fast-acting cochlear compression associated with SNHL exaggerates the perceived fluctuations in intensity in amplitude-modulated sounds. SNHL-induced changes in the coding of amplitude-modulated sounds may have a detrimental effect on the ability of SNHL listeners to understand speech in the presence of modulated background noise. To date, direct evidence for a link between magnified envelope coding and deficits in speech identification in modulated noise has been absent. Here, magnetoencephalography was used to quantify the effects of SNHL on phase locking to the temporal envelope of modulated noise (envelope coding) in human auditory cortex. Our results show that SNHL enhances the amplitude of envelope coding in posteromedial auditory cortex, whereas it enhances the fidelity of envelope coding in posteromedial and posterolateral auditory cortex. This dissociation was more evident in the right hemisphere, demonstrating functional lateralization in enhanced envelope coding in SNHL listeners. However, enhanced envelope coding was not perceptually beneficial. Our results also show that both hearing thresholds and, to a lesser extent, magnified cortical envelope coding in left posteromedial auditory cortex predict speech identification in modulated background noise. We propose a framework in which magnified envelope coding in posteromedial auditory cortex disrupts the segregation of speech from background noise, leading to deficits in speech perception in modulated background noise. SIGNIFICANCE STATEMENT People with hearing loss struggle to follow conversations in noisy environments. Background noise that fluctuates in intensity over time poses a particular challenge. Using magnetoencephalography, we demonstrate

  5. Do North Carolina Students Have Freedom of Speech? A Review of Campus Speech Codes

    Science.gov (United States)

    Robinson, Jenna Ashley

    2010-01-01

    America's colleges and universities are supposed to be strongholds of classically liberal ideals, including the protection of individual rights and openness to debate and inquiry. Too often, this is not the case. Across the country, universities deny students and faculty their fundamental rights to freedom of speech and expression. The report…

  6. Detecting Abnormal Word Utterances in Children With Autism Spectrum Disorders: Machine-Learning-Based Voice Analysis Versus Speech Therapists.

    Science.gov (United States)

    Nakai, Yasushi; Takiguchi, Tetsuya; Matsui, Gakuyo; Yamaoka, Noriko; Takada, Satoshi

    2017-10-01

    Abnormal prosody is often evident in the voice intonations of individuals with autism spectrum disorders. We compared a machine-learning-based voice analysis with human hearing judgments made by 10 speech therapists for classifying children with autism spectrum disorders ( n = 30) and typical development ( n = 51). Using stimuli limited to single-word utterances, machine-learning-based voice analysis was superior to speech therapist judgments. There was a significantly higher true-positive than false-negative rate for machine-learning-based voice analysis but not for speech therapists. Results are discussed in terms of some artificiality of clinician judgments based on single-word utterances, and the objectivity machine-learning-based voice analysis adds to judging abnormal prosody.

  7. Look at the Gato! Code-Switching in Speech to Toddlers

    Science.gov (United States)

    Bail, Amelie; Morini, Giovanna; Newman, Rochelle S.

    2015-01-01

    We examined code-switching (CS) in the speech of twenty-four bilingual caregivers when speaking with their 18- to 24-month-old children. All parents CS at least once in a short play session, and some code-switched quite often (over 1/3 of utterances). This CS included both inter-sentential and intra-sentential switches, suggesting that at least…

  8. Improving Language Models in Speech-Based Human-Machine Interaction

    Directory of Open Access Journals (Sweden)

    Raquel Justo

    2013-02-01

    Full Text Available This work focuses on speech-based human-machine interaction. Specifically, a Spoken Dialogue System (SDS that could be integrated into a robot is considered. Since Automatic Speech Recognition is one of the most sensitive tasks that must be confronted in such systems, the goal of this work is to improve the results obtained by this specific module. In order to do so, a hierarchical Language Model (LM is considered. Different series of experiments were carried out using the proposed models over different corpora and tasks. The results obtained show that these models provide greater accuracy in the recognition task. Additionally, the influence of the Acoustic Modelling (AM in the improvement percentage of the Language Models has also been explored. Finally the use of hierarchical Language Models in a language understanding task has been successfully employed, as shown in an additional series of experiments.

  9. Speech Analysis and Synthesis and Man-Machine Speech Communications for Air Operations. (Synthese et Analyse de la Parole et Liaisons Vocales Homme- Machine dans les Operations Aeriennes)

    Science.gov (United States)

    1990-05-01

    speech processing area are faced . He presents speech communication as an interactive process, in which the listener actively reconstructs the message...speech produced by these systems. Finally, perhaps the greatest recent impetus in advancing digital Finally, in the area of speech and speaker recognitio

  10. A Digital Liquid State Machine With Biologically Inspired Learning and Its Application to Speech Recognition.

    Science.gov (United States)

    Zhang, Yong; Li, Peng; Jin, Yingyezhe; Choe, Yoonsuck

    2015-11-01

    This paper presents a bioinspired digital liquid-state machine (LSM) for low-power very-large-scale-integration (VLSI)-based machine learning applications. To the best of the authors' knowledge, this is the first work that employs a bioinspired spike-based learning algorithm for the LSM. With the proposed online learning, the LSM extracts information from input patterns on the fly without needing intermediate data storage as required in offline learning methods such as ridge regression. The proposed learning rule is local such that each synaptic weight update is based only upon the firing activities of the corresponding presynaptic and postsynaptic neurons without incurring global communications across the neural network. Compared with the backpropagation-based learning, the locality of computation in the proposed approach lends itself to efficient parallel VLSI implementation. We use subsets of the TI46 speech corpus to benchmark the bioinspired digital LSM. To reduce the complexity of the spiking neural network model without performance degradation for speech recognition, we study the impacts of synaptic models on the fading memory of the reservoir and hence the network performance. Moreover, we examine the tradeoffs between synaptic weight resolution, reservoir size, and recognition performance and present techniques to further reduce the overhead of hardware implementation. Our simulation results show that in terms of isolated word recognition evaluated using the TI46 speech corpus, the proposed digital LSM rivals the state-of-the-art hidden Markov-model-based recognizer Sphinx-4 and outperforms all other reported recognizers including the ones that are based upon the LSM or neural networks.

  11. Speech Compression

    Directory of Open Access Journals (Sweden)

    Jerry D. Gibson

    2016-06-01

    Full Text Available Speech compression is a key technology underlying digital cellular communications, VoIP, voicemail, and voice response systems. We trace the evolution of speech coding based on the linear prediction model, highlight the key milestones in speech coding, and outline the structures of the most important speech coding standards. Current challenges, future research directions, fundamental limits on performance, and the critical open problem of speech coding for emergency first responders are all discussed.

  12. The classification problem in machine learning: an overview with study cases in emotion recognition and music-speech differentiation

    OpenAIRE

    Rodríguez Cadavid, Santiago

    2015-01-01

    This work addresses the well-known classification problem in machine learning -- The goal of this study is to approach the reader to the methodological aspects of the feature extraction, feature selection and classifier performance through simple and understandable theoretical aspects and two study cases -- Finally, a very good classification performance was obtained for the emotion recognition from speech

  13. Contributions of speech science to the technology of man-machine voice interactions

    Science.gov (United States)

    Lea, Wayne A.

    1977-01-01

    Research in speech understanding was reviewed. Plans which include prosodics research, phonological rules for speech understanding systems, and continued interdisciplinary phonetics research are discussed. Improved acoustic phonetic analysis capabilities in speech recognizers are suggested.

  14. Machine-Learning Algorithms to Code Public Health Spending Accounts.

    Science.gov (United States)

    Brady, Eoghan S; Leider, Jonathon P; Resnick, Beth A; Alfonso, Y Natalia; Bishai, David

    Government public health expenditure data sets require time- and labor-intensive manipulation to summarize results that public health policy makers can use. Our objective was to compare the performances of machine-learning algorithms with manual classification of public health expenditures to determine if machines could provide a faster, cheaper alternative to manual classification. We used machine-learning algorithms to replicate the process of manually classifying state public health expenditures, using the standardized public health spending categories from the Foundational Public Health Services model and a large data set from the US Census Bureau. We obtained a data set of 1.9 million individual expenditure items from 2000 to 2013. We collapsed these data into 147 280 summary expenditure records, and we followed a standardized method of manually classifying each expenditure record as public health, maybe public health, or not public health. We then trained 9 machine-learning algorithms to replicate the manual process. We calculated recall, precision, and coverage rates to measure the performance of individual and ensembled algorithms. Compared with manual classification, the machine-learning random forests algorithm produced 84% recall and 91% precision. With algorithm ensembling, we achieved our target criterion of 90% recall by using a consensus ensemble of ≥6 algorithms while still retaining 93% coverage, leaving only 7% of the summary expenditure records unclassified. Machine learning can be a time- and cost-saving tool for estimating public health spending in the United States. It can be used with standardized public health spending categories based on the Foundational Public Health Services model to help parse public health expenditure information from other types of health-related spending, provide data that are more comparable across public health organizations, and evaluate the impact of evidence-based public health resource allocation.

  15. Towards a universal code formatter through machine learning

    NARCIS (Netherlands)

    Parr, T. (Terence); J.J. Vinju (Jurgen)

    2016-01-01

    textabstractThere are many declarative frameworks that allow us to implement code formatters relatively easily for any specific language, but constructing them is cumbersome. The first problem is that "everybody" wants to format their code differently, leading to either many formatter variants or a

  16. Code-Expanded Random Access for Machine-Type Communications

    DEFF Research Database (Denmark)

    Kiilerich Pratas, Nuno; Thomsen, Henning; Stefanovic, Cedomir

    2012-01-01

    Abstract—The random access methods used for support of machine-type communications (MTC) in current cellular standards are derivatives of traditional framed slotted ALOHA and therefore do not support high user loads efficiently. Motivated by the random access method employed in LTE, we propose...

  17. Bilingual Voicing: A Study of Code-Switching in the Reported Speech of Finnish Immigrants in Estonia

    Science.gov (United States)

    Frick, Maria; Riionheimo, Helka

    2013-01-01

    Through a conversation analytic investigation of Finnish-Estonian bilingual (direct) reported speech (i.e., voicing) by Finns who live in Estonia, this study shows how code-switching is used as a double contextualization device. The code-switched voicings are shaped by the on-going interactional situation, serving its needs by opening up a context…

  18. Code-expanded radio access protocol for machine-to-machine communications

    DEFF Research Database (Denmark)

    Thomsen, Henning; Kiilerich Pratas, Nuno; Stefanovic, Cedomir

    2013-01-01

    The random access methods used for support of machine-to-machine, also referred to as Machine-Type Communications, in current cellular standards are derivatives of traditional framed slotted ALOHA and therefore do not support high user loads efficiently. We propose an approach that is motivated b...... subframes and orthogonal preambles, the amount of available contention resources is drastically increased, enabling the massive support of Machine-Type Communication users that is beyond the reach of current systems.......The random access methods used for support of machine-to-machine, also referred to as Machine-Type Communications, in current cellular standards are derivatives of traditional framed slotted ALOHA and therefore do not support high user loads efficiently. We propose an approach that is motivated...... by the random access method employed in LTE, which significantly increases the amount of contention resources without increasing the system resources, such as contention subframes and preambles. This is accomplished by a logical, rather than physical, extension of the access method in which the available system...

  19. The semaphore codes attached to a Turing machine via resets and their various limits

    OpenAIRE

    Rhodes, John; Schilling, Anne; Silva, Pedro V.

    2016-01-01

    We introduce semaphore codes associated to a Turing machine via resets. Semaphore codes provide an approximation theory for resets. In this paper we generalize the set-up of our previous paper "Random walks on semaphore codes and delay de Bruijn semigroups" to the infinite case by taking the profinite limit of $k$-resets to obtain $(-\\omega)$-resets. We mention how this opens new avenues to attack the P versus NP problem.

  20. A Machine Learning Perspective on Predictive Coding with PAQ

    OpenAIRE

    Knoll, Byron; de Freitas, Nando

    2011-01-01

    PAQ8 is an open source lossless data compression algorithm that currently achieves the best compression rates on many benchmarks. This report presents a detailed description of PAQ8 from a statistical machine learning perspective. It shows that it is possible to understand some of the modules of PAQ8 and use this understanding to improve the method. However, intuitive statistical explanations of the behavior of other modules remain elusive. We hope the description in this report will be a sta...

  1. Improving on hidden Markov models: An articulatorily constrained, maximum likelihood approach to speech recognition and speech coding

    Energy Technology Data Exchange (ETDEWEB)

    Hogden, J.

    1996-11-05

    The goal of the proposed research is to test a statistical model of speech recognition that incorporates the knowledge that speech is produced by relatively slow motions of the tongue, lips, and other speech articulators. This model is called Maximum Likelihood Continuity Mapping (Malcom). Many speech researchers believe that by using constraints imposed by articulator motions, we can improve or replace the current hidden Markov model based speech recognition algorithms. Unfortunately, previous efforts to incorporate information about articulation into speech recognition algorithms have suffered because (1) slight inaccuracies in our knowledge or the formulation of our knowledge about articulation may decrease recognition performance, (2) small changes in the assumptions underlying models of speech production can lead to large changes in the speech derived from the models, and (3) collecting measurements of human articulator positions in sufficient quantity for training a speech recognition algorithm is still impractical. The most interesting (and in fact, unique) quality of Malcom is that, even though Malcom makes use of a mapping between acoustics and articulation, Malcom can be trained to recognize speech using only acoustic data. By learning the mapping between acoustics and articulation using only acoustic data, Malcom avoids the difficulties involved in collecting articulator position measurements and does not require an articulatory synthesizer model to estimate the mapping between vocal tract shapes and speech acoustics. Preliminary experiments that demonstrate that Malcom can learn the mapping between acoustics and articulation are discussed. Potential applications of Malcom aside from speech recognition are also discussed. Finally, specific deliverables resulting from the proposed research are described.

  2. A Bit Stream Scalable Speech/Audio Coder Combining Enhanced Regular Pulse Excitation and Parametric Coding

    Science.gov (United States)

    Riera-Palou, Felip; den Brinker, Albertus C.

    2007-12-01

    This paper introduces a new audio and speech broadband coding technique based on the combination of a pulse excitation coder and a standardized parametric coder, namely, MPEG-4 high-quality parametric coder. After presenting a series of enhancements to regular pulse excitation (RPE) to make it suitable for the modeling of broadband signals, it is shown how pulse and parametric codings complement each other and how they can be merged to yield a layered bit stream scalable coder able to operate at different points in the quality bit rate plane. The performance of the proposed coder is evaluated in a listening test. The major result is that the extra functionality of the bit stream scalability does not come at the price of a reduced performance since the coder is competitive with standardized coders (MP3, AAC, SSC).

  3. A Bit Stream Scalable Speech/Audio Coder Combining Enhanced Regular Pulse Excitation and Parametric Coding

    Directory of Open Access Journals (Sweden)

    Albertus C. den Brinker

    2007-01-01

    Full Text Available This paper introduces a new audio and speech broadband coding technique based on the combination of a pulse excitation coder and a standardized parametric coder, namely, MPEG-4 high-quality parametric coder. After presenting a series of enhancements to regular pulse excitation (RPE to make it suitable for the modeling of broadband signals, it is shown how pulse and parametric codings complement each other and how they can be merged to yield a layered bit stream scalable coder able to operate at different points in the quality bit rate plane. The performance of the proposed coder is evaluated in a listening test. The major result is that the extra functionality of the bit stream scalability does not come at the price of a reduced performance since the coder is competitive with standardized coders (MP3, AAC, SSC.

  4. An Automatic Instruction-Level Parallelization of Machine Code

    Directory of Open Access Journals (Sweden)

    MARINKOVIC, V.

    2018-02-01

    Full Text Available Prevailing multicores and novel manycores have made a great challenge of modern day - parallelization of embedded software that is still written as sequential. In this paper, automatic code parallelization is considered, focusing on developing a parallelization tool at the binary level as well as on the validation of this approach. The novel instruction-level parallelization algorithm for assembly code which uses the register names after SSA to find independent blocks of code and then to schedule independent blocks using METIS to achieve good load balance is developed. The sequential consistency is verified and the validation is done by measuring the program execution time on the target architecture. Great speedup, taken as the performance measure in the validation process, and optimal load balancing are achieved for multicore RISC processors with 2 to 16 cores (e.g. MIPS, MicroBlaze, etc.. In particular, for 16 cores, the average speedup is 7.92x, while in some cases it reaches 14x. An approach to automatic parallelization provided by this paper is useful to researchers and developers in the area of parallelization as the basis for further optimizations, as the back-end of a compiler, or as the code parallelization tool for an embedded system.

  5. Lean coding machine. Facilities target productivity and job satisfaction with coding automation.

    Science.gov (United States)

    Rollins, Genna

    2010-07-01

    Facilities are turning to coding automation to help manage the volume of electronic documentation, streamlining workflow, boosting productivity, and increasing job satisfaction. As EHR adoption increases, computer-assisted coding may become a necessity, not an option.

  6. Machine-Checked Sequencer for Critical Embedded Code Generator

    Science.gov (United States)

    Izerrouken, Nassima; Pantel, Marc; Thirioux, Xavier

    This paper presents the development of a correct-by-construction block sequencer for GeneAuto a qualifiable (according to DO178B/ED12B recommendation) automatic code generator. It transforms Simulink models to MISRA C code for safety critical systems. Our approach which combines classical development process and formal specification and verification using proof-assistants, led to preliminary fruitful exchanges with certification authorities. We present parts of the classical user and tools requirements and derived formal specifications, implementation and verification for the correctness and termination of the block sequencer. This sequencer has been successfully applied to real-size industrial use cases from various transportation domain partners and led to requirement errors detection and a correct-by-construction implementation.

  7. CNC LATHE MACHINE PRODUCING NC CODE BY USING DIALOG METHOD

    Directory of Open Access Journals (Sweden)

    Yakup TURGUT

    2004-03-01

    Full Text Available In this study, an NC code generation program utilising Dialog Method was developed for turning centres. Initially, CNC lathes turning methods and tool path development techniques were reviewed briefly. By using geometric definition methods, tool path was generated and CNC part program was developed for FANUC control unit. The developed program made CNC part program generation process easy. The program was developed using BASIC 6.0 programming language while the material and cutting tool database were and supported with the help of ACCESS 7.0.

  8. Accuracy comparison among different machine learning techniques for detecting malicious codes

    Science.gov (United States)

    Narang, Komal

    2016-03-01

    In this paper, a machine learning based model for malware detection is proposed. It can detect newly released malware i.e. zero day attack by analyzing operation codes on Android operating system. The accuracy of Naïve Bayes, Support Vector Machine (SVM) and Neural Network for detecting malicious code has been compared for the proposed model. In the experiment 400 benign files, 100 system files and 500 malicious files have been used to construct the model. The model yields the best accuracy 88.9% when neural network is used as classifier and achieved 95% and 82.8% accuracy for sensitivity and specificity respectively.

  9. Using machine-coded event data for the micro-level study of political violence

    Directory of Open Access Journals (Sweden)

    Jesse Hammond

    2014-07-01

    Full Text Available Machine-coded datasets likely represent the future of event data analysis. We assess the use of one of these datasets—Global Database of Events, Language and Tone (GDELT—for the micro-level study of political violence by comparing it to two hand-coded conflict event datasets. Our findings indicate that GDELT should be used with caution for geo-spatial analyses at the subnational level: its overall correlation with hand-coded data is mediocre, and at the local level major issues of geographic bias exist in how events are reported. Overall, our findings suggest that due to these issues, researchers studying local conflict processes may want to wait for a more reliable geocoding method before relying too heavily on this set of machine-coded data.

  10. An Efficient VQ Codebook Search Algorithm Applied to AMR-WB Speech Coding

    Directory of Open Access Journals (Sweden)

    Cheng-Yu Yeh

    2017-04-01

    Full Text Available The adaptive multi-rate wideband (AMR-WB speech codec is widely used in modern mobile communication systems for high speech quality in handheld devices. Nonetheless, a major disadvantage is that vector quantization (VQ of immittance spectral frequency (ISF coefficients takes a considerable computational load in the AMR-WB coding. Accordingly, a binary search space-structured VQ (BSS-VQ algorithm is adopted to efficiently reduce the complexity of ISF quantization in AMR-WB. This search algorithm is done through a fast locating technique combined with lookup tables, such that an input vector is efficiently assigned to a subspace where relatively few codeword searches are required to be executed. In terms of overall search performance, this work is experimentally validated as a superior search algorithm relative to a multiple triangular inequality elimination (MTIE, a TIE with dynamic and intersection mechanisms (DI-TIE, and an equal-average equal-variance equal-norm nearest neighbor search (EEENNS approach. With a full search algorithm as a benchmark for overall search load comparison, this work provides an 87% search load reduction at a threshold of quantization accuracy of 0.96, a figure far beyond 55% in the MTIE, 76% in the EEENNS approach, and 83% in the DI-TIE approach.

  11. Temporal Fine-Structure Coding and Lateralized Speech Perception in Normal-Hearing and Hearing-Impaired Listeners

    DEFF Research Database (Denmark)

    Locsei, Gusztav; Pedersen, Julie Hefting; Laugesen, Søren

    2016-01-01

    This study investigated the relationship between speech perception performance in spatially complex, lateralized listening scenarios and temporal fine-structure (TFS) coding at low frequencies. Young normal-hearing (NH) and two groups of elderly hearing-impaired (HI) listeners with mild or moderate...... hearing loss above 1.5 kHz participated in the study. Speech reception thresholds (SRTs) were estimated in the presence of either speech-shaped noise, two-, four-, or eight-talker babble played reversed, or a nonreversed two-talker masker. Target audibility was ensured by applying individualized linear...... threshold nor the interaural phase difference threshold tasks showed a correlation with the SRTs or with the amount of masking release due to binaural unmasking, respectively. The results suggest that, although HI listeners with normal hearing thresholds below 1.5 kHz experienced difficulties with speech...

  12. Reading your own lips: common-coding theory and visual speech perception.

    Science.gov (United States)

    Tye-Murray, Nancy; Spehar, Brent P; Myerson, Joel; Hale, Sandra; Sommers, Mitchell S

    2013-02-01

    Common-coding theory posits that (1) perceiving an action activates the same representations of motor plans that are activated by actually performing that action, and (2) because of individual differences in the ways that actions are performed, observing recordings of one's own previous behavior activates motor plans to an even greater degree than does observing someone else's behavior. We hypothesized that if observing oneself activates motor plans to a greater degree than does observing others, and if these activated plans contribute to perception, then people should be able to lipread silent video clips of their own previous utterances more accurately than they can lipread video clips of other talkers. As predicted, two groups of participants were able to lipread video clips of themselves, recorded more than two weeks earlier, significantly more accurately than video clips of others. These results suggest that visual input activates speech motor activity that links to word representations in the mental lexicon.

  13. Numerical code to determine the particle trapping region in the LISA machine

    International Nuclear Information System (INIS)

    Azevedo, M.T. de; Raposo, C.C. de; Tomimura, A.

    1984-01-01

    A numerical code is constructed to determine the trapping region in machine like LISA. The variable magnetic field is two deimensional and is coupled to the Runge-Kutta through the Tchebichev polynomial. Various particle orbits including particle interactions were analysed. Beside this, a strong electric field is introduced to see the possible effects happening inside the plasma. (Author) [pt

  14. A friend man-machine interface for thermo-hydraulic simulation codes of nuclear installations

    International Nuclear Information System (INIS)

    Araujo Filho, F. de; Belchior Junior, A.; Barroso, A.C.O.; Gebrim, A.

    1994-01-01

    This work presents the development of a Man-Machine Interface to the TRAC-PF1 code, a computer program to perform best estimate analysis of transients and accidents at nuclear power plants. The results were considered satisfactory and a considerable productivity gain was achieved in the activity of preparing and analyzing simulations. (author)

  15. Using supervised machine learning to code policy issues: Can classifiers generalize across contexts?

    NARCIS (Netherlands)

    Burscher, B.; Vliegenthart, R.; de Vreese, C.H.

    2015-01-01

    Content analysis of political communication usually covers large amounts of material and makes the study of dynamics in issue salience a costly enterprise. In this article, we present a supervised machine learning approach for the automatic coding of policy issues, which we apply to news articles

  16. The vector and parallel processing of MORSE code on Monte Carlo Machine

    International Nuclear Information System (INIS)

    Hasegawa, Yukihiro; Higuchi, Kenji.

    1995-11-01

    Multi-group Monte Carlo Code for particle transport, MORSE is modified for high performance computing on Monte Carlo Machine Monte-4. The method and the results are described. Monte-4 was specially developed to realize high performance computing of Monte Carlo codes for particle transport, which have been difficult to obtain high performance in vector processing on conventional vector processors. Monte-4 has four vector processor units with the special hardware called Monte Carlo pipelines. The vectorization and parallelization of MORSE code and the performance evaluation on Monte-4 are described. (author)

  17. Speech recognition technology: an outlook for human-to-machine interaction.

    Science.gov (United States)

    Erdel, T; Crooks, S

    2000-01-01

    Speech recognition, as an enabling technology in healthcare-systems computing, is a topic that has been discussed for quite some time, but is just now coming to fruition. Traditionally, speech-recognition software has been constrained by hardware, but improved processors and increased memory capacities are starting to remove some of these limitations. With these barriers removed, companies that create software for the healthcare setting have the opportunity to write more successful applications. Among the criticisms of speech-recognition applications are the high rates of error and steep training curves. However, even in the face of such negative perceptions, there remains significant opportunities for speech recognition to allow healthcare providers and, more specifically, physicians, to work more efficiently and ultimately spend more time with their patients and less time completing necessary documentation. This article will identify opportunities for inclusion of speech-recognition technology in the healthcare setting and examine major categories of speech-recognition software--continuous speech recognition, command and control, and text-to-speech. We will discuss the advantages and disadvantages of each area, the limitations of the software today, and how future trends might affect them.

  18. Machine-learning-assisted correction of correlated qubit errors in a topological code

    Directory of Open Access Journals (Sweden)

    Paul Baireuther

    2018-01-01

    Full Text Available A fault-tolerant quantum computation requires an efficient means to detect and correct errors that accumulate in encoded quantum information. In the context of machine learning, neural networks are a promising new approach to quantum error correction. Here we show that a recurrent neural network can be trained, using only experimentally accessible data, to detect errors in a widely used topological code, the surface code, with a performance above that of the established minimum-weight perfect matching (or blossom decoder. The performance gain is achieved because the neural network decoder can detect correlations between bit-flip (X and phase-flip (Z errors. The machine learning algorithm adapts to the physical system, hence no noise model is needed. The long short-term memory layers of the recurrent neural network maintain their performance over a large number of quantum error correction cycles, making it a practical decoder for forthcoming experimental realizations of the surface code.

  19. Support vector machine and mel frequency Cepstral coefficient based algorithm for hand gestures and bidirectional speech to text device

    Science.gov (United States)

    Balbin, Jessie R.; Padilla, Dionis A.; Fausto, Janette C.; Vergara, Ernesto M.; Garcia, Ramon G.; Delos Angeles, Bethsedea Joy S.; Dizon, Neil John A.; Mardo, Mark Kevin N.

    2017-02-01

    This research is about translating series of hand gesture to form a word and produce its equivalent sound on how it is read and said in Filipino accent using Support Vector Machine and Mel Frequency Cepstral Coefficient analysis. The concept is to detect Filipino speech input and translate the spoken words to their text form in Filipino. This study is trying to help the Filipino deaf community to impart their thoughts through the use of hand gestures and be able to communicate to people who do not know how to read hand gestures. This also helps literate deaf to simply read the spoken words relayed to them using the Filipino speech to text system.

  20. Implementation of support vector machine for classification of speech marked hijaiyah letters based on Mel frequency cepstrum coefficient feature extraction

    Science.gov (United States)

    Adhi Pradana, Wisnu; Adiwijaya; Novia Wisesty, Untari

    2018-03-01

    Support Vector Machine or commonly called SVM is one method that can be used to process the classification of a data. SVM classifies data from 2 different classes with hyperplane. In this study, the system was built using SVM to develop Arabic Speech Recognition. In the development of the system, there are 2 kinds of speakers that have been tested that is dependent speakers and independent speakers. The results from this system is an accuracy of 85.32% for speaker dependent and 61.16% for independent speakers.

  1. Development of non-linear vibration analysis code for CANDU fuelling machine

    International Nuclear Information System (INIS)

    Murakami, Hajime; Hirai, Takeshi; Horikoshi, Kiyomi; Mizukoshi, Kaoru; Takenaka, Yasuo; Suzuki, Norio.

    1988-01-01

    This paper describes the development of a non-linear, dynamic analysis code for the CANDU 600 fuelling machine (F-M), which includes a number of non-linearities such as gap with or without Coulomb friction, special multi-linear spring connections, etc. The capabilities and features of the code and the mathematical treatment for the non-linearities are explained. The modeling and numerical methodology for the non-linearities employed in the code are verified experimentally. Finally, the simulation analyses for the full-scale F-M vibration testing are carried out, and the applicability of the code to such multi-degree of freedom systems as F-M is demonstrated. (author)

  2. Experiences and results multitasking a hydrodynamics code on global and local memory machines

    International Nuclear Information System (INIS)

    Mandell, D.

    1987-01-01

    A one-dimensional, time-dependent Lagrangian hydrodynamics code using a Godunov solution method has been multimasked for the Cray X-MP/48, the Intel iPSC hypercube, the Alliant FX series and the IBM RP3 computers. Actual multitasking results have been obtained for the Cray, Intel and Alliant computers and simulated results were obtained for the Cray and RP3 machines. The differences in the methods required to multitask on each of the machines is discussed. Results are presented for a sample problem involving a shock wave moving down a channel. Comparisons are made between theoretical speedups, predicted by Amdahl's law, and the actual speedups obtained. The problems of debugging on the different machines are also described

  3. A video, text, and speech-driven realistic 3-d virtual head for human-machine interface.

    Science.gov (United States)

    Yu, Jun; Wang, Zeng-Fu

    2015-05-01

    A multiple inputs-driven realistic facial animation system based on 3-D virtual head for human-machine interface is proposed. The system can be driven independently by video, text, and speech, thus can interact with humans through diverse interfaces. The combination of parameterized model and muscular model is used to obtain a tradeoff between computational efficiency and high realism of 3-D facial animation. The online appearance model is used to track 3-D facial motion from video in the framework of particle filtering, and multiple measurements, i.e., pixel color value of input image and Gabor wavelet coefficient of illumination ratio image, are infused to reduce the influence of lighting and person dependence for the construction of online appearance model. The tri-phone model is used to reduce the computational consumption of visual co-articulation in speech synchronized viseme synthesis without sacrificing any performance. The objective and subjective experiments show that the system is suitable for human-machine interaction.

  4. Parallelization of MCNP Monte Carlo neutron and photon transport code in parallel virtual machine and message passing interface

    International Nuclear Information System (INIS)

    Deng Li; Xie Zhongsheng

    1999-01-01

    The coupled neutron and photon transport Monte Carlo code MCNP (version 3B) has been parallelized in parallel virtual machine (PVM) and message passing interface (MPI) by modifying a previous serial code. The new code has been verified by solving sample problems. The speedup increases linearly with the number of processors and the average efficiency is up to 99% for 12-processor. (author)

  5. Joint Machine Learning and Game Theory for Rate Control in High Efficiency Video Coding.

    Science.gov (United States)

    Gao, Wei; Kwong, Sam; Jia, Yuheng

    2017-08-25

    In this paper, a joint machine learning and game theory modeling (MLGT) framework is proposed for inter frame coding tree unit (CTU) level bit allocation and rate control (RC) optimization in High Efficiency Video Coding (HEVC). First, a support vector machine (SVM) based multi-classification scheme is proposed to improve the prediction accuracy of CTU-level Rate-Distortion (R-D) model. The legacy "chicken-and-egg" dilemma in video coding is proposed to be overcome by the learning-based R-D model. Second, a mixed R-D model based cooperative bargaining game theory is proposed for bit allocation optimization, where the convexity of the mixed R-D model based utility function is proved, and Nash bargaining solution (NBS) is achieved by the proposed iterative solution search method. The minimum utility is adjusted by the reference coding distortion and frame-level Quantization parameter (QP) change. Lastly, intra frame QP and inter frame adaptive bit ratios are adjusted to make inter frames have more bit resources to maintain smooth quality and bit consumption in the bargaining game optimization. Experimental results demonstrate that the proposed MLGT based RC method can achieve much better R-D performances, quality smoothness, bit rate accuracy, buffer control results and subjective visual quality than the other state-of-the-art one-pass RC methods, and the achieved R-D performances are very close to the performance limits from the FixedQP method.

  6. A Navier-Strokes Chimera Code on the Connection Machine CM-5: Design and Performance

    Science.gov (United States)

    Jespersen, Dennis C.; Levit, Creon; Kwak, Dochan (Technical Monitor)

    1994-01-01

    We have implemented a three-dimensional compressible Navier-Stokes code on the Connection Machine CM-5. The code is set up for implicit time-stepping on single or multiple structured grids. For multiple grids and geometrically complex problems, we follow the 'chimera' approach, where flow data on one zone is interpolated onto another in the region of overlap. We will describe our design philosophy and give some timing results for the current code. A parallel machine like the CM-5 is well-suited for finite-difference methods on structured grids. The regular pattern of connections of a structured mesh maps well onto the architecture of the machine. So the first design choice, finite differences on a structured mesh, is natural. We use centered differences in space, with added artificial dissipation terms. When numerically solving the Navier-Stokes equations, there are liable to be some mesh cells near a solid body that are small in at least one direction. This mesh cell geometry can impose a very severe CFL (Courant-Friedrichs-Lewy) condition on the time step for explicit time-stepping methods. Thus, though explicit time-stepping is well-suited to the architecture of the machine, we have adopted implicit time-stepping. We have further taken the approximate factorization approach. This creates the need to solve large banded linear systems and creates the first possible barrier to an efficient algorithm. To overcome this first possible barrier we have considered two options. The first is just to solve the banded linear systems with data spread over the whole machine, using whatever fast method is available. This option is adequate for solving scalar tridiagonal systems, but for scalar pentadiagonal or block tridiagonal systems it is somewhat slower than desired. The second option is to 'transpose' the flow and geometry variables as part of the time-stepping process: Start with x-lines of data in-processor. Form explicit terms in x, then transpose so y-lines of data are

  7. Evaluation of pitch coding alternatives for vibrotactile stimulation in speech training of the deaf

    International Nuclear Information System (INIS)

    Barbacena, I L; Barros, A T; Freire, R C S; Vieira, E C A

    2007-01-01

    Use of vibrotactile feedback stimulation as an aid for speech vocalization by the hearing impaired or deaf is reviewed. Architecture of a vibrotactile based speech therapy system is proposed. Different formulations for encoding the fundamental frequency of the vocalized speech into the pulsed stimulation frequency are proposed and investigated. Simulation results are also presented to obtain a comparative evaluation of the effectiveness of the different formulated transformations. Results of the perception sensitivity to the vibrotactile stimulus frequency to verify effectiveness of the above transformations are included

  8. Evaluation of pitch coding alternatives for vibrotactile stimulation in speech training of the deaf

    Energy Technology Data Exchange (ETDEWEB)

    Barbacena, I L; Barros, A T [CEFET/PB, Joao Pessoa - PB (Brazil); Freire, R C S [DEE, UFCG, Campina Grande-PB (Brazil); Vieira, E C A [CEFET/PB, Joao Pessoa - PB (Brazil)

    2007-11-15

    Use of vibrotactile feedback stimulation as an aid for speech vocalization by the hearing impaired or deaf is reviewed. Architecture of a vibrotactile based speech therapy system is proposed. Different formulations for encoding the fundamental frequency of the vocalized speech into the pulsed stimulation frequency are proposed and investigated. Simulation results are also presented to obtain a comparative evaluation of the effectiveness of the different formulated transformations. Results of the perception sensitivity to the vibrotactile stimulus frequency to verify effectiveness of the above transformations are included.

  9. Brain cells in the avian 'prefrontal cortex' code for features of slot-machine-like gambling.

    Directory of Open Access Journals (Sweden)

    Damian Scarf

    2011-01-01

    Full Text Available Slot machines are the most common and addictive form of gambling. In the current study, we recorded from single neurons in the 'prefrontal cortex' of pigeons while they played a slot-machine-like task. We identified four categories of neurons that coded for different aspects of our slot-machine-like task. Reward-Proximity neurons showed a linear increase in activity as the opportunity for a reward drew near. I-Won neurons fired only when the fourth stimulus of a winning (four-of-a-kind combination was displayed. I-Lost neurons changed their firing rate at the presentation of the first nonidentical stimulus, that is, when it was apparent that no reward was forthcoming. Finally, Near-Miss neurons also changed their activity the moment it was recognized that a reward was no longer available, but more importantly, the activity level was related to whether the trial contained one, two, or three identical stimuli prior to the display of the nonidentical stimulus. These findings not only add to recent neurophysiological research employing simulated gambling paradigms, but also add to research addressing the functional correspondence between the avian NCL and primate PFC.

  10. From Vocal Replication to Shared Combinatorial Speech Codes: A Small Step for Evolution, A Big Step for Language

    Science.gov (United States)

    Oudeyer, Pierre-Yves

    Humans use spoken vocalizations, or their signed equivalent, as a physical support to carry language. This support is highly organized: vocalizations are built with the re-use of a small number of articulatory units, which are themselves discrete elements carved up by each linguistic community in the articulatory continuum. Moreover, the repertoires of these elementary units (the gestures, the phonemes, the morphemes) have a number of structural regularities: for example, while our vocal tract allows physically the production of hundreds of vowels, each language uses most often 5, and never more than 20 of them. Also, certain vowels are very frequent, like /a,e,i,o,u/, and some others are very rare, like /en/. All the speakers of a given linguistic community categorize the speech sounds in the same manner, and share the same repertoire of vocalizations. Speakers of different communities may have very different ways of categorizing sounds (for example, Chinese use tones to distinguish sounds), and repertoires of vocalizations. Such an organized physical support of language is crucial for the existence of language, and thus asking how it may have appeared in the biological and/or cultural history of humans is a fundamental questions. In particular, one can wonder how much the evolution of human speech codes relied on specific evolutionary innovations, and thus how difficult (or not) it was for speech to appear.

  11. Breaking the language barrier: machine assisted diagnosis using the medical speech translator.

    Science.gov (United States)

    Starlander, Marianne; Bouillon, Pierrette; Rayner, Manny; Chatzichrisafis, Nikos; Hockey, Beth Ann; Isahara, Hitoshi; Kanzaki, Kyoko; Nakao, Yukie; Santaholma, Marianne

    2005-01-01

    In this paper, we describe and evaluate an Open Source medical speech translation system (MedSLT) intended for safety-critical applications. The aim of this system is to eliminate the language barriers in emergency situation. It translates spoken questions from English into French, Japanese and Finnish in three medical subdomains (headache, chest pain and abdominal pain), using a vocabulary of about 250-400 words per sub-domain. The architecture is a compromise between fixed-phrase translation on one hand and complex linguistically-based systems on the other. Recognition is guided by a Context Free Grammar Language Model compiled from a general unification grammar, automatically specialised for the domain. We present an evaluation of this initial prototype that shows the advantages of this grammar-based approach for this particular translation task in term of both reliability and use.

  12. Sparse coding of the modulation spectrum for noise-robust automatic speech recognition

    NARCIS (Netherlands)

    Ahmadi, S.; Ahadi, S.M.; Cranen, B.; Boves, L.W.J.

    2014-01-01

    The full modulation spectrum is a high-dimensional representation of one-dimensional audio signals. Most previous research in automatic speech recognition converted this very rich representation into the equivalent of a sequence of short-time power spectra, mainly to simplify the computation of the

  13. The development of speech coding and the first standard coder for public mobile telephony

    NARCIS (Netherlands)

    Sluijter, R.J.

    2005-01-01

    This thesis describes in its core chapter (Chapter 4) the original algorithmic and design features of the ??rst coder for public mobile telephony, the GSM full-rate speech coder, as standardized in 1988. It has never been described in so much detail as presented here. The coder is put in a

  14. Particle-in-cell plasma simulation codes on the connection machine

    International Nuclear Information System (INIS)

    Walker, D.W.

    1991-01-01

    Methods for implementing three-dimensional, electromagnetic, relativistic PIC plasma simulation codes on the Connection Machine (CM-2) are discussed. The gather and scatter phases of the PIC algorithm involve indirect indexing of data, which results in a large amount of communication on the CM-2. Different data decompositions are described that seek to reduce the amount of communication while maintaining good load balance. These methods require the particles to be spatially sorted at the start of each time step, which introduced another form of overhead. The different methods are implemented in CM Fortran on the CM-2 and compared. It was found that the general router is slow in performing the communication in the gather and scatter steps, which precludes an efficient CM Fortran implementation. An alternative method that uses PARIS calls and the NEWS communication network to pipeline data along the axes of the VP set is suggested as a more efficient algorithm

  15. An Adaptive Approach to a 2.4 kb/s LPC Speech Coding System.

    Science.gov (United States)

    1985-07-01

    laryngeal cancer ). Spectral estimation is at the foundation of speech analysis for all these goals and accurate AR model estimation in noise is...S ,5 mWnL NrinKt ) o ,-G p (d va Rmea.imn flU: 5() WOM Lu M(G)INUNM 40 4KeemS! MU= 1 UD M5) SIGHSM A SO= WAGe . M. (d) I U NS maIm ( IW vis MAMA

  16. A Support Vector Machine-Based Gender Identification Using Speech Signal

    Science.gov (United States)

    Lee, Kye-Hwan; Kang, Sang-Ick; Kim, Deok-Hwan; Chang, Joon-Hyuk

    We propose an effective voice-based gender identification method using a support vector machine (SVM). The SVM is a binary classification algorithm that classifies two groups by finding the voluntary nonlinear boundary in a feature space and is known to yield high classification performance. In the present work, we compare the identification performance of the SVM with that of a Gaussian mixture model (GMM)-based method using the mel frequency cepstral coefficients (MFCC). A novel approach of incorporating a features fusion scheme based on a combination of the MFCC and the fundamental frequency is proposed with the aim of improving the performance of gender identification. Experimental results demonstrate that the gender identification performance using the SVM is significantly better than that of the GMM-based scheme. Moreover, the performance is substantially improved when the proposed features fusion technique is applied.

  17. A sparse neural code for some speech sounds but not for others.

    Directory of Open Access Journals (Sweden)

    Mathias Scharinger

    Full Text Available The precise neural mechanisms underlying speech sound representations are still a matter of debate. Proponents of 'sparse representations' assume that on the level of speech sounds, only contrastive or otherwise not predictable information is stored in long-term memory. Here, in a passive oddball paradigm, we challenge the neural foundations of such a 'sparse' representation; we use words that differ only in their penultimate consonant ("coronal" [t] vs. "dorsal" [k] place of articulation and for example distinguish between the German nouns Latz ([lats]; bib and Lachs ([laks]; salmon. Changes from standard [t] to deviant [k] and vice versa elicited a discernible Mismatch Negativity (MMN response. Crucially, however, the MMN for the deviant [lats] was stronger than the MMN for the deviant [laks]. Source localization showed this difference to be due to enhanced brain activity in right superior temporal cortex. These findings reflect a difference in phonological 'sparsity': Coronal [t] segments, but not dorsal [k] segments, are based on more sparse representations and elicit less specific neural predictions; sensory deviations from this prediction are more readily 'tolerated' and accordingly trigger weaker MMNs. The results foster the neurocomputational reality of 'representationally sparse' models of speech perception that are compatible with more general predictive mechanisms in auditory perception.

  18. Event-related potential evidence of form and meaning coding during online speech recognition.

    Science.gov (United States)

    Friedrich, Claudia K; Kotz, Sonja A

    2007-04-01

    It is still a matter of debate whether initial analysis of speech is independent of contextual influences or whether meaning can modulate word activation directly. Utilizing event-related brain potentials (ERPs), we tested the neural correlates of speech recognition by presenting sentences that ended with incomplete words, such as To light up the dark she needed her can-. Immediately following the incomplete words, subjects saw visual words that (i) matched form and meaning, such as candle; (ii) matched meaning but not form, such as lantern; (iii) matched form but not meaning, such as candy; or (iv) mismatched form and meaning, such as number. We report ERP evidence for two distinct cohorts of lexical tokens: (a) a left-lateralized effect, the P250, differentiates form-matching words (i, iii) and form-mismatching words (ii, iv); (b) a right-lateralized effect, the P220, differentiates words that match in form and/or meaning (i, ii, iii) from mismatching words (iv). Lastly, fully matching words (i) reduce the amplitude of the N400. These results accommodate bottom-up and top-down accounts of human speech recognition. They suggest that neural representations of form and meaning are activated independently early on and are integrated at a later stage during sentence comprehension.

  19. Verbal Short-Term Memory Span in Speech-Disordered Children: Implications for Articulatory Coding in Short-Term Memory.

    Science.gov (United States)

    Raine, Adrian; And Others

    1991-01-01

    Children with speech disorders had lower short-term memory capacity and smaller word length effect than control children. Children with speech disorders also had reduced speech-motor activity during rehearsal. Results suggest that speech rate may be a causal determinant of verbal short-term memory capacity. (BC)

  20. Optimization and Openmp Parallelization of a Discrete Element Code for Convex Polyhedra on Multi-Core Machines

    Science.gov (United States)

    Chen, Jian; Matuttis, Hans-Georg

    2013-02-01

    We report our experiences with the optimization and parallelization of a discrete element code for convex polyhedra on multi-core machines and introduce a novel variant of the sort-and-sweep neighborhood algorithm. While in theory the whole code in itself parallelizes ideally, in practice the results on different architectures with different compilers and performance measurement tools depend very much on the particle number and optimization of the code. After difficulties with the interpretation of the data for speedup and efficiency are overcome, respectable parallelization speedups could be obtained.

  1. Stochastic algorithm for channel optimized vector quantization: application to robust narrow-band speech coding

    International Nuclear Information System (INIS)

    Bouzid, M.; Benkherouf, H.; Benzadi, K.

    2011-01-01

    In this paper, we propose a stochastic joint source-channel scheme developed for efficient and robust encoding of spectral speech LSF parameters. The encoding system, named LSF-SSCOVQ-RC, is an LSF encoding scheme based on a reduced complexity stochastic split vector quantizer optimized for noisy channel. For transmissions over noisy channel, we will show first that our LSF-SSCOVQ-RC encoder outperforms the conventional LSF encoder designed by the split vector quantizer. After that, we applied the LSF-SSCOVQ-RC encoder (with weighted distance) for the robust encoding of LSF parameters of the 2.4 Kbits/s MELP speech coder operating over a noisy/noiseless channel. The simulation results will show that the proposed LSF encoder, incorporated in the MELP, ensure better performances than the original MELP MSVQ of 25 bits/frame; especially when the transmission channel is highly disturbed. Indeed, we will show that the LSF-SSCOVQ-RC yields significant improvement to the LSFs encoding performances by ensuring reliable transmissions over noisy channel.

  2. Virtual machine provisioning, code management, and data movement design for the Fermilab HEPCloud Facility

    Science.gov (United States)

    Timm, S.; Cooper, G.; Fuess, S.; Garzoglio, G.; Holzman, B.; Kennedy, R.; Grassano, D.; Tiradani, A.; Krishnamurthy, R.; Vinayagam, S.; Raicu, I.; Wu, H.; Ren, S.; Noh, S.-Y.

    2017-10-01

    The Fermilab HEPCloud Facility Project has as its goal to extend the current Fermilab facility interface to provide transparent access to disparate resources including commercial and community clouds, grid federations, and HPC centers. This facility enables experiments to perform the full spectrum of computing tasks, including data-intensive simulation and reconstruction. We have evaluated the use of the commercial cloud to provide elasticity to respond to peaks of demand without overprovisioning local resources. Full scale data-intensive workflows have been successfully completed on Amazon Web Services for two High Energy Physics Experiments, CMS and NOνA, at the scale of 58000 simultaneous cores. This paper describes the significant improvements that were made to the virtual machine provisioning system, code caching system, and data movement system to accomplish this work. The virtual image provisioning and contextualization service was extended to multiple AWS regions, and to support experiment-specific data configurations. A prototype Decision Engine was written to determine the optimal availability zone and instance type to run on, minimizing cost and job interruptions. We have deployed a scalable on-demand caching service to deliver code and database information to jobs running on the commercial cloud. It uses the frontiersquid server and CERN VM File System (CVMFS) clients on EC2 instances and utilizes various services provided by AWS to build the infrastructure (stack). We discuss the architecture and load testing benchmarks on the squid servers. We also describe various approaches that were evaluated to transport experimental data to and from the cloud, and the optimal solutions that were used for the bulk of the data transport. Finally, we summarize lessons learned from this scale test, and our future plans to expand and improve the Fermilab HEP Cloud Facility.

  3. Virtual Machine Provisioning, Code Management, and Data Movement Design for the Fermilab HEPCloud Facility

    Energy Technology Data Exchange (ETDEWEB)

    Timm, S. [Fermilab; Cooper, G. [Fermilab; Fuess, S. [Fermilab; Garzoglio, G. [Fermilab; Holzman, B. [Fermilab; Kennedy, R. [Fermilab; Grassano, D. [Fermilab; Tiradani, A. [Fermilab; Krishnamurthy, R. [IIT, Chicago; Vinayagam, S. [IIT, Chicago; Raicu, I. [IIT, Chicago; Wu, H. [IIT, Chicago; Ren, S. [IIT, Chicago; Noh, S. Y. [KISTI, Daejeon

    2017-11-22

    The Fermilab HEPCloud Facility Project has as its goal to extend the current Fermilab facility interface to provide transparent access to disparate resources including commercial and community clouds, grid federations, and HPC centers. This facility enables experiments to perform the full spectrum of computing tasks, including data-intensive simulation and reconstruction. We have evaluated the use of the commercial cloud to provide elasticity to respond to peaks of demand without overprovisioning local resources. Full scale data-intensive workflows have been successfully completed on Amazon Web Services for two High Energy Physics Experiments, CMS and NOνA, at the scale of 58000 simultaneous cores. This paper describes the significant improvements that were made to the virtual machine provisioning system, code caching system, and data movement system to accomplish this work. The virtual image provisioning and contextualization service was extended to multiple AWS regions, and to support experiment-specific data configurations. A prototype Decision Engine was written to determine the optimal availability zone and instance type to run on, minimizing cost and job interruptions. We have deployed a scalable on-demand caching service to deliver code and database information to jobs running on the commercial cloud. It uses the frontiersquid server and CERN VM File System (CVMFS) clients on EC2 instances and utilizes various services provided by AWS to build the infrastructure (stack). We discuss the architecture and load testing benchmarks on the squid servers. We also describe various approaches that were evaluated to transport experimental data to and from the cloud, and the optimal solutions that were used for the bulk of the data transport. Finally, we summarize lessons learned from this scale test, and our future plans to expand and improve the Fermilab HEP Cloud Facility.

  4. A new method for species identification via protein-coding and non-coding DNA barcodes by combining machine learning with bioinformatic methods.

    Directory of Open Access Journals (Sweden)

    Ai-bing Zhang

    Full Text Available Species identification via DNA barcodes is contributing greatly to current bioinventory efforts. The initial, and widely accepted, proposal was to use the protein-coding cytochrome c oxidase subunit I (COI region as the standard barcode for animals, but recently non-coding internal transcribed spacer (ITS genes have been proposed as candidate barcodes for both animals and plants. However, achieving a robust alignment for non-coding regions can be problematic. Here we propose two new methods (DV-RBF and FJ-RBF to address this issue for species assignment by both coding and non-coding sequences that take advantage of the power of machine learning and bioinformatics. We demonstrate the value of the new methods with four empirical datasets, two representing typical protein-coding COI barcode datasets (neotropical bats and marine fish and two representing non-coding ITS barcodes (rust fungi and brown algae. Using two random sub-sampling approaches, we demonstrate that the new methods significantly outperformed existing Neighbor-joining (NJ and Maximum likelihood (ML methods for both coding and non-coding barcodes when there was complete species coverage in the reference dataset. The new methods also out-performed NJ and ML methods for non-coding sequences in circumstances of potentially incomplete species coverage, although then the NJ and ML methods performed slightly better than the new methods for protein-coding barcodes. A 100% success rate of species identification was achieved with the two new methods for 4,122 bat queries and 5,134 fish queries using COI barcodes, with 95% confidence intervals (CI of 99.75-100%. The new methods also obtained a 96.29% success rate (95%CI: 91.62-98.40% for 484 rust fungi queries and a 98.50% success rate (95%CI: 96.60-99.37% for 1094 brown algae queries, both using ITS barcodes.

  5. A new method for species identification via protein-coding and non-coding DNA barcodes by combining machine learning with bioinformatic methods.

    Science.gov (United States)

    Zhang, Ai-bing; Feng, Jie; Ward, Robert D; Wan, Ping; Gao, Qiang; Wu, Jun; Zhao, Wei-zhong

    2012-01-01

    Species identification via DNA barcodes is contributing greatly to current bioinventory efforts. The initial, and widely accepted, proposal was to use the protein-coding cytochrome c oxidase subunit I (COI) region as the standard barcode for animals, but recently non-coding internal transcribed spacer (ITS) genes have been proposed as candidate barcodes for both animals and plants. However, achieving a robust alignment for non-coding regions can be problematic. Here we propose two new methods (DV-RBF and FJ-RBF) to address this issue for species assignment by both coding and non-coding sequences that take advantage of the power of machine learning and bioinformatics. We demonstrate the value of the new methods with four empirical datasets, two representing typical protein-coding COI barcode datasets (neotropical bats and marine fish) and two representing non-coding ITS barcodes (rust fungi and brown algae). Using two random sub-sampling approaches, we demonstrate that the new methods significantly outperformed existing Neighbor-joining (NJ) and Maximum likelihood (ML) methods for both coding and non-coding barcodes when there was complete species coverage in the reference dataset. The new methods also out-performed NJ and ML methods for non-coding sequences in circumstances of potentially incomplete species coverage, although then the NJ and ML methods performed slightly better than the new methods for protein-coding barcodes. A 100% success rate of species identification was achieved with the two new methods for 4,122 bat queries and 5,134 fish queries using COI barcodes, with 95% confidence intervals (CI) of 99.75-100%. The new methods also obtained a 96.29% success rate (95%CI: 91.62-98.40%) for 484 rust fungi queries and a 98.50% success rate (95%CI: 96.60-99.37%) for 1094 brown algae queries, both using ITS barcodes.

  6. The Interpretation Of Speech Code In A Communication Ethnographic Context For Outsider Students Of Graduate Communication Science Universitas Sumatera Utara In Medan

    Directory of Open Access Journals (Sweden)

    Fauzi Eka Putra

    2017-06-01

    Full Text Available Interpreting the typical Medan speech code is something unique and distinctive which could create confusion for the outsider students because of the speech code similarities and differences in Medan. Therefore the graduate students of communication science Universitas Sumatera Utara whose originated from outside of North Sumatera needs to learn comprehend and aware in order to perform effective communication. The purpose of this research is to discover how the interpretation of speech code for the graduate students of communication science Universitas Sumatera Utara whose originated from outside of North Sumatera in adapting themselves in Medan. This research uses qualitative method with the study of ethnography and acculturation communication. The subject of this research is the graduate students of communication science Universitas Sumatera Utara whose originated from outside of North Sumatera in adapting themselves in Medan. Data were collected through interviews observation and documentation. The conclusion of this research shows that speech code interpretation by students from outside of North Sumatera in adapting themselves in Medan leads to an acculturation process of assimilation and integration.

  7. Superwideband Bandwidth Extension Using Normalized MDCT Coefficients for Scalable Speech and Audio Coding

    Directory of Open Access Journals (Sweden)

    Young Han Lee

    2013-01-01

    Full Text Available A bandwidth extension (BWE algorithm from wideband to superwideband (SWB is proposed for a scalable speech/audio codec that uses modified discrete cosine transform (MDCT coefficients as spectral parameters. The superwideband is first split into several subbands that are represented as gain parameters and normalized MDCT coefficients in the proposed BWE algorithm. We then estimate normalized MDCT coefficients of the wideband to be fetched for the superwideband and quantize the fetch indices. After that, we quantize gain parameters by using relative ratios between adjacent subbands. The proposed BWE algorithm is embedded into a standard superwideband codec, the SWB extension of G.729.1 Annex E, and its bitrate and quality are compared with those of the BWE algorithm already employed in the standard superwideband codec. It is shown from the comparison that the proposed BWE algorithm relatively reduces the bitrate by around 19% with better quality, compared to the BWE algorithm in the SWB extension of G.729.1 Annex E.

  8. Predicting automatic speech recognition performance over communication channels from instrumental speech quality and intelligibility scores

    NARCIS (Netherlands)

    Gallardo, L.F.; Möller, S.; Beerends, J.

    2017-01-01

    The performance of automatic speech recognition based on coded-decoded speech heavily depends on the quality of the transmitted signals, determined by channel impairments. This paper examines relationships between speech recognition performance and measurements of speech quality and intelligibility

  9. Vector Quantization of Harmonic Magnitudes in Speech Coding Applications—A Survey and New Technique

    Directory of Open Access Journals (Sweden)

    Wai C. Chu

    2004-12-01

    Full Text Available A harmonic coder extracts the harmonic components of a signal and represents them efficiently using a few parameters. The principles of harmonic coding have become quite successful and several standardized speech and audio coders are based on it. One of the key issues in harmonic coder design is in the quantization of harmonic magnitudes, where many propositions have appeared in the literature. The objective of this paper is to provide a survey of the various techniques that have appeared in the literature for vector quantization of harmonic magnitudes, with emphasis on those adopted by the major speech coding standards; these include constant magnitude approximation, partial quantization, dimension conversion, and variable-dimension vector quantization (VDVQ. In addition, a refined VDVQ technique is proposed where experimental data are provided to demonstrate its effectiveness.

  10. Speech enhancement

    CERN Document Server

    Benesty, Jacob; Chen, Jingdong

    2006-01-01

    We live in a noisy world! In all applications (telecommunications, hands-free communications, recording, human-machine interfaces, etc.) that require at least one microphone, the signal of interest is usually contaminated by noise and reverberation. As a result, the microphone signal has to be ""cleaned"" with digital signal processing tools before it is played out, transmitted, or stored.This book is about speech enhancement. Different well-known and state-of-the-art methods for noise reduction, with one or multiple microphones, are discussed. By speech enhancement, we mean not only noise red

  11. Multilevel Analysis in Analyzing Speech Data

    Science.gov (United States)

    Guddattu, Vasudeva; Krishna, Y.

    2011-01-01

    The speech produced by human vocal tract is a complex acoustic signal, with diverse applications in phonetics, speech synthesis, automatic speech recognition, speaker identification, communication aids, speech pathology, speech perception, machine translation, hearing research, rehabilitation and assessment of communication disorders and many…

  12. Extraction of state machines of legacy C code with Cpp2XMI

    NARCIS (Netherlands)

    Brand, van den M.G.J.; Serebrenik, A.; Zeeland, van D.; Serebrenik, A.

    2008-01-01

    Analysis of legacy code is often focussed on extracting either metrics or relations, e.g. call relations or structure relations. For object-oriented programs, e.g. Java or C++ code, such relations are commonly represented as UML diagrams: e.g., such tools as Columbus [1] and Cpp2XMI [2] are capable

  13. On Coding the States of Sequential Machines with the Use of Partition Pairs

    DEFF Research Database (Denmark)

    Zahle, Torben U.

    1966-01-01

    This article introduces a new technique of making state assignment for sequential machines. The technique is in line with the approach used by Hartmanis [l], Stearns and Hartmanis [3], and Curtis [4]. It parallels the work of Dolotta and McCluskey [7], although it was developed independently...

  14. Speech Alarms Pilot Study

    Science.gov (United States)

    Sandor, Aniko; Moses, Haifa

    2016-01-01

    Speech alarms have been used extensively in aviation and included in International Building Codes (IBC) and National Fire Protection Association's (NFPA) Life Safety Code. However, they have not been implemented on space vehicles. Previous studies conducted at NASA JSC showed that speech alarms lead to faster identification and higher accuracy. This research evaluated updated speech and tone alerts in a laboratory environment and in the Human Exploration Research Analog (HERA) in a realistic setup.

  15. Monte Carlo simulation of a multi-leaf collimator design for telecobalt machine using BEAMnrc code

    Directory of Open Access Journals (Sweden)

    Ayyangar Komanduri

    2010-01-01

    Full Text Available This investigation aims to design a practical multi-leaf collimator (MLC system for the cobalt teletherapy machine and check its radiation properties using the Monte Carlo (MC method. The cobalt machine was modeled using the BEAMnrc Omega-Beam MC system, which could be freely downloaded from the website of the National Research Council (NRC, Canada. Comparison with standard depth dose data tables and the theoretically modeled beam showed good agreement within 2%. An MLC design with low melting point alloy (LMPA was tested for leakage properties of leaves. The LMPA leaves with a width of 7 mm and height of 6 cm, with tongue and groove of size 2 mm wide by 4 cm height, produced only 4% extra leakage compared to 10 cm height tungsten leaves. With finite 60 Co source size, the interleaf leakage was insignificant. This analysis helped to design a prototype MLC as an accessory mount on a cobalt machine. The complete details of the simulation process and analysis of results are discussed.

  16. Monte Carlo simulation of a multi-leaf collimator design for telecobalt machine using BEAMnrc code

    International Nuclear Information System (INIS)

    Ayyangar, Komanduri M.; Narayan, Pradush; Jesuraj, Fenedit; Raju, M.R.; Dinesh Kumar, M.

    2010-01-01

    This investigation aims to design a practical multi-leaf collimator (MLC) system for the cobalt teletherapy machine and check its radiation properties using the Monte Carlo (MC) method. The cobalt machine was modeled using the BEAMnrc Omega-Beam MC system, which could be freely downloaded from the website of the National Research Council (NRC), Canada. Comparison with standard depth dose data tables and the theoretically modeled beam showed good agreement within 2%. An MLC design with low melting point alloy (LMPA) was tested for leakage properties of leaves. The LMPA leaves with a width of 7 mm and height of 6 cm, with tongue and groove of size 2 mm wide by 4 cm height, produced only 4% extra leakage compared to 10 cm height tungsten leaves. With finite 60 Co source size, the interleaf leakage was insignificant. This analysis helped to design a prototype MLC as an accessory mount on a cobalt machine. The complete details of the simulation process and analysis of results are discussed. (author)

  17. Computer codes for the calculation of vibrations in machines and structures

    International Nuclear Information System (INIS)

    1989-01-01

    After an introductory paper on the typical requirements to be met by vibration calculations, the first two sections of the conference papers present universal as well as specific finite-element codes tailored to solve individual problems. The calculation of dynamic processes increasingly now in addition to the finite elements applies the method of multi-component systems which takes into account rigid bodies or partial structures and linking and joining elements. This method, too, is explained referring to universal computer codes and to special versions. In mechanical engineering, rotary vibrations are a major problem, and under this topic, conference papers exclusively deal with codes that also take into account special effects such as electromechanical coupling, non-linearities in clutches, etc. (orig./HP) [de

  18. lncRScan-SVM: A Tool for Predicting Long Non-Coding RNAs Using Support Vector Machine.

    Science.gov (United States)

    Sun, Lei; Liu, Hui; Zhang, Lin; Meng, Jia

    2015-01-01

    Functional long non-coding RNAs (lncRNAs) have been bringing novel insight into biological study, however it is still not trivial to accurately distinguish the lncRNA transcripts (LNCTs) from the protein coding ones (PCTs). As various information and data about lncRNAs are preserved by previous studies, it is appealing to develop novel methods to identify the lncRNAs more accurately. Our method lncRScan-SVM aims at classifying PCTs and LNCTs using support vector machine (SVM). The gold-standard datasets for lncRScan-SVM model training, lncRNA prediction and method comparison were constructed according to the GENCODE gene annotations of human and mouse respectively. By integrating features derived from gene structure, transcript sequence, potential codon sequence and conservation, lncRScan-SVM outperforms other approaches, which is evaluated by several criteria such as sensitivity, specificity, accuracy, Matthews correlation coefficient (MCC) and area under curve (AUC). In addition, several known human lncRNA datasets were assessed using lncRScan-SVM. LncRScan-SVM is an efficient tool for predicting the lncRNAs, and it is quite useful for current lncRNA study.

  19. Practical speech user interface design

    CERN Document Server

    Lewis, James R

    2010-01-01

    Although speech is the most natural form of communication between humans, most people find using speech to communicate with machines anything but natural. Drawing from psychology, human-computer interaction, linguistics, and communication theory, Practical Speech User Interface Design provides a comprehensive yet concise survey of practical speech user interface (SUI) design. It offers practice-based and research-based guidance on how to design effective, efficient, and pleasant speech applications that people can really use. Focusing on the design of speech user interfaces for IVR application

  20. Voice Activity Detection. Fundamentals and Speech Recognition System Robustness

    OpenAIRE

    Ramirez, J.; Gorriz, J. M.; Segura, J. C.

    2007-01-01

    This chapter has shown an overview of the main challenges in robust speech detection and a review of the state of the art and applications. VADs are frequently used in a number of applications including speech coding, speech enhancement and speech recognition. A precise VAD extracts a set of discriminative speech features from the noisy speech and formulates the decision in terms of well defined rule. The chapter has summarized three robust VAD methods that yield high speech/non-speech discri...

  1. Human phoneme recognition depending on speech-intrinsic variability.

    Science.gov (United States)

    Meyer, Bernd T; Jürgens, Tim; Wesker, Thorsten; Brand, Thomas; Kollmeier, Birger

    2010-11-01

    The influence of different sources of speech-intrinsic variation (speaking rate, effort, style and dialect or accent) on human speech perception was investigated. In listening experiments with 16 listeners, confusions of consonant-vowel-consonant (CVC) and vowel-consonant-vowel (VCV) sounds in speech-weighted noise were analyzed. Experiments were based on the OLLO logatome speech database, which was designed for a man-machine comparison. It contains utterances spoken by 50 speakers from five dialect/accent regions and covers several intrinsic variations. By comparing results depending on intrinsic and extrinsic variations (i.e., different levels of masking noise), the degradation induced by variabilities can be expressed in terms of the SNR. The spectral level distance between the respective speech segment and the long-term spectrum of the masking noise was found to be a good predictor for recognition rates, while phoneme confusions were influenced by the distance to spectrally close phonemes. An analysis based on transmitted information of articulatory features showed that voicing and manner of articulation are comparatively robust cues in the presence of intrinsic variations, whereas the coding of place is more degraded. The database and detailed results have been made available for comparisons between human speech recognition (HSR) and automatic speech recognizers (ASR).

  2. From Phonemes to Articulatory Codes: An fMRI Study of the Role of Broca's Area in Speech Production

    Science.gov (United States)

    de Zwart, Jacco A.; Jansma, J. Martijn; Pickering, Martin J.; Bednar, James A.; Horwitz, Barry

    2009-01-01

    We used event-related functional magnetic resonance imaging to investigate the neuroanatomical substrates of phonetic encoding and the generation of articulatory codes from phonological representations. Our focus was on the role of the left inferior frontal gyrus (LIFG) and in particular whether the LIFG plays a role in sublexical phonological processing such as syllabification or whether it is directly involved in phonetic encoding and the generation of articulatory codes. To answer this question, we contrasted the brain activation patterns elicited by pseudowords with high– or low–sublexical frequency components, which we expected would reveal areas related to the generation of articulatory codes but not areas related to phonological encoding. We found significant activation of a premotor network consisting of the dorsal precentral gyrus, the inferior frontal gyrus bilaterally, and the supplementary motor area for low– versus high–sublexical frequency pseudowords. Based on our hypothesis, we concluded that these areas and in particular the LIFG are involved in phonetic and not phonological encoding. We further discuss our findings with respect to the mechanisms of phonetic encoding and provide evidence in support of a functional segregation of the posterior part of Broca's area, the pars opercularis. PMID:19181696

  3. Potential role of monkey inferior parietal neurons coding action semantic equivalences as precursors of parts of speech.

    Science.gov (United States)

    Yamazaki, Yumiko; Yokochi, Hiroko; Tanaka, Michio; Okanoya, Kazuo; Iriki, Atsushi

    2010-01-01

    The anterior portion of the inferior parietal cortex possesses comprehensive representations of actions embedded in behavioural contexts. Mirror neurons, which respond to both self-executed and observed actions, exist in this brain region in addition to those originally found in the premotor cortex. We found that parietal mirror neurons responded differentially to identical actions embedded in different contexts. Another type of parietal mirror neuron represents an inverse and complementary property of responding equally to dissimilar actions made by itself and others for an identical purpose. Here, we propose a hypothesis that these sets of inferior parietal neurons constitute a neural basis for encoding the semantic equivalence of various actions across different agents and contexts. The neurons have mirror neuron properties, and they encoded generalization of agents, differentiation of outcomes, and categorization of actions that led to common functions. By integrating the activities of these mirror neurons with various codings, we further suggest that in the ancestral primates' brains, these various representations of meaningful action led to the gradual establishment of equivalence relations among the different types of actions, by sharing common action semantics. Such differential codings of the components of actions might represent precursors to the parts of protolanguage, such as gestural communication, which are shared among various members of a society. Finally, we suggest that the inferior parietal cortex serves as an interface between this action semantics system and other higher semantic systems, through common structures of action representation that mimic language syntax.

  4. Speech Problems

    Science.gov (United States)

    ... Staying Safe Videos for Educators Search English Español Speech Problems KidsHealth / For Teens / Speech Problems What's in ... a person's ability to speak clearly. Some Common Speech and Language Disorders Stuttering is a problem that ...

  5. The development of a speech act coding scheme to characterize communication patterns under an off-normal situation in nuclear power plants

    International Nuclear Information System (INIS)

    Kim, Seung Hwan; Park, Jin Kyun

    2009-01-01

    Since communication is an important means to exchange information between individuals/teams or auxiliary means to share resources and information given in the team and group activity, effective communication is the prerequisite for construct powerful teamwork by a sharing mental model. Therefore, unless communication is performed efficiently, the quality of task and performance of team lower. Furthermore, since communication is highly related to situation awareness during team activities, inappropriate communication causes a lack of situation awareness and tension and stress are intensified and errors are increased. According to lesson learned from several accidents that have actually occurred in nuclear power plant (NPP), consequence of accident leads most critical results and is more dangerous than those of other industries. In order to improve operator's cope ability and operation ability through simulation training with various off-normal condition, the operation groups are trained regularly every 6 months in the training center of reference NPP. The objective of this study is to suggest modified speech act coding scheme and to elucidate the communication pattern characteristics of an operator's conversation during an abnormal situation in NPP

  6. The development of a speech act coding scheme to characterize communication patterns under an off-normal situation in nuclear power plants

    Energy Technology Data Exchange (ETDEWEB)

    Kim, Seung Hwan; Park, Jin Kyun [Korea Atomic Energy Research Institute, Daejeon (Korea, Republic of)

    2009-05-15

    Since communication is an important means to exchange information between individuals/teams or auxiliary means to share resources and information given in the team and group activity, effective communication is the prerequisite for construct powerful teamwork by a sharing mental model. Therefore, unless communication is performed efficiently, the quality of task and performance of team lower. Furthermore, since communication is highly related to situation awareness during team activities, inappropriate communication causes a lack of situation awareness and tension and stress are intensified and errors are increased. According to lesson learned from several accidents that have actually occurred in nuclear power plant (NPP), consequence of accident leads most critical results and is more dangerous than those of other industries. In order to improve operator's cope ability and operation ability through simulation training with various off-normal condition, the operation groups are trained regularly every 6 months in the training center of reference NPP. The objective of this study is to suggest modified speech act coding scheme and to elucidate the communication pattern characteristics of an operator's conversation during an abnormal situation in NPP.

  7. Use of system code to estimate equilibrium tritium inventory in fusion DT machines, such as ARIES-AT and components testing facilities

    International Nuclear Information System (INIS)

    Wong, C.P.C.; Merrill, B.

    2014-01-01

    Highlights: • With the use of a system code, tritium burn-up fraction (f burn ) can be determined. • Initial tritium inventory for steady state DT machines can be estimated. • f burn of ARIES-AT, CFETR and FNSF-AT are in the range of 1–2.8%. • Respective total tritium inventories of are 7.6 kg, 6.1 kg, and 5.2 kg. - Abstract: ITER is under construction and will begin operation in 2020. This is the first 500 MW fusion class DT device, and since it is not going to breed tritium, it will consume most of the limited supply of tritium resources in the world. Yet, in parallel, DT fusion nuclear component testing machines will be needed to provide technical data for the design of DEMO. It becomes necessary to estimate the tritium burn-up fraction and corresponding initial tritium inventory and the doubling time of these machines for the planning of future supply and utilization of tritium. With the use of a system code, tritium burn-up fraction and initial tritium inventory for steady state DT machines can be estimated. Estimated tritium burn-up fractions of FNSF-AT, CFETR-R and ARIES-AT are in the range of 1–2.8%. Corresponding total equilibrium tritium inventories of the plasma flow and tritium processing system, and with the DCLL blanket option are 7.6 kg, 6.1 kg, and 5.2 kg for ARIES-AT, CFETR-R and FNSF-AT, respectively

  8. Use of system code to estimate equilibrium tritium inventory in fusion DT machines, such as ARIES-AT and components testing facilities

    Energy Technology Data Exchange (ETDEWEB)

    Wong, C.P.C., E-mail: wongc@fusion.gat.com [General Atomics, San Diego, CA (United States); Merrill, B. [Idaho National Laboratory, Idaho Falls, ID (United States)

    2014-10-15

    Highlights: • With the use of a system code, tritium burn-up fraction (f{sub burn}) can be determined. • Initial tritium inventory for steady state DT machines can be estimated. • f{sub burn} of ARIES-AT, CFETR and FNSF-AT are in the range of 1–2.8%. • Respective total tritium inventories of are 7.6 kg, 6.1 kg, and 5.2 kg. - Abstract: ITER is under construction and will begin operation in 2020. This is the first 500 MW{sub fusion} class DT device, and since it is not going to breed tritium, it will consume most of the limited supply of tritium resources in the world. Yet, in parallel, DT fusion nuclear component testing machines will be needed to provide technical data for the design of DEMO. It becomes necessary to estimate the tritium burn-up fraction and corresponding initial tritium inventory and the doubling time of these machines for the planning of future supply and utilization of tritium. With the use of a system code, tritium burn-up fraction and initial tritium inventory for steady state DT machines can be estimated. Estimated tritium burn-up fractions of FNSF-AT, CFETR-R and ARIES-AT are in the range of 1–2.8%. Corresponding total equilibrium tritium inventories of the plasma flow and tritium processing system, and with the DCLL blanket option are 7.6 kg, 6.1 kg, and 5.2 kg for ARIES-AT, CFETR-R and FNSF-AT, respectively.

  9. A study of speech interfaces for the vehicle environment.

    Science.gov (United States)

    2013-05-01

    Over the past few years, there has been a shift in automotive human machine interfaces from : visual-manual interactions (pushing buttons and rotating knobs) to speech interaction. In terms of : distraction, the industry views speech interaction as a...

  10. Improved Methods for Pitch Synchronous Linear Prediction Analysis of Speech

    OpenAIRE

    劉, 麗清

    2015-01-01

    Linear prediction (LP) analysis has been applied to speech system over the last few decades. LP technique is well-suited for speech analysis due to its ability to model speech production process approximately. Hence LP analysis has been widely used for speech enhancement, low-bit-rate speech coding in cellular telephony, speech recognition, characteristic parameter extraction (vocal tract resonances frequencies, fundamental frequency called pitch) and so on. However, the performance of the co...

  11. Speech Matters

    DEFF Research Database (Denmark)

    Hasse Jørgensen, Stina

    2011-01-01

    About Speech Matters - Katarina Gregos, the Greek curator's exhibition at the Danish Pavillion, the Venice Biannual 2011.......About Speech Matters - Katarina Gregos, the Greek curator's exhibition at the Danish Pavillion, the Venice Biannual 2011....

  12. Speech-to-Speech Relay Service

    Science.gov (United States)

    Consumer Guide Speech to Speech Relay Service Speech-to-Speech (STS) is one form of Telecommunications Relay Service (TRS). TRS is a service that allows persons with hearing and speech disabilities ...

  13. Objective Prediction of Hearing Aid Benefit Across Listener Groups Using Machine Learning: Speech Recognition Performance With Binaural Noise-Reduction Algorithms

    Science.gov (United States)

    Schädler, Marc R.; Warzybok, Anna; Kollmeier, Birger

    2018-01-01

    The simulation framework for auditory discrimination experiments (FADE) was adopted and validated to predict the individual speech-in-noise recognition performance of listeners with normal and impaired hearing with and without a given hearing-aid algorithm. FADE uses a simple automatic speech recognizer (ASR) to estimate the lowest achievable speech reception thresholds (SRTs) from simulated speech recognition experiments in an objective way, independent from any empirical reference data. Empirical data from the literature were used to evaluate the model in terms of predicted SRTs and benefits in SRT with the German matrix sentence recognition test when using eight single- and multichannel binaural noise-reduction algorithms. To allow individual predictions of SRTs in binaural conditions, the model was extended with a simple better ear approach and individualized by taking audiograms into account. In a realistic binaural cafeteria condition, FADE explained about 90% of the variance of the empirical SRTs for a group of normal-hearing listeners and predicted the corresponding benefits with a root-mean-square prediction error of 0.6 dB. This highlights the potential of the approach for the objective assessment of benefits in SRT without prior knowledge about the empirical data. The predictions for the group of listeners with impaired hearing explained 75% of the empirical variance, while the individual predictions explained less than 25%. Possibly, additional individual factors should be considered for more accurate predictions with impaired hearing. A competing talker condition clearly showed one limitation of current ASR technology, as the empirical performance with SRTs lower than −20 dB could not be predicted. PMID:29692200

  14. Objective Prediction of Hearing Aid Benefit Across Listener Groups Using Machine Learning: Speech Recognition Performance With Binaural Noise-Reduction Algorithms.

    Science.gov (United States)

    Schädler, Marc R; Warzybok, Anna; Kollmeier, Birger

    2018-01-01

    The simulation framework for auditory discrimination experiments (FADE) was adopted and validated to predict the individual speech-in-noise recognition performance of listeners with normal and impaired hearing with and without a given hearing-aid algorithm. FADE uses a simple automatic speech recognizer (ASR) to estimate the lowest achievable speech reception thresholds (SRTs) from simulated speech recognition experiments in an objective way, independent from any empirical reference data. Empirical data from the literature were used to evaluate the model in terms of predicted SRTs and benefits in SRT with the German matrix sentence recognition test when using eight single- and multichannel binaural noise-reduction algorithms. To allow individual predictions of SRTs in binaural conditions, the model was extended with a simple better ear approach and individualized by taking audiograms into account. In a realistic binaural cafeteria condition, FADE explained about 90% of the variance of the empirical SRTs for a group of normal-hearing listeners and predicted the corresponding benefits with a root-mean-square prediction error of 0.6 dB. This highlights the potential of the approach for the objective assessment of benefits in SRT without prior knowledge about the empirical data. The predictions for the group of listeners with impaired hearing explained 75% of the empirical variance, while the individual predictions explained less than 25%. Possibly, additional individual factors should be considered for more accurate predictions with impaired hearing. A competing talker condition clearly showed one limitation of current ASR technology, as the empirical performance with SRTs lower than -20 dB could not be predicted.

  15. Apraxia of Speech

    Science.gov (United States)

    ... Health Info » Voice, Speech, and Language Apraxia of Speech On this page: What is apraxia of speech? ... about apraxia of speech? What is apraxia of speech? Apraxia of speech (AOS)—also known as acquired ...

  16. Introductory speeches

    International Nuclear Information System (INIS)

    2001-01-01

    This CD is multimedia presentation of programme safety upgrading of Bohunice V1 NPP. This chapter consist of introductory commentary and 4 introductory speeches (video records): (1) Introductory speech of Vincent Pillar, Board chairman and director general of Slovak electric, Plc. (SE); (2) Introductory speech of Stefan Schmidt, director of SE - Bohunice Nuclear power plants; (3) Introductory speech of Jan Korec, Board chairman and director general of VUJE Trnava, Inc. - Engineering, Design and Research Organisation, Trnava; Introductory speech of Dietrich Kuschel, Senior vice-president of FRAMATOME ANP Project and Engineering

  17. Cracking the code: a decode strategy for the international business machines punch cards of Korean war soldiers.

    Science.gov (United States)

    Mitsunaga, Erin M

    2006-05-01

    During the Korean War, International Business Machines (IBM) punch cards were created for every individual involved in military combat. Each card contained all pertinent personal information about the individual and was utilized to keep track of all soldiers involved. However, at present, all of the information known about these punch cards reveals only their format and their significance; there is little to no information on how these cards were created or how to interpret the information contained without the aid of the computer system used during the war. Today, it is believed there is no one available to explain this computerized system, nor do the original computers exist. This decode strategy is the result of an attempt to decipher the information on these cards through the use of all available medical and dental records for each individual examined. By cross-referencing the relevant personal information with the known format of the cards, a basic guess-and-check method was utilized. After examining hundreds of IBM punch cards, however, it has become clear that the punch card method of recording information was not infallible. In some cases, there are gaps of information on cards where there are data recorded on personal records; in others, information is punched incorrectly onto the cards, perhaps as the result of a transcription error. Taken all together, it is clear that the information contained on each individual's card should be taken solely as another form of personal documentation.

  18. Machine medical ethics

    CERN Document Server

    Pontier, Matthijs

    2015-01-01

    The essays in this book, written by researchers from both humanities and sciences, describe various theoretical and experimental approaches to adding medical ethics to a machine in medical settings. Medical machines are in close proximity with human beings, and getting closer: with patients who are in vulnerable states of health, who have disabilities of various kinds, with the very young or very old, and with medical professionals. In such contexts, machines are undertaking important medical tasks that require emotional sensitivity, knowledge of medical codes, human dignity, and privacy. As machine technology advances, ethical concerns become more urgent: should medical machines be programmed to follow a code of medical ethics? What theory or theories should constrain medical machine conduct? What design features are required? Should machines share responsibility with humans for the ethical consequences of medical actions? How ought clinical relationships involving machines to be modeled? Is a capacity for e...

  19. Examining the Role of Orthographic Coding Ability in Elementary Students with Previously Identified Reading Disability, Speech or Language Impairment, or Comorbid Language and Learning Disabilities

    Science.gov (United States)

    Haugh, Erin Kathleen

    2017-01-01

    The purpose of this study was to examine the role orthographic coding might play in distinguishing between membership in groups of language-based disability types. The sample consisted of 36 second and third-grade subjects who were administered the PAL-II Receptive Coding and Word Choice Accuracy subtest as a measure of orthographic coding…

  20. Hateful Help--A Practical Look at the Issue of Hate Speech.

    Science.gov (United States)

    Shelton, Michael W.

    Many college and university administrators have responded to the recent increase in hateful incidents on campus by putting hate speech codes into place. The establishment of speech codes has sparked a heated debate over the impact that such codes have upon free speech and First Amendment values. Some commentators have suggested that viewing hate…

  1. Neural Entrainment to Speech Modulates Speech Intelligibility

    NARCIS (Netherlands)

    Riecke, Lars; Formisano, Elia; Sorger, Bettina; Baskent, Deniz; Gaudrain, Etienne

    2018-01-01

    Speech is crucial for communication in everyday life. Speech-brain entrainment, the alignment of neural activity to the slow temporal fluctuations (envelope) of acoustic speech input, is a ubiquitous element of current theories of speech processing. Associations between speech-brain entrainment and

  2. Noise-robust speech recognition through auditory feature detection and spike sequence decoding.

    Science.gov (United States)

    Schafer, Phillip B; Jin, Dezhe Z

    2014-03-01

    Speech recognition in noisy conditions is a major challenge for computer systems, but the human brain performs it routinely and accurately. Automatic speech recognition (ASR) systems that are inspired by neuroscience can potentially bridge the performance gap between humans and machines. We present a system for noise-robust isolated word recognition that works by decoding sequences of spikes from a population of simulated auditory feature-detecting neurons. Each neuron is trained to respond selectively to a brief spectrotemporal pattern, or feature, drawn from the simulated auditory nerve response to speech. The neural population conveys the time-dependent structure of a sound by its sequence of spikes. We compare two methods for decoding the spike sequences--one using a hidden Markov model-based recognizer, the other using a novel template-based recognition scheme. In the latter case, words are recognized by comparing their spike sequences to template sequences obtained from clean training data, using a similarity measure based on the length of the longest common sub-sequence. Using isolated spoken digits from the AURORA-2 database, we show that our combined system outperforms a state-of-the-art robust speech recognizer at low signal-to-noise ratios. Both the spike-based encoding scheme and the template-based decoding offer gains in noise robustness over traditional speech recognition methods. Our system highlights potential advantages of spike-based acoustic coding and provides a biologically motivated framework for robust ASR development.

  3. Incorporating Speech Recognition into a Natural User Interface

    Science.gov (United States)

    Chapa, Nicholas

    2017-01-01

    The Augmented/ Virtual Reality (AVR) Lab has been working to study the applicability of recent virtual and augmented reality hardware and software to KSC operations. This includes the Oculus Rift, HTC Vive, Microsoft HoloLens, and Unity game engine. My project in this lab is to integrate voice recognition and voice commands into an easy to modify system that can be added to an existing portion of a Natural User Interface (NUI). A NUI is an intuitive and simple to use interface incorporating visual, touch, and speech recognition. The inclusion of speech recognition capability will allow users to perform actions or make inquiries using only their voice. The simplicity of needing only to speak to control an on-screen object or enact some digital action means that any user can quickly become accustomed to using this system. Multiple programs were tested for use in a speech command and recognition system. Sphinx4 translates speech to text using a Hidden Markov Model (HMM) based Language Model, an Acoustic Model, and a word Dictionary running on Java. PocketSphinx had similar functionality to Sphinx4 but instead ran on C. However, neither of these programs were ideal as building a Java or C wrapper slowed performance. The most ideal speech recognition system tested was the Unity Engine Grammar Recognizer. A Context Free Grammar (CFG) structure is written in an XML file to specify the structure of phrases and words that will be recognized by Unity Grammar Recognizer. Using Speech Recognition Grammar Specification (SRGS) 1.0 makes modifying the recognized combinations of words and phrases very simple and quick to do. With SRGS 1.0, semantic information can also be added to the XML file, which allows for even more control over how spoken words and phrases are interpreted by Unity. Additionally, using a CFG with SRGS 1.0 produces a Finite State Machine (FSM) functionality limiting the potential for incorrectly heard words or phrases. The purpose of my project was to

  4. Speech Research

    Science.gov (United States)

    Several articles addressing topics in speech research are presented. The topics include: exploring the functional significance of physiological tremor: A biospectroscopic approach; differences between experienced and inexperienced listeners to deaf speech; a language-oriented view of reading and its disabilities; Phonetic factors in letter detection; categorical perception; Short-term recall by deaf signers of American sign language; a common basis for auditory sensory storage in perception and immediate memory; phonological awareness and verbal short-term memory; initiation versus execution time during manual and oral counting by stutterers; trading relations in the perception of speech by five-year-old children; the role of the strap muscles in pitch lowering; phonetic validation of distinctive features; consonants and syllable boundaires; and vowel information in postvocalic frictions.

  5. Hate speech

    Directory of Open Access Journals (Sweden)

    Anne Birgitta Nilsen

    2014-12-01

    Full Text Available The manifesto of the Norwegian terrorist Anders Behring Breivik is based on the “Eurabia” conspiracy theory. This theory is a key starting point for hate speech amongst many right-wing extremists in Europe, but also has ramifications beyond these environments. In brief, proponents of the Eurabia theory claim that Muslims are occupying Europe and destroying Western culture, with the assistance of the EU and European governments. By contrast, members of Al-Qaeda and other extreme Islamists promote the conspiracy theory “the Crusade” in their hate speech directed against the West. Proponents of the latter theory argue that the West is leading a crusade to eradicate Islam and Muslims, a crusade that is similarly facilitated by their governments. This article presents analyses of texts written by right-wing extremists and Muslim extremists in an effort to shed light on how hate speech promulgates conspiracy theories in order to spread hatred and intolerance.The aim of the article is to contribute to a more thorough understanding of hate speech’s nature by applying rhetorical analysis. Rhetorical analysis is chosen because it offers a means of understanding the persuasive power of speech. It is thus a suitable tool to describe how hate speech works to convince and persuade. The concepts from rhetorical theory used in this article are ethos, logos and pathos. The concept of ethos is used to pinpoint factors that contributed to Osama bin Laden's impact, namely factors that lent credibility to his promotion of the conspiracy theory of the Crusade. In particular, Bin Laden projected common sense, good morals and good will towards his audience. He seemed to have coherent and relevant arguments; he appeared to possess moral credibility; and his use of language demonstrated that he wanted the best for his audience.The concept of pathos is used to define hate speech, since hate speech targets its audience's emotions. In hate speech it is the

  6. The DNA of prophetic speech

    African Journals Online (AJOL)

    2014-03-04

    Mar 4, 2014 ... It is expected that people will be drawn into the reality of God by authentic prophetic speech, .... strands of the DNA molecule show themselves to be arranged ... explains, chemical patterns act like the letters of a code, .... viewing the self-reflection regarding the ministry of renewal from the .... Irresistible force.

  7. Soft computing in machine learning

    CERN Document Server

    Park, Jooyoung; Inoue, Atsushi

    2014-01-01

    As users or consumers are now demanding smarter devices, intelligent systems are revolutionizing by utilizing machine learning. Machine learning as part of intelligent systems is already one of the most critical components in everyday tools ranging from search engines and credit card fraud detection to stock market analysis. You can train machines to perform some things, so that they can automatically detect, diagnose, and solve a variety of problems. The intelligent systems have made rapid progress in developing the state of the art in machine learning based on smart and deep perception. Using machine learning, the intelligent systems make widely applications in automated speech recognition, natural language processing, medical diagnosis, bioinformatics, and robot locomotion. This book aims at introducing how to treat a substantial amount of data, to teach machines and to improve decision making models. And this book specializes in the developments of advanced intelligent systems through machine learning. It...

  8. Speech Intelligibility

    Science.gov (United States)

    Brand, Thomas

    Speech intelligibility (SI) is important for different fields of research, engineering and diagnostics in order to quantify very different phenomena like the quality of recordings, communication and playback devices, the reverberation of auditoria, characteristics of hearing impairment, benefit using hearing aids or combinations of these things.

  9. Machine Translation from Text

    Science.gov (United States)

    Habash, Nizar; Olive, Joseph; Christianson, Caitlin; McCary, John

    Machine translation (MT) from text, the topic of this chapter, is perhaps the heart of the GALE project. Beyond being a well defined application that stands on its own, MT from text is the link between the automatic speech recognition component and the distillation component. The focus of MT in GALE is on translating from Arabic or Chinese to English. The three languages represent a wide range of linguistic diversity and make the GALE MT task rather challenging and exciting.

  10. Identifying Deceptive Speech Across Cultures

    Science.gov (United States)

    2016-06-25

    enough from the truth. Subjects were then interviewed individually in a sound booth to obtain “norming” speech data, pre- interview. We also...e.g. pitch, intensity, speaking rate, voice quality), gender, ethnicity and personality information, our machine learning experiments can classify...Have you ever been in trouble with the police?” vs. open-ended (e.g. “What is the last movie you saw that you really hated ?”) DISTRIBUTION A

  11. Speech emotion recognition methods: A literature review

    Science.gov (United States)

    Basharirad, Babak; Moradhaseli, Mohammadreza

    2017-10-01

    Recently, attention of the emotional speech signals research has been boosted in human machine interfaces due to availability of high computation capability. There are many systems proposed in the literature to identify the emotional state through speech. Selection of suitable feature sets, design of a proper classifications methods and prepare an appropriate dataset are the main key issues of speech emotion recognition systems. This paper critically analyzed the current available approaches of speech emotion recognition methods based on the three evaluating parameters (feature set, classification of features, accurately usage). In addition, this paper also evaluates the performance and limitations of available methods. Furthermore, it highlights the current promising direction for improvement of speech emotion recognition systems.

  12. Personality in speech assessment and automatic classification

    CERN Document Server

    Polzehl, Tim

    2015-01-01

    This work combines interdisciplinary knowledge and experience from research fields of psychology, linguistics, audio-processing, machine learning, and computer science. The work systematically explores a novel research topic devoted to automated modeling of personality expression from speech. For this aim, it introduces a novel personality assessment questionnaire and presents the results of extensive labeling sessions to annotate the speech data with personality assessments. It provides estimates of the Big 5 personality traits, i.e. openness, conscientiousness, extroversion, agreeableness, and neuroticism. Based on a database built on the questionnaire, the book presents models to tell apart different personality types or classes from speech automatically.

  13. 78 FR 49693 - Speech-to-Speech and Internet Protocol (IP) Speech-to-Speech Telecommunications Relay Services...

    Science.gov (United States)

    2013-08-15

    ...-Speech Services for Individuals with Hearing and Speech Disabilities, Report and Order (Order), document...] Speech-to-Speech and Internet Protocol (IP) Speech-to-Speech Telecommunications Relay Services; Telecommunications Relay Services and Speech-to-Speech Services for Individuals With Hearing and Speech Disabilities...

  14. Autocoding State Machine in Erlang

    DEFF Research Database (Denmark)

    Guo, Yu; Hoffman, Torben; Gunder, Nicholas

    2008-01-01

    This paper presents an autocoding tool suit, which supports development of state machine in a model-driven fashion, where models are central to all phases of the development process. The tool suit, which is built on the Eclipse platform, provides facilities for the graphical specification...... of a state machine model. Once the state machine is specified, it is used as input to a code generation engine that generates source code in Erlang....

  15. Shared acoustic codes underlie emotional communication in music and speech—Evidence from deep transfer learning

    Science.gov (United States)

    Schuller, Björn

    2017-01-01

    Music and speech exhibit striking similarities in the communication of emotions in the acoustic domain, in such a way that the communication of specific emotions is achieved, at least to a certain extent, by means of shared acoustic patterns. From an Affective Sciences points of view, determining the degree of overlap between both domains is fundamental to understand the shared mechanisms underlying such phenomenon. From a Machine learning perspective, the overlap between acoustic codes for emotional expression in music and speech opens new possibilities to enlarge the amount of data available to develop music and speech emotion recognition systems. In this article, we investigate time-continuous predictions of emotion (Arousal and Valence) in music and speech, and the Transfer Learning between these domains. We establish a comparative framework including intra- (i.e., models trained and tested on the same modality, either music or speech) and cross-domain experiments (i.e., models trained in one modality and tested on the other). In the cross-domain context, we evaluated two strategies—the direct transfer between domains, and the contribution of Transfer Learning techniques (feature-representation-transfer based on Denoising Auto Encoders) for reducing the gap in the feature space distributions. Our results demonstrate an excellent cross-domain generalisation performance with and without feature representation transfer in both directions. In the case of music, cross-domain approaches outperformed intra-domain models for Valence estimation, whereas for Speech intra-domain models achieve the best performance. This is the first demonstration of shared acoustic codes for emotional expression in music and speech in the time-continuous domain. PMID:28658285

  16. Random Deep Belief Networks for Recognizing Emotions from Speech Signals.

    Science.gov (United States)

    Wen, Guihua; Li, Huihui; Huang, Jubing; Li, Danyang; Xun, Eryang

    2017-01-01

    Now the human emotions can be recognized from speech signals using machine learning methods; however, they are challenged by the lower recognition accuracies in real applications due to lack of the rich representation ability. Deep belief networks (DBN) can automatically discover the multiple levels of representations in speech signals. To make full of its advantages, this paper presents an ensemble of random deep belief networks (RDBN) method for speech emotion recognition. It firstly extracts the low level features of the input speech signal and then applies them to construct lots of random subspaces. Each random subspace is then provided for DBN to yield the higher level features as the input of the classifier to output an emotion label. All outputted emotion labels are then fused through the majority voting to decide the final emotion label for the input speech signal. The conducted experimental results on benchmark speech emotion databases show that RDBN has better accuracy than the compared methods for speech emotion recognition.

  17. Speech disorders - children

    Science.gov (United States)

    ... disorder; Voice disorders; Vocal disorders; Disfluency; Communication disorder - speech disorder; Speech disorder - stuttering ... evaluation tools that can help identify and diagnose speech disorders: Denver Articulation Screening Examination Goldman-Fristoe Test of ...

  18. Steganalysis of recorded speech

    Science.gov (United States)

    Johnson, Micah K.; Lyu, Siwei; Farid, Hany

    2005-03-01

    Digital audio provides a suitable cover for high-throughput steganography. At 16 bits per sample and sampled at a rate of 44,100 Hz, digital audio has the bit-rate to support large messages. In addition, audio is often transient and unpredictable, facilitating the hiding of messages. Using an approach similar to our universal image steganalysis, we show that hidden messages alter the underlying statistics of audio signals. Our statistical model begins by building a linear basis that captures certain statistical properties of audio signals. A low-dimensional statistical feature vector is extracted from this basis representation and used by a non-linear support vector machine for classification. We show the efficacy of this approach on LSB embedding and Hide4PGP. While no explicit assumptions about the content of the audio are made, our technique has been developed and tested on high-quality recorded speech.

  19. Speech Processing.

    Science.gov (United States)

    1983-05-01

    The VDE system developed had the capability of recognizing up to 248 separate words in syntactic structures. 4 The two systems described are isolated...AND SPEAKER RECOGNITION by M.J.Hunt 5 ASSESSMENT OF SPEECH SYSTEMS ’ ..- * . by R.K.Moore 6 A SURVEY OF CURRENT EQUIPMENT AND RESEARCH’ by J.S.Bridle...TECHNOLOGY IN NAVY TRAINING SYSTEMS by R.Breaux, M.Blind and R.Lynchard 10 9 I-I GENERAL REVIEW OF MILITARY APPLICATIONS OF VOICE PROCESSING DR. BRUNO

  20. Speech Recognition

    Directory of Open Access Journals (Sweden)

    Adrian Morariu

    2009-01-01

    Full Text Available This paper presents a method of speech recognition by pattern recognition techniques. Learning consists in determining the unique characteristics of a word (cepstral coefficients by eliminating those characteristics that are different from one word to another. For learning and recognition, the system will build a dictionary of words by determining the characteristics of each word to be used in the recognition. Determining the characteristics of an audio signal consists in the following steps: noise removal, sampling it, applying Hamming window, switching to frequency domain through Fourier transform, calculating the magnitude spectrum, filtering data, determining cepstral coefficients.

  1. Focal versus distributed temporal cortex activity for speech sound category assignment

    Science.gov (United States)

    Bouton, Sophie; Chambon, Valérian; Tyrand, Rémi; Seeck, Margitta; Karkar, Sami; van de Ville, Dimitri; Giraud, Anne-Lise

    2018-01-01

    Percepts and words can be decoded from distributed neural activity measures. However, the existence of widespread representations might conflict with the more classical notions of hierarchical processing and efficient coding, which are especially relevant in speech processing. Using fMRI and magnetoencephalography during syllable identification, we show that sensory and decisional activity colocalize to a restricted part of the posterior superior temporal gyrus (pSTG). Next, using intracortical recordings, we demonstrate that early and focal neural activity in this region distinguishes correct from incorrect decisions and can be machine-decoded to classify syllables. Crucially, significant machine decoding was possible from neuronal activity sampled across different regions of the temporal and frontal lobes, despite weak or absent sensory or decision-related responses. These findings show that speech-sound categorization relies on an efficient readout of focal pSTG neural activity, while more distributed activity patterns, although classifiable by machine learning, instead reflect collateral processes of sensory perception and decision. PMID:29363598

  2. Fishing for meaningful units in connected speech

    DEFF Research Database (Denmark)

    Henrichsen, Peter Juel; Christiansen, Thomas Ulrich

    2009-01-01

    In many branches of spoken language analysis including ASR, the set of smallest meaningful units of speech is taken to coincide with the set of phones or phonemes. However, fishing for phones is difficult, error-prone, and computationally expensive. We present an experiment, based on machine...

  3. Machine Shop Grinding Machines.

    Science.gov (United States)

    Dunn, James

    This curriculum manual is one in a series of machine shop curriculum manuals intended for use in full-time secondary and postsecondary classes, as well as part-time adult classes. The curriculum can also be adapted to open-entry, open-exit programs. Its purpose is to equip students with basic knowledge and skills that will enable them to enter the…

  4. Speech and Language Delay

    Science.gov (United States)

    ... OTC Relief for Diarrhea Home Diseases and Conditions Speech and Language Delay Condition Speech and Language Delay Share Print Table of Contents1. ... Treatment6. Everyday Life7. Questions8. Resources What is a speech and language delay? A speech and language delay ...

  5. Spoken Language Understanding Systems for Extracting Semantic Information from Speech

    CERN Document Server

    Tur, Gokhan

    2011-01-01

    Spoken language understanding (SLU) is an emerging field in between speech and language processing, investigating human/ machine and human/ human communication by leveraging technologies from signal processing, pattern recognition, machine learning and artificial intelligence. SLU systems are designed to extract the meaning from speech utterances and its applications are vast, from voice search in mobile devices to meeting summarization, attracting interest from both commercial and academic sectors. Both human/machine and human/human communications can benefit from the application of SLU, usin

  6. Panda code

    International Nuclear Information System (INIS)

    Altomare, S.; Minton, G.

    1975-02-01

    PANDA is a new two-group one-dimensional (slab/cylinder) neutron diffusion code designed to replace and extend the FAB series. PANDA allows for the nonlinear effects of xenon, enthalpy and Doppler. Fuel depletion is allowed. PANDA has a completely general search facility which will seek criticality, maximize reactivity, or minimize peaking. Any single parameter may be varied in a search. PANDA is written in FORTRAN IV, and as such is nearly machine independent. However, PANDA has been written with the present limitations of the Westinghouse CDC-6600 system in mind. Most computation loops are very short, and the code is less than half the useful 6600 memory size so that two jobs can reside in the core at once. (auth)

  7. The cognitive approach to conscious machines

    CERN Document Server

    Haikonen, Pentti O

    2003-01-01

    Could a machine have an immaterial mind? The author argues that true conscious machines can be built, but rejects artificial intelligence and classical neural networks in favour of the emulation of the cognitive processes of the brain-the flow of inner speech, inner imagery and emotions. This results in a non-numeric meaning-processing machine with distributed information representation and system reactions. It is argued that this machine would be conscious; it would be aware of its own existence and its mental content and perceive this as immaterial. Novel views on consciousness and the mind-

  8. Sustainable machining

    CERN Document Server

    2017-01-01

    This book provides an overview on current sustainable machining. Its chapters cover the concept in economic, social and environmental dimensions. It provides the reader with proper ways to handle several pollutants produced during the machining process. The book is useful on both undergraduate and postgraduate levels and it is of interest to all those working with manufacturing and machining technology.

  9. Speech Acquisition and Automatic Speech Recognition for Integrated Spacesuit Audio Systems

    Science.gov (United States)

    Huang, Yiteng; Chen, Jingdong; Chen, Shaoyan

    2010-01-01

    A voice-command human-machine interface system has been developed for spacesuit extravehicular activity (EVA) missions. A multichannel acoustic signal processing method has been created for distant speech acquisition in noisy and reverberant environments. This technology reduces noise by exploiting differences in the statistical nature of signal (i.e., speech) and noise that exists in the spatial and temporal domains. As a result, the automatic speech recognition (ASR) accuracy can be improved to the level at which crewmembers would find the speech interface useful. The developed speech human/machine interface will enable both crewmember usability and operational efficiency. It can enjoy a fast rate of data/text entry, small overall size, and can be lightweight. In addition, this design will free the hands and eyes of a suited crewmember. The system components and steps include beam forming/multi-channel noise reduction, single-channel noise reduction, speech feature extraction, feature transformation and normalization, feature compression, model adaption, ASR HMM (Hidden Markov Model) training, and ASR decoding. A state-of-the-art phoneme recognizer can obtain an accuracy rate of 65 percent when the training and testing data are free of noise. When it is used in spacesuits, the rate drops to about 33 percent. With the developed microphone array speech-processing technologies, the performance is improved and the phoneme recognition accuracy rate rises to 44 percent. The recognizer can be further improved by combining the microphone array and HMM model adaptation techniques and using speech samples collected from inside spacesuits. In addition, arithmetic complexity models for the major HMMbased ASR components were developed. They can help real-time ASR system designers select proper tasks when in the face of constraints in computational resources.

  10. Speech and Communication Disorders

    Science.gov (United States)

    ... to being completely unable to speak or understand speech. Causes include Hearing disorders and deafness Voice problems, ... or those caused by cleft lip or palate Speech problems like stuttering Developmental disabilities Learning disorders Autism ...

  11. Cultural and biological evolution of phonemic speech

    NARCIS (Netherlands)

    de Boer, B.; Freitas, A.A.; Capcarrere, M.S.; Bentley, Peter J.; Johnson, Colin G.; Timmis, Jon

    2005-01-01

    This paper investigates the interaction between cultural evolution and biological evolution in the emergence of phonemic coding in speech. It is observed that our nearest relatives, the primates, use holistic utterances, whereas humans use phonemic utterances. It can therefore be argued that our

  12. OLIVE: Speech-Based Video Retrieval

    NARCIS (Netherlands)

    de Jong, Franciska M.G.; Gauvain, Jean-Luc; den Hartog, Jurgen; den Hartog, Jeremy; Netter, Klaus

    1999-01-01

    This paper describes the Olive project which aims to support automated indexing of video material by use of human language technologies. Olive is making use of speech recognition to automatically derive transcriptions of the sound tracks, generating time-coded linguistic elements which serve as the

  13. Freedom of Speech Wins in Wisconsin

    Science.gov (United States)

    Downs, Donald Alexander

    2006-01-01

    One might derive, from the eradication of a particularly heinous speech code, some encouragement that all is not lost in the culture wars. A core of dedicated scholars, working from within, made it obvious, to all but the most radical left, that imposing social justice by restricting thought and expression was a recipe for tyranny. Donald…

  14. Free Speech Yearbook 1978.

    Science.gov (United States)

    Phifer, Gregg, Ed.

    The 17 articles in this collection deal with theoretical and practical freedom of speech issues. The topics include: freedom of speech in Marquette Park, Illinois; Nazis in Skokie, Illinois; freedom of expression in the Confederate States of America; Robert M. LaFollette's arguments for free speech and the rights of Congress; the United States…

  15. Analysis of Feature Extraction Methods for Speaker Dependent Speech Recognition

    Directory of Open Access Journals (Sweden)

    Gurpreet Kaur

    2017-02-01

    Full Text Available Speech recognition is about what is being said, irrespective of who is saying. Speech recognition is a growing field. Major progress is taking place on the technology of automatic speech recognition (ASR. Still, there are lots of barriers in this field in terms of recognition rate, background noise, speaker variability, speaking rate, accent etc. Speech recognition rate mainly depends on the selection of features and feature extraction methods. This paper outlines the feature extraction techniques for speaker dependent speech recognition for isolated words. A brief survey of different feature extraction techniques like Mel-Frequency Cepstral Coefficients (MFCC, Linear Predictive Coding Coefficients (LPCC, Perceptual Linear Prediction (PLP, Relative Spectra Perceptual linear Predictive (RASTA-PLP analysis are presented and evaluation is done. Speech recognition has various applications from daily use to commercial use. We have made a speaker dependent system and this system can be useful in many areas like controlling a patient vehicle using simple commands.

  16. An introduction to quantum machine learning

    OpenAIRE

    Schuld, M.; Sinayskiy, I.; Petruccione, F.

    2014-01-01

    Machine learning algorithms learn a desired input-output relation from examples in order to interpret new inputs. This is important for tasks such as image and speech recognition or strategy optimisation, with growing applications in the IT industry. In the last couple of years, researchers investigated if quantum computing can help to improve classical machine learning algorithms. Ideas range from running computationally costly algorithms or their subroutines efficiently on a quantum compute...

  17. ACOUSTIC SPEECH RECOGNITION FOR MARATHI LANGUAGE USING SPHINX

    Directory of Open Access Journals (Sweden)

    Aman Ankit

    2016-09-01

    Full Text Available Speech recognition or speech to text processing, is a process of recognizing human speech by the computer and converting into text. In speech recognition, transcripts are created by taking recordings of speech as audio and their text transcriptions. Speech based applications which include Natural Language Processing (NLP techniques are popular and an active area of research. Input to such applications is in natural language and output is obtained in natural language. Speech recognition mostly revolves around three approaches namely Acoustic phonetic approach, Pattern recognition approach and Artificial intelligence approach. Creation of acoustic model requires a large database of speech and training algorithms. The output of an ASR system is recognition and translation of spoken language into text by computers and computerized devices. ASR today finds enormous application in tasks that require human machine interfaces like, voice dialing, and etc. Our key contribution in this paper is to create corpora for Marathi language and explore the use of Sphinx engine for automatic speech recognition

  18. Speech in spinocerebellar ataxia.

    Science.gov (United States)

    Schalling, Ellika; Hartelius, Lena

    2013-12-01

    Spinocerebellar ataxias (SCAs) are a heterogeneous group of autosomal dominant cerebellar ataxias clinically characterized by progressive ataxia, dysarthria and a range of other concomitant neurological symptoms. Only a few studies include detailed characterization of speech symptoms in SCA. Speech symptoms in SCA resemble ataxic dysarthria but symptoms related to phonation may be more prominent. One study to date has shown an association between differences in speech and voice symptoms related to genotype. More studies of speech and voice phenotypes are motivated, to possibly aid in clinical diagnosis. In addition, instrumental speech analysis has been demonstrated to be a reliable measure that may be used to monitor disease progression or therapy outcomes in possible future pharmacological treatments. Intervention by speech and language pathologists should go beyond assessment. Clinical guidelines for management of speech, communication and swallowing need to be developed for individuals with progressive cerebellar ataxia. Copyright © 2013 Elsevier Inc. All rights reserved.

  19. [Prosody, speech input and language acquisition].

    Science.gov (United States)

    Jungheim, M; Miller, S; Kühn, D; Ptok, M

    2014-04-01

    In order to acquire language, children require speech input. The prosody of the speech input plays an important role. In most cultures adults modify their code when communicating with children. Compared to normal speech this code differs especially with regard to prosody. For this review a selective literature search in PubMed and Scopus was performed. Prosodic characteristics are a key feature of spoken language. By analysing prosodic features, children gain knowledge about underlying grammatical structures. Child-directed speech (CDS) is modified in a way that meaningful sequences are highlighted acoustically so that important information can be extracted from the continuous speech flow more easily. CDS is said to enhance the representation of linguistic signs. Taking into consideration what has previously been described in the literature regarding the perception of suprasegmentals, CDS seems to be able to support language acquisition due to the correspondence of prosodic and syntactic units. However, no findings have been reported, stating that the linguistically reduced CDS could hinder first language acquisition.

  20. Non linear analyses of speech and prosody in Asperger's syndrome

    DEFF Research Database (Denmark)

    Fusaroli, Riccardo; Bang, Dan; Weed, Ethan

    It is widely acknowledged that people on the ASD spectrum behave atypically in the way they modulate aspects of speech and voice, including pitch, fluency, and voice quality. ASD speech has been described at times as “odd”, “mechanical”, or “monotone”. However, it has proven difficult to quantify...... the results in a supervised machine-learning process to classify speech production as either belonging to the control or the AS group as well as to assess the severity of the disorder (as measured by Autism Spectrum Quotient), based solely on acoustic features....

  1. Simple machines

    CERN Document Server

    Graybill, George

    2007-01-01

    Just how simple are simple machines? With our ready-to-use resource, they are simple to teach and easy to learn! Chocked full of information and activities, we begin with a look at force, motion and work, and examples of simple machines in daily life are given. With this background, we move on to different kinds of simple machines including: Levers, Inclined Planes, Wedges, Screws, Pulleys, and Wheels and Axles. An exploration of some compound machines follows, such as the can opener. Our resource is a real time-saver as all the reading passages, student activities are provided. Presented in s

  2. Digital speech processing using Matlab

    CERN Document Server

    Gopi, E S

    2014-01-01

    Digital Speech Processing Using Matlab deals with digital speech pattern recognition, speech production model, speech feature extraction, and speech compression. The book is written in a manner that is suitable for beginners pursuing basic research in digital speech processing. Matlab illustrations are provided for most topics to enable better understanding of concepts. This book also deals with the basic pattern recognition techniques (illustrated with speech signals using Matlab) such as PCA, LDA, ICA, SVM, HMM, GMM, BPN, and KSOM.

  3. Speech parts as Poisson processes.

    Science.gov (United States)

    Badalamenti, A F

    2001-09-01

    This paper presents evidence that six of the seven parts of speech occur in written text as Poisson processes, simple or recurring. The six major parts are nouns, verbs, adjectives, adverbs, prepositions, and conjunctions, with the interjection occurring too infrequently to support a model. The data consist of more than the first 5000 words of works by four major authors coded to label the parts of speech, as well as periods (sentence terminators). Sentence length is measured via the period and found to be normally distributed with no stochastic model identified for its occurrence. The models for all six speech parts but the noun significantly distinguish some pairs of authors and likewise for the joint use of all words types. Any one author is significantly distinguished from any other by at least one word type and sentence length very significantly distinguishes each from all others. The variety of word type use, measured by Shannon entropy, builds to about 90% of its maximum possible value. The rate constants for nouns are close to the fractions of maximum entropy achieved. This finding together with the stochastic models and the relations among them suggest that the noun may be a primitive organizer of written text.

  4. Quantum Virtual Machine (QVM)

    Energy Technology Data Exchange (ETDEWEB)

    2016-11-18

    There is a lack of state-of-the-art HPC simulation tools for simulating general quantum computing. Furthermore, there are no real software tools that integrate current quantum computers into existing classical HPC workflows. This product, the Quantum Virtual Machine (QVM), solves this problem by providing an extensible framework for pluggable virtual, or physical, quantum processing units (QPUs). It enables the execution of low level quantum assembly codes and returns the results of such executions.

  5. Applications of Hilbert Spectral Analysis for Speech and Sound Signals

    Science.gov (United States)

    Huang, Norden E.

    2003-01-01

    A new method for analyzing nonlinear and nonstationary data has been developed, and the natural applications are to speech and sound signals. The key part of the method is the Empirical Mode Decomposition method with which any complicated data set can be decomposed into a finite and often small number of Intrinsic Mode Functions (IMF). An IMF is defined as any function having the same numbers of zero-crossing and extrema, and also having symmetric envelopes defined by the local maxima and minima respectively. The IMF also admits well-behaved Hilbert transform. This decomposition method is adaptive, and, therefore, highly efficient. Since the decomposition is based on the local characteristic time scale of the data, it is applicable to nonlinear and nonstationary processes. With the Hilbert transform, the Intrinsic Mode Functions yield instantaneous frequencies as functions of time, which give sharp identifications of imbedded structures. This method invention can be used to process all acoustic signals. Specifically, it can process the speech signals for Speech synthesis, Speaker identification and verification, Speech recognition, and Sound signal enhancement and filtering. Additionally, as the acoustical signals from machinery are essentially the way the machines are talking to us. Therefore, the acoustical signals, from the machines, either from sound through air or vibration on the machines, can tell us the operating conditions of the machines. Thus, we can use the acoustic signal to diagnosis the problems of machines.

  6. Face machines

    Energy Technology Data Exchange (ETDEWEB)

    Hindle, D.

    1999-06-01

    The article surveys latest equipment available from the world`s manufacturers of a range of machines for tunnelling. These are grouped under headings: excavators; impact hammers; road headers; and shields and tunnel boring machines. Products of thirty manufacturers are referred to. Addresses and fax numbers of companies are supplied. 5 tabs., 13 photos.

  7. Electric machine

    Science.gov (United States)

    El-Refaie, Ayman Mohamed Fawzi [Niskayuna, NY; Reddy, Patel Bhageerath [Madison, WI

    2012-07-17

    An interior permanent magnet electric machine is disclosed. The interior permanent magnet electric machine comprises a rotor comprising a plurality of radially placed magnets each having a proximal end and a distal end, wherein each magnet comprises a plurality of magnetic segments and at least one magnetic segment towards the distal end comprises a high resistivity magnetic material.

  8. Machine Learning.

    Science.gov (United States)

    Kirrane, Diane E.

    1990-01-01

    As scientists seek to develop machines that can "learn," that is, solve problems by imitating the human brain, a gold mine of information on the processes of human learning is being discovered, expert systems are being improved, and human-machine interactions are being enhanced. (SK)

  9. Nonplanar machines

    International Nuclear Information System (INIS)

    Ritson, D.

    1989-05-01

    This talk examines methods available to minimize, but never entirely eliminate, degradation of machine performance caused by terrain following. Breaking of planar machine symmetry for engineering convenience and/or monetary savings must be balanced against small performance degradation, and can only be decided on a case-by-case basis. 5 refs

  10. Objective voice and speech analysis of persons with chronic hoarseness by prosodic analysis of speech samples.

    Science.gov (United States)

    Haderlein, Tino; Döllinger, Michael; Matoušek, Václav; Nöth, Elmar

    2016-10-01

    Automatic voice assessment is often performed using sustained vowels. In contrast, speech analysis of read-out texts can be applied to voice and speech assessment. Automatic speech recognition and prosodic analysis were used to find regression formulae between automatic and perceptual assessment of four voice and four speech criteria. The regression was trained with 21 men and 62 women (average age 49.2 years) and tested with another set of 24 men and 49 women (48.3 years), all suffering from chronic hoarseness. They read the text 'Der Nordwind und die Sonne' ('The North Wind and the Sun'). Five voice and speech therapists evaluated the data on 5-point Likert scales. Ten prosodic and recognition accuracy measures (features) were identified which describe all the examined criteria. Inter-rater correlation within the expert group was between r = 0.63 for the criterion 'match of breath and sense units' and r = 0.87 for the overall voice quality. Human-machine correlation was between r = 0.40 for the match of breath and sense units and r = 0.82 for intelligibility. The perceptual ratings of different criteria were highly correlated with each other. Likewise, the feature sets modeling the criteria were very similar. The automatic method is suitable for assessing chronic hoarseness in general and for subgroups of functional and organic dysphonia. In its current version, it is almost as reliable as a randomly picked rater from a group of voice and speech therapists.

  11. Towards advanced code simulators

    International Nuclear Information System (INIS)

    Scriven, A.H.

    1990-01-01

    The Central Electricity Generating Board (CEGB) uses advanced thermohydraulic codes extensively to support PWR safety analyses. A system has been developed to allow fully interactive execution of any code with graphical simulation of the operator desk and mimic display. The system operates in a virtual machine environment, with the thermohydraulic code executing in one virtual machine, communicating via interrupts with any number of other virtual machines each running other programs and graphics drivers. The driver code itself does not have to be modified from its normal batch form. Shortly following the release of RELAP5 MOD1 in IBM compatible form in 1983, this code was used as the driver for this system. When RELAP5 MOD2 became available, it was adopted with no changes needed in the basic system. Overall the system has been used for some 5 years for the analysis of LOBI tests, full scale plant studies and for simple what-if studies. For gaining rapid understanding of system dependencies it has proved invaluable. The graphical mimic system, being independent of the driver code, has also been used with other codes to study core rewetting, to replay results obtained from batch jobs on a CRAY2 computer system and to display suitably processed experimental results from the LOBI facility to aid interpretation. For the above work real-time execution was not necessary. Current work now centers on implementing the RELAP 5 code on a true parallel architecture machine. Marconi Simulation have been contracted to investigate the feasibility of using upwards of 100 processors, each capable of a peak of 30 MIPS to run a highly detailed RELAP5 model in real time, complete with specially written 3D core neutronics and balance of plant models. This paper describes the experience of using RELAP5 as an analyzer/simulator, and outlines the proposed methods and problems associated with parallel execution of RELAP5

  12. The Machine within the Machine

    CERN Multimedia

    Katarina Anthony

    2014-01-01

    Although Virtual Machines are widespread across CERN, you probably won't have heard of them unless you work for an experiment. Virtual machines - known as VMs - allow you to create a separate machine within your own, allowing you to run Linux on your Mac, or Windows on your Linux - whatever combination you need.   Using a CERN Virtual Machine, a Linux analysis software runs on a Macbook. When it comes to LHC data, one of the primary issues collaborations face is the diversity of computing environments among collaborators spread across the world. What if an institute cannot run the analysis software because they use different operating systems? "That's where the CernVM project comes in," says Gerardo Ganis, PH-SFT staff member and leader of the CernVM project. "We were able to respond to experimentalists' concerns by providing a virtual machine package that could be used to run experiment software. This way, no matter what hardware they have ...

  13. Ear, Hearing and Speech

    DEFF Research Database (Denmark)

    Poulsen, Torben

    2000-01-01

    An introduction is given to the the anatomy and the function of the ear, basic psychoacoustic matters (hearing threshold, loudness, masking), the speech signal and speech intelligibility. The lecture note is written for the course: Fundamentals of Acoustics and Noise Control (51001)......An introduction is given to the the anatomy and the function of the ear, basic psychoacoustic matters (hearing threshold, loudness, masking), the speech signal and speech intelligibility. The lecture note is written for the course: Fundamentals of Acoustics and Noise Control (51001)...

  14. Speech disorder prevention

    Directory of Open Access Journals (Sweden)

    Miladis Fornaris-Méndez

    2017-04-01

    Full Text Available Language therapy has trafficked from a medical focus until a preventive focus. However, difficulties are evidenced in the development of this last task, because he is devoted bigger space to the correction of the disorders of the language. Because the speech disorders is the dysfunction with more frequently appearance, acquires special importance the preventive work that is developed to avoid its appearance. Speech education since early age of the childhood makes work easier for prevent the appearance of speech disorders in the children. The present work has as objective to offer different activities for the prevention of the speech disorders.

  15. Machine translation

    Energy Technology Data Exchange (ETDEWEB)

    Nagao, M

    1982-04-01

    Each language has its own structure. In translating one language into another one, language attributes and grammatical interpretation must be defined in an unambiguous form. In order to parse a sentence, it is necessary to recognize its structure. A so-called context-free grammar can help in this respect for machine translation and machine-aided translation. Problems to be solved in studying machine translation are taken up in the paper, which discusses subjects for semantics and for syntactic analysis and translation software. 14 references.

  16. Bandwidth extension of speech using perceptual criteria

    CERN Document Server

    Berisha, Visar; Liss, Julie

    2013-01-01

    Bandwidth extension of speech is used in the International Telecommunication Union G.729.1 standard in which the narrowband bitstream is combined with quantized high-band parameters. Although this system produces high-quality wideband speech, the additional bits used to represent the high band can be further reduced. In addition to the algorithm used in the G.729.1 standard, bandwidth extension methods based on spectrum prediction have also been proposed. Although these algorithms do not require additional bits, they perform poorly when the correlation between the low and the high band is weak. In this book, two wideband speech coding algorithms that rely on bandwidth extension are developed. The algorithms operate as wrappers around existing narrowband compression schemes. More specifically, in these algorithms, the low band is encoded using an existing toll-quality narrowband system, whereas the high band is generated using the proposed extension techniques. The first method relies only on transmitted high-...

  17. Code Cactus; Code Cactus

    Energy Technology Data Exchange (ETDEWEB)

    Fajeau, M; Nguyen, L T; Saunier, J [Commissariat a l' Energie Atomique, Centre d' Etudes Nucleaires de Saclay, 91 - Gif-sur-Yvette (France)

    1966-09-01

    This code handles the following problems: -1) Analysis of thermal experiments on a water loop at high or low pressure; steady state or transient behavior; -2) Analysis of thermal and hydrodynamic behavior of water-cooled and moderated reactors, at either high or low pressure, with boiling permitted; fuel elements are assumed to be flat plates: - Flowrate in parallel channels coupled or not by conduction across plates, with conditions of pressure drops or flowrate, variable or not with respect to time is given; the power can be coupled to reactor kinetics calculation or supplied by the code user. The code, containing a schematic representation of safety rod behavior, is a one dimensional, multi-channel code, and has as its complement (FLID), a one-channel, two-dimensional code. (authors) [French] Ce code permet de traiter les problemes ci-dessous: 1. Depouillement d'essais thermiques sur boucle a eau, haute ou basse pression, en regime permanent ou transitoire; 2. Etudes thermiques et hydrauliques de reacteurs a eau, a plaques, a haute ou basse pression, ebullition permise: - repartition entre canaux paralleles, couples on non par conduction a travers plaques, pour des conditions de debit ou de pertes de charge imposees, variables ou non dans le temps; - la puissance peut etre couplee a la neutronique et une representation schematique des actions de securite est prevue. Ce code (Cactus) a une dimension d'espace et plusieurs canaux, a pour complement Flid qui traite l'etude d'un seul canal a deux dimensions. (auteurs)

  18. Machine learning and medical imaging

    CERN Document Server

    Shen, Dinggang; Sabuncu, Mert

    2016-01-01

    Machine Learning and Medical Imaging presents state-of- the-art machine learning methods in medical image analysis. It first summarizes cutting-edge machine learning algorithms in medical imaging, including not only classical probabilistic modeling and learning methods, but also recent breakthroughs in deep learning, sparse representation/coding, and big data hashing. In the second part leading research groups around the world present a wide spectrum of machine learning methods with application to different medical imaging modalities, clinical domains, and organs. The biomedical imaging modalities include ultrasound, magnetic resonance imaging (MRI), computed tomography (CT), histology, and microscopy images. The targeted organs span the lung, liver, brain, and prostate, while there is also a treatment of examining genetic associations. Machine Learning and Medical Imaging is an ideal reference for medical imaging researchers, industry scientists and engineers, advanced undergraduate and graduate students, a...

  19. Machine Learning an algorithmic perspective

    CERN Document Server

    Marsland, Stephen

    2009-01-01

    Traditional books on machine learning can be divided into two groups - those aimed at advanced undergraduates or early postgraduates with reasonable mathematical knowledge and those that are primers on how to code algorithms. The field is ready for a text that not only demonstrates how to use the algorithms that make up machine learning methods, but also provides the background needed to understand how and why these algorithms work. Machine Learning: An Algorithmic Perspective is that text.Theory Backed up by Practical ExamplesThe book covers neural networks, graphical models, reinforcement le

  20. Using the TED Talks to Evaluate Spoken Post-editing of Machine Translation

    DEFF Research Database (Denmark)

    Liyanapathirana, Jeevanthi; Popescu-Belis, Andrei

    2016-01-01

    This paper presents a solution to evaluate spoken post-editing of imperfect machine translation output by a human translator. We compare two approaches to the combination of machine translation (MT) and automatic speech recognition (ASR): a heuristic algorithm and a machine learning method...

  1. Machine Learning

    CERN Multimedia

    CERN. Geneva

    2017-01-01

    Machine learning, which builds on ideas in computer science, statistics, and optimization, focuses on developing algorithms to identify patterns and regularities in data, and using these learned patterns to make predictions on new observations. Boosted by its industrial and commercial applications, the field of machine learning is quickly evolving and expanding. Recent advances have seen great success in the realms of computer vision, natural language processing, and broadly in data science. Many of these techniques have already been applied in particle physics, for instance for particle identification, detector monitoring, and the optimization of computer resources. Modern machine learning approaches, such as deep learning, are only just beginning to be applied to the analysis of High Energy Physics data to approach more and more complex problems. These classes will review the framework behind machine learning and discuss recent developments in the field.

  2. Machine Translation

    Indian Academy of Sciences (India)

    Research Mt System Example: The 'Janus' Translating Phone Project. The Janus ... based on laptops, and simultaneous translation of two speakers in a dialogue. For more ..... The current focus in MT research is on using machine learning.

  3. Using Peephole Optimization on Intermediate Code

    NARCIS (Netherlands)

    Tanenbaum, A.S.; van Staveren, H.; Stevenson, J.W.

    1982-01-01

    Many portable compilers generate an intermediate code that is subsequently translated into the target machine's assembly language. In this paper a stack-machine-based intermediate code suitable for algebraic languages (e.g., PASCAL, C, FORTRAN) and most byte-addressed mini- and microcomputers is

  4. Collective speech acts

    NARCIS (Netherlands)

    Meijers, A.W.M.; Tsohatzidis, S.L.

    2007-01-01

    From its early development in the 1960s, speech act theory always had an individualistic orientation. It focused exclusively on speech acts performed by individual agents. Paradigmatic examples are ‘I promise that p’, ‘I order that p’, and ‘I declare that p’. There is a single speaker and a single

  5. Private Speech in Ballet

    Science.gov (United States)

    Johnston, Dale

    2006-01-01

    Authoritarian teaching practices in ballet inhibit the use of private speech. This paper highlights the critical importance of private speech in the cognitive development of young ballet students, within what is largely a non-verbal art form. It draws upon research by Russian psychologist Lev Vygotsky and contemporary socioculturalists, to…

  6. Free Speech Yearbook 1980.

    Science.gov (United States)

    Kane, Peter E., Ed.

    The 11 articles in this collection deal with theoretical and practical freedom of speech issues. The topics covered are (1) the United States Supreme Court and communication theory; (2) truth, knowledge, and a democratic respect for diversity; (3) denial of freedom of speech in Jock Yablonski's campaign for the presidency of the United Mine…

  7. Illustrated Speech Anatomy.

    Science.gov (United States)

    Shearer, William M.

    Written for students in the fields of speech correction and audiology, the text deals with the following: structures involved in respiration; the skeleton and the processes of inhalation and exhalation; phonation and pitch, the larynx, and esophageal speech; muscles involved in articulation; muscles involved in resonance; and the anatomy of the…

  8. Free Speech. No. 38.

    Science.gov (United States)

    Kane, Peter E., Ed.

    This issue of "Free Speech" contains the following articles: "Daniel Schoor Relieved of Reporting Duties" by Laurence Stern, "The Sellout at CBS" by Michael Harrington, "Defending Dan Schorr" by Tome Wicker, "Speech to the Washington Press Club, February 25, 1976" by Daniel Schorr, "Funds…

  9. Criteria for Labelling Prosodic Aspects of English Speech.

    Science.gov (United States)

    Bagshaw, Paul C.; Williams, Briony J.

    A study reports a set of labelling criteria which have been developed to label prosodic events in clear, continuous speech, and proposes a scheme whereby this information can be transcribed in a machine readable format. A prosody in a syllabic domain which is synchronized with a phonemic segmentation was annotated. A procedural definition of…

  10. SII-Based Speech Prepocessing for Intelligibility Improvement in Noise

    DEFF Research Database (Denmark)

    Taal, Cees H.; Jensen, Jesper

    2013-01-01

    filter sets certain frequency bands to zero when they do not contribute to intelligibility anymore. Experiments show large intelligibility improvements with the proposed method when used in stationary speech-shaped noise. However, it was also found that the method does not perform well for speech...... corrupted by a competing speaker. This is due to the fact that the SII is not a reliable intelligibility predictor for fluctuating noise sources. MATLAB code is provided....

  11. Marital conflict and adjustment: speech nonfluencies in intimate disclosure.

    Science.gov (United States)

    Paul, E L; White, K M; Speisman, J C; Costos, D

    1988-06-01

    Speech nonfluency in response to questions about the marital relationship was used to assess anxiety. Subjects were 31 husbands and 31 wives, all white, college educated, from middle- to lower-middle-class families, and ranging from 20 to 30 years of age. Three types of nonfluencies were coded: filled pauses, unfilled pauses, and repetitions. Speech-disturbance ratios were computed by dividing the sum of speech nonfluencies by the total words spoken. The results support the notion that some issues within marriage are more sensitive and/or problematic than others, and that, in an interview situation, gender interacts with question content in the production of nonfluencies.

  12. Musician advantage for speech-on-speech perception

    NARCIS (Netherlands)

    Başkent, Deniz; Gaudrain, Etienne

    Evidence for transfer of musical training to better perception of speech in noise has been mixed. Unlike speech-in-noise, speech-on-speech perception utilizes many of the skills that musical training improves, such as better pitch perception and stream segregation, as well as use of higher-level

  13. Speech Production and Speech Discrimination by Hearing-Impaired Children.

    Science.gov (United States)

    Novelli-Olmstead, Tina; Ling, Daniel

    1984-01-01

    Seven hearing impaired children (five to seven years old) assigned to the Speakers group made highly significant gains in speech production and auditory discrimination of speech, while Listeners made only slight speech production gains and no gains in auditory discrimination. Combined speech and auditory training was more effective than auditory…

  14. Hybrid methodological approach to context-dependent speech recognition

    Directory of Open Access Journals (Sweden)

    Dragiša Mišković

    2017-01-01

    Full Text Available Although the importance of contextual information in speech recognition has been acknowledged for a long time now, it has remained clearly underutilized even in state-of-the-art speech recognition systems. This article introduces a novel, methodologically hybrid approach to the research question of context-dependent speech recognition in human–machine interaction. To the extent that it is hybrid, the approach integrates aspects of both statistical and representational paradigms. We extend the standard statistical pattern-matching approach with a cognitively inspired and analytically tractable model with explanatory power. This methodological extension allows for accounting for contextual information which is otherwise unavailable in speech recognition systems, and using it to improve post-processing of recognition hypotheses. The article introduces an algorithm for evaluation of recognition hypotheses, illustrates it for concrete interaction domains, and discusses its implementation within two prototype conversational agents.

  15. Neural Decoder for Topological Codes

    Science.gov (United States)

    Torlai, Giacomo; Melko, Roger G.

    2017-07-01

    We present an algorithm for error correction in topological codes that exploits modern machine learning techniques. Our decoder is constructed from a stochastic neural network called a Boltzmann machine, of the type extensively used in deep learning. We provide a general prescription for the training of the network and a decoding strategy that is applicable to a wide variety of stabilizer codes with very little specialization. We demonstrate the neural decoder numerically on the well-known two-dimensional toric code with phase-flip errors.

  16. Machine Protection

    International Nuclear Information System (INIS)

    Zerlauth, Markus; Schmidt, Rüdiger; Wenninger, Jörg

    2012-01-01

    The present architecture of the machine protection system is being recalled and the performance of the associated systems during the 2011 run will be briefly summarized. An analysis of the causes of beam dumps as well as an assessment of the dependability of the machine protection systems (MPS) itself is being presented. Emphasis will be given to events that risked exposing parts of the machine to damage. Further improvements and mitigations of potential holes in the protection systems will be evaluated along with their impact on the 2012 run. The role of rMPP during the various operational phases (commissioning, intensity ramp up, MDs...) will be discussed along with a proposal for the intensity ramp up for the start of beam operation in 2012

  17. Machine Learning

    Energy Technology Data Exchange (ETDEWEB)

    Chikkagoudar, Satish; Chatterjee, Samrat; Thomas, Dennis G.; Carroll, Thomas E.; Muller, George

    2017-04-21

    The absence of a robust and unified theory of cyber dynamics presents challenges and opportunities for using machine learning based data-driven approaches to further the understanding of the behavior of such complex systems. Analysts can also use machine learning approaches to gain operational insights. In order to be operationally beneficial, cybersecurity machine learning based models need to have the ability to: (1) represent a real-world system, (2) infer system properties, and (3) learn and adapt based on expert knowledge and observations. Probabilistic models and Probabilistic graphical models provide these necessary properties and are further explored in this chapter. Bayesian Networks and Hidden Markov Models are introduced as an example of a widely used data driven classification/modeling strategy.

  18. Machine Protection

    CERN Document Server

    Zerlauth, Markus; Wenninger, Jörg

    2012-01-01

    The present architecture of the machine protection system is being recalled and the performance of the associated systems during the 2011 run will be briefly summarized. An analysis of the causes of beam dumps as well as an assessment of the dependability of the machine protection systems (MPS) itself is being presented. Emphasis will be given to events that risked exposing parts of the machine to damage. Further improvements and mitigations of potential holes in the protection systems will be evaluated along with their impact on the 2012 run. The role of rMPP during the various operational phases (commissioning, intensity ramp up, MDs...) will be discussed along with a proposal for the intensity ramp up for the start of beam operation in 2012.

  19. Machine Protection

    Energy Technology Data Exchange (ETDEWEB)

    Zerlauth, Markus; Schmidt, Rüdiger; Wenninger, Jörg [European Organization for Nuclear Research, Geneva (Switzerland)

    2012-07-01

    The present architecture of the machine protection system is being recalled and the performance of the associated systems during the 2011 run will be briefly summarized. An analysis of the causes of beam dumps as well as an assessment of the dependability of the machine protection systems (MPS) itself is being presented. Emphasis will be given to events that risked exposing parts of the machine to damage. Further improvements and mitigations of potential holes in the protection systems will be evaluated along with their impact on the 2012 run. The role of rMPP during the various operational phases (commissioning, intensity ramp up, MDs...) will be discussed along with a proposal for the intensity ramp up for the start of beam operation in 2012.

  20. Perception of words and pitch patterns in song and speech

    Directory of Open Access Journals (Sweden)

    Julia eMerrill

    2012-03-01

    Full Text Available This fMRI study examines shared and distinct cortical areas involved in the auditory perception of song and speech at the level of their underlying constituents: words, pitch and rhythm. Univariate and multivariate analyses were performed on the brain activity patterns of six conditions, arranged in a subtractive hierarchy: sung sentences including words, pitch and rhythm; hummed speech prosody and song melody containing only pitch patterns and rhythm; as well as the pure musical or speech rhythm.Systematic contrasts between these balanced conditions following their hierarchical organization showed a great overlap between song and speech at all levels in the bilateral temporal lobe, but suggested a differential role of the inferior frontal gyrus (IFG and intraparietal sulcus (IPS in processing song and speech. The left IFG was involved in word- and pitch-related processing in speech, the right IFG in processing pitch in song.Furthermore, the IPS showed sensitivity to discrete pitch relations in song as opposed to the gliding pitch in speech. Finally, the superior temporal gyrus and premotor cortex coded for general differences between words and pitch patterns, irrespective of whether they were sung or spoken. Thus, song and speech share many features which are reflected in a fundamental similarity of brain areas involved in their perception. However, fine-grained acoustic differences on word and pitch level are reflected in the activity of IFG and IPS.

  1. Random Deep Belief Networks for Recognizing Emotions from Speech Signals

    Directory of Open Access Journals (Sweden)

    Guihua Wen

    2017-01-01

    Full Text Available Now the human emotions can be recognized from speech signals using machine learning methods; however, they are challenged by the lower recognition accuracies in real applications due to lack of the rich representation ability. Deep belief networks (DBN can automatically discover the multiple levels of representations in speech signals. To make full of its advantages, this paper presents an ensemble of random deep belief networks (RDBN method for speech emotion recognition. It firstly extracts the low level features of the input speech signal and then applies them to construct lots of random subspaces. Each random subspace is then provided for DBN to yield the higher level features as the input of the classifier to output an emotion label. All outputted emotion labels are then fused through the majority voting to decide the final emotion label for the input speech signal. The conducted experimental results on benchmark speech emotion databases show that RDBN has better accuracy than the compared methods for speech emotion recognition.

  2. Unvoiced Speech Recognition Using Tissue-Conductive Acoustic Sensor

    Directory of Open Access Journals (Sweden)

    Heracleous Panikos

    2007-01-01

    Full Text Available We present the use of stethoscope and silicon NAM (nonaudible murmur microphones in automatic speech recognition. NAM microphones are special acoustic sensors, which are attached behind the talker's ear and can capture not only normal (audible speech, but also very quietly uttered speech (nonaudible murmur. As a result, NAM microphones can be applied in automatic speech recognition systems when privacy is desired in human-machine communication. Moreover, NAM microphones show robustness against noise and they might be used in special systems (speech recognition, speech transform, etc. for sound-impaired people. Using adaptation techniques and a small amount of training data, we achieved for a 20 k dictation task a word accuracy for nonaudible murmur recognition in a clean environment. In this paper, we also investigate nonaudible murmur recognition in noisy environments and the effect of the Lombard reflex on nonaudible murmur recognition. We also propose three methods to integrate audible speech and nonaudible murmur recognition using a stethoscope NAM microphone with very promising results.

  3. Unvoiced Speech Recognition Using Tissue-Conductive Acoustic Sensor

    Directory of Open Access Journals (Sweden)

    Hiroshi Saruwatari

    2007-01-01

    Full Text Available We present the use of stethoscope and silicon NAM (nonaudible murmur microphones in automatic speech recognition. NAM microphones are special acoustic sensors, which are attached behind the talker's ear and can capture not only normal (audible speech, but also very quietly uttered speech (nonaudible murmur. As a result, NAM microphones can be applied in automatic speech recognition systems when privacy is desired in human-machine communication. Moreover, NAM microphones show robustness against noise and they might be used in special systems (speech recognition, speech transform, etc. for sound-impaired people. Using adaptation techniques and a small amount of training data, we achieved for a 20 k dictation task a 93.9% word accuracy for nonaudible murmur recognition in a clean environment. In this paper, we also investigate nonaudible murmur recognition in noisy environments and the effect of the Lombard reflex on nonaudible murmur recognition. We also propose three methods to integrate audible speech and nonaudible murmur recognition using a stethoscope NAM microphone with very promising results.

  4. Inner Speech's Relationship With Overt Speech in Poststroke Aphasia.

    Science.gov (United States)

    Stark, Brielle C; Geva, Sharon; Warburton, Elizabeth A

    2017-09-18

    Relatively preserved inner speech alongside poor overt speech has been documented in some persons with aphasia (PWA), but the relationship of overt speech with inner speech is still largely unclear, as few studies have directly investigated these factors. The present study investigates the relationship of relatively preserved inner speech in aphasia with selected measures of language and cognition. Thirty-eight persons with chronic aphasia (27 men, 11 women; average age 64.53 ± 13.29 years, time since stroke 8-111 months) were classified as having relatively preserved inner and overt speech (n = 21), relatively preserved inner speech with poor overt speech (n = 8), or not classified due to insufficient measurements of inner and/or overt speech (n = 9). Inner speech scores (by group) were correlated with selected measures of language and cognition from the Comprehensive Aphasia Test (Swinburn, Porter, & Al, 2004). The group with poor overt speech showed a significant relationship of inner speech with overt naming (r = .95, p speech and language and cognition factors were not significant for the group with relatively good overt speech. As in previous research, we show that relatively preserved inner speech is found alongside otherwise severe production deficits in PWA. PWA with poor overt speech may rely more on preserved inner speech for overt picture naming (perhaps due to shared resources with verbal working memory) and for written picture description (perhaps due to reliance on inner speech due to perceived task difficulty). Assessments of inner speech may be useful as a standard component of aphasia screening, and therapy focused on improving and using inner speech may prove clinically worthwhile. https://doi.org/10.23641/asha.5303542.

  5. Teletherapy machine

    International Nuclear Information System (INIS)

    Panyam, Vinatha S.; Rakshit, Sougata; Kulkarni, M.S.; Pradeepkumar, K.S.

    2017-01-01

    Radiation Standards Section (RSS), RSSD, BARC is the national metrology institute for ionizing radiation. RSS develops and maintains radiation standards for X-ray, beta, gamma and neutron radiations. In radiation dosimetry, traceability, accuracy and consistency of radiation measurements is very important especially in radiotherapy where the success of patient treatment is dependent on the accuracy of the dose delivered to the tumour. Cobalt teletherapy machines have been used in the treatment of cancer since the early 1950s and India had its first cobalt teletherapy machine installed at the Cancer Institute, Chennai in 1956

  6. Environmental Contamination of Normal Speech.

    Science.gov (United States)

    Harley, Trevor A.

    1990-01-01

    Environmentally contaminated speech errors (irrelevant words or phrases derived from the speaker's environment and erroneously incorporated into speech) are hypothesized to occur at a high level of speech processing, but with a relatively late insertion point. The data indicate that speech production processes are not independent of other…

  7. APPRECIATING SPEECH THROUGH GAMING

    Directory of Open Access Journals (Sweden)

    Mario T Carreon

    2014-06-01

    Full Text Available This paper discusses the Speech and Phoneme Recognition as an Educational Aid for the Deaf and Hearing Impaired (SPREAD application and the ongoing research on its deployment as a tool for motivating deaf and hearing impaired students to learn and appreciate speech. This application uses the Sphinx-4 voice recognition system to analyze the vocalization of the student and provide prompt feedback on their pronunciation. The packaging of the application as an interactive game aims to provide additional motivation for the deaf and hearing impaired student through visual motivation for them to learn and appreciate speech.

  8. Global Freedom of Speech

    DEFF Research Database (Denmark)

    Binderup, Lars Grassme

    2007-01-01

    , as opposed to a legal norm, that curbs exercises of the right to free speech that offend the feelings or beliefs of members from other cultural groups. The paper rejects the suggestion that acceptance of such a norm is in line with liberal egalitarian thinking. Following a review of the classical liberal...... egalitarian reasons for free speech - reasons from overall welfare, from autonomy and from respect for the equality of citizens - it is argued that these reasons outweigh the proposed reasons for curbing culturally offensive speech. Currently controversial cases such as that of the Danish Cartoon Controversy...

  9. Using others' words: conversational use of reported speech by individuals with aphasia and their communication partners.

    Science.gov (United States)

    Hengst, Julie A; Frame, Simone R; Neuman-Stritzel, Tiffany; Gannaway, Rachel

    2005-02-01

    Reported speech, wherein one quotes or paraphrases the speech of another, has been studied extensively as a set of linguistic and discourse practices. Researchers agree that reported speech is pervasive, found across languages, and used in diverse contexts. However, to date, there have been no studies of the use of reported speech among individuals with aphasia. Grounded in an interactional sociolinguistic perspective, the study presented here documents and analyzes the use of reported speech by 7 adults with mild to moderately severe aphasia and their routine communication partners. Each of the 7 pairs was videotaped in 4 everyday activities at home or around the community, yielding over 27 hr of conversational interaction for analysis. A coding scheme was developed that identified 5 types of explicitly marked reported speech: direct, indirect, projected, indexed, and undecided. Analysis of the data documented reported speech as a common discourse practice used successfully by the individuals with aphasia and their communication partners. All participants produced reported speech at least once, and across all observations the target pairs produced 400 reported speech episodes (RSEs), 149 by individuals with aphasia and 251 by their communication partners. For all participants, direct and indirect forms were the most prevalent (70% of RSEs). Situated discourse analysis of specific episodes of reported speech used by 3 of the pairs provides detailed portraits of the diverse interactional, referential, social, and discourse functions of reported speech and explores ways that the pairs used reported speech to successfully frame talk despite their ongoing management of aphasia.

  10. Neuroscience-inspired computational systems for speech recognition under noisy conditions

    Science.gov (United States)

    Schafer, Phillip B.

    Humans routinely recognize speech in challenging acoustic environments with background music, engine sounds, competing talkers, and other acoustic noise. However, today's automatic speech recognition (ASR) systems perform poorly in such environments. In this dissertation, I present novel methods for ASR designed to approach human-level performance by emulating the brain's processing of sounds. I exploit recent advances in auditory neuroscience to compute neuron-based representations of speech, and design novel methods for decoding these representations to produce word transcriptions. I begin by considering speech representations modeled on the spectrotemporal receptive fields of auditory neurons. These representations can be tuned to optimize a variety of objective functions, which characterize the response properties of a neural population. I propose an objective function that explicitly optimizes the noise invariance of the neural responses, and find that it gives improved performance on an ASR task in noise compared to other objectives. The method as a whole, however, fails to significantly close the performance gap with humans. I next consider speech representations that make use of spiking model neurons. The neurons in this method are feature detectors that selectively respond to spectrotemporal patterns within short time windows in speech. I consider a number of methods for training the response properties of the neurons. In particular, I present a method using linear support vector machines (SVMs) and show that this method produces spikes that are robust to additive noise. I compute the spectrotemporal receptive fields of the neurons for comparison with previous physiological results. To decode the spike-based speech representations, I propose two methods designed to work on isolated word recordings. The first method uses a classical ASR technique based on the hidden Markov model. The second method is a novel template-based recognition scheme that takes

  11. Machine testning

    DEFF Research Database (Denmark)

    De Chiffre, Leonardo

    This document is used in connection with a laboratory exercise of 3 hours duration as a part of the course GEOMETRICAL METROLOGY AND MACHINE TESTING. The exercise includes a series of tests carried out by the student on a conventional and a numerically controled lathe, respectively. This document...

  12. Machine rates for selected forest harvesting machines

    Science.gov (United States)

    R.W. Brinker; J. Kinard; Robert Rummer; B. Lanford

    2002-01-01

    Very little new literature has been published on the subject of machine rates and machine cost analysis since 1989 when the Alabama Agricultural Experiment Station Circular 296, Machine Rates for Selected Forest Harvesting Machines, was originally published. Many machines discussed in the original publication have undergone substantial changes in various aspects, not...

  13. Audiovisual speech facilitates voice learning.

    Science.gov (United States)

    Sheffert, Sonya M; Olson, Elizabeth

    2004-02-01

    In this research, we investigated the effects of voice and face information on the perceptual learning of talkers and on long-term memory for spoken words. In the first phase, listeners were trained over several days to identify voices from words presented auditorily or audiovisually. The training data showed that visual information about speakers enhanced voice learning, revealing cross-modal connections in talker processing akin to those observed in speech processing. In the second phase, the listeners completed an auditory or audiovisual word recognition memory test in which equal numbers of words were spoken by familiar and unfamiliar talkers. The data showed that words presented by familiar talkers were more likely to be retrieved from episodic memory, regardless of modality. Together, these findings provide new information about the representational code underlying familiar talker recognition and the role of stimulus familiarity in episodic word recognition.

  14. Charisma in business speeches

    DEFF Research Database (Denmark)

    Niebuhr, Oliver; Brem, Alexander; Novák-Tót, Eszter

    2016-01-01

    to business speeches. Consistent with the public opinion, our findings are indicative of Steve Jobs being a more charismatic speaker than Mark Zuckerberg. Beyond previous studies, our data suggest that rhythm and emphatic accentuation are also involved in conveying charisma. Furthermore, the differences...... between Steve Jobs and Mark Zuckerberg and the investor- and customer-related sections of their speeches support the modern understanding of charisma as a gradual, multiparametric, and context-sensitive concept....

  15. Speech spectrum envelope modeling

    Czech Academy of Sciences Publication Activity Database

    Vích, Robert; Vondra, Martin

    Vol. 4775, - (2007), s. 129-137 ISSN 0302-9743. [COST Action 2102 International Workshop. Vietri sul Mare, 29.03.2007-31.03.2007] R&D Projects: GA AV ČR(CZ) 1ET301710509 Institutional research plan: CEZ:AV0Z20670512 Keywords : speech * speech processing * cepstral analysis Subject RIV: JA - Electronics ; Optoelectronics, Electrical Engineering Impact factor: 0.302, year: 2005

  16. Induction technology optimization code

    International Nuclear Information System (INIS)

    Caporaso, G.J.; Brooks, A.L.; Kirbie, H.C.

    1992-01-01

    A code has been developed to evaluate relative costs of induction accelerator driver systems for relativistic klystrons. The code incorporates beam generation, transport and pulsed power system constraints to provide an integrated design tool. The code generates an injector/accelerator combination which satisfies the top level requirements and all system constraints once a small number of design choices have been specified (rise time of the injector voltage and aspect ratio of the ferrite induction cores, for example). The code calculates dimensions of accelerator mechanical assemblies and values of all electrical components. Cost factors for machined parts, raw materials and components are applied to yield a total system cost. These costs are then plotted as a function of the two design choices to enable selection of an optimum design based on various criteria. (Author) 11 refs., 3 figs

  17. Memory for speech and speech for memory.

    Science.gov (United States)

    Locke, J L; Kutz, K J

    1975-03-01

    Thirty kindergarteners, 15 who substituted /w/ for /r/ and 15 with correct articulation, received two perception tests and a memory test that included /w/ and /r/ in minimally contrastive syllables. Although both groups had nearly perfect perception of the experimenter's productions of /w/ and /r/, misarticulating subjects perceived their own tape-recorded w/r productions as /w/. In the memory task these same misarticulating subjects committed significantly more /w/-/r/ confusions in unspoken recall. The discussion considers why people subvocally rehearse; a developmental period in which children do not rehearse; ways subvocalization may aid recall, including motor and acoustic encoding; an echoic store that provides additional recall support if subjects rehearse vocally, and perception of self- and other- produced phonemes by misarticulating children-including its relevance to a motor theory of perception. Evidence is presented that speech for memory can be sufficiently impaired to cause memory disorder. Conceptions that restrict speech disorder to an impairment of communication are challenged.

  18. Predicting speech intelligibility in conditions with nonlinearly processed noisy speech

    DEFF Research Database (Denmark)

    Jørgensen, Søren; Dau, Torsten

    2013-01-01

    The speech-based envelope power spectrum model (sEPSM; [1]) was proposed in order to overcome the limitations of the classical speech transmission index (STI) and speech intelligibility index (SII). The sEPSM applies the signal-tonoise ratio in the envelope domain (SNRenv), which was demonstrated...... to successfully predict speech intelligibility in conditions with nonlinearly processed noisy speech, such as processing with spectral subtraction. Moreover, a multiresolution version (mr-sEPSM) was demonstrated to account for speech intelligibility in various conditions with stationary and fluctuating...

  19. Modelling the Architecture of Phonetic Plans: Evidence from Apraxia of Speech

    Science.gov (United States)

    Ziegler, Wolfram

    2009-01-01

    In theories of spoken language production, the gestural code prescribing the movements of the speech organs is usually viewed as a linear string of holistic, encapsulated, hard-wired, phonetic plans, e.g., of the size of phonemes or syllables. Interactions between phonetic units on the surface of overt speech are commonly attributed to either the…

  20. Subspace-Based Noise Reduction for Speech Signals via Diagonal and Triangular Matrix Decompositions

    DEFF Research Database (Denmark)

    Hansen, Per Christian; Jensen, Søren Holdt

    2007-01-01

    We survey the definitions and use of rank-revealing matrix decompositions in single-channel noise reduction algorithms for speech signals. Our algorithms are based on the rank-reduction paradigm and, in particular, signal subspace techniques. The focus is on practical working algorithms, using both...... with working Matlab code and applications in speech processing....

  1. Family Worlds: Couple Satisfaction, Parenting Style, and Mothers' and Fathers' Speech to Young Children.

    Science.gov (United States)

    Pratt, Michael W.; And Others

    1992-01-01

    Investigated relations between certain family context variables and the conversational behavior of 36 parents who were playing with their 3 year olds. Transcripts were coded for types of conversational functions and structure of parent speech. Marital satisfaction was associated with aspects of parent speech. (LB)

  2. Electric machines

    CERN Document Server

    Gross, Charles A

    2006-01-01

    BASIC ELECTROMAGNETIC CONCEPTSBasic Magnetic ConceptsMagnetically Linear Systems: Magnetic CircuitsVoltage, Current, and Magnetic Field InteractionsMagnetic Properties of MaterialsNonlinear Magnetic Circuit AnalysisPermanent MagnetsSuperconducting MagnetsThe Fundamental Translational EM MachineThe Fundamental Rotational EM MachineMultiwinding EM SystemsLeakage FluxThe Concept of Ratings in EM SystemsSummaryProblemsTRANSFORMERSThe Ideal n-Winding TransformerTransformer Ratings and Per-Unit ScalingThe Nonideal Three-Winding TransformerThe Nonideal Two-Winding TransformerTransformer Efficiency and Voltage RegulationPractical ConsiderationsThe AutotransformerOperation of Transformers in Three-Phase EnvironmentsSequence Circuit Models for Three-Phase Transformer AnalysisHarmonics in TransformersSummaryProblemsBASIC MECHANICAL CONSIDERATIONSSome General PerspectivesEfficiencyLoad Torque-Speed CharacteristicsMass Polar Moment of InertiaGearingOperating ModesTranslational SystemsA Comprehensive Example: The ElevatorP...

  3. Charging machine

    International Nuclear Information System (INIS)

    Medlin, J.B.

    1976-01-01

    A charging machine for loading fuel slugs into the process tubes of a nuclear reactor includes a tubular housing connected to the process tube, a charging trough connected to the other end of the tubular housing, a device for loading the charging trough with a group of fuel slugs, means for equalizing the coolant pressure in the charging trough with the pressure in the process tubes, means for pushing the group of fuel slugs into the process tube and a latch and a seal engaging the last object in the group of fuel slugs to prevent the fuel slugs from being ejected from the process tube when the pusher is removed and to prevent pressure liquid from entering the charging machine. 3 claims, 11 drawing figures

  4. Implications of Sepedi/English code switching for ASR systems

    CSIR Research Space (South Africa)

    Modipa, TI

    2013-12-01

    Full Text Available . We also perform an initial acoustic analysis to determine the impact of such code switching on speech recognition performance. We nd that the frequency of code switching is unexpectedly high, and that the continuum of code switching (from unmodi ed...

  5. Genesis machines

    CERN Document Server

    Amos, Martyn

    2014-01-01

    Silicon chips are out. Today's scientists are using real, wet, squishy, living biology to build the next generation of computers. Cells, gels and DNA strands are the 'wetware' of the twenty-first century. Much smaller and more intelligent, these organic computers open up revolutionary possibilities. Tracing the history of computing and revealing a brave new world to come, Genesis Machines describes how this new technology will change the way we think not just about computers - but about life itself.

  6. Music and Speech Perception in Children Using Sung Speech.

    Science.gov (United States)

    Nie, Yingjiu; Galvin, John J; Morikawa, Michael; André, Victoria; Wheeler, Harley; Fu, Qian-Jie

    2018-01-01

    This study examined music and speech perception in normal-hearing children with some or no musical training. Thirty children (mean age = 11.3 years), 15 with and 15 without formal music training participated in the study. Music perception was measured using a melodic contour identification (MCI) task; stimuli were a piano sample or sung speech with a fixed timbre (same word for each note) or a mixed timbre (different words for each note). Speech perception was measured in quiet and in steady noise using a matrix-styled sentence recognition task; stimuli were naturally intonated speech or sung speech with a fixed pitch (same note for each word) or a mixed pitch (different notes for each word). Significant musician advantages were observed for MCI and speech in noise but not for speech in quiet. MCI performance was significantly poorer with the mixed timbre stimuli. Speech performance in noise was significantly poorer with the fixed or mixed pitch stimuli than with spoken speech. Across all subjects, age at testing and MCI performance were significantly correlated with speech performance in noise. MCI and speech performance in quiet was significantly poorer for children than for adults from a related study using the same stimuli and tasks; speech performance in noise was significantly poorer for young than for older children. Long-term music training appeared to benefit melodic pitch perception and speech understanding in noise in these pediatric listeners.

  7. Non-linear Dynamics of Speech in Schizophrenia

    DEFF Research Database (Denmark)

    Fusaroli, Riccardo; Simonsen, Arndis; Weed, Ethan

    (regularity and complexity) of speech. Our aims are (1) to achieve a more fine-grained understanding of the speech patterns in schizophrenia than has previously been achieved using traditional, linear measures of prosody and fluency, and (2) to employ the results in a supervised machine-learning process......-effects inference. SANS and SAPS scores were predicted using a 10-fold cross-validated multiple linear regression. Both analyses were iterated 1000 to test for stability of results. Results: Voice dynamics allowed discrimination of patients with schizophrenia from healthy controls with a balanced accuracy of 85...

  8. Under-resourced speech recognition based on the speech manifold

    CSIR Research Space (South Africa)

    Sahraeian, R

    2015-09-01

    Full Text Available Conventional acoustic modeling involves estimating many parameters to effectively model feature distributions. The sparseness of speech and text data, however, degrades the reliability of the estimation process and makes speech recognition a...

  9. Speech Alarms Pilot Study

    Science.gov (United States)

    Sandor, A.; Moses, H. R.

    2016-01-01

    Currently on the International Space Station (ISS) and other space vehicles Caution & Warning (C&W) alerts are represented with various auditory tones that correspond to the type of event. This system relies on the crew's ability to remember what each tone represents in a high stress, high workload environment when responding to the alert. Furthermore, crew receive a year or more in advance of the mission that makes remembering the semantic meaning of the alerts more difficult. The current system works for missions conducted close to Earth where ground operators can assist as needed. On long duration missions, however, they will need to work off-nominal events autonomously. There is evidence that speech alarms may be easier and faster to recognize, especially during an off-nominal event. The Information Presentation Directed Research Project (FY07-FY09) funded by the Human Research Program included several studies investigating C&W alerts. The studies evaluated tone alerts currently in use with NASA flight deck displays along with candidate speech alerts. A follow-on study used four types of speech alerts to investigate how quickly various types of auditory alerts with and without a speech component - either at the beginning or at the end of the tone - can be identified. Even though crew were familiar with the tone alert from training or direct mission experience, alerts starting with a speech component were identified faster than alerts starting with a tone. The current study replicated the results from the previous study in a more rigorous experimental design to determine if the candidate speech alarms are ready for transition to operations or if more research is needed. Four types of alarms (caution, warning, fire, and depressurization) were presented to participants in both tone and speech formats in laboratory settings and later in the Human Exploration Research Analog (HERA). In the laboratory study, the alerts were presented by software and participants were

  10. Intelligibility of speech of children with speech and sound disorders

    OpenAIRE

    Ivetac, Tina

    2014-01-01

    The purpose of this study is to examine speech intelligibility of children with primary speech and sound disorders aged 3 to 6 years in everyday life. The research problem is based on the degree to which parents or guardians, immediate family members (sister, brother, grandparents), extended family members (aunt, uncle, cousin), child's friends, other acquaintances, child's teachers and strangers understand the speech of children with speech sound disorders. We examined whether the level ...

  11. Speech Recognition for the iCub Platform

    Directory of Open Access Journals (Sweden)

    Bertrand Higy

    2018-02-01

    Full Text Available This paper describes open source software (available at https://github.com/robotology/natural-speech to build automatic speech recognition (ASR systems and run them within the YARP platform. The toolkit is designed (i to allow non-ASR experts to easily create their own ASR system and run it on iCub and (ii to build deep learning-based models specifically addressing the main challenges an ASR system faces in the context of verbal human–iCub interactions. The toolkit mostly consists of Python, C++ code and shell scripts integrated in YARP. As additional contribution, a second codebase (written in Matlab is provided for more expert ASR users who want to experiment with bio-inspired and developmental learning-inspired ASR systems. Specifically, we provide code for two distinct kinds of speech recognition: “articulatory” and “unsupervised” speech recognition. The first is largely inspired by influential neurobiological theories of speech perception which assume speech perception to be mediated by brain motor cortex activities. Our articulatory systems have been shown to outperform strong deep learning-based baselines. The second type of recognition systems, the “unsupervised” systems, do not use any supervised information (contrary to most ASR systems, including our articulatory systems. To some extent, they mimic an infant who has to discover the basic speech units of a language by herself. In addition, we provide resources consisting of pre-trained deep learning models for ASR, and a 2.5-h speech dataset of spoken commands, the VoCub dataset, which can be used to adapt an ASR system to the typical acoustic environments in which iCub operates.

  12. Gesture-speech integration in children with specific language impairment.

    Science.gov (United States)

    Mainela-Arnold, Elina; Alibali, Martha W; Hostetter, Autumn B; Evans, Julia L

    2014-11-01

    Previous research suggests that speakers are especially likely to produce manual communicative gestures when they have relative ease in thinking about the spatial elements of what they are describing, paired with relative difficulty organizing those elements into appropriate spoken language. Children with specific language impairment (SLI) exhibit poor expressive language abilities together with within-normal-range nonverbal IQs. This study investigated whether weak spoken language abilities in children with SLI influence their reliance on gestures to express information. We hypothesized that these children would rely on communicative gestures to express information more often than their age-matched typically developing (TD) peers, and that they would sometimes express information in gestures that they do not express in the accompanying speech. Participants were 15 children with SLI (aged 5;6-10;0) and 18 age-matched TD controls. Children viewed a wordless cartoon and retold the story to a listener unfamiliar with the story. Children's gestures were identified and coded for meaning using a previously established system. Speech-gesture combinations were coded as redundant if the information conveyed in speech and gesture was the same, and non-redundant if the information conveyed in speech was different from the information conveyed in gesture. Children with SLI produced more gestures than children in the TD group; however, the likelihood that speech-gesture combinations were non-redundant did not differ significantly across the SLI and TD groups. In both groups, younger children were significantly more likely to produce non-redundant speech-gesture combinations than older children. The gesture-speech integration system functions similarly in children with SLI and TD, but children with SLI rely more on gesture to help formulate, conceptualize or express the messages they want to convey. This provides motivation for future research examining whether interventions

  13. Auditory-neurophysiological responses to speech during early childhood: Effects of background noise.

    Science.gov (United States)

    White-Schwoch, Travis; Davies, Evan C; Thompson, Elaine C; Woodruff Carr, Kali; Nicol, Trent; Bradlow, Ann R; Kraus, Nina

    2015-10-01

    Early childhood is a critical period of auditory learning, during which children are constantly mapping sounds to meaning. But this auditory learning rarely occurs in ideal listening conditions-children are forced to listen against a relentless din. This background noise degrades the neural coding of these critical sounds, in turn interfering with auditory learning. Despite the importance of robust and reliable auditory processing during early childhood, little is known about the neurophysiology underlying speech processing in children so young. To better understand the physiological constraints these adverse listening scenarios impose on speech sound coding during early childhood, auditory-neurophysiological responses were elicited to a consonant-vowel syllable in quiet and background noise in a cohort of typically-developing preschoolers (ages 3-5 yr). Overall, responses were degraded in noise: they were smaller, less stable across trials, slower, and there was poorer coding of spectral content and the temporal envelope. These effects were exacerbated in response to the consonant transition relative to the vowel, suggesting that the neural coding of spectrotemporally-dynamic speech features is more tenuous in noise than the coding of static features-even in children this young. Neural coding of speech temporal fine structure, however, was more resilient to the addition of background noise than coding of temporal envelope information. Taken together, these results demonstrate that noise places a neurophysiological constraint on speech processing during early childhood by causing a breakdown in neural processing of speech acoustics. These results may explain why some listeners have inordinate difficulties understanding speech in noise. Speech-elicited auditory-neurophysiological responses offer objective insight into listening skills during early childhood by reflecting the integrity of neural coding in quiet and noise; this paper documents typical response

  14. Robust Speech/Non-Speech Classification in Heterogeneous Multimedia Content

    NARCIS (Netherlands)

    Huijbregts, M.A.H.; de Jong, Franciska M.G.

    In this paper we present a speech/non-speech classification method that allows high quality classification without the need to know in advance what kinds of audible non-speech events are present in an audio recording and that does not require a single parameter to be tuned on in-domain data. Because

  15. Untyped Memory in the Java Virtual Machine

    DEFF Research Database (Denmark)

    Gal, Andreas; Probst, Christian; Franz, Michael

    2005-01-01

    We have implemented a virtual execution environment that executes legacy binary code on top of the type-safe Java Virtual Machine by recompiling native code instructions to type-safe bytecode. As it is essentially impossible to infer static typing into untyped machine code, our system emulates...... untyped memory on top of Java’s type system. While this approach allows to execute native code on any off-the-shelf JVM, the resulting runtime performance is poor. We propose a set of virtual machine extensions that add type-unsafe memory objects to JVM. We contend that these JVM extensions do not relax...... Java’s type system as the same functionality can be achieved in pure Java, albeit much less efficiently....

  16. Is talking to an automated teller machine natural and fun?

    Science.gov (United States)

    Chan, F Y; Khalid, H M

    Usability and affective issues of using automatic speech recognition technology to interact with an automated teller machine (ATM) are investigated in two experiments. The first uncovered dialogue patterns of ATM users for the purpose of designing the user interface for a simulated speech ATM system. Applying the Wizard-of-Oz methodology, multiple mapping and word spotting techniques, the speech driven ATM accommodates bilingual users of Bahasa Melayu and English. The second experiment evaluates the usability of a hybrid speech ATM, comparing it with a simulated manual ATM. The aim is to investigate how natural and fun can talking to a speech ATM be for these first-time users. Subjects performed the withdrawal and balance enquiry tasks. The ANOVA was performed on the usability and affective data. The results showed significant differences between systems in the ability to complete the tasks as well as in transaction errors. Performance was measured on the time taken by subjects to complete the task and the number of speech recognition errors that occurred. On the basis of user emotions, it can be said that the hybrid speech system enabled pleasurable interaction. Despite the limitations of speech recognition technology, users are set to talk to the ATM when it becomes available for public use.

  17. Representational Machines

    DEFF Research Database (Denmark)

    Photography not only represents space. Space is produced photographically. Since its inception in the 19th century, photography has brought to light a vast array of represented subjects. Always situated in some spatial order, photographic representations have been operatively underpinned by social...... to the enterprises of the medium. This is the subject of Representational Machines: How photography enlists the workings of institutional technologies in search of establishing new iconic and social spaces. Together, the contributions to this edited volume span historical epochs, social environments, technological...... possibilities, and genre distinctions. Presenting several distinct ways of producing space photographically, this book opens a new and important field of inquiry for photography research....

  18. Shear machines

    International Nuclear Information System (INIS)

    Astill, M.; Sunderland, A.; Waine, M.G.

    1980-01-01

    A shear machine for irradiated nuclear fuel elements has a replaceable shear assembly comprising a fuel element support block, a shear blade support and a clamp assembly which hold the fuel element to be sheared in contact with the support block. A first clamp member contacts the fuel element remote from the shear blade and a second clamp member contacts the fuel element adjacent the shear blade and is advanced towards the support block during shearing to compensate for any compression of the fuel element caused by the shear blade (U.K.)

  19. Variable Frame Rate and Length Analysis for Data Compression in Distributed Speech Recognition

    DEFF Research Database (Denmark)

    Kraljevski, Ivan; Tan, Zheng-Hua

    2014-01-01

    This paper addresses the issue of data compression in distributed speech recognition on the basis of a variable frame rate and length analysis method. The method first conducts frame selection by using a posteriori signal-to-noise ratio weighted energy distance to find the right time resolution...... length for steady regions. The method is applied to scalable source coding in distributed speech recognition where the target bitrate is met by adjusting the frame rate. Speech recognition results show that the proposed approach outperforms other compression methods in terms of recognition accuracy...... for noisy speech while achieving higher compression rates....

  20. Human speech articulator measurements using low power, 2GHz Homodyne sensors

    International Nuclear Information System (INIS)

    Barnes, T; Burnett, G C; Holzrichter, J F

    1999-01-01

    Very low power, short-range microwave ''radar-like'' sensors can measure the motions and vibrations of internal human speech articulators as speech is produced. In these animate (and also in inanimate acoustic systems) microwave sensors can measure vibration information associated with excitation sources and other interfaces. These data, together with the corresponding acoustic data, enable the calculation of system transfer functions. This information appears to be useful for a surprisingly wide range of applications such as speech coding and recognition, speaker or object identification, speech and musical instrument synthesis, noise cancellation, and other applications

  1. Human speech articulator measurements using low power, 2GHz Homodyne sensors

    Energy Technology Data Exchange (ETDEWEB)

    Barnes, T; Burnett, G C; Holzrichter, J F

    1999-06-29

    Very low power, short-range microwave ''radar-like'' sensors can measure the motions and vibrations of internal human speech articulators as speech is produced. In these animate (and also in inanimate acoustic systems) microwave sensors can measure vibration information associated with excitation sources and other interfaces. These data, together with the corresponding acoustic data, enable the calculation of system transfer functions. This information appears to be useful for a surprisingly wide range of applications such as speech coding and recognition, speaker or object identification, speech and musical instrument synthesis, noise cancellation, and other applications.

  2. Tackling the complexity in speech

    DEFF Research Database (Denmark)

    section includes four carefully selected chapters. They deal with facets of speech production, speech acoustics, and/or speech perception or recognition, place them in an integrated phonetic-phonological perspective, and relate them in more or less explicit ways to aspects of speech technology. Therefore......, we hope that this volume can help speech scientists with traditional training in phonetics and phonology to keep up with the latest developments in speech technology. In the opposite direction, speech researchers starting from a technological perspective will hopefully get inspired by reading about...... the questions, phenomena, and communicative functions that are currently addressed in phonetics and phonology. Either way, the future of speech research lies in international, interdisciplinary collaborations, and our volume is meant to reflect and facilitate such collaborations...

  3. Model-based machine learning.

    Science.gov (United States)

    Bishop, Christopher M

    2013-02-13

    Several decades of research in the field of machine learning have resulted in a multitude of different algorithms for solving a broad range of problems. To tackle a new application, a researcher typically tries to map their problem onto one of these existing methods, often influenced by their familiarity with specific algorithms and by the availability of corresponding software implementations. In this study, we describe an alternative methodology for applying machine learning, in which a bespoke solution is formulated for each new application. The solution is expressed through a compact modelling language, and the corresponding custom machine learning code is then generated automatically. This model-based approach offers several major advantages, including the opportunity to create highly tailored models for specific scenarios, as well as rapid prototyping and comparison of a range of alternative models. Furthermore, newcomers to the field of machine learning do not have to learn about the huge range of traditional methods, but instead can focus their attention on understanding a single modelling environment. In this study, we show how probabilistic graphical models, coupled with efficient inference algorithms, provide a very flexible foundation for model-based machine learning, and we outline a large-scale commercial application of this framework involving tens of millions of users. We also describe the concept of probabilistic programming as a powerful software environment for model-based machine learning, and we discuss a specific probabilistic programming language called Infer.NET, which has been widely used in practical applications.

  4. Electricity of machine tool

    International Nuclear Information System (INIS)

    Gijeon media editorial department

    1977-10-01

    This book is divided into three parts. The first part deals with electricity machine, which can taints from generator to motor, motor a power source of machine tool, electricity machine for machine tool such as switch in main circuit, automatic machine, a knife switch and pushing button, snap switch, protection device, timer, solenoid, and rectifier. The second part handles wiring diagram. This concludes basic electricity circuit of machine tool, electricity wiring diagram in your machine like milling machine, planer and grinding machine. The third part introduces fault diagnosis of machine, which gives the practical solution according to fault diagnosis and the diagnostic method with voltage and resistance measurement by tester.

  5. Environmentally Friendly Machining

    CERN Document Server

    Dixit, U S; Davim, J Paulo

    2012-01-01

    Environment-Friendly Machining provides an in-depth overview of environmentally-friendly machining processes, covering numerous different types of machining in order to identify which practice is the most environmentally sustainable. The book discusses three systems at length: machining with minimal cutting fluid, air-cooled machining and dry machining. Also covered is a way to conserve energy during machining processes, along with useful data and detailed descriptions for developing and utilizing the most efficient modern machining tools. Researchers and engineers looking for sustainable machining solutions will find Environment-Friendly Machining to be a useful volume.

  6. Machine Protection

    CERN Document Server

    Schmidt, R

    2014-01-01

    The protection of accelerator equipment is as old as accelerator technology and was for many years related to high-power equipment. Examples are the protection of powering equipment from overheating (magnets, power converters, high-current cables), of superconducting magnets from damage after a quench and of klystrons. The protection of equipment from beam accidents is more recent. It is related to the increasing beam power of high-power proton accelerators such as ISIS, SNS, ESS and the PSI cyclotron, to the emission of synchrotron light by electron–positron accelerators and FELs, and to the increase of energy stored in the beam (in particular for hadron colliders such as LHC). Designing a machine protection system requires an excellent understanding of accelerator physics and operation to anticipate possible failures that could lead to damage. Machine protection includes beam and equipment monitoring, a system to safely stop beam operation (e.g. dumping the beam or stopping the beam at low energy) and an ...

  7. Innovative Speech Reconstructive Surgery

    OpenAIRE

    Hashem Shemshadi

    2003-01-01

    Proper speech functioning in human being, depends on the precise coordination and timing balances in a series of complex neuro nuscular movements and actions. Starting from the prime organ of energy source of expelled air from respirato y system; deliver such air to trigger vocal cords; swift changes of this phonatory episode to a comprehensible sound in RESONACE and final coordination of all head and neck structures to elicit final speech in ...

  8. The chairman's speech

    International Nuclear Information System (INIS)

    Allen, A.M.

    1986-01-01

    The paper contains a transcript of a speech by the chairman of the UKAEA, to mark the publication of the 1985/6 annual report. The topics discussed in the speech include: the Chernobyl accident and its effect on public attitudes to nuclear power, management and disposal of radioactive waste, the operation of UKAEA as a trading fund, and the UKAEA development programmes. The development programmes include work on the following: fast reactor technology, thermal reactors, reactor safety, health and safety aspects of water cooled reactors, the Joint European Torus, and under-lying research. (U.K.)

  9. Visualizing structures of speech expressiveness

    DEFF Research Database (Denmark)

    Herbelin, Bruno; Jensen, Karl Kristoffer; Graugaard, Lars

    2008-01-01

    Speech is both beautiful and informative. In this work, a conceptual study of the speech, through investigation of the tower of Babel, the archetypal phonemes, and a study of the reasons of uses of language is undertaken in order to create an artistic work investigating the nature of speech. The ....... The artwork is presented at the Re:New festival in May 2008....

  10. Social Robotics in Therapy of Apraxia of Speech

    Directory of Open Access Journals (Sweden)

    José Carlos Castillo

    2018-01-01

    Full Text Available Apraxia of speech is a motor speech disorder in which messages from the brain to the mouth are disrupted, resulting in an inability for moving lips or tongue to the right place to pronounce sounds correctly. Current therapies for this condition involve a therapist that in one-on-one sessions conducts the exercises. Our aim is to work in the line of robotic therapies in which a robot is able to perform partially or autonomously a therapy session, endowing a social robot with the ability of assisting therapists in apraxia of speech rehabilitation exercises. Therefore, we integrate computer vision and machine learning techniques to detect the mouth pose of the user and, on top of that, our social robot performs autonomously the different steps of the therapy using multimodal interaction.

  11. Normativity in 18th century discourse on speech.

    Science.gov (United States)

    MacNamee, T

    1984-11-01

    Eighteenth century phoneticians, such as Dodart, Ferrein, and Hellwag, extended the taxonomy of visible articulatory processes into the realm of the invisible, notably with the exploration of the voicing mechanism. Remedial initiatives were not simply confined to consideration of the outward manifestations of speech and its disorders: The work of Haller, Kuestner, and Morgagni shows an acute awareness of the nervous organization underlying verbal behavior. There was a characteristic preoccupation with mechanical models of speech, which led to the attempts of Kempelen and other investigators to construct actual "speaking machines." Eighteenth century scholars regarded language as not only an innate capacity peculiar to human nature, but also as a bodily habit learned by experience. The function of the orthoepist was to teach the right speech habits, and the upward mobility of the bourgeoisie created a demand for his services.

  12. Coding Partitions

    Directory of Open Access Journals (Sweden)

    Fabio Burderi

    2007-05-01

    Full Text Available Motivated by the study of decipherability conditions for codes weaker than Unique Decipherability (UD, we introduce the notion of coding partition. Such a notion generalizes that of UD code and, for codes that are not UD, allows to recover the ``unique decipherability" at the level of the classes of the partition. By tacking into account the natural order between the partitions, we define the characteristic partition of a code X as the finest coding partition of X. This leads to introduce the canonical decomposition of a code in at most one unambiguouscomponent and other (if any totally ambiguouscomponents. In the case the code is finite, we give an algorithm for computing its canonical partition. This, in particular, allows to decide whether a given partition of a finite code X is a coding partition. This last problem is then approached in the case the code is a rational set. We prove its decidability under the hypothesis that the partition contains a finite number of classes and each class is a rational set. Moreover we conjecture that the canonical partition satisfies such a hypothesis. Finally we consider also some relationships between coding partitions and varieties of codes.

  13. Reviewing the connection between speech and obstructive sleep apnea.

    Science.gov (United States)

    Espinoza-Cuadros, Fernando; Fernández-Pozo, Rubén; Toledano, Doroteo T; Alcázar-Ramírez, José D; López-Gonzalo, Eduardo; Hernández-Gómez, Luis A

    2016-02-20

    Sleep apnea (OSA) is a common sleep disorder characterized by recurring breathing pauses during sleep caused by a blockage of the upper airway (UA). The altered UA structure or function in OSA speakers has led to hypothesize the automatic analysis of speech for OSA assessment. In this paper we critically review several approaches using speech analysis and machine learning techniques for OSA detection, and discuss the limitations that can arise when using machine learning techniques for diagnostic applications. A large speech database including 426 male Spanish speakers suspected to suffer OSA and derived to a sleep disorders unit was used to study the clinical validity of several proposals using machine learning techniques to predict the apnea-hypopnea index (AHI) or classify individuals according to their OSA severity. AHI describes the severity of patients' condition. We first evaluate AHI prediction using state-of-the-art speaker recognition technologies: speech spectral information is modelled using supervectors or i-vectors techniques, and AHI is predicted through support vector regression (SVR). Using the same database we then critically review several OSA classification approaches previously proposed. The influence and possible interference of other clinical variables or characteristics available for our OSA population: age, height, weight, body mass index, and cervical perimeter, are also studied. The poor results obtained when estimating AHI using supervectors or i-vectors followed by SVR contrast with the positive results reported by previous research. This fact prompted us to a careful review of these approaches, also testing some reported results over our database. Several methodological limitations and deficiencies were detected that may have led to overoptimistic results. The methodological deficiencies observed after critically reviewing previous research can be relevant examples of potential pitfalls when using machine learning techniques for

  14. Analysis of machining and machine tools

    CERN Document Server

    Liang, Steven Y

    2016-01-01

    This book delivers the fundamental science and mechanics of machining and machine tools by presenting systematic and quantitative knowledge in the form of process mechanics and physics. It gives readers a solid command of machining science and engineering, and familiarizes them with the geometry and functionality requirements of creating parts and components in today’s markets. The authors address traditional machining topics, such as: single and multiple point cutting processes grinding components accuracy and metrology shear stress in cutting cutting temperature and analysis chatter They also address non-traditional machining, such as: electrical discharge machining electrochemical machining laser and electron beam machining A chapter on biomedical machining is also included. This book is appropriate for advanced undergraduate and graduate mechani cal engineering students, manufacturing engineers, and researchers. Each chapter contains examples, exercises and their solutions, and homework problems that re...

  15. Workshop: Welcoming speech

    International Nuclear Information System (INIS)

    Lummerzheim, D.

    1994-01-01

    The welcoming speech underlines the fact that any validation process starting with calculation methods and ending with studies on the long-term behaviour of a repository system can only be effected through laboratory, field and natural-analogue studies. The use of natural analogues (NA) is to secure the biosphere and to verify whether this safety really exists. (HP) [de

  16. Hearing speech in music.

    Science.gov (United States)

    Ekström, Seth-Reino; Borg, Erik

    2011-01-01

    The masking effect of a piano composition, played at different speeds and in different octaves, on speech-perception thresholds was investigated in 15 normal-hearing and 14 moderately-hearing-impaired subjects. Running speech (just follow conversation, JFC) testing and use of hearing aids increased the everyday validity of the findings. A comparison was made with standard audiometric noises [International Collegium of Rehabilitative Audiology (ICRA) noise and speech spectrum-filtered noise (SPN)]. All masking sounds, music or noise, were presented at the same equivalent sound level (50 dBA). The results showed a significant effect of piano performance speed and octave (Ptempo had the largest effect; and high octave and slow tempo, the smallest. Music had a lower masking effect than did ICRA noise with two or six speakers at normal vocal effort (Pmusic offers an interesting opportunity for studying masking under realistic conditions, where spectral and temporal features can be varied independently. The results have implications for composing music with vocal parts, designing acoustic environments and creating a balance between speech perception and privacy in social settings.

  17. Hearing speech in music

    Directory of Open Access Journals (Sweden)

    Seth-Reino Ekström

    2011-01-01

    Full Text Available The masking effect of a piano composition, played at different speeds and in different octaves, on speech-perception thresholds was investigated in 15 normal-hearing and 14 moderately-hearing-impaired subjects. Running speech (just follow conversation, JFC testing and use of hearing aids increased the everyday validity of the findings. A comparison was made with standard audiometric noises [International Collegium of Rehabilitative Audiology (ICRA noise and speech spectrum-filtered noise (SPN]. All masking sounds, music or noise, were presented at the same equivalent sound level (50 dBA. The results showed a significant effect of piano performance speed and octave (P<.01. Low octave and fast tempo had the largest effect; and high octave and slow tempo, the smallest. Music had a lower masking effect than did ICRA noise with two or six speakers at normal vocal effort (P<.01 and SPN (P<.05. Subjects with hearing loss had higher masked thresholds than the normal-hearing subjects (P<.01, but there were smaller differences between masking conditions (P<.01. It is pointed out that music offers an interesting opportunity for studying masking under realistic conditions, where spectral and temporal features can be varied independently. The results have implications for composing music with vocal parts, designing acoustic environments and creating a balance between speech perception and privacy in social settings.

  18. Free Speech Yearbook 1979.

    Science.gov (United States)

    Kane, Peter E., Ed.

    The seven articles in this collection deal with theoretical and practical freedom of speech issues. Topics covered are: the United States Supreme Court, motion picture censorship, and the color line; judicial decision making; the established scientific community's suppression of the ideas of Immanuel Velikovsky; the problems of avant-garde jazz,…

  19. Kurzweil Reading Machine: A Partial Evaluation of Its Optical Character Recognition Error Rate.

    Science.gov (United States)

    Goodrich, Gregory L.; And Others

    1979-01-01

    A study designed to assess the ability of the Kurzweil reading machine (a speech reading device for the visually handicapped) to read three different type styles produced by five different means indicated that the machines tested had different error rates depending upon the means of producing the copy and upon the type style used. (Author/CL)

  20. Machine Protection

    International Nuclear Information System (INIS)

    Schmidt, R

    2014-01-01

    The protection of accelerator equipment is as old as accelerator technology and was for many years related to high-power equipment. Examples are the protection of powering equipment from overheating (magnets, power converters, high-current cables), of superconducting magnets from damage after a quench and of klystrons. The protection of equipment from beam accidents is more recent. It is related to the increasing beam power of high-power proton accelerators such as ISIS, SNS, ESS and the PSI cyclotron, to the emission of synchrotron light by electron–positron accelerators and FELs, and to the increase of energy stored in the beam (in particular for hadron colliders such as LHC). Designing a machine protection system requires an excellent understanding of accelerator physics and operation to anticipate possible failures that could lead to damage. Machine protection includes beam and equipment monitoring, a system to safely stop beam operation (e.g. dumping the beam or stopping the beam at low energy) and an interlock system providing the glue between these systems. The most recent accelerator, the LHC, will operate with about 3 × 10 14 protons per beam, corresponding to an energy stored in each beam of 360 MJ. This energy can cause massive damage to accelerator equipment in case of uncontrolled beam loss, and a single accident damaging vital parts of the accelerator could interrupt operation for years. This article provides an overview of the requirements for protection of accelerator equipment and introduces the various protection systems. Examples are mainly from LHC, SNS and ESS

  1. Automated analysis of free speech predicts psychosis onset in high-risk youths

    Science.gov (United States)

    Bedi, Gillinder; Carrillo, Facundo; Cecchi, Guillermo A; Slezak, Diego Fernández; Sigman, Mariano; Mota, Natália B; Ribeiro, Sidarta; Javitt, Daniel C; Copelli, Mauro; Corcoran, Cheryl M

    2015-01-01

    Background/Objectives: Psychiatry lacks the objective clinical tests routinely used in other specializations. Novel computerized methods to characterize complex behaviors such as speech could be used to identify and predict psychiatric illness in individuals. AIMS: In this proof-of-principle study, our aim was to test automated speech analyses combined with Machine Learning to predict later psychosis onset in youths at clinical high-risk (CHR) for psychosis. Methods: Thirty-four CHR youths (11 females) had baseline interviews and were assessed quarterly for up to 2.5 years; five transitioned to psychosis. Using automated analysis, transcripts of interviews were evaluated for semantic and syntactic features predicting later psychosis onset. Speech features were fed into a convex hull classification algorithm with leave-one-subject-out cross-validation to assess their predictive value for psychosis outcome. The canonical correlation between the speech features and prodromal symptom ratings was computed. Results: Derived speech features included a Latent Semantic Analysis measure of semantic coherence and two syntactic markers of speech complexity: maximum phrase length and use of determiners (e.g., which). These speech features predicted later psychosis development with 100% accuracy, outperforming classification from clinical interviews. Speech features were significantly correlated with prodromal symptoms. Conclusions: Findings support the utility of automated speech analysis to measure subtle, clinically relevant mental state changes in emergent psychosis. Recent developments in computer science, including natural language processing, could provide the foundation for future development of objective clinical tests for psychiatry. PMID:27336038

  2. What makes an automated teller machine usable by blind users?

    Science.gov (United States)

    Manzke, J M; Egan, D H; Felix, D; Krueger, H

    1998-07-01

    Fifteen blind and sighted subjects, who featured as a control group for acceptance, were asked for their requirements for automated teller machines (ATMs). Both groups also tested the usability of a partially operational ATM mock-up. This machine was based on an existing cash dispenser, providing natural speech output, different function menus and different key arrangements. Performance and subjective evaluation data of blind and sighted subjects were collected. All blind subjects were able to operate the ATM successfully. The implemented speech output was the main usability factor for them. The different interface designs did not significantly affect performance and subjective evaluation. Nevertheless, design recommendations can be derived from the requirement assessment. The sighted subjects were rather open for design modifications, especially the implementation of speech output. However, there was also a mismatch of the requirements of the two subject groups, mainly concerning the key arrangement.

  3. Metallizing of machinable glass ceramic

    International Nuclear Information System (INIS)

    Seigal, P.K.

    1976-02-01

    A satisfactory technique has been developed for metallizing Corning (Code 9658) machinable glass ceramic for brazing. Analyses of several bonding materials suitable for metallizing were made using microprobe analysis, optical metallography, and tensile strength tests. The effect of different cleaning techniques on the microstructure and the effect of various firing temperatures on the bonding interface were also investigated. A nickel paste, used for thick-film application, has been applied to obtain braze joints with strength in excess of 2000 psi

  4. Current trends in small vocabulary speech recognition for equipment control

    Science.gov (United States)

    Doukas, Nikolaos; Bardis, Nikolaos G.

    2017-09-01

    Speech recognition systems allow human - machine communication to acquire an intuitive nature that approaches the simplicity of inter - human communication. Small vocabulary speech recognition is a subset of the overall speech recognition problem, where only a small number of words need to be recognized. Speaker independent small vocabulary recognition can find significant applications in field equipment used by military personnel. Such equipment may typically be controlled by a small number of commands that need to be given quickly and accurately, under conditions where delicate manual operations are difficult to achieve. This type of application could hence significantly benefit by the use of robust voice operated control components, as they would facilitate the interaction with their users and render it much more reliable in times of crisis. This paper presents current challenges involved in attaining efficient and robust small vocabulary speech recognition. These challenges concern feature selection, classification techniques, speaker diversity and noise effects. A state machine approach is presented that facilitates the voice guidance of different equipment in a variety of situations.

  5. Nobel peace speech

    Directory of Open Access Journals (Sweden)

    Joshua FRYE

    2017-07-01

    Full Text Available The Nobel Peace Prize has long been considered the premier peace prize in the world. According to Geir Lundestad, Secretary of the Nobel Committee, of the 300 some peace prizes awarded worldwide, “none is in any way as well known and as highly respected as the Nobel Peace Prize” (Lundestad, 2001. Nobel peace speech is a unique and significant international site of public discourse committed to articulating the universal grammar of peace. Spanning over 100 years of sociopolitical history on the world stage, Nobel Peace Laureates richly represent an important cross-section of domestic and international issues increasingly germane to many publics. Communication scholars’ interest in this rhetorical genre has increased in the past decade. Yet, the norm has been to analyze a single speech artifact from a prestigious or controversial winner rather than examine the collection of speeches for generic commonalities of import. In this essay, we analyze the discourse of Nobel peace speech inductively and argue that the organizing principle of the Nobel peace speech genre is the repetitive form of normative liberal principles and values that function as rhetorical topoi. These topoi include freedom and justice and appeal to the inviolable, inborn right of human beings to exercise certain political and civil liberties and the expectation of equality of protection from totalitarian and tyrannical abuses. The significance of this essay to contemporary communication theory is to expand our theoretical understanding of rhetoric’s role in the maintenance and development of an international and cross-cultural vocabulary for the grammar of peace.

  6. Oscillatory Brain Responses Reflect Anticipation during Comprehension of Speech Acts in Spoken Dialog

    Directory of Open Access Journals (Sweden)

    Rosa S. Gisladottir

    2018-02-01

    Full Text Available Everyday conversation requires listeners to quickly recognize verbal actions, so-called speech acts, from the underspecified linguistic code and prepare a relevant response within the tight time constraints of turn-taking. The goal of this study was to determine the time-course of speech act recognition by investigating oscillatory EEG activity during comprehension of spoken dialog. Participants listened to short, spoken dialogs with target utterances that delivered three distinct speech acts (Answers, Declinations, Pre-offers. The targets were identical across conditions at lexico-syntactic and phonetic/prosodic levels but differed in the pragmatic interpretation of the speech act performed. Speech act comprehension was associated with reduced power in the alpha/beta bands just prior to Declination speech acts, relative to Answers and Pre-offers. In addition, we observed reduced power in the theta band during the beginning of Declinations, relative to Answers. Based on the role of alpha and beta desynchronization in anticipatory processes, the results are taken to indicate that anticipation plays a role in speech act recognition. Anticipation of speech acts could be critical for efficient turn-taking, allowing interactants to quickly recognize speech acts and respond within the tight time frame characteristic of conversation. The results show that anticipatory processes can be triggered by the characteristics of the interaction, including the speech act type.

  7. Speech reconstruction using a deep partially supervised neural network.

    Science.gov (United States)

    McLoughlin, Ian; Li, Jingjie; Song, Yan; Sharifzadeh, Hamid R

    2017-08-01

    Statistical speech reconstruction for larynx-related dysphonia has achieved good performance using Gaussian mixture models and, more recently, restricted Boltzmann machine arrays; however, deep neural network (DNN)-based systems have been hampered by the limited amount of training data available from individual voice-loss patients. The authors propose a novel DNN structure that allows a partially supervised training approach on spectral features from smaller data sets, yielding very good results compared with the current state-of-the-art.

  8. Machine terms dictionary

    Energy Technology Data Exchange (ETDEWEB)

    NONE

    1979-04-15

    This book gives descriptions of machine terms which includes machine design, drawing, the method of machine, machine tools, machine materials, automobile, measuring and controlling, electricity, basic of electron, information technology, quality assurance, Auto CAD and FA terms and important formula of mechanical engineering.

  9. Metaheuristic applications to speech enhancement

    CERN Document Server

    Kunche, Prajna

    2016-01-01

    This book serves as a basic reference for those interested in the application of metaheuristics to speech enhancement. The major goal of the book is to explain the basic concepts of optimization methods and their use in heuristic optimization in speech enhancement to scientists, practicing engineers, and academic researchers in speech processing. The authors discuss why it has been a challenging problem for researchers to develop new enhancement algorithms that aid in the quality and intelligibility of degraded speech. They present powerful optimization methods to speech enhancement that can help to solve the noise reduction problems. Readers will be able to understand the fundamentals of speech processing as well as the optimization techniques, how the speech enhancement algorithms are implemented by utilizing optimization methods, and will be given the tools to develop new algorithms. The authors also provide a comprehensive literature survey regarding the topic.

  10. An introduction to quantum machine learning

    Science.gov (United States)

    Schuld, Maria; Sinayskiy, Ilya; Petruccione, Francesco

    2015-04-01

    Machine learning algorithms learn a desired input-output relation from examples in order to interpret new inputs. This is important for tasks such as image and speech recognition or strategy optimisation, with growing applications in the IT industry. In the last couple of years, researchers investigated if quantum computing can help to improve classical machine learning algorithms. Ideas range from running computationally costly algorithms or their subroutines efficiently on a quantum computer to the translation of stochastic methods into the language of quantum theory. This contribution gives a systematic overview of the emerging field of quantum machine learning. It presents the approaches as well as technical details in an accessible way, and discusses the potential of a future theory of quantum learning.

  11. Probability Machines: Consistent Probability Estimation Using Nonparametric Learning Machines

    Science.gov (United States)

    Malley, J. D.; Kruppa, J.; Dasgupta, A.; Malley, K. G.; Ziegler, A.

    2011-01-01

    Summary Background Most machine learning approaches only provide a classification for binary responses. However, probabilities are required for risk estimation using individual patient characteristics. It has been shown recently that every statistical learning machine known to be consistent for a nonparametric regression problem is a probability machine that is provably consistent for this estimation problem. Objectives The aim of this paper is to show how random forests and nearest neighbors can be used for consistent estimation of individual probabilities. Methods Two random forest algorithms and two nearest neighbor algorithms are described in detail for estimation of individual probabilities. We discuss the consistency of random forests, nearest neighbors and other learning machines in detail. We conduct a simulation study to illustrate the validity of the methods. We exemplify the algorithms by analyzing two well-known data sets on the diagnosis of appendicitis and the diagnosis of diabetes in Pima Indians. Results Simulations demonstrate the validity of the method. With the real data application, we show the accuracy and practicality of this approach. We provide sample code from R packages in which the probability estimation is already available. This means that all calculations can be performed using existing software. Conclusions Random forest algorithms as well as nearest neighbor approaches are valid machine learning methods for estimating individual probabilities for binary responses. Freely available implementations are available in R and may be used for applications. PMID:21915433

  12. Digitised evaluation of speech intelligibility using vowels in maxillectomy patients.

    Science.gov (United States)

    Sumita, Y I; Hattori, M; Murase, M; Elbashti, M E; Taniguchi, H

    2018-03-01

    Among the functional disabilities that patients face following maxillectomy, speech impairment is a major factor influencing quality of life. Proper rehabilitation of speech, which may include prosthodontic and surgical treatments and speech therapy, requires accurate evaluation of speech intelligibility (SI). A simple, less time-consuming yet accurate evaluation is desirable both for maxillectomy patients and the various clinicians providing maxillofacial treatment. This study sought to determine the utility of digital acoustic analysis of vowels for the prediction of SI in maxillectomy patients, based on a comprehensive understanding of speech production in the vocal tract of maxillectomy patients and its perception. Speech samples were collected from 33 male maxillectomy patients (mean age 57.4 years) in two conditions, without and with a maxillofacial prosthesis, and formant data for the vowels /a/,/e/,/i/,/o/, and /u/ were calculated based on linear predictive coding. The frequency range of formant 2 (F2) was determined by differences between the minimum and maximum frequency. An SI test was also conducted to reveal the relationship between SI score and F2 range. Statistical analyses were applied. F2 range and SI score were significantly different between the two conditions without and with a prosthesis (both P maxillectomy. © 2017 John Wiley & Sons Ltd.

  13. FCG: a code generator for lazy functional languages

    NARCIS (Netherlands)

    Kastens, U.; Langendoen, K.G.; Hartel, Pieter H.; Pfahler, P.

    1992-01-01

    The FCGcode generator produces portable code that supports efficient two-space copying garbage collection. The code generator transforms the output of the FAST compiler front end into an abstract machine code. This code explicitly uses a call stack, which is accessible to the garbage collector. In

  14. Improved part-of-speech prediction in suffix analysis.

    Directory of Open Access Journals (Sweden)

    Mario Fruzangohar

    Full Text Available MOTIVATION: Predicting the part of speech (POS tag of an unknown word in a sentence is a significant challenge. This is particularly difficult in biomedicine, where POS tags serve as an input to training sophisticated literature summarization techniques, such as those based on Hidden Markov Models (HMM. Different approaches have been taken to deal with the POS tagger challenge, but with one exception--the TnT POS tagger--previous publications on POS tagging have omitted details of the suffix analysis used for handling unknown words. The suffix of an English word is a strong predictor of a POS tag for that word. As a pre-requisite for an accurate HMM POS tagger for biomedical publications, we present an efficient suffix prediction method for integration into a POS tagger. RESULTS: We have implemented a fully functional HMM POS tagger using experimentally optimised suffix based prediction. Our simple suffix analysis method, significantly outperformed the probability interpolation based TnT method. We have also shown how important suffix analysis can be for probability estimation of a known word (in the training corpus with an unseen POS tag; a common scenario with a small training corpus. We then integrated this simple method in our POS tagger and determined an optimised parameter set for both methods, which can help developers to optimise their current algorithm, based on our results. We also introduce the concept of counting methods in maximum likelihood estimation for the first time and show how counting methods can affect the prediction result. Finally, we describe how machine-learning techniques were applied to identify words, for which prediction of POS tags were always incorrect and propose a method to handle words of this type. AVAILABILITY AND IMPLEMENTATION: Java source code, binaries and setup instructions are freely available at http://genomes.sapac.edu.au/text_mining/pos_tagger.zip.

  15. Python for probability, statistics, and machine learning

    CERN Document Server

    Unpingco, José

    2016-01-01

    This book covers the key ideas that link probability, statistics, and machine learning illustrated using Python modules in these areas. The entire text, including all the figures and numerical results, is reproducible using the Python codes and their associated Jupyter/IPython notebooks, which are provided as supplementary downloads. The author develops key intuitions in machine learning by working meaningful examples using multiple analytical methods and Python codes, thereby connecting theoretical concepts to concrete implementations. Modern Python modules like Pandas, Sympy, and Scikit-learn are applied to simulate and visualize important machine learning concepts like the bias/variance trade-off, cross-validation, and regularization. Many abstract mathematical ideas, such as convergence in probability theory, are developed and illustrated with numerical examples. This book is suitable for anyone with an undergraduate-level exposure to probability, statistics, or machine learning and with rudimentary knowl...

  16. Machine learning topological states

    Science.gov (United States)

    Deng, Dong-Ling; Li, Xiaopeng; Das Sarma, S.

    2017-11-01

    Artificial neural networks and machine learning have now reached a new era after several decades of improvement where applications are to explode in many fields of science, industry, and technology. Here, we use artificial neural networks to study an intriguing phenomenon in quantum physics—the topological phases of matter. We find that certain topological states, either symmetry-protected or with intrinsic topological order, can be represented with classical artificial neural networks. This is demonstrated by using three concrete spin systems, the one-dimensional (1D) symmetry-protected topological cluster state and the 2D and 3D toric code states with intrinsic topological orders. For all three cases, we show rigorously that the topological ground states can be represented by short-range neural networks in an exact and efficient fashion—the required number of hidden neurons is as small as the number of physical spins and the number of parameters scales only linearly with the system size. For the 2D toric-code model, we find that the proposed short-range neural networks can describe the excited states with Abelian anyons and their nontrivial mutual statistics as well. In addition, by using reinforcement learning we show that neural networks are capable of finding the topological ground states of nonintegrable Hamiltonians with strong interactions and studying their topological phase transitions. Our results demonstrate explicitly the exceptional power of neural networks in describing topological quantum states, and at the same time provide valuable guidance to machine learning of topological phases in generic lattice models.

  17. A qualitative analysis of hate speech reported to the Romanian National Council for Combating Discrimination (2003‑2015)

    OpenAIRE

    Adriana Iordache

    2015-01-01

    The article analyzes the specificities of Romanian hate speech over a period of twelve years through a qualitative analysis of 384 Decisions of the National Council for Combating Discrimination. The study employs a coding methodology which allows one to separate decisions according to the group that was the victim of hate speech. The article finds that stereotypes employed are similar to those encountered in the international literature. The main target of hate speech is the Roma, who are ...

  18. Automatic Speech Acquisition and Recognition for Spacesuit Audio Systems

    Science.gov (United States)

    Ye, Sherry

    2015-01-01

    NASA has a widely recognized but unmet need for novel human-machine interface technologies that can facilitate communication during astronaut extravehicular activities (EVAs), when loud noises and strong reverberations inside spacesuits make communication challenging. WeVoice, Inc., has developed a multichannel signal-processing method for speech acquisition in noisy and reverberant environments that enables automatic speech recognition (ASR) technology inside spacesuits. The technology reduces noise by exploiting differences between the statistical nature of signals (i.e., speech) and noise that exists in the spatial and temporal domains. As a result, ASR accuracy can be improved to the level at which crewmembers will find the speech interface useful. System components and features include beam forming/multichannel noise reduction, single-channel noise reduction, speech feature extraction, feature transformation and normalization, feature compression, and ASR decoding. Arithmetic complexity models were developed and will help designers of real-time ASR systems select proper tasks when confronted with constraints in computational resources. In Phase I of the project, WeVoice validated the technology. The company further refined the technology in Phase II and developed a prototype for testing and use by suited astronauts.

  19. Voice Activity Detection Using Fuzzy Entropy and Support Vector Machine

    Directory of Open Access Journals (Sweden)

    R. Johny Elton

    2016-08-01

    Full Text Available This paper proposes support vector machine (SVM based voice activity detection using FuzzyEn to improve detection performance under noisy conditions. The proposed voice activity detection (VAD uses fuzzy entropy (FuzzyEn as a feature extracted from noise-reduced speech signals to train an SVM model for speech/non-speech classification. The proposed VAD method was tested by conducting various experiments by adding real background noises of different signal-to-noise ratios (SNR ranging from −10 dB to 10 dB to actual speech signals collected from the TIMIT database. The analysis proves that FuzzyEn feature shows better results in discriminating noise and corrupted noisy speech. The efficacy of the SVM classifier was validated using 10-fold cross validation. Furthermore, the results obtained by the proposed method was compared with those of previous standardized VAD algorithms as well as recently developed methods. Performance comparison suggests that the proposed method is proven to be more efficient in detecting speech under various noisy environments with an accuracy of 93.29%, and the FuzzyEn feature detects speech efficiently even at low SNR levels.

  20. Speech Motor Control in Fluent and Dysfluent Speech Production of an Individual with Apraxia of Speech and Broca's Aphasia

    Science.gov (United States)

    van Lieshout, Pascal H. H. M.; Bose, Arpita; Square, Paula A.; Steele, Catriona M.

    2007-01-01

    Apraxia of speech (AOS) is typically described as a motor-speech disorder with clinically well-defined symptoms, but without a clear understanding of the underlying problems in motor control. A number of studies have compared the speech of subjects with AOS to the fluent speech of controls, but only a few have included speech movement data and if…

  1. Addiction Machines

    Directory of Open Access Journals (Sweden)

    James Godley

    2011-10-01

    Full Text Available Entry into the crypt William Burroughs shared with his mother opened and shut around a failed re-enactment of William Tell’s shot through the prop placed upon a loved one’s head. The accidental killing of his wife Joan completed the installation of the addictation machine that spun melancholia as manic dissemination. An early encryptment to which was added the audio portion of abuse deposited an undeliverable message in WB. Wil- liam could never tell, although his corpus bears the in- scription of this impossibility as another form of pos- sibility. James Godley is currently a doctoral candidate in Eng- lish at SUNY Buffalo, where he studies psychoanalysis, Continental philosophy, and nineteenth-century litera- ture and poetry (British and American. His work on the concept of mourning and “the dead” in Freudian and Lacanian approaches to psychoanalytic thought and in Gothic literature has also spawned an essay on zombie porn. Since entering the Academy of Fine Arts Karlsruhe in 2007, Valentin Hennig has studied in the classes of Sil- via Bächli, Claudio Moser, and Corinne Wasmuht. In 2010 he spent a semester at the Dresden Academy of Fine Arts. His work has been shown in group exhibi- tions in Freiburg and Karlsruhe.

  2. Machine musicianship

    Science.gov (United States)

    Rowe, Robert

    2002-05-01

    The training of musicians begins by teaching basic musical concepts, a collection of knowledge commonly known as musicianship. Computer programs designed to implement musical skills (e.g., to make sense of what they hear, perform music expressively, or compose convincing pieces) can similarly benefit from access to a fundamental level of musicianship. Recent research in music cognition, artificial intelligence, and music theory has produced a repertoire of techniques that can make the behavior of computer programs more musical. Many of these were presented in a recently published book/CD-ROM entitled Machine Musicianship. For use in interactive music systems, we are interested in those which are fast enough to run in real time and that need only make reference to the material as it appears in sequence. This talk will review several applications that are able to identify the tonal center of musical material during performance. Beyond this specific task, the design of real-time algorithmic listening through the concurrent operation of several connected analyzers is examined. The presentation includes discussion of a library of C++ objects that can be combined to perform interactive listening and a demonstration of their capability.

  3. Speech is Golden

    DEFF Research Database (Denmark)

    Juel Henrichsen, Peter

    2014-01-01

    on the supply side. The present article reports on a new public action strategy which has taken shape in the course of 2013-14. While Denmark is a small language area, our public sector is well organised and has considerable purchasing power. Across this past year, Danish local authorities have organised around......Most of the Danish municipalities are ready to begin to adopt automatic speech recognition, but at the same time remain nervous following a long series of bad business cases in the recent past. Complaints are voiced over costly licences and low service levels, typical effects of a de facto monopoly...... the speech technology challenge, they have formulated a number of joint questions and new requirements to be met by suppliers and have deliberately worked towards formulating tendering material which will allow fair competition. Public researchers have contributed to this work, including the author...

  4. Trunnion Collar Removal Machine - Gap Analysis Table

    International Nuclear Information System (INIS)

    Johnson, M.

    2005-01-01

    The purpose of this document is to review the existing the trunnion collar removal machine against the ''Nuclear Safety Design Bases for License Application'' (NSDB) [Ref. 10] requirements and to identify codes and standards and supplemental requirements to meet these requirements. If these codes and standards can not fully meet these requirements then a ''gap'' is identified. These gaps will be identified here and addressed using the ''Trunnion Collar Removal Machine Design Development Plan'' [Ref. 15]. The codes and standards, supplemental requirements, and design development requirements for the trunnion collar removal machine are provided in the gap analysis table (Appendix A, Table 1). Because the trunnion collar removal machine is credited with performing functions important to safety (ITS) in the NSDB [Ref. 10], design basis requirements are applicable to ensure equipment is available and performs required safety functions when needed. The gap analysis table is used to identify design objectives and provide a means to satisfy safety requirements. To ensure that the trunnion collar removal machine performs required safety functions and meets performance criteria, this portion of the gap analysis tables supplies codes and standards sections and the supplemental requirements and identifies design development requirements, if needed

  5. Speech-Language Therapy (For Parents)

    Science.gov (United States)

    ... Staying Safe Videos for Educators Search English Español Speech-Language Therapy KidsHealth / For Parents / Speech-Language Therapy ... most kids with speech and/or language disorders. Speech Disorders, Language Disorders, and Feeding Disorders A speech ...

  6. [Improving speech comprehension using a new cochlear implant speech processor].

    Science.gov (United States)

    Müller-Deile, J; Kortmann, T; Hoppe, U; Hessel, H; Morsnowski, A

    2009-06-01

    The aim of this multicenter clinical field study was to assess the benefits of the new Freedom 24 sound processor for cochlear implant (CI) users implanted with the Nucleus 24 cochlear implant system. The study included 48 postlingually profoundly deaf experienced CI users who demonstrated speech comprehension performance with their current speech processor on the Oldenburg sentence test (OLSA) in quiet conditions of at least 80% correct scores and who were able to perform adaptive speech threshold testing using the OLSA in noisy conditions. Following baseline measures of speech comprehension performance with their current speech processor, subjects were upgraded to the Freedom 24 speech processor. After a take-home trial period of at least 2 weeks, subject performance was evaluated by measuring the speech reception threshold with the Freiburg multisyllabic word test and speech intelligibility with the Freiburg monosyllabic word test at 50 dB and 70 dB in the sound field. The results demonstrated highly significant benefits for speech comprehension with the new speech processor. Significant benefits for speech comprehension were also demonstrated with the new speech processor when tested in competing background noise.In contrast, use of the Abbreviated Profile of Hearing Aid Benefit (APHAB) did not prove to be a suitably sensitive assessment tool for comparative subjective self-assessment of hearing benefits with each processor. Use of the preprocessing algorithm known as adaptive dynamic range optimization (ADRO) in the Freedom 24 led to additional improvements over the standard upgrade map for speech comprehension in quiet and showed equivalent performance in noise. Through use of the preprocessing beam-forming algorithm BEAM, subjects demonstrated a highly significant improved signal-to-noise ratio for speech comprehension thresholds (i.e., signal-to-noise ratio for 50% speech comprehension scores) when tested with an adaptive procedure using the Oldenburg

  7. Neurophysiology of speech differences in childhood apraxia of speech.

    Science.gov (United States)

    Preston, Jonathan L; Molfese, Peter J; Gumkowski, Nina; Sorcinelli, Andrea; Harwood, Vanessa; Irwin, Julia R; Landi, Nicole

    2014-01-01

    Event-related potentials (ERPs) were recorded during a picture naming task of simple and complex words in children with typical speech and with childhood apraxia of speech (CAS). Results reveal reduced amplitude prior to speaking complex (multisyllabic) words relative to simple (monosyllabic) words for the CAS group over the right hemisphere during a time window thought to reflect phonological encoding of word forms. Group differences were also observed prior to production of spoken tokens regardless of word complexity during a time window just prior to speech onset (thought to reflect motor planning/programming). Results suggest differences in pre-speech neurolinguistic processes.

  8. Asymmetric Dynamic Attunement of Speech and Gestures in the Construction of Children's Understanding.

    Science.gov (United States)

    De Jonge-Hoekstra, Lisette; Van der Steen, Steffie; Van Geert, Paul; Cox, Ralf F A

    2016-01-01

    As children learn they use their speech to express words and their hands to gesture. This study investigates the interplay between real-time gestures and speech as children construct cognitive understanding during a hands-on science task. 12 children (M = 6, F = 6) from Kindergarten (n = 5) and first grade (n = 7) participated in this study. Each verbal utterance and gesture during the task were coded, on a complexity scale derived from dynamic skill theory. To explore the interplay between speech and gestures, we applied a cross recurrence quantification analysis (CRQA) to the two coupled time series of the skill levels of verbalizations and gestures. The analysis focused on (1) the temporal relation between gestures and speech, (2) the relative strength and direction of the interaction between gestures and speech, (3) the relative strength and direction between gestures and speech for different levels of understanding, and (4) relations between CRQA measures and other child characteristics. The results show that older and younger children differ in the (temporal) asymmetry in the gestures-speech interaction. For younger children, the balance leans more toward gestures leading speech in time, while the balance leans more toward speech leading gestures for older children. Secondly, at the group level, speech attracts gestures in a more dynamically stable fashion than vice versa, and this asymmetry in gestures and speech extends to lower and higher understanding levels. Yet, for older children, the mutual coupling between gestures and speech is more dynamically stable regarding the higher understanding levels. Gestures and speech are more synchronized in time as children are older. A higher score on schools' language tests is related to speech attracting gestures more rigidly and more asymmetry between gestures and speech, only for the less difficult understanding levels. A higher score on math or past science tasks is related to less asymmetry between gestures and

  9. Asymmetric dynamic attunement of speech and gestures in the construction of children’s understanding

    Directory of Open Access Journals (Sweden)

    Lisette eDe Jonge-Hoekstra

    2016-03-01

    Full Text Available As children learn they use their speech to express words and their hands to gesture. This study investigates the interplay between real-time gestures and speech as children construct cognitive understanding during a hands-on science task. 12 children (M = 6, F = 6 from Kindergarten (n = 5 and first grade (n = 7 participated in this study. Each verbal utterance and gesture during the task were coded, on a complexity scale derived from dynamic skill theory. To explore the interplay between speech and gestures, we applied a cross recurrence quantification analysis (CRQA to the two coupled time series of the skill levels of verbalizations and gestures. The analysis focused on 1 the temporal relation between gestures and speech, 2 the relative strength and direction of the interaction between gestures and speech, 3 the relative strength and direction between gestures and speech for different levels of understanding, and 4 relations between CRQA measures and other child characteristics. The results show that older and younger children differ in the (temporal asymmetry in the gestures-speech interaction. For younger children, the balance leans more towards gestures leading speech in time, while the balance leans more towards speech leading gestures for older children. Secondly, at the group level, speech attracts gestures in a more dynamically stable fashion than vice versa, and this asymmetry in gestures and speech extends to lower and higher understanding levels. Yet, for older children, the mutual coupling between gestures and speech is more dynamically stable regarding the higher understanding levels. Gestures and speech are more synchronized in time as children are older. A higher score on schools’ language tests is related to speech attracting gestures more rigidly and more asymmetry between gestures and speech, only for the less difficult understanding levels. A higher score on math or past science tasks is related to less asymmetry between

  10. Bacteriological quality of drinks from vending machines.

    Science.gov (United States)

    Hunter, P. R.; Burge, S. H.

    1986-01-01

    A survey on the bacteriological quality of both drinking water and flavoured drinks from coin-operated vending machines is reported. Forty-four per cent of 25 drinking water samples examined contained coliforms and 84% had viable counts of greater than 1000 organisms ml at 30 degrees C. Thirty-one flavoured drinks were examined; 6% contained coliforms and 39% had total counts greater than 1000 organisms ml. It is suggested that the D.H.S.S. code of practice on coin-operated vending machines is not being followed. It is also suggested that drinking water alone should not be dispensed from such machines. PMID:3794325

  11. Speech endpoint detection with non-language speech sounds for generic speech processing applications

    Science.gov (United States)

    McClain, Matthew; Romanowski, Brian

    2009-05-01

    Non-language speech sounds (NLSS) are sounds produced by humans that do not carry linguistic information. Examples of these sounds are coughs, clicks, breaths, and filled pauses such as "uh" and "um" in English. NLSS are prominent in conversational speech, but can be a significant source of errors in speech processing applications. Traditionally, these sounds are ignored by speech endpoint detection algorithms, where speech regions are identified in the audio signal prior to processing. The ability to filter NLSS as a pre-processing step can significantly enhance the performance of many speech processing applications, such as speaker identification, language identification, and automatic speech recognition. In order to be used in all such applications, NLSS detection must be performed without the use of language models that provide knowledge of the phonology and lexical structure of speech. This is especially relevant to situations where the languages used in the audio are not known apriori. We present the results of preliminary experiments using data from American and British English speakers, in which segments of audio are classified as language speech sounds (LSS) or NLSS using a set of acoustic features designed for language-agnostic NLSS detection and a hidden-Markov model (HMM) to model speech generation. The results of these experiments indicate that the features and model used are capable of detection certain types of NLSS, such as breaths and clicks, while detection of other types of NLSS such as filled pauses will require future research.

  12. Abortion and compelled physician speech.

    Science.gov (United States)

    Orentlicher, David

    2015-01-01

    Informed consent mandates for abortion providers may infringe the First Amendment's freedom of speech. On the other hand, they may reinforce the physician's duty to obtain informed consent. Courts can promote both doctrines by ensuring that compelled physician speech pertains to medical facts about abortion rather than abortion ideology and that compelled speech is truthful and not misleading. © 2015 American Society of Law, Medicine & Ethics, Inc.

  13. Speech-To-Text Conversion STT System Using Hidden Markov Model HMM

    Directory of Open Access Journals (Sweden)

    Su Myat Mon

    2015-06-01

    Full Text Available Abstract Speech is an easiest way to communicate with each other. Speech processing is widely used in many applications like security devices household appliances cellular phones ATM machines and computers. The human computer interface has been developed to communicate or interact conveniently for one who is suffering from some kind of disabilities. Speech-to-Text Conversion STT systems have a lot of benefits for the deaf or dumb people and find their applications in our daily lives. In the same way the aim of the system is to convert the input speech signals into the text output for the deaf or dumb students in the educational fields. This paper presents an approach to extract features by using Mel Frequency Cepstral Coefficients MFCC from the speech signals of isolated spoken words. And Hidden Markov Model HMM method is applied to train and test the audio files to get the recognized spoken word. The speech database is created by using MATLAB.Then the original speech signals are preprocessed and these speech samples are extracted to the feature vectors which are used as the observation sequences of the Hidden Markov Model HMM recognizer. The feature vectors are analyzed in the HMM depending on the number of states.

  14. Bi-Modal Face and Speech Authentication: a BioLogin Demonstration System

    OpenAIRE

    Marcel, Sébastien; Mariéthoz, Johnny; Rodriguez, Yann; Cardinaux, Fabien

    2006-01-01

    This paper presents a bi-modal (face and speech) authentication demonstration system that simulates the login of a user using its face and its voice. This demonstration is called BioLogin. It runs both on Linux and Windows and the Windows version is freely available for download. Bio\\-Login is implemented using an open source machine learning library and its machine vision package.

  15. Speech Recognition on Mobile Devices

    DEFF Research Database (Denmark)

    Tan, Zheng-Hua; Lindberg, Børge

    2010-01-01

    in the mobile context covering motivations, challenges, fundamental techniques and applications. Three ASR architectures are introduced: embedded speech recognition, distributed speech recognition and network speech recognition. Their pros and cons and implementation issues are discussed. Applications within......The enthusiasm of deploying automatic speech recognition (ASR) on mobile devices is driven both by remarkable advances in ASR technology and by the demand for efficient user interfaces on such devices as mobile phones and personal digital assistants (PDAs). This chapter presents an overview of ASR...

  16. Quantification and Systematic Characterization of Stuttering-Like Disfluencies in Acquired Apraxia of Speech.

    Science.gov (United States)

    Bailey, Dallin J; Blomgren, Michael; DeLong, Catharine; Berggren, Kiera; Wambaugh, Julie L

    2017-06-22

    The purpose of this article is to quantify and describe stuttering-like disfluencies in speakers with acquired apraxia of speech (AOS), utilizing the Lidcombe Behavioural Data Language (LBDL). Additional purposes include measuring test-retest reliability and examining the effect of speech sample type on disfluency rates. Two types of speech samples were elicited from 20 persons with AOS and aphasia: repetition of mono- and multisyllabic words from a protocol for assessing AOS (Duffy, 2013), and connected speech tasks (Nicholas & Brookshire, 1993). Sampling was repeated at 1 and 4 weeks following initial sampling. Stuttering-like disfluencies were coded using the LBDL, which is a taxonomy that focuses on motoric aspects of stuttering. Disfluency rates ranged from 0% to 13.1% for the connected speech task and from 0% to 17% for the word repetition task. There was no significant effect of speech sampling time on disfluency rate in the connected speech task, but there was a significant effect of time for the word repetition task. There was no significant effect of speech sample type. Speakers demonstrated both major types of stuttering-like disfluencies as categorized by the LBDL (fixed postures and repeated movements). Connected speech samples yielded more reliable tallies over repeated measurements. Suggestions are made for modifying the LBDL for use in AOS in order to further add to systematic descriptions of motoric disfluencies in this disorder.

  17. Speaking Code

    DEFF Research Database (Denmark)

    Cox, Geoff

    Speaking Code begins by invoking the “Hello World” convention used by programmers when learning a new language, helping to establish the interplay of text and code that runs through the book. Interweaving the voice of critical writing from the humanities with the tradition of computing and software...

  18. Machine technology: a survey

    International Nuclear Information System (INIS)

    Barbier, M.M.

    1981-01-01

    An attempt was made to find existing machines that have been upgraded and that could be used for large-scale decontamination operations outdoors. Such machines are in the building industry, the mining industry, and the road construction industry. The road construction industry has yielded the machines in this presentation. A review is given of operations that can be done with the machines available

  19. Machine Shop Lathes.

    Science.gov (United States)

    Dunn, James

    This guide, the second in a series of five machine shop curriculum manuals, was designed for use in machine shop courses in Oklahoma. The purpose of the manual is to equip students with basic knowledge and skills that will enable them to enter the machine trade at the machine-operator level. The curriculum is designed so that it can be used in…

  20. Superconducting rotating machines

    International Nuclear Information System (INIS)

    Smith, J.L. Jr.; Kirtley, J.L. Jr.; Thullen, P.

    1975-01-01

    The opportunities and limitations of the applications of superconductors in rotating electric machines are given. The relevant properties of superconductors and the fundamental requirements for rotating electric machines are discussed. The current state-of-the-art of superconducting machines is reviewed. Key problems, future developments and the long range potential of superconducting machines are assessed

  1. Current trends in multilingual speech processing

    Indian Academy of Sciences (India)

    2016-08-26

    ; speech-to-speech translation; language identification. ... interest owing to two strong driving forces. Firstly, technical advances in speech recognition and synthesis are posing new challenges and opportunities to researchers.

  2. Machine-to-machine communications architectures, technology, standards, and applications

    CERN Document Server

    Misic, Vojislav B

    2014-01-01

    With the number of machine-to-machine (M2M)-enabled devices projected to reach 20 to 50 billion by 2020, there is a critical need to understand the demands imposed by such systems. Machine-to-Machine Communications: Architectures, Technology, Standards, and Applications offers rigorous treatment of the many facets of M2M communication, including its integration with current technology.Presenting the work of a different group of international experts in each chapter, the book begins by supplying an overview of M2M technology. It considers proposed standards, cutting-edge applications, architectures, and traffic modeling and includes case studies that highlight the differences between traditional and M2M communications technology.Details a practical scheme for the forward error correction code designInvestigates the effectiveness of the IEEE 802.15.4 low data rate wireless personal area network standard for use in M2M communicationsIdentifies algorithms that will ensure functionality, performance, reliability, ...

  3. Multimodal Speech Capture System for Speech Rehabilitation and Learning.

    Science.gov (United States)

    Sebkhi, Nordine; Desai, Dhyey; Islam, Mohammad; Lu, Jun; Wilson, Kimberly; Ghovanloo, Maysam

    2017-11-01

    Speech-language pathologists (SLPs) are trained to correct articulation of people diagnosed with motor speech disorders by analyzing articulators' motion and assessing speech outcome while patients speak. To assist SLPs in this task, we are presenting the multimodal speech capture system (MSCS) that records and displays kinematics of key speech articulators, the tongue and lips, along with voice, using unobtrusive methods. Collected speech modalities, tongue motion, lips gestures, and voice are visualized not only in real-time to provide patients with instant feedback but also offline to allow SLPs to perform post-analysis of articulators' motion, particularly the tongue, with its prominent but hardly visible role in articulation. We describe the MSCS hardware and software components, and demonstrate its basic visualization capabilities by a healthy individual repeating the words "Hello World." A proof-of-concept prototype has been successfully developed for this purpose, and will be used in future clinical studies to evaluate its potential impact on accelerating speech rehabilitation by enabling patients to speak naturally. Pattern matching algorithms to be applied to the collected data can provide patients with quantitative and objective feedback on their speech performance, unlike current methods that are mostly subjective, and may vary from one SLP to another.

  4. Measurement of speech parameters in casual speech of dementia patients

    NARCIS (Netherlands)

    Ossewaarde, Roelant; Jonkers, Roel; Jalvingh, Fedor; Bastiaanse, Yvonne

    Measurement of speech parameters in casual speech of dementia patients Roelant Adriaan Ossewaarde1,2, Roel Jonkers1, Fedor Jalvingh1,3, Roelien Bastiaanse1 1CLCG, University of Groningen (NL); 2HU University of Applied Sciences Utrecht (NL); 33St. Marienhospital - Vechta, Geriatric Clinic Vechta

  5. Alternative Speech Communication System for Persons with Severe Speech Disorders

    Science.gov (United States)

    Selouani, Sid-Ahmed; Sidi Yakoub, Mohammed; O'Shaughnessy, Douglas

    2009-12-01

    Assistive speech-enabled systems are proposed to help both French and English speaking persons with various speech disorders. The proposed assistive systems use automatic speech recognition (ASR) and speech synthesis in order to enhance the quality of communication. These systems aim at improving the intelligibility of pathologic speech making it as natural as possible and close to the original voice of the speaker. The resynthesized utterances use new basic units, a new concatenating algorithm and a grafting technique to correct the poorly pronounced phonemes. The ASR responses are uttered by the new speech synthesis system in order to convey an intelligible message to listeners. Experiments involving four American speakers with severe dysarthria and two Acadian French speakers with sound substitution disorders (SSDs) are carried out to demonstrate the efficiency of the proposed methods. An improvement of the Perceptual Evaluation of the Speech Quality (PESQ) value of 5% and more than 20% is achieved by the speech synthesis systems that deal with SSD and dysarthria, respectively.

  6. LINGUISTIC ANALYSIS FOR THE BELARUSIAN CORPUS WITH THE APPLICATION OF NATURAL LANGUAGE PROCESSING AND MACHINE LEARNING TECHNIQUES

    Directory of Open Access Journals (Sweden)

    Yu. S. Hetsevich

    2017-01-01

    Full Text Available The article focuses on the problems existing in text-to-speech synthesis. Different morphological, lexical and syntactical elements were localized with the help of the Belarusian unit of NooJ program. Those types of errors, which occur in Belarusian texts, were analyzed and corrected. Language model and part of speech tagging model were built. The natural language processing of Belarusian corpus with the help of developed algorithm using machine learning was carried out. The precision of developed models of machine learning has been 80–90 %. The dictionary was enriched with new words for the further using it in the systems of Belarusian speech synthesis.

  7. Speech Perception as a Multimodal Phenomenon

    OpenAIRE

    Rosenblum, Lawrence D.

    2008-01-01

    Speech perception is inherently multimodal. Visual speech (lip-reading) information is used by all perceivers and readily integrates with auditory speech. Imaging research suggests that the brain treats auditory and visual speech similarly. These findings have led some researchers to consider that speech perception works by extracting amodal information that takes the same form across modalities. From this perspective, speech integration is a property of the input information itself. Amodal s...

  8. VOA: a 2-d plasma physics code

    International Nuclear Information System (INIS)

    Eltgroth, P.G.

    1975-12-01

    A 2-dimensional relativistic plasma physics code was written and tested. The non-thermal components of the particle distribution functions are represented by expansion into moments in momentum space. These moments are computed directly from numerical equations. Currently three species are included - electrons, ions and ''beam electrons''. The computer code runs on either the 7600 or STAR machines at LLL. Both the physics and the operation of the code are discussed

  9. Discrete Sparse Coding.

    Science.gov (United States)

    Exarchakis, Georgios; Lücke, Jörg

    2017-11-01

    Sparse coding algorithms with continuous latent variables have been the subject of a large number of studies. However, discrete latent spaces for sparse coding have been largely ignored. In this work, we study sparse coding with latents described by discrete instead of continuous prior distributions. We consider the general case in which the latents (while being sparse) can take on any value of a finite set of possible values and in which we learn the prior probability of any value from data. This approach can be applied to any data generated by discrete causes, and it can be applied as an approximation of continuous causes. As the prior probabilities are learned, the approach then allows for estimating the prior shape without assuming specific functional forms. To efficiently train the parameters of our probabilistic generative model, we apply a truncated expectation-maximization approach (expectation truncation) that we modify to work with a general discrete prior. We evaluate the performance of the algorithm by applying it to a variety of tasks: (1) we use artificial data to verify that the algorithm can recover the generating parameters from a random initialization, (2) use image patches of natural images and discuss the role of the prior for the extraction of image components, (3) use extracellular recordings of neurons to present a novel method of analysis for spiking neurons that includes an intuitive discretization strategy, and (4) apply the algorithm on the task of encoding audio waveforms of human speech. The diverse set of numerical experiments presented in this letter suggests that discrete sparse coding algorithms can scale efficiently to work with realistic data sets and provide novel statistical quantities to describe the structure of the data.

  10. Auditory Modeling for Noisy Speech Recognition

    National Research Council Canada - National Science Library

    2000-01-01

    ... digital filtering for noise cancellation which interfaces to speech recognition software. It uses auditory features in speech recognition training, and provides applications to multilingual spoken language translation...

  11. Teaching Speech Acts

    Directory of Open Access Journals (Sweden)

    Teaching Speech Acts

    2007-01-01

    Full Text Available In this paper I argue that pragmatic ability must become part of what we teach in the classroom if we are to realize the goals of communicative competence for our students. I review the research on pragmatics, especially those articles that point to the effectiveness of teaching pragmatics in an explicit manner, and those that posit methods for teaching. I also note two areas of scholarship that address classroom needs—the use of authentic data and appropriate assessment tools. The essay concludes with a summary of my own experience teaching speech acts in an advanced-level Portuguese class.

  12. Assembly processor program converts symbolic programming language to machine language

    Science.gov (United States)

    Pelto, E. V.

    1967-01-01

    Assembly processor program converts symbolic programming language to machine language. This program translates symbolic codes into computer understandable instructions, assigns locations in storage for successive instructions, and computer locations from symbolic addresses.

  13. Psychoacoustic cues to emotion in speech prosody and music.

    Science.gov (United States)

    Coutinho, Eduardo; Dibben, Nicola

    2013-01-01

    There is strong evidence of shared acoustic profiles common to the expression of emotions in music and speech, yet relatively limited understanding of the specific psychoacoustic features involved. This study combined a controlled experiment and computational modelling to investigate the perceptual codes associated with the expression of emotion in the acoustic domain. The empirical stage of the study provided continuous human ratings of emotions perceived in excerpts of film music and natural speech samples. The computational stage created a computer model that retrieves the relevant information from the acoustic stimuli and makes predictions about the emotional expressiveness of speech and music close to the responses of human subjects. We show that a significant part of the listeners' second-by-second reported emotions to music and speech prosody can be predicted from a set of seven psychoacoustic features: loudness, tempo/speech rate, melody/prosody contour, spectral centroid, spectral flux, sharpness, and roughness. The implications of these results are discussed in the context of cross-modal similarities in the communication of emotion in the acoustic domain.

  14. Speech Training for Inmate Rehabilitation.

    Science.gov (United States)

    Parkinson, Michael G.; Dobkins, David H.

    1982-01-01

    Using a computerized content analysis, the authors demonstrate changes in speech behaviors of prison inmates. They conclude that two to four hours of public speaking training can have only limited effect on students who live in a culture in which "prison speech" is the expected and rewarded form of behavior. (PD)

  15. Separating Underdetermined Convolutive Speech Mixtures

    DEFF Research Database (Denmark)

    Pedersen, Michael Syskind; Wang, DeLiang; Larsen, Jan

    2006-01-01

    a method for underdetermined blind source separation of convolutive mixtures. The proposed framework is applicable for separation of instantaneous as well as convolutive speech mixtures. It is possible to iteratively extract each speech signal from the mixture by combining blind source separation...

  16. Speech recognition from spectral dynamics

    Indian Academy of Sciences (India)

    Some of the history of gradual infusion of the modulation spectrum concept into Automatic recognition of speech (ASR) comes next, pointing to the relationship of modulation spectrum processing to wellaccepted ASR techniques such as dynamic speech features or RelAtive SpecTrAl (RASTA) filtering. Next, the frequency ...

  17. Speech Prosody in Cerebellar Ataxia

    Science.gov (United States)

    Casper, Maureen A.; Raphael, Lawrence J.; Harris, Katherine S.; Geibel, Jennifer M.

    2007-01-01

    Persons with cerebellar ataxia exhibit changes in physical coordination and speech and voice production. Previously, these alterations of speech and voice production were described primarily via perceptual coordinates. In this study, the spatial-temporal properties of syllable production were examined in 12 speakers, six of whom were healthy…

  18. Code portability and data management considerations in the SAS3D LMFBR accident-analysis code

    International Nuclear Information System (INIS)

    Dunn, F.E.

    1981-01-01

    The SAS3D code was produced from a predecessor in order to reduce or eliminate interrelated problems in the areas of code portability, the large size of the code, inflexibility in the use of memory and the size of cases that can be run, code maintenance, and running speed. Many conventional solutions, such as variable dimensioning, disk storage, virtual memory, and existing code-maintenance utilities were not feasible or did not help in this case. A new data management scheme was developed, coding standards and procedures were adopted, special machine-dependent routines were written, and a portable source code processing code was written. The resulting code is quite portable, quite flexible in the use of memory and the size of cases that can be run, much easier to maintain, and faster running. SAS3D is still a large, long running code that only runs well if sufficient main memory is available

  19. Speech Enhancement by Classification of Noisy Signals Decomposed Using NMF and Wiener Filtering

    DEFF Research Database (Denmark)

    Fakhry, Mahmoud; Poorjam, Amir Hossein; Christensen, Mads Græsbøll

    2018-01-01

    are identified in the cepstral domain using the trained classifier. We apply unsupervised NMF followed by Wiener filtering for the decomposition, and use a support vector machine trained on the mel-frequency cepstral coefficients of the parts of training speech and noise signals for the classification...

  20. Machine learning a probabilistic perspective

    CERN Document Server

    Murphy, Kevin P

    2012-01-01

    Today's Web-enabled deluge of electronic data calls for automated methods of data analysis. Machine learning provides these, developing methods that can automatically detect patterns in data and then use the uncovered patterns to predict future data. This textbook offers a comprehensive and self-contained introduction to the field of machine learning, based on a unified, probabilistic approach. The coverage combines breadth and depth, offering necessary background material on such topics as probability, optimization, and linear algebra as well as discussion of recent developments in the field, including conditional random fields, L1 regularization, and deep learning. The book is written in an informal, accessible style, complete with pseudo-code for the most important algorithms. All topics are copiously illustrated with color images and worked examples drawn from such application domains as biology, text processing, computer vision, and robotics. Rather than providing a cookbook of different heuristic method...

  1. Coding Labour

    Directory of Open Access Journals (Sweden)

    Anthony McCosker

    2014-03-01

    Full Text Available As well as introducing the Coding Labour section, the authors explore the diffusion of code across the material contexts of everyday life, through the objects and tools of mediation, the systems and practices of cultural production and organisational management, and in the material conditions of labour. Taking code beyond computation and software, their specific focus is on the increasingly familiar connections between code and labour with a focus on the codification and modulation of affect through technologies and practices of management within the contemporary work organisation. In the grey literature of spreadsheets, minutes, workload models, email and the like they identify a violence of forms through which workplace affect, in its constant flux of crisis and ‘prodromal’ modes, is regulated and governed.

  2. On speech recognition during anaesthesia

    DEFF Research Database (Denmark)

    Alapetite, Alexandre

    2007-01-01

    This PhD thesis in human-computer interfaces (informatics) studies the case of the anaesthesia record used during medical operations and the possibility to supplement it with speech recognition facilities. Problems and limitations have been identified with the traditional paper-based anaesthesia...... and inaccuracies in the anaesthesia record. Supplementing the electronic anaesthesia record interface with speech input facilities is proposed as one possible solution to a part of the problem. The testing of the various hypotheses has involved the development of a prototype of an electronic anaesthesia record...... interface with speech input facilities in Danish. The evaluation of the new interface was carried out in a full-scale anaesthesia simulator. This has been complemented by laboratory experiments on several aspects of speech recognition for this type of use, e.g. the effects of noise on speech recognition...

  3. From Gesture to Speech

    Directory of Open Access Journals (Sweden)

    Maurizio Gentilucci

    2012-11-01

    Full Text Available One of the major problems concerning the evolution of human language is to understand how sounds became associated to meaningful gestures. It has been proposed that the circuit controlling gestures and speech evolved from a circuit involved in the control of arm and mouth movements related to ingestion. This circuit contributed to the evolution of spoken language, moving from a system of communication based on arm gestures. The discovery of the mirror neurons has provided strong support for the gestural theory of speech origin because they offer a natural substrate for the embodiment of language and create a direct link between sender and receiver of a message. Behavioural studies indicate that manual gestures are linked to mouth movements used for syllable emission. Grasping with the hand selectively affected movement of inner or outer parts of the mouth according to syllable pronunciation and hand postures, in addition to hand actions, influenced the control of mouth grasp and vocalization. Gestures and words are also related to each other. It was found that when producing communicative gestures (emblems the intention to interact directly with a conspecific was transferred from gestures to words, inducing modification in voice parameters. Transfer effects of the meaning of representational gestures were found on both vocalizations and meaningful words. It has been concluded that the results of our studies suggest the existence of a system relating gesture to vocalization which was precursor of a more general system reciprocally relating gesture to word.

  4. SwingStates: adding state machines to the swing toolkit

    OpenAIRE

    Appert , Caroline; Beaudouin-Lafon , Michel

    2006-01-01

    International audience; This article describes SwingStates, a library that adds state machines to the Java Swing user interface toolkit. Unlike traditional approaches, which use callbacks or listeners to define interaction, state machines provide a powerful control structure and localize all of the interaction code in one place. SwingStates takes advantage of Java's inner classes, providing programmers with a natural syntax and making it easier to follow and debug the resulting code. SwingSta...

  5. Stable 1-Norm Error Minimization Based Linear Predictors for Speech Modeling

    DEFF Research Database (Denmark)

    Giacobello, Daniele; Christensen, Mads Græsbøll; Jensen, Tobias Lindstrøm

    2014-01-01

    In linear prediction of speech, the 1-norm error minimization criterion has been shown to provide a valid alternative to the 2-norm minimization criterion. However, unlike 2-norm minimization, 1-norm minimization does not guarantee the stability of the corresponding all-pole filter and can generate...... saturations when this is used to synthesize speech. In this paper, we introduce two new methods to obtain intrinsically stable predictors with the 1-norm minimization. The first method is based on constraining the roots of the predictor to lie within the unit circle by reducing the numerical range...... based linear prediction for modeling and coding of speech....

  6. Machine tool structures

    CERN Document Server

    Koenigsberger, F

    1970-01-01

    Machine Tool Structures, Volume 1 deals with fundamental theories and calculation methods for machine tool structures. Experimental investigations into stiffness are discussed, along with the application of the results to the design of machine tool structures. Topics covered range from static and dynamic stiffness to chatter in metal cutting, stability in machine tools, and deformations of machine tool structures. This volume is divided into three sections and opens with a discussion on stiffness specifications and the effect of stiffness on the behavior of the machine under forced vibration c

  7. Detecting Parkinson's disease from sustained phonation and speech signals.

    Directory of Open Access Journals (Sweden)

    Evaldas Vaiciukynas

    Full Text Available This study investigates signals from sustained phonation and text-dependent speech modalities for Parkinson's disease screening. Phonation corresponds to the vowel /a/ voicing task and speech to the pronunciation of a short sentence in Lithuanian language. Signals were recorded through two channels simultaneously, namely, acoustic cardioid (AC and smart phone (SP microphones. Additional modalities were obtained by splitting speech recording into voiced and unvoiced parts. Information in each modality is summarized by 18 well-known audio feature sets. Random forest (RF is used as a machine learning algorithm, both for individual feature sets and for decision-level fusion. Detection performance is measured by the out-of-bag equal error rate (EER and the cost of log-likelihood-ratio. Essentia audio feature set was the best using the AC speech modality and YAAFE audio feature set was the best using the SP unvoiced modality, achieving EER of 20.30% and 25.57%, respectively. Fusion of all feature sets and modalities resulted in EER of 19.27% for the AC and 23.00% for the SP channel. Non-linear projection of a RF-based proximity matrix into the 2D space enriched medical decision support by visualization.

  8. Research on the optoacoustic communication system for speech transmission by variable laser-pulse repetition rates

    Science.gov (United States)

    Jiang, Hongyan; Qiu, Hongbing; He, Ning; Liao, Xin

    2018-06-01

    For the optoacoustic communication from in-air platforms to submerged apparatus, a method based on speech recognition and variable laser-pulse repetition rates is proposed, which realizes character encoding and transmission for speech. Firstly, the theories and spectrum characteristics of the laser-generated underwater sound are analyzed; and moreover character conversion and encoding for speech as well as the pattern of codes for laser modulation is studied; lastly experiments to verify the system design are carried out. Results show that the optoacoustic system, where laser modulation is controlled by speech-to-character baseband codes, is beneficial to improve flexibility in receiving location for underwater targets as well as real-time performance in information transmission. In the overwater transmitter, a pulse laser is controlled to radiate by speech signals with several repetition rates randomly selected in the range of one to fifty Hz, and then in the underwater receiver laser pulse repetition rate and data can be acquired by the preamble and information codes of the corresponding laser-generated sound. When the energy of the laser pulse is appropriate, real-time transmission for speaker-independent speech can be realized in that way, which solves the problem of underwater bandwidth resource and provides a technical approach for the air-sea communication.

  9. Stuttering Frequency, Speech Rate, Speech Naturalness, and Speech Effort During the Production of Voluntary Stuttering.

    Science.gov (United States)

    Davidow, Jason H; Grossman, Heather L; Edge, Robin L

    2018-05-01

    Voluntary stuttering techniques involve persons who stutter purposefully interjecting disfluencies into their speech. Little research has been conducted on the impact of these techniques on the speech pattern of persons who stutter. The present study examined whether changes in the frequency of voluntary stuttering accompanied changes in stuttering frequency, articulation rate, speech naturalness, and speech effort. In total, 12 persons who stutter aged 16-34 years participated. Participants read four 300-syllable passages during a control condition, and three voluntary stuttering conditions that involved attempting to produce purposeful, tension-free repetitions of initial sounds or syllables of a word for two or more repetitions (i.e., bouncing). The three voluntary stuttering conditions included bouncing on 5%, 10%, and 15% of syllables read. Friedman tests and follow-up Wilcoxon signed ranks tests were conducted for the statistical analyses. Stuttering frequency, articulation rate, and speech naturalness were significantly different between the voluntary stuttering conditions. Speech effort did not differ between the voluntary stuttering conditions. Stuttering frequency was significantly lower during the three voluntary stuttering conditions compared to the control condition, and speech effort was significantly lower during two of the three voluntary stuttering conditions compared to the control condition. Due to changes in articulation rate across the voluntary stuttering conditions, it is difficult to conclude, as has been suggested previously, that voluntary stuttering is the reason for stuttering reductions found when using voluntary stuttering techniques. Additionally, future investigations should examine different types of voluntary stuttering over an extended period of time to determine their impact on stuttering frequency, speech rate, speech naturalness, and speech effort.

  10. Electronic data processing codes for California wildland plants

    Science.gov (United States)

    Merton J. Reed; W. Robert Powell; Bur S. Bal

    1963-01-01

    Systematized codes for plant names are helpful to a wide variety of workers who must record the identity of plants in the field. We have developed such codes for a majority of the vascular plants encountered on California wildlands and have published the codes in pocket size, using photo-reductions of the output from data processing machines. A limited number of the...

  11. Small intragenic deletion in FOXP2 associated with childhood apraxia of speech and dysarthria.

    Science.gov (United States)

    Turner, Samantha J; Hildebrand, Michael S; Block, Susan; Damiano, John; Fahey, Michael; Reilly, Sheena; Bahlo, Melanie; Scheffer, Ingrid E; Morgan, Angela T

    2013-09-01

    Relatively little is known about the neurobiological basis of speech disorders although genetic determinants are increasingly recognized. The first gene for primary speech disorder was FOXP2, identified in a large, informative family with verbal and oral dyspraxia. Subsequently, many de novo and familial cases with a severe speech disorder associated with FOXP2 mutations have been reported. These mutations include sequencing alterations, translocations, uniparental disomy, and genomic copy number variants. We studied eight probands with speech disorder and their families. Family members were phenotyped using a comprehensive assessment of speech, oral motor function, language, literacy skills, and cognition. Coding regions of FOXP2 were screened to identify novel variants. Segregation of the variant was determined in the probands' families. Variants were identified in two probands. One child with severe motor speech disorder had a small de novo intragenic FOXP2 deletion. His phenotype included features of childhood apraxia of speech and dysarthria, oral motor dyspraxia, receptive and expressive language disorder, and literacy difficulties. The other variant was found in a family in two of three family members with stuttering, and also in the mother with oral motor impairment. This variant was considered a benign polymorphism as it was predicted to be non-pathogenic with in silico tools and found in database controls. This is the first report of a small intragenic deletion of FOXP2 that is likely to be the cause of severe motor speech disorder associated with language and literacy problems. Copyright © 2013 Wiley Periodicals, Inc.

  12. Optimal codes as Tanner codes with cyclic component codes

    DEFF Research Database (Denmark)

    Høholdt, Tom; Pinero, Fernando; Zeng, Peng

    2014-01-01

    In this article we study a class of graph codes with cyclic code component codes as affine variety codes. Within this class of Tanner codes we find some optimal binary codes. We use a particular subgraph of the point-line incidence plane of A(2,q) as the Tanner graph, and we are able to describe ...

  13. Emotion Recognition from Chinese Speech for Smart Affective Services Using a Combination of SVM and DBN

    Science.gov (United States)

    Zhu, Lianzhang; Chen, Leiming; Zhao, Dehai

    2017-01-01

    Accurate emotion recognition from speech is important for applications like smart health care, smart entertainment, and other smart services. High accuracy emotion recognition from Chinese speech is challenging due to the complexities of the Chinese language. In this paper, we explore how to improve the accuracy of speech emotion recognition, including speech signal feature extraction and emotion classification methods. Five types of features are extracted from a speech sample: mel frequency cepstrum coefficient (MFCC), pitch, formant, short-term zero-crossing rate and short-term energy. By comparing statistical features with deep features extracted by a Deep Belief Network (DBN), we attempt to find the best features to identify the emotion status for speech. We propose a novel classification method that combines DBN and SVM (support vector machine) instead of using only one of them. In addition, a conjugate gradient method is applied to train DBN in order to speed up the training process. Gender-dependent experiments are conducted using an emotional speech database created by the Chinese Academy of Sciences. The results show that DBN features can reflect emotion status better than artificial features, and our new classification approach achieves an accuracy of 95.8%, which is higher than using either DBN or SVM separately. Results also show that DBN can work very well for small training databases if it is properly designed. PMID:28737705

  14. Emotion Recognition from Chinese Speech for Smart Affective Services Using a Combination of SVM and DBN.

    Science.gov (United States)

    Zhu, Lianzhang; Chen, Leiming; Zhao, Dehai; Zhou, Jiehan; Zhang, Weishan

    2017-07-24

    Accurate emotion recognition from speech is important for applications like smart health care, smart entertainment, and other smart services. High accuracy emotion recognition from Chinese speech is challenging due to the complexities of the Chinese language. In this paper, we explore how to improve the accuracy of speech emotion recognition, including speech signal feature extraction and emotion classification methods. Five types of features are extracted from a speech sample: mel frequency cepstrum coefficient (MFCC), pitch, formant, short-term zero-crossing rate and short-term energy. By comparing statistical features with deep features extracted by a Deep Belief Network (DBN), we attempt to find the best features to identify the emotion status for speech. We propose a novel classification method that combines DBN and SVM (support vector machine) instead of using only one of them. In addition, a conjugate gradient method is applied to train DBN in order to speed up the training process. Gender-dependent experiments are conducted using an emotional speech database created by the Chinese Academy of Sciences. The results show that DBN features can reflect emotion status better than artificial features, and our new classification approach achieves an accuracy of 95.8%, which is higher than using either DBN or SVM separately. Results also show that DBN can work very well for small training databases if it is properly designed.

  15. Deriving Word Order in Code-Switching: Feature Inheritance and Light Verbs

    Science.gov (United States)

    Shim, Ji Young

    2013-01-01

    This dissertation investigates code-switching (CS), the concurrent use of more than one language in conversation, commonly observed in bilingual speech. Assuming that code-switching is subject to universal principles, just like monolingual grammar, the dissertation provides a principled account of code-switching, with particular emphasis on OV~VO…

  16. Visual Speech Fills in Both Discrimination and Identification of Non-Intact Auditory Speech in Children

    Science.gov (United States)

    Jerger, Susan; Damian, Markus F.; McAlpine, Rachel P.; Abdi, Herve

    2018-01-01

    To communicate, children must discriminate and identify speech sounds. Because visual speech plays an important role in this process, we explored how visual speech influences phoneme discrimination and identification by children. Critical items had intact visual speech (e.g. baez) coupled to non-intact (excised onsets) auditory speech (signified…

  17. Speech enhancement using emotion dependent codebooks

    NARCIS (Netherlands)

    Naidu, D.H.R.; Srinivasan, S.

    2012-01-01

    Several speech enhancement approaches utilize trained models of clean speech data, such as codebooks, Gaussian mixtures, and hidden Markov models. These models are typically trained on neutral clean speech data, without any emotion. However, in practical scenarios, emotional speech is a common

  18. Automated Speech Rate Measurement in Dysarthria

    Science.gov (United States)

    Martens, Heidi; Dekens, Tomas; Van Nuffelen, Gwen; Latacz, Lukas; Verhelst, Werner; De Bodt, Marc

    2015-01-01

    Purpose: In this study, a new algorithm for automated determination of speech rate (SR) in dysarthric speech is evaluated. We investigated how reliably the algorithm calculates the SR of dysarthric speech samples when compared with calculation performed by speech-language pathologists. Method: The new algorithm was trained and tested using Dutch…

  19. Is Birdsong More Like Speech or Music?

    Science.gov (United States)

    Shannon, Robert V

    2016-04-01

    Music and speech share many acoustic cues but not all are equally important. For example, harmonic pitch is essential for music but not for speech. When birds communicate is their song more like speech or music? A new study contrasting pitch and spectral patterns shows that birds perceive their song more like humans perceive speech. Copyright © 2016 Elsevier Ltd. All rights reserved.

  20. Freedom of Speech Newsletter, September, 1975.

    Science.gov (United States)

    Allen, Winfred G., Jr., Ed.

    The Freedom of Speech Newsletter is the communication medium for the Freedom of Speech Interest Group of the Western Speech Communication Association. The newsletter contains such features as a statement of concern by the National Ad Hoc Committee Against Censorship; Reticence and Free Speech, an article by James F. Vickrey discussing the subtle…

  1. MITS machine operations

    International Nuclear Information System (INIS)

    Flinchem, J.

    1980-01-01

    This document contains procedures which apply to operations performed on individual P-1c machines in the Machine Interface Test System (MITS) at AiResearch Manufacturing Company's Torrance, California Facility

  2. Brain versus Machine Control.

    Directory of Open Access Journals (Sweden)

    Jose M Carmena

    2004-12-01

    Full Text Available Dr. Octopus, the villain of the movie "Spiderman 2", is a fusion of man and machine. Neuroscientist Jose Carmena examines the facts behind this fictional account of a brain- machine interface

  3. Applied machining technology

    CERN Document Server

    Tschätsch, Heinz

    2010-01-01

    Machining and cutting technologies are still crucial for many manufacturing processes. This reference presents all important machining processes in a comprehensive and coherent way. It includes many examples of concrete calculations, problems and solutions.

  4. Machining with abrasives

    CERN Document Server

    Jackson, Mark J

    2011-01-01

    Abrasive machining is key to obtaining the desired geometry and surface quality in manufacturing. This book discusses the fundamentals and advances in the abrasive machining processes. It provides a complete overview of developing areas in the field.

  5. QCD on the connection machine

    International Nuclear Information System (INIS)

    Gupta, R.

    1990-01-01

    In this talk I give a brief introduction to the standard model of particle interactions and illustrate why analytical methods fail to solve QCD. I then give some details of our implementation of the high performance QCD code on the CM2 and highlight the important lessons learned. The sustained speed of the code at the time of this conference is 5.2 Gigaflops (scaled to a full 64K machine). Since this is a conference dedicated to computing in the 21st century, I will tailor my expectations (somewhat idiosyncratic) of the physics objectives to reflect what we will be able to do in 10 years time, extrapolating from where we stand today. This work is being done under a joint LANL-TMC collaboration consisting of C. Baillie, R. Brickner, D. Daniel, G. Kilcup, L. Johnson, A. Patel. S. Sharpe and myself. 5 refs

  6. Aztheca Code

    International Nuclear Information System (INIS)

    Quezada G, S.; Espinosa P, G.; Centeno P, J.; Sanchez M, H.

    2017-09-01

    This paper presents the Aztheca code, which is formed by the mathematical models of neutron kinetics, power generation, heat transfer, core thermo-hydraulics, recirculation systems, dynamic pressure and level models and control system. The Aztheca code is validated with plant data, as well as with predictions from the manufacturer when the reactor operates in a stationary state. On the other hand, to demonstrate that the model is applicable during a transient, an event occurred in a nuclear power plant with a BWR reactor is selected. The plant data are compared with the results obtained with RELAP-5 and the Aztheca model. The results show that both RELAP-5 and the Aztheca code have the ability to adequately predict the behavior of the reactor. (Author)

  7. Speech recovery device

    Energy Technology Data Exchange (ETDEWEB)

    Frankle, Christen M.

    2004-04-20

    There is provided an apparatus and method for assisting speech recovery in people with inability to speak due to aphasia, apraxia or another condition with similar effect. A hollow, rigid, thin-walled tube with semi-circular or semi-elliptical cut out shapes at each open end is positioned such that one end mates with the throat/voice box area of the neck of the assistor and the other end mates with the throat/voice box area of the assisted. The speaking person (assistor) makes sounds that produce standing wave vibrations at the same frequency in the vocal cords of the assisted person. Driving the assisted person's vocal cords with the assisted person being able to hear the correct tone enables the assisted person to speak by simply amplifying the vibration of membranes in their throat.

  8. Speech recovery device

    Energy Technology Data Exchange (ETDEWEB)

    Frankle, Christen M.

    2000-10-19

    There is provided an apparatus and method for assisting speech recovery in people with inability to speak due to aphasia, apraxia or another condition with similar effect. A hollow, rigid, thin-walled tube with semi-circular or semi-elliptical cut out shapes at each open end is positioned such that one end mates with the throat/voice box area of the neck of the assistor and the other end mates with the throat/voice box area of the assisted. The speaking person (assistor) makes sounds that produce standing wave vibrations at the same frequency in the vocal cords of the assisted person. Driving the assisted person's vocal cords with the assisted person being able to hear the correct tone enables the assisted person to speak by simply amplifying the vibration of membranes in their throat.

  9. Emotion Recognition of Speech Signals Based on Filter Methods

    Directory of Open Access Journals (Sweden)

    Narjes Yazdanian

    2016-10-01

    Full Text Available Speech is the basic mean of communication among human beings.With the increase of transaction between human and machine, necessity of automatic dialogue and removing human factor has been considered. The aim of this study was to determine a set of affective features the speech signal is based on emotions. In this study system was designs that include three mains sections, features extraction, features selection and classification. After extraction of useful features such as, mel frequency cepstral coefficient (MFCC, linear prediction cepstral coefficients (LPC, perceptive linear prediction coefficients (PLP, ferment frequency, zero crossing rate, cepstral coefficients and pitch frequency, Mean, Jitter, Shimmer, Energy, Minimum, Maximum, Amplitude, Standard Deviation, at a later stage with filter methods such as Pearson Correlation Coefficient, t-test, relief and information gain, we came up with a method to rank and select effective features in emotion recognition. Then Result, are given to the classification system as a subset of input. In this classification stage, multi support vector machine are used to classify seven type of emotion. According to the results, that method of relief, together with multi support vector machine, has the most classification accuracy with emotion recognition rate of 93.94%.

  10. Machine protection systems

    CERN Document Server

    Macpherson, A L

    2010-01-01

    A summary of the Machine Protection System of the LHC is given, with particular attention given to the outstanding issues to be addressed, rather than the successes of the machine protection system from the 2009 run. In particular, the issues of Safe Machine Parameter system, collimation and beam cleaning, the beam dump system and abort gap cleaning, injection and dump protection, and the overall machine protection program for the upcoming run are summarised.

  11. Speech enhancement theory and practice

    CERN Document Server

    Loizou, Philipos C

    2013-01-01

    With the proliferation of mobile devices and hearing devices, including hearing aids and cochlear implants, there is a growing and pressing need to design algorithms that can improve speech intelligibility without sacrificing quality. Responding to this need, Speech Enhancement: Theory and Practice, Second Edition introduces readers to the basic problems of speech enhancement and the various algorithms proposed to solve these problems. Updated and expanded, this second edition of the bestselling textbook broadens its scope to include evaluation measures and enhancement algorithms aimed at impr

  12. Speaker diarization and speech recognition in the semi-automatization of audio description: An exploratory study on future possibilities?

    Directory of Open Access Journals (Sweden)

    Héctor Delgado

    2015-12-01

    Full Text Available This article presents an overview of the technological components used in the process of audio description, and suggests a new scenario in which speech recognition, machine translation, and text-to-speech, with the corresponding human revision, could be used to increase audio description provision. The article focuses on a process in which both speaker diarization and speech recognition are used in order to obtain a semi-automatic transcription of the audio description track. The technical process is presented and experimental results are summarized.

  13. Speaker diarization and speech recognition in the semi-automatization of audio description: An exploratory study on future possibilities?

    Directory of Open Access Journals (Sweden)

    Héctor Delgado

    2015-06-01

    This article presents an overview of the technological components used in the process of audio description, and suggests a new scenario in which speech recognition, machine translation, and text-to-speech, with the corresponding human revision, could be used to increase audio description provision. The article focuses on a process in which both speaker diarization and speech recognition are used in order to obtain a semi-automatic transcription of the audio description track. The technical process is presented and experimental results are summarized.

  14. Vocable Code

    DEFF Research Database (Denmark)

    Soon, Winnie; Cox, Geoff

    2018-01-01

    a computational and poetic composition for two screens: on one of these, texts and voices are repeated and disrupted by mathematical chaos, together exploring the performativity of code and language; on the other, is a mix of a computer programming syntax and human language. In this sense queer code can...... be understood as both an object and subject of study that intervenes in the world’s ‘becoming' and how material bodies are produced via human and nonhuman practices. Through mixing the natural and computer language, this article presents a script in six parts from a performative lecture for two persons...

  15. NSURE code

    International Nuclear Information System (INIS)

    Rattan, D.S.

    1993-11-01

    NSURE stands for Near-Surface Repository code. NSURE is a performance assessment code. developed for the safety assessment of near-surface disposal facilities for low-level radioactive waste (LLRW). Part one of this report documents the NSURE model, governing equations and formulation of the mathematical models, and their implementation under the SYVAC3 executive. The NSURE model simulates the release of nuclides from an engineered vault, their subsequent transport via the groundwater and surface water pathways tot he biosphere, and predicts the resulting dose rate to a critical individual. Part two of this report consists of a User's manual, describing simulation procedures, input data preparation, output and example test cases

  16. Quantum neural network based machine translator for Hindi to English.

    Science.gov (United States)

    Narayan, Ravi; Singh, V P; Chakraverty, S

    2014-01-01

    This paper presents the machine learning based machine translation system for Hindi to English, which learns the semantically correct corpus. The quantum neural based pattern recognizer is used to recognize and learn the pattern of corpus, using the information of part of speech of individual word in the corpus, like a human. The system performs the machine translation using its knowledge gained during the learning by inputting the pair of sentences of Devnagri-Hindi and English. To analyze the effectiveness of the proposed approach, 2600 sentences have been evaluated during simulation and evaluation. The accuracy achieved on BLEU score is 0.7502, on NIST score is 6.5773, on ROUGE-L score is 0.9233, and on METEOR score is 0.5456, which is significantly higher in comparison with Google Translation and Bing Translation for Hindi to English Machine Translation.

  17. Dictionary of machine terms

    International Nuclear Information System (INIS)

    1990-06-01

    This book has introduction of dictionary of machine terms, and a compilation committee and introductory remarks. It gives descriptions of the machine terms in alphabetical order from a to Z and also includes abbreviation of machine terms and symbol table, way to read mathematical symbols and abbreviation and terms of drawings.

  18. Mankind, machines and people

    Energy Technology Data Exchange (ETDEWEB)

    Hugli, A

    1984-01-01

    The following questions are addressed: is there a difference between machines and men, between human communication and communication with machines. Will we ever reach the point when the dream of artificial intelligence becomes a reality. Will thinking machines be able to replace the human spirit in all its aspects. Social consequences and philosophical aspects are addressed. 8 references.

  19. A Universal Reactive Machine

    DEFF Research Database (Denmark)

    Andersen, Henrik Reif; Mørk, Simon; Sørensen, Morten U.

    1997-01-01

    Turing showed the existence of a model universal for the set of Turing machines in the sense that given an encoding of any Turing machine asinput the universal Turing machine simulates it. We introduce the concept of universality for reactive systems and construct a CCS processuniversal...

  20. HTS machine laboratory prototype

    DEFF Research Database (Denmark)

    machine. The machine comprises six stationary HTS field windings wound from both YBCO and BiSCOO tape operated at liquid nitrogen temperature and enclosed in a cryostat, and a three phase armature winding spinning at up to 300 rpm. This design has full functionality of HTS synchronous machines. The design...

  1. Your Sewing Machine.

    Science.gov (United States)

    Peacock, Marion E.

    The programed instruction manual is designed to aid the student in learning the parts, uses, and operation of the sewing machine. Drawings of sewing machine parts are presented, and space is provided for the student's written responses. Following an introductory section identifying sewing machine parts, the manual deals with each part and its…

  2. Speech of people with autism: Echolalia and echolalic speech

    OpenAIRE

    Błeszyński, Jacek Jarosław

    2013-01-01

    Speech of people with autism is recognised as one of the basic diagnostic, therapeutic and theoretical problems. One of the most common symptoms of autism in children is echolalia, described here as being of different types and severity. This paper presents the results of studies into different levels of echolalia, both in normally developing children and in children diagnosed with autism, discusses the differences between simple echolalia and echolalic speech - which can be considered to b...

  3. Advocate: A Distributed Architecture for Speech-to-Speech Translation

    Science.gov (United States)

    2009-01-01

    tecture, are either wrapped natural-language processing ( NLP ) components or objects developed from scratch using the architecture’s API. GATE is...framework, we put together a demonstration Arabic -to- English speech translation system using both internally developed ( Arabic speech recognition and MT...conditions of our Arabic S2S demonstration system described earlier. Once again, the data size was varied and eighty identical requests were

  4. Manifold learning in machine vision and robotics

    Science.gov (United States)

    Bernstein, Alexander

    2017-02-01

    Smart algorithms are used in Machine vision and Robotics to organize or extract high-level information from the available data. Nowadays, Machine learning is an essential and ubiquitous tool to automate extraction patterns or regularities from data (images in Machine vision; camera, laser, and sonar sensors data in Robotics) in order to solve various subject-oriented tasks such as understanding and classification of images content, navigation of mobile autonomous robot in uncertain environments, robot manipulation in medical robotics and computer-assisted surgery, and other. Usually such data have high dimensionality, however, due to various dependencies between their components and constraints caused by physical reasons, all "feasible and usable data" occupy only a very small part in high dimensional "observation space" with smaller intrinsic dimensionality. Generally accepted model of such data is manifold model in accordance with which the data lie on or near an unknown manifold (surface) of lower dimensionality embedded in an ambient high dimensional observation space; real-world high-dimensional data obtained from "natural" sources meet, as a rule, this model. The use of Manifold learning technique in Machine vision and Robotics, which discovers a low-dimensional structure of high dimensional data and results in effective algorithms for solving of a large number of various subject-oriented tasks, is the content of the conference plenary speech some topics of which are in the paper.

  5. The Aster code; Code Aster

    Energy Technology Data Exchange (ETDEWEB)

    Delbecq, J.M

    1999-07-01

    The Aster code is a 2D or 3D finite-element calculation code for structures developed by the R and D direction of Electricite de France (EdF). This dossier presents a complete overview of the characteristics and uses of the Aster code: introduction of version 4; the context of Aster (organisation of the code development, versions, systems and interfaces, development tools, quality assurance, independent validation); static mechanics (linear thermo-elasticity, Euler buckling, cables, Zarka-Casier method); non-linear mechanics (materials behaviour, big deformations, specific loads, unloading and loss of load proportionality indicators, global algorithm, contact and friction); rupture mechanics (G energy restitution level, restitution level in thermo-elasto-plasticity, 3D local energy restitution level, KI and KII stress intensity factors, calculation of limit loads for structures), specific treatments (fatigue, rupture, wear, error estimation); meshes and models (mesh generation, modeling, loads and boundary conditions, links between different modeling processes, resolution of linear systems, display of results etc..); vibration mechanics (modal and harmonic analysis, dynamics with shocks, direct transient dynamics, seismic analysis and aleatory dynamics, non-linear dynamics, dynamical sub-structuring); fluid-structure interactions (internal acoustics, mass, rigidity and damping); linear and non-linear thermal analysis; steels and metal industry (structure transformations); coupled problems (internal chaining, internal thermo-hydro-mechanical coupling, chaining with other codes); products and services. (J.S.)

  6. Coding Class

    DEFF Research Database (Denmark)

    Ejsing-Duun, Stine; Hansbøl, Mikala

    Denne rapport rummer evaluering og dokumentation af Coding Class projektet1. Coding Class projektet blev igangsat i skoleåret 2016/2017 af IT-Branchen i samarbejde med en række medlemsvirksomheder, Københavns kommune, Vejle Kommune, Styrelsen for IT- og Læring (STIL) og den frivillige forening...... Coding Pirates2. Rapporten er forfattet af Docent i digitale læringsressourcer og forskningskoordinator for forsknings- og udviklingsmiljøet Digitalisering i Skolen (DiS), Mikala Hansbøl, fra Institut for Skole og Læring ved Professionshøjskolen Metropol; og Lektor i læringsteknologi, interaktionsdesign......, design tænkning og design-pædagogik, Stine Ejsing-Duun fra Forskningslab: It og Læringsdesign (ILD-LAB) ved Institut for kommunikation og psykologi, Aalborg Universitet i København. Vi har fulgt og gennemført evaluering og dokumentation af Coding Class projektet i perioden november 2016 til maj 2017...

  7. Uplink Coding

    Science.gov (United States)

    Andrews, Ken; Divsalar, Dariush; Dolinar, Sam; Moision, Bruce; Hamkins, Jon; Pollara, Fabrizio

    2007-01-01

    This slide presentation reviews the objectives, meeting goals and overall NASA goals for the NASA Data Standards Working Group. The presentation includes information on the technical progress surrounding the objective, short LDPC codes, and the general results on the Pu-Pw tradeoff.

  8. ANIMAL code

    International Nuclear Information System (INIS)

    Lindemuth, I.R.

    1979-01-01

    This report describes ANIMAL, a two-dimensional Eulerian magnetohydrodynamic computer code. ANIMAL's physical model also appears. Formulated are temporal and spatial finite-difference equations in a manner that facilitates implementation of the algorithm. Outlined are the functions of the algorithm's FORTRAN subroutines and variables

  9. Network Coding

    Indian Academy of Sciences (India)

    Home; Journals; Resonance – Journal of Science Education; Volume 15; Issue 7. Network Coding. K V Rashmi Nihar B Shah P Vijay Kumar. General Article Volume 15 Issue 7 July 2010 pp 604-621. Fulltext. Click here to view fulltext PDF. Permanent link: https://www.ias.ac.in/article/fulltext/reso/015/07/0604-0621 ...

  10. MCNP code

    International Nuclear Information System (INIS)

    Cramer, S.N.

    1984-01-01

    The MCNP code is the major Monte Carlo coupled neutron-photon transport research tool at the Los Alamos National Laboratory, and it represents the most extensive Monte Carlo development program in the United States which is available in the public domain. The present code is the direct descendent of the original Monte Carlo work of Fermi, von Neumaum, and Ulam at Los Alamos in the 1940s. Development has continued uninterrupted since that time, and the current version of MCNP (or its predecessors) has always included state-of-the-art methods in the Monte Carlo simulation of radiation transport, basic cross section data, geometry capability, variance reduction, and estimation procedures. The authors of the present code have oriented its development toward general user application. The documentation, though extensive, is presented in a clear and simple manner with many examples, illustrations, and sample problems. In addition to providing the desired results, the output listings give a a wealth of detailed information (some optional) concerning each state of the calculation. The code system is continually updated to take advantage of advances in computer hardware and software, including interactive modes of operation, diagnostic interrupts and restarts, and a variety of graphical and video aids

  11. Expander Codes

    Indian Academy of Sciences (India)

    Home; Journals; Resonance – Journal of Science Education; Volume 10; Issue 1. Expander Codes - The Sipser–Spielman Construction. Priti Shankar. General Article Volume 10 ... Author Affiliations. Priti Shankar1. Department of Computer Science and Automation, Indian Institute of Science Bangalore 560 012, India.

  12. Comprehension of synthetic speech and digitized natural speech by adults with aphasia.

    Science.gov (United States)

    Hux, Karen; Knollman-Porter, Kelly; Brown, Jessica; Wallace, Sarah E

    2017-09-01

    Using text-to-speech technology to provide simultaneous written and auditory content presentation may help compensate for chronic reading challenges if people with aphasia can understand synthetic speech output; however, inherent auditory comprehension challenges experienced by people with aphasia may make understanding synthetic speech difficult. This study's purpose was to compare the preferences and auditory comprehension accuracy of people with aphasia when listening to sentences generated with digitized natural speech, Alex synthetic speech (i.e., Macintosh platform), or David synthetic speech (i.e., Windows platform). The methodology required each of 20 participants with aphasia to select one of four images corresponding in meaning to each of 60 sentences comprising three stimulus sets. Results revealed significantly better accuracy given digitized natural speech than either synthetic speech option; however, individual participant performance analyses revealed three patterns: (a) comparable accuracy regardless of speech condition for 30% of participants, (b) comparable accuracy between digitized natural speech and one, but not both, synthetic speech option for 45% of participants, and (c) greater accuracy with digitized natural speech than with either synthetic speech option for remaining participants. Ranking and Likert-scale rating data revealed a preference for digitized natural speech and David synthetic speech over Alex synthetic speech. Results suggest many individuals with aphasia can comprehend synthetic speech options available on popular operating systems. Further examination of synthetic speech use to support reading comprehension through text-to-speech technology is thus warranted. Copyright © 2017 Elsevier Inc. All rights reserved.

  13. Computation of the bounce-average code

    International Nuclear Information System (INIS)

    Cutler, T.A.; Pearlstein, L.D.; Rensink, M.E.

    1977-01-01

    The bounce-average computer code simulates the two-dimensional velocity transport of ions in a mirror machine. The code evaluates and bounce-averages the collision operator and sources along the field line. A self-consistent equilibrium magnetic field is also computed using the long-thin approximation. Optionally included are terms that maintain μ, J invariance as the magnetic field changes in time. The assumptions and analysis that form the foundation of the bounce-average code are described. When references can be cited, the required results are merely stated and explained briefly. A listing of the code is appended

  14. Impact of dynamic rate coding aspects of mobile phone networks on forensic voice comparison.

    Science.gov (United States)

    Alzqhoul, Esam A S; Nair, Balamurali B T; Guillemin, Bernard J

    2015-09-01

    Previous studies have shown that landline and mobile phone networks are different in their ways of handling the speech signal, and therefore in their impact on it. But the same is also true of the different networks within the mobile phone arena. There are two major mobile phone technologies currently in use today, namely the global system for mobile communications (GSM) and code division multiple access (CDMA) and these are fundamentally different in their design. For example, the quality of the coded speech in the GSM network is a function of channel quality, whereas in the CDMA network it is determined by channel capacity (i.e., the number of users sharing a cell site). This paper examines the impact on the speech signal of a key feature of these networks, namely dynamic rate coding, and its subsequent impact on the task of likelihood-ratio-based forensic voice comparison (FVC). Surprisingly, both FVC accuracy and precision are found to be better for both GSM- and CDMA-coded speech than for uncoded. Intuitively one expects FVC accuracy to increase with increasing coded speech quality. This trend is shown to occur for the CDMA network, but, surprisingly, not for the GSM network. Further, in respect to comparisons between these two networks, FVC accuracy for CDMA-coded speech is shown to be slightly better than for GSM-coded speech, particularly when the coded-speech quality is high, but in terms of FVC precision the two networks are shown to be very similar. Copyright © 2015 The Chartered Society of Forensic Sciences. Published by Elsevier Ireland Ltd. All rights reserved.

  15. Speech Mannerisms: Games Clients Play

    Science.gov (United States)

    Morgan, Lewis B.

    1978-01-01

    This article focuses on speech mannerisms often employed by clients in a helping relationship. Eight mannerisms are presented and discussed, as well as possible interpretations. Suggestions are given to help counselors respond to them. (Author)

  16. Speech recognition from spectral dynamics

    Indian Academy of Sciences (India)

    Carrier nature of speech; modulation spectrum; spectral dynamics ... the relationships between phonetic values of sounds and their short-term spectral envelopes .... the number of free parameters that need to be estimated from training data.

  17. Stability analysis by ERATO code

    International Nuclear Information System (INIS)

    Tsunematsu, Toshihide; Takeda, Tatsuoki; Matsuura, Toshihiko; Azumi, Masafumi; Kurita, Gen-ichi

    1979-12-01

    Problems in MHD stability calculations by ERATO code are described; which concern convergence property of results, equilibrium codes, and machine optimization of ERATO code. It is concluded that irregularity on a convergence curve is not due to a fault of the ERATO code itself but due to inappropriate choice of the equilibrium calculation meshes. Also described are a code to calculate an equilibrium as a quasi-inverse problem and a code to calculate an equilibrium as a result of a transport process. Optimization of the code with respect to I/O operations reduced both CPU time and I/O time considerably. With the FACOM230-75 APU/CPU multiprocessor system, the performance is about 6 times as high as with the FACOM230-75 CPU, showing the effectiveness of a vector processing computer for the kind of MHD computations. This report is a summary of the material presented at the ERATO workshop 1979(ORNL), supplemented with some details. (author)

  18. Quantum machine learning.

    Science.gov (United States)

    Biamonte, Jacob; Wittek, Peter; Pancotti, Nicola; Rebentrost, Patrick; Wiebe, Nathan; Lloyd, Seth

    2017-09-13

    Fuelled by increasing computer power and algorithmic advances, machine learning techniques have become powerful tools for finding patterns in data. Quantum systems produce atypical patterns that classical systems are thought not to produce efficiently, so it is reasonable to postulate that quantum computers may outperform classical computers on machine learning tasks. The field of quantum machine learning explores how to devise and implement quantum software that could enable machine learning that is faster than that of classical computers. Recent work has produced quantum algorithms that could act as the building blocks of machine learning programs, but the hardware and software challenges are still considerable.

  19. Asynchronized synchronous machines

    CERN Document Server

    Botvinnik, M M

    1964-01-01

    Asynchronized Synchronous Machines focuses on the theoretical research on asynchronized synchronous (AS) machines, which are "hybrids” of synchronous and induction machines that can operate with slip. Topics covered in this book include the initial equations; vector diagram of an AS machine; regulation in cases of deviation from the law of full compensation; parameters of the excitation system; and schematic diagram of an excitation regulator. The possible applications of AS machines and its calculations in certain cases are also discussed. This publication is beneficial for students and indiv

  20. Designing speech for a recipient

    DEFF Research Database (Denmark)

    Fischer, Kerstin

    This study asks how speakers adjust their speech to their addressees, focusing on the potential roles of cognitive representations such as partner models, automatic processes such as interactive alignment, and social processes such as interactional negotiation. The nature of addressee orientation......, psycholinguistics and conversation analysis, and offers both overviews of child-directed, foreigner-directed and robot-directed speech and in-depth analyses of the processes involved in adjusting to a communication partner....

  1. National features of speech etiquette

    OpenAIRE

    Nacafova S.

    2017-01-01

    The article shows the differences between the speech etiquette of different peoples. The most important thing is to find a common language with this or that interlocutor. Knowledge of national etiquette, national character helps to learn the principles of speech of another nation. The article indicates in which cases certain forms of etiquette considered acceptable. At the same time, the rules of etiquette emphasized in the conduct of a dialogue in official meetings and for example, in the ex...

  2. Censored: Whistleblowers and impossible speech

    OpenAIRE

    Kenny, Kate

    2017-01-01

    What happens to a person who speaks out about corruption in their organization, and finds themselves excluded from their profession? In this article, I argue that whistleblowers experience exclusions because they have engaged in ‘impossible speech’, that is, a speech act considered to be unacceptable or illegitimate. Drawing on Butler’s theories of recognition and censorship, I show how norms of acceptable speech working through recruitment practices, alongside the actions of colleagues, can ...

  3. An object-oriented extension for debugging the virtual machine

    Energy Technology Data Exchange (ETDEWEB)

    Pizzi, Jr, Robert G. [Univ. of California, Davis, CA (United States)

    1994-12-01

    A computer is nothing more then a virtual machine programmed by source code to perform a task. The program`s source code expresses abstract constructs which are compiled into some lower level target language. When a virtual machine breaks, it can be very difficult to debug because typical debuggers provide only low-level target implementation information to the software engineer. We believe that the debugging task can be simplified by introducing aspects of the abstract design and data into the source code. We introduce OODIE, an object-oriented extension to programming languages that allows programmers to specify a virtual environment by describing the meaning of the design and data of a virtual machine. This specification is translated into symbolic information such that an augmented debugger can present engineers with a programmable debugging environment specifically tailored for the virtual machine that is to be debugged.

  4. Graphic man-machine interface applied to nuclear reactor designs

    International Nuclear Information System (INIS)

    Pereira, Claudio M.N.A; Mol, Antonio Carlos A.

    1999-01-01

    The Man-Machine Interfaces have been of interest of many researchers in the area of nuclear human factors engineering, principally applied to monitoring systems. The clarity of information provides best adaptation of the men to the machine. This work proposes the development of a Graphic Man-Machine Interface applied to nuclear reactor designs as a tool to optimize them. Here is present a prototype of a graphic man-machine interface for the Hammer code developed for PC under the Windows environment. The results of its application are commented. (author)

  5. Filtering, Coding, and Compression with Malvar Wavelets

    Science.gov (United States)

    1993-12-01

    speech coding techniques being investigated by the military (38). Imagery: Space imagery often requires adaptive restoration to deblur out-of-focus...and blurred image, find an estimate of the ideal image using a priori information about the blur, noise , and the ideal image" (12). The research for...recording can be described as the original signal convolved with impulses , which appear as echoes in the seismic event. The term deconvolution indicates

  6. Speech Function and Speech Role in Carl Fredricksen's Dialogue on Up Movie

    OpenAIRE

    Rehana, Ridha; Silitonga, Sortha

    2013-01-01

    One aim of this article is to show through a concrete example how speech function and speech role used in movie. The illustrative example is taken from the dialogue of Up movie. Central to the analysis proper form of dialogue on Up movie that contain of speech function and speech role; i.e. statement, offer, question, command, giving, and demanding. 269 dialogue were interpreted by actor, and it was found that the use of speech function and speech role.

  7. Effect of speech-intrinsic variations on human and automatic recognition of spoken phonemes.

    Science.gov (United States)

    Meyer, Bernd T; Brand, Thomas; Kollmeier, Birger

    2011-01-01

    The aim of this study is to quantify the gap between the recognition performance of human listeners and an automatic speech recognition (ASR) system with special focus on intrinsic variations of speech, such as speaking rate and effort, altered pitch, and the presence of dialect and accent. Second, it is investigated if the most common ASR features contain all information required to recognize speech in noisy environments by using resynthesized ASR features in listening experiments. For the phoneme recognition task, the ASR system achieved the human performance level only when the signal-to-noise ratio (SNR) was increased by 15 dB, which is an estimate for the human-machine gap in terms of the SNR. The major part of this gap is attributed to the feature extraction stage, since human listeners achieve comparable recognition scores when the SNR difference between unaltered and resynthesized utterances is 10 dB. Intrinsic variabilities result in strong increases of error rates, both in human speech recognition (HSR) and ASR (with a relative increase of up to 120%). An analysis of phoneme duration and recognition rates indicates that human listeners are better able to identify temporal cues than the machine at low SNRs, which suggests incorporating information about the temporal dynamics of speech into ASR systems.

  8. Exploring Australian speech-language pathologists' use and perceptions ofnon-speech oral motor exercises.

    Science.gov (United States)

    Rumbach, Anna F; Rose, Tanya A; Cheah, Mynn

    2018-01-29

    To explore Australian speech-language pathologists' use of non-speech oral motor exercises, and rationales for using/not using non-speech oral motor exercises in clinical practice. A total of 124 speech-language pathologists practising in Australia, working with paediatric and/or adult clients with speech sound difficulties, completed an online survey. The majority of speech-language pathologists reported that they did not use non-speech oral motor exercises when working with paediatric or adult clients with speech sound difficulties. However, more than half of the speech-language pathologists working with adult clients who have dysarthria reported using non-speech oral motor exercises with this population. The most frequently reported rationale for using non-speech oral motor exercises in speech sound difficulty management was to improve awareness/placement of articulators. The majority of speech-language pathologists agreed there is no clear clinical or research evidence base to support non-speech oral motor exercise use with clients who have speech sound difficulties. This study provides an overview of Australian speech-language pathologists' reported use and perceptions of non-speech oral motor exercises' applicability and efficacy in treating paediatric and adult clients who have speech sound difficulties. The research findings provide speech-language pathologists with insight into how and why non-speech oral motor exercises are currently used, and adds to the knowledge base regarding Australian speech-language pathology practice of non-speech oral motor exercises in the treatment of speech sound difficulties. Implications for Rehabilitation Non-speech oral motor exercises refer to oral motor activities which do not involve speech, but involve the manipulation or stimulation of oral structures including the lips, tongue, jaw, and soft palate. Non-speech oral motor exercises are intended to improve the function (e.g., movement, strength) of oral structures. The

  9. Subjective and objective measurement of the intelligibility of synthesized speech impaired by the very low bit rate STANAG 4591 codec including packet loss

    NARCIS (Netherlands)

    Počta, P.; Beerends, J.G.

    2017-01-01

    This paper deals with the intelligibility of speech coded by the STANAG 4591 standard codec, including packet loss, using synthesized speech input. Both subjective and objective assessments are used. It is shown that this codec significantly degrades intelligibility when compared to a standard

  10. CANAL code

    International Nuclear Information System (INIS)

    Gara, P.; Martin, E.

    1983-01-01

    The CANAL code presented here optimizes a realistic iron free extraction channel which has to provide a given transversal magnetic field law in the median plane: the current bars may be curved, have finite lengths and cooling ducts and move in a restricted transversal area; terminal connectors may be added, images of the bars in pole pieces may be included. A special option optimizes a real set of circular coils [fr

  11. Novel Techniques for Dialectal Arabic Speech Recognition

    CERN Document Server

    Elmahdy, Mohamed; Minker, Wolfgang

    2012-01-01

    Novel Techniques for Dialectal Arabic Speech describes approaches to improve automatic speech recognition for dialectal Arabic. Since speech resources for dialectal Arabic speech recognition are very sparse, the authors describe how existing Modern Standard Arabic (MSA) speech data can be applied to dialectal Arabic speech recognition, while assuming that MSA is always a second language for all Arabic speakers. In this book, Egyptian Colloquial Arabic (ECA) has been chosen as a typical Arabic dialect. ECA is the first ranked Arabic dialect in terms of number of speakers, and a high quality ECA speech corpus with accurate phonetic transcription has been collected. MSA acoustic models were trained using news broadcast speech. In order to cross-lingually use MSA in dialectal Arabic speech recognition, the authors have normalized the phoneme sets for MSA and ECA. After this normalization, they have applied state-of-the-art acoustic model adaptation techniques like Maximum Likelihood Linear Regression (MLLR) and M...

  12. Auditory analysis for speech recognition based on physiological models

    Science.gov (United States)

    Jeon, Woojay; Juang, Biing-Hwang

    2004-05-01

    To address the limitations of traditional cepstrum or LPC based front-end processing methods for automatic speech recognition, more elaborate methods based on physiological models of the human auditory system may be used to achieve more robust speech recognition in adverse environments. For this purpose, a modified version of a model of the primary auditory cortex featuring a three dimensional mapping of auditory spectra [Wang and Shamma, IEEE Trans. Speech Audio Process. 3, 382-395 (1995)] is adopted and investigated for its use as an improved front-end processing method. The study is conducted in two ways: first, by relating the model's redundant representation to traditional spectral representations and showing that the former not only encompasses information provided by the latter, but also reveals more relevant information that makes it superior in describing the identifying features of speech signals; and second, by observing the statistical features of the representation for various classes of sound to show how different identifying features manifest themselves as specific patterns on the cortical map, thereby becoming a place-coded data set on which detection theory could be applied to simulate auditory perception and cognition.

  13. Coding strategies for cochlear implants under adverse environments

    Science.gov (United States)

    Tahmina, Qudsia

    Cochlear implants are electronic prosthetic devices that restores partial hearing in patients with severe to profound hearing loss. Although most coding strategies have significantly improved the perception of speech in quite listening conditions, there remains limitations on speech perception under adverse environments such as in background noise, reverberation and band-limited channels, and we propose strategies that improve the intelligibility of speech transmitted over the telephone networks, reverberated speech and speech in the presence of background noise. For telephone processed speech, we propose to examine the effects of adding low-frequency and high- frequency information to the band-limited telephone speech. Four listening conditions were designed to simulate the receiving frequency characteristics of telephone handsets. Results indicated improvement in cochlear implant and bimodal listening when telephone speech was augmented with high frequency information and therefore this study provides support for design of algorithms to extend the bandwidth towards higher frequencies. The results also indicated added benefit from hearing aids for bimodal listeners in all four types of listening conditions. Speech understanding in acoustically reverberant environments is always a difficult task for hearing impaired listeners. Reverberated sounds consists of direct sound, early reflections and late reflections. Late reflections are known to be detrimental to speech intelligibility. In this study, we propose a reverberation suppression strategy based on spectral subtraction to suppress the reverberant energies from late reflections. Results from listening tests for two reverberant conditions (RT60 = 0.3s and 1.0s) indicated significant improvement when stimuli was processed with SS strategy. The proposed strategy operates with little to no prior information on the signal and the room characteristics and therefore, can potentially be implemented in real-time CI

  14. Experimental comparison between speech transmission index, rapid speech transmission index, and speech intelligibility index.

    Science.gov (United States)

    Larm, Petra; Hongisto, Valtteri

    2006-02-01

    During the acoustical design of, e.g., auditoria or open-plan offices, it is important to know how speech can be perceived in various parts of the room. Different objective methods have been developed to measure and predict speech intelligibility, and these have been extensively used in various spaces. In this study, two such methods were compared, the speech transmission index (STI) and the speech intelligibility index (SII). Also the simplification of the STI, the room acoustics speech transmission index (RASTI), was considered. These quantities are all based on determining an apparent speech-to-noise ratio on selected frequency bands and summing them using a specific weighting. For comparison, some data were needed on the possible differences of these methods resulting from the calculation scheme and also measuring equipment. Their prediction accuracy was also of interest. Measurements were made in a laboratory having adjustable noise level and absorption, and in a real auditorium. It was found that the measurement equipment, especially the selection of the loudspeaker, can greatly affect the accuracy of the results. The prediction accuracy of the RASTI was found acceptable, if the input values for the prediction are accurately known, even though the studied space was not ideally diffuse.

  15. Neural pathways for visual speech perception

    Directory of Open Access Journals (Sweden)

    Lynne E Bernstein

    2014-12-01

    Full Text Available This paper examines the questions, what levels of speech can be perceived visually, and how is visual speech represented by the brain? Review of the literature leads to the conclusions that every level of psycholinguistic speech structure (i.e., phonetic features, phonemes, syllables, words, and prosody can be perceived visually, although individuals differ in their abilities to do so; and that there are visual modality-specific representations of speech qua speech in higher-level vision brain areas. That is, the visual system represents the modal patterns of visual speech. The suggestion that the auditory speech pathway receives and represents visual speech is examined in light of neuroimaging evidence on the auditory speech pathways. We outline the generally agreed-upon organization of the visual ventral and dorsal pathways and examine several types of visual processing that might be related to speech through those pathways, specifically, face and body, orthography, and sign language processing. In this context, we examine the visual speech processing literature, which reveals widespread diverse patterns activity in posterior temporal cortices in response to visual speech stimuli. We outline a model of the visual and auditory speech pathways and make several suggestions: (1 The visual perception of speech relies on visual pathway representations of speech qua speech. (2 A proposed site of these representations, the temporal visual speech area (TVSA has been demonstrated in posterior temporal cortex, ventral and posterior to multisensory posterior superior temporal sulcus (pSTS. (3 Given that visual speech has dynamic and configural features, its representations in feedforward visual pathways are expected to integrate these features, possibly in TVSA.

  16. Performance evaluation based on data from code reviews

    OpenAIRE

    Andrej, Sekáč

    2016-01-01

    Context. Modern code review tools such as Gerrit have made available great amounts of code review data from different open source projects as well as other commercial projects. Code reviews are used to keep the quality of produced source code under control but the stored data could also be used for evaluation of the software development process. Objectives. This thesis uses machine learning methods for an approximation of review expert’s performance evaluation function. Due to limitations in ...

  17. High Order Tensor Formulation for Convolutional Sparse Coding

    KAUST Repository

    Bibi, Adel Aamer; Ghanem, Bernard

    2017-01-01

    Convolutional sparse coding (CSC) has gained attention for its successful role as a reconstruction and a classification tool in the computer vision and machine learning community. Current CSC methods can only reconstruct singlefeature 2D images

  18. Blind estimation of the number of speech source in reverberant multisource scenarios based on binaural signals

    DEFF Research Database (Denmark)

    May, Tobias; van de Par, Steven

    2012-01-01

    In this paper we present a new approach for estimating the number of active speech sources in the presence of interfering noise sources and reverberation. First, a binaural front-end is used to detect the spatial positions of all active sound sources, resulting in a binary mask for each candidate...... on a support vector machine (SVM) classifier. A systematic analysis shows that the proposed algorithm is able to blindly determine the number and the corresponding spatial positions of speech sources in multisource scenarios and generalizes well to unknown acoustic conditions...

  19. I2D: code for conversion of ISOTXS structured data to DTF and ANISN structured tables

    International Nuclear Information System (INIS)

    Resnik, W.M. II.

    1977-06-01

    The I2D code converts neutron cross-section data written in the standard interface file format called ISOTXS to a matrix structured format commonly called DTF tables. Several BCD and binary output options are available including FIDO (ANISN) format. The I2D code adheres to the guidelines established by the Committee on Computer Code Coordination for standardized code development. Since some machine dependency is inherent regardless of the degree of standardization, provisions have been made in the I2D code for easy implementation on either short-word machines (IBM) or on long-word machines (CDC). 3 figures, 5 tables

  20. Private speech in teacher-learner interactions in an EFL context: A sociocultural perspective

    Directory of Open Access Journals (Sweden)

    Nouzar Gheisari

    2017-07-01

    Full Text Available Theoretically framed within Vygotskyan sociocultural theory (SCT of mind, the present study investigated resurfacing of private speech markers by Iranian elementary female EFL learners in teacher-learner interactions. To this end, an elementary EFL class including 12 female learners and a same-sex teacher were selected as the participants of the study. As for the data, six 30-minute reading comprehension tasks with the interval of every two weeks were videotaped, while each participant was provided with a sensitive MP3 player to keep track of very low private speech markers. Instances of externalized private speech markers were coded and reports were generated for the patterns of private speech markers regarding their form and content. While a high number of literal translation, metalanguage, and switching to L1 mid-utterance were reported, the generated number of such private markers as self-directed questions, reading aloud, reviewing, and self-explanations in L2 was comparatively less which could be due to low L2 proficiency of the learners. The findings of the study, besides highlighting the importance of paying more attention to private speech as a mediating tool in cognitive regulation of learners in doing tasks in L2, suggest that teachers’ type of classroom practice is effective in production of private speech. Pedagogically speaking, the results suggest that instead of seeing L1 private speech markers as detrimental to L2 learning, they should be seen as signs of cognitive regulation when facing challenging tasks.

  1. Influence of musical training on understanding voiced and whispered speech in noise.

    Science.gov (United States)

    Ruggles, Dorea R; Freyman, Richard L; Oxenham, Andrew J

    2014-01-01

    This study tested the hypothesis that the previously reported advantage of musicians over non-musicians in understanding speech in noise arises from more efficient or robust coding of periodic voiced speech, particularly in fluctuating backgrounds. Speech intelligibility was measured in listeners with extensive musical training, and in those with very little musical training or experience, using normal (voiced) or whispered (unvoiced) grammatically correct nonsense sentences in noise that was spectrally shaped to match the long-term spectrum of the speech, and was either continuous or gated with a 16-Hz square wave. Performance was also measured in clinical speech-in-noise tests and in pitch discrimination. Musicians exhibited enhanced pitch discrimination, as expected. However, no systematic or statistically significant advantage for musicians over non-musicians was found in understanding either voiced or whispered sentences in either continuous or gated noise. Musicians also showed no statistically significant advantage in the clinical speech-in-noise tests. Overall, the results provide no evidence for a significant difference between young adult musicians and non-musicians in their ability to understand speech in noise.

  2. Detection of target phonemes in spontaneous and read speech

    NARCIS (Netherlands)

    Mehta, G.; Cutler, A.

    1988-01-01

    Although spontaneous speech occurs more frequently in most listeners' experience than read speech, laboratory studies of human speech recognition typically use carefully controlled materials read from a script. The phonological and prosodic characteristics of spontaneous and read speech differ

  3. A Very Fast and Angular Momentum Conserving Tree Code

    International Nuclear Information System (INIS)

    Marcello, Dominic C.

    2017-01-01

    There are many methods used to compute the classical gravitational field in astrophysical simulation codes. With the exception of the typically impractical method of direct computation, none ensure conservation of angular momentum to machine precision. Under uniform time-stepping, the Cartesian fast multipole method of Dehnen (also known as the very fast tree code) conserves linear momentum to machine precision. We show that it is possible to modify this method in a way that conserves both angular and linear momenta.

  4. A Very Fast and Angular Momentum Conserving Tree Code

    Energy Technology Data Exchange (ETDEWEB)

    Marcello, Dominic C., E-mail: dmarce504@gmail.com [Department of Physics and Astronomy, and Center for Computation and Technology Louisiana State University, Baton Rouge, LA 70803 (United States)

    2017-09-01

    There are many methods used to compute the classical gravitational field in astrophysical simulation codes. With the exception of the typically impractical method of direct computation, none ensure conservation of angular momentum to machine precision. Under uniform time-stepping, the Cartesian fast multipole method of Dehnen (also known as the very fast tree code) conserves linear momentum to machine precision. We show that it is possible to modify this method in a way that conserves both angular and linear momenta.

  5. Pattern recognition & machine learning

    CERN Document Server

    Anzai, Y

    1992-01-01

    This is the first text to provide a unified and self-contained introduction to visual pattern recognition and machine learning. It is useful as a general introduction to artifical intelligence and knowledge engineering, and no previous knowledge of pattern recognition or machine learning is necessary. Basic for various pattern recognition and machine learning methods. Translated from Japanese, the book also features chapter exercises, keywords, and summaries.

  6. Support vector machines applications

    CERN Document Server

    Guo, Guodong

    2014-01-01

    Support vector machines (SVM) have both a solid mathematical background and good performance in practical applications. This book focuses on the recent advances and applications of the SVM in different areas, such as image processing, medical practice, computer vision, pattern recognition, machine learning, applied statistics, business intelligence, and artificial intelligence. The aim of this book is to create a comprehensive source on support vector machine applications, especially some recent advances.

  7. The Newest Machine Material

    International Nuclear Information System (INIS)

    Seo, Yeong Seop; Choe, Byeong Do; Bang, Meong Sung

    2005-08-01

    This book gives descriptions of machine material with classification of machine material and selection of machine material, structure and connection of material, coagulation of metal and crystal structure, equilibrium diagram, properties of metal material, elasticity and plasticity, biopsy of metal, material test and nondestructive test. It also explains steel material such as heat treatment of steel, cast iron and cast steel, nonferrous metal materials, non metallic materials, and new materials.

  8. Introduction to machine learning

    OpenAIRE

    Baştanlar, Yalın; Özuysal, Mustafa

    2014-01-01

    The machine learning field, which can be briefly defined as enabling computers make successful predictions using past experiences, has exhibited an impressive development recently with the help of the rapid increase in the storage capacity and processing power of computers. Together with many other disciplines, machine learning methods have been widely employed in bioinformatics. The difficulties and cost of biological analyses have led to the development of sophisticated machine learning app...

  9. Machinability of advanced materials

    CERN Document Server

    Davim, J Paulo

    2014-01-01

    Machinability of Advanced Materials addresses the level of difficulty involved in machining a material, or multiple materials, with the appropriate tooling and cutting parameters.  A variety of factors determine a material's machinability, including tool life rate, cutting forces and power consumption, surface integrity, limiting rate of metal removal, and chip shape. These topics, among others, and multiple examples comprise this research resource for engineering students, academics, and practitioners.

  10. Machining of titanium alloys

    CERN Document Server

    2014-01-01

    This book presents a collection of examples illustrating the resent research advances in the machining of titanium alloys. These materials have excellent strength and fracture toughness as well as low density and good corrosion resistance; however, machinability is still poor due to their low thermal conductivity and high chemical reactivity with cutting tool materials. This book presents solutions to enhance machinability in titanium-based alloys and serves as a useful reference to professionals and researchers in aerospace, automotive and biomedical fields.

  11. Difficulty understanding speech in noise by the hearing impaired: underlying causes and technological solutions.

    Science.gov (United States)

    Healy, Eric W; Yoho, Sarah E

    2016-08-01

    A primary complaint of hearing-impaired individuals involves poor speech understanding when background noise is present. Hearing aids and cochlear implants often allow good speech understanding in quiet backgrounds. But hearing-impaired individuals are highly noise intolerant, and existing devices are not very effective at combating background noise. As a result, speech understanding in noise is often quite poor. In accord with the significance of the problem, considerable effort has been expended toward understanding and remedying this issue. Fortunately, our understanding of the underlying issues is reasonably good. In sharp contrast, effective solutions have remained elusive. One solution that seems promising involves a single-microphone machine-learning algorithm to extract speech from background noise. Data from our group indicate that the algorithm is capable of producing vast increases in speech understanding by hearing-impaired individuals. This paper will first provide an overview of the speech-in-noise problem and outline why hearing-impaired individuals are so noise intolerant. An overview of our approach to solving this problem will follow.

  12. Development an Automatic Speech to Facial Animation Conversion for Improve Deaf Lives

    Directory of Open Access Journals (Sweden)

    S. Hamidreza Kasaei

    2011-05-01

    Full Text Available In this paper, we propose design and initial implementation of a robust system which can automatically translates voice into text and text to sign language animations. Sign Language
    Translation Systems could significantly improve deaf lives especially in communications, exchange of information and employment of machine for translation conversations from one language to another has. Therefore, considering these points, it seems necessary to study the speech recognition. Usually, the voice recognition algorithms address three major challenges. The first is extracting feature form speech and the second is when limited sound gallery are available for recognition, and the final challenge is to improve speaker dependent to speaker independent voice recognition. Extracting feature form speech is an important stage in our method. Different procedures are available for extracting feature form speech. One of the commonest of which used in speech
    recognition systems is Mel-Frequency Cepstral Coefficients (MFCCs. The algorithm starts with preprocessing and signal conditioning. Next extracting feature form speech using Cepstral coefficients will be done. Then the result of this process sends to segmentation part. Finally recognition part recognizes the words and then converting word recognized to facial animation. The project is still in progress and some new interesting methods are described in the current report.

  13. Tribology in machine design

    CERN Document Server

    Stolarski, Tadeusz

    1999-01-01

    ""Tribology in Machine Design is strongly recommended for machine designers, and engineers and scientists interested in tribology. It should be in the engineering library of companies producing mechanical equipment.""Applied Mechanics ReviewTribology in Machine Design explains the role of tribology in the design of machine elements. It shows how algorithms developed from the basic principles of tribology can be used in a range of practical applications within mechanical devices and systems.The computer offers today's designer the possibility of greater stringen

  14. Induction machine handbook

    CERN Document Server

    Boldea, Ion

    2002-01-01

    Often called the workhorse of industry, the advent of power electronics and advances in digital control are transforming the induction motor into the racehorse of industrial motion control. Now, the classic texts on induction machines are nearly three decades old, while more recent books on electric motors lack the necessary depth and detail on induction machines.The Induction Machine Handbook fills industry's long-standing need for a comprehensive treatise embracing the many intricate facets of induction machine analysis and design. Moving gradually from simple to complex and from standard to

  15. Chaotic Boltzmann machines

    Science.gov (United States)

    Suzuki, Hideyuki; Imura, Jun-ichi; Horio, Yoshihiko; Aihara, Kazuyuki

    2013-01-01

    The chaotic Boltzmann machine proposed in this paper is a chaotic pseudo-billiard system that works as a Boltzmann machine. Chaotic Boltzmann machines are shown numerically to have computing abilities comparable to conventional (stochastic) Boltzmann machines. Since no randomness is required, efficient hardware implementation is expected. Moreover, the ferromagnetic phase transition of the Ising model is shown to be characterised by the largest Lyapunov exponent of the proposed system. In general, a method to relate probabilistic models to nonlinear dynamics by derandomising Gibbs sampling is presented. PMID:23558425

  16. Electrical machines & drives

    CERN Document Server

    Hammond, P

    1985-01-01

    Containing approximately 200 problems (100 worked), the text covers a wide range of topics concerning electrical machines, placing particular emphasis upon electrical-machine drive applications. The theory is concisely reviewed and focuses on features common to all machine types. The problems are arranged in order of increasing levels of complexity and discussions of the solutions are included where appropriate to illustrate the engineering implications. This second edition includes an important new chapter on mathematical and computer simulation of machine systems and revised discussions o

  17. Nanocomposites for Machining Tools

    Directory of Open Access Journals (Sweden)

    Daria Sidorenko

    2017-10-01

    Full Text Available Machining tools are used in many areas of production. To a considerable extent, the performance characteristics of the tools determine the quality and cost of obtained products. The main materials used for producing machining tools are steel, cemented carbides, ceramics and superhard materials. A promising way to improve the performance characteristics of these materials is to design new nanocomposites based on them. The application of micromechanical modeling during the elaboration of composite materials for machining tools can reduce the financial and time costs for development of new tools, with enhanced performance. This article reviews the main groups of nanocomposites for machining tools and their performance.

  18. Machine listening intelligence

    Science.gov (United States)

    Cella, C. E.

    2017-05-01

    This manifesto paper will introduce machine listening intelligence, an integrated research framework for acoustic and musical signals modelling, based on signal processing, deep learning and computational musicology.

  19. Machine learning with R

    CERN Document Server

    Lantz, Brett

    2013-01-01

    Written as a tutorial to explore and understand the power of R for machine learning. This practical guide that covers all of the need to know topics in a very systematic way. For each machine learning approach, each step in the process is detailed, from preparing the data for analysis to evaluating the results. These steps will build the knowledge you need to apply them to your own data science tasks.Intended for those who want to learn how to use R's machine learning capabilities and gain insight from your data. Perhaps you already know a bit about machine learning, but have never used R; or

  20. Rotating electrical machines

    CERN Document Server

    Le Doeuff, René

    2013-01-01

    In this book a general matrix-based approach to modeling electrical machines is promulgated. The model uses instantaneous quantities for key variables and enables the user to easily take into account associations between rotating machines and static converters (such as in variable speed drives).   General equations of electromechanical energy conversion are established early in the treatment of the topic and then applied to synchronous, induction and DC machines. The primary characteristics of these machines are established for steady state behavior as well as for variable speed scenarios. I

  1. Noise-robust speech triage.

    Science.gov (United States)

    Bartos, Anthony L; Cipr, Tomas; Nelson, Douglas J; Schwarz, Petr; Banowetz, John; Jerabek, Ladislav

    2018-04-01

    A method is presented in which conventional speech algorithms are applied, with no modifications, to improve their performance in extremely noisy environments. It has been demonstrated that, for eigen-channel algorithms, pre-training multiple speaker identification (SID) models at a lattice of signal-to-noise-ratio (SNR) levels and then performing SID using the appropriate SNR dependent model was successful in mitigating noise at all SNR levels. In those tests, it was found that SID performance was optimized when the SNR of the testing and training data were close or identical. In this current effort multiple i-vector algorithms were used, greatly improving both processing throughput and equal error rate classification accuracy. Using identical approaches in the same noisy environment, performance of SID, language identification, gender identification, and diarization were significantly improved. A critical factor in this improvement is speech activity detection (SAD) that performs reliably in extremely noisy environments, where the speech itself is barely audible. To optimize SAD operation at all SNR levels, two algorithms were employed. The first maximized detection probability at low levels (-10 dB ≤ SNR < +10 dB) using just the voiced speech envelope, and the second exploited features extracted from the original speech to improve overall accuracy at higher quality levels (SNR ≥ +10 dB).

  2. No, there is no 150 ms lead of visual speech on auditory speech, but a range of audiovisual asynchronies varying from small audio lead to large audio lag.

    Directory of Open Access Journals (Sweden)

    Jean-Luc Schwartz

    2014-07-01

    Full Text Available An increasing number of neuroscience papers capitalize on the assumption published in this journal that visual speech would be typically 150 ms ahead of auditory speech. It happens that the estimation of audiovisual asynchrony in the reference paper is valid only in very specific cases, for isolated consonant-vowel syllables or at the beginning of a speech utterance, in what we call "preparatory gestures". However, when syllables are chained in sequences, as they are typically in most parts of a natural speech utterance, asynchrony should be defined in a different way. This is what we call "comodulatory gestures" providing auditory and visual events more or less in synchrony. We provide audiovisual data on sequences of plosive-vowel syllables (pa, ta, ka, ba, da, ga, ma, na showing that audiovisual synchrony is actually rather precise, varying between 20 ms audio lead and 70 ms audio lag. We show how more complex speech material should result in a range typically varying between 40 ms audio lead and 200 ms audio lag, and we discuss how this natural coordination is reflected in the so-called temporal integration window for audiovisual speech perception. Finally we present a toy model of auditory and audiovisual predictive coding, showing that visual lead is actually not necessary for visual prediction.

  3. Making extreme computations possible with virtual machines

    International Nuclear Information System (INIS)

    Reuter, J.; Chokoufe Nejad, B.

    2016-02-01

    State-of-the-art algorithms generate scattering amplitudes for high-energy physics at leading order for high-multiplicity processes as compiled code (in Fortran, C or C++). For complicated processes the size of these libraries can become tremendous (many GiB). We show that amplitudes can be translated to byte-code instructions, which even reduce the size by one order of magnitude. The byte-code is interpreted by a Virtual Machine with runtimes comparable to compiled code and a better scaling with additional legs. We study the properties of this algorithm, as an extension of the Optimizing Matrix Element Generator (O'Mega). The bytecode matrix elements are available as alternative input for the event generator WHIZARD. The bytecode interpreter can be implemented very compactly, which will help with a future implementation on massively parallel GPUs.

  4. From concatenated codes to graph codes

    DEFF Research Database (Denmark)

    Justesen, Jørn; Høholdt, Tom

    2004-01-01

    We consider codes based on simple bipartite expander graphs. These codes may be seen as the first step leading from product type concatenated codes to more complex graph codes. We emphasize constructions of specific codes of realistic lengths, and study the details of decoding by message passing...

  5. Are there intelligent Turing machines?

    OpenAIRE

    Bátfai, Norbert

    2015-01-01

    This paper introduces a new computing model based on the cooperation among Turing machines called orchestrated machines. Like universal Turing machines, orchestrated machines are also designed to simulate Turing machines but they can also modify the original operation of the included Turing machines to create a new layer of some kind of collective behavior. Using this new model we can define some interested notions related to cooperation ability of Turing machines such as the intelligence quo...

  6. Real time implementation of a linear predictive coding algorithm on digital signal processor DSP32C

    International Nuclear Information System (INIS)

    Sheikh, N.M.; Usman, S.R.; Fatima, S.

    2002-01-01

    Pulse Code Modulation (PCM) has been widely used in speech coding. However, due to its high bit rate. PCM has severe limitations in application where high spectral efficiency is desired, for example, in mobile communication, CD quality broadcasting system etc. These limitation have motivated research in bit rate reduction techniques. Linear predictive coding (LPC) is one of the most powerful complex techniques for bit rate reduction. With the introduction of powerful digital signal processors (DSP) it is possible to implement the complex LPC algorithm in real time. In this paper we present a real time implementation of the LPC algorithm on AT and T's DSP32C at a sampling frequency of 8192 HZ. Application of the LPC algorithm on two speech signals is discussed. Using this implementation , a bit rate reduction of 1:3 is achieved for better than tool quality speech, while a reduction of 1.16 is possible for speech quality required in military applications. (author)

  7. Speech Inconsistency in Children with Childhood Apraxia of Speech, Language Impairment, and Speech Delay: Depends on the Stimuli

    Science.gov (United States)

    Iuzzini-Seigel, Jenya; Hogan, Tiffany P.; Green, Jordan R.

    2017-01-01

    Purpose: The current research sought to determine (a) if speech inconsistency is a core feature of childhood apraxia of speech (CAS) or if it is driven by comorbid language impairment that affects a large subset of children with CAS and (b) if speech inconsistency is a sensitive and specific diagnostic marker that can differentiate between CAS and…

  8. Beam Dynamics Studies in Recirculating Machines

    CERN Document Server

    Pellegrini, Dario; Latina, A

    The LHeC and the CLIC Drive Beam share not only the high-current beams that make them prone to show instabilities, but also unconventional lattice topologies and operational schemes in which the time sequence of the bunches varies along the machine. In order to asses the feasibility of these projects, realistic simulations taking into account the most worrisome effects and their interplays, are crucial. These include linear and non-linear optics with time dependent elements, incoherent and coherent synchrotron radiation, short and long-range wakefields, beam-beam effect and ion cloud. In order to investigate multi-bunch effects in recirculating machines, a new version of the tracking code PLACET has been developed from scratch. PLACET2, already integrates most of the effects mentioned before and can easily receive additional physics. Its innovative design allows to describe complex lattices and track one or more bunches accordingly to the machine operation, reproducing the bunch train splitting and recombinat...

  9. Voice based gender classification using machine learning

    Science.gov (United States)

    Raahul, A.; Sapthagiri, R.; Pankaj, K.; Vijayarajan, V.

    2017-11-01

    Gender identification is one of the major problem speech analysis today. Tracing the gender from acoustic data i.e., pitch, median, frequency etc. Machine learning gives promising results for classification problem in all the research domains. There are several performance metrics to evaluate algorithms of an area. Our Comparative model algorithm for evaluating 5 different machine learning algorithms based on eight different metrics in gender classification from acoustic data. Agenda is to identify gender, with five different algorithms: Linear Discriminant Analysis (LDA), K-Nearest Neighbour (KNN), Classification and Regression Trees (CART), Random Forest (RF), and Support Vector Machine (SVM) on basis of eight different metrics. The main parameter in evaluating any algorithms is its performance. Misclassification rate must be less in classification problems, which says that the accuracy rate must be high. Location and gender of the person have become very crucial in economic markets in the form of AdSense. Here with this comparative model algorithm, we are trying to assess the different ML algorithms and find the best fit for gender classification of acoustic data.

  10. Variable Span Filters for Speech Enhancement

    DEFF Research Database (Denmark)

    Jensen, Jesper Rindom; Benesty, Jacob; Christensen, Mads Græsbøll

    2016-01-01

    In this work, we consider enhancement of multichannel speech recordings. Linear filtering and subspace approaches have been considered previously for solving the problem. The current linear filtering methods, although many variants exist, have limited control of noise reduction and speech...

  11. Represented Speech in Qualitative Health Research

    DEFF Research Database (Denmark)

    Musaeus, Peter

    2017-01-01

    Represented speech refers to speech where we reference somebody. Represented speech is an important phenomenon in everyday conversation, health care communication, and qualitative research. This case will draw first from a case study on physicians’ workplace learning and second from a case study...... on nurses’ apprenticeship learning. The aim of the case is to guide the qualitative researcher to use own and others’ voices in the interview and to be sensitive to represented speech in everyday conversation. Moreover, reported speech matters to health professionals who aim to represent the voice...... of their patients. Qualitative researchers and students might learn to encourage interviewees to elaborate different voices or perspectives. Qualitative researchers working with natural speech might pay attention to how people talk and use represented speech. Finally, represented speech might be relevant...

  12. Quick Statistics about Voice, Speech, and Language

    Science.gov (United States)

    ... here Home » Health Info » Statistics and Epidemiology Quick Statistics About Voice, Speech, Language Voice, Speech, Language, and ... no 205. Hyattsville, MD: National Center for Health Statistics. 2015. Hoffman HJ, Li C-M, Losonczy K, ...

  13. A NOVEL APPROACH TO STUTTERED SPEECH CORRECTION

    Directory of Open Access Journals (Sweden)

    Alim Sabur Ajibola

    2016-06-01

    Full Text Available Stuttered speech is a dysfluency rich speech, more prevalent in males than females. It has been associated with insufficient air pressure or poor articulation, even though the root causes are more complex. The primary features include prolonged speech and repetitive speech, while some of its secondary features include, anxiety, fear, and shame. This study used LPC analysis and synthesis algorithms to reconstruct the stuttered speech. The results were evaluated using cepstral distance, Itakura-Saito distance, mean square error, and likelihood ratio. These measures implied perfect speech reconstruction quality. ASR was used for further testing, and the results showed that all the reconstructed speech samples were perfectly recognized while only three samples of the original speech were perfectly recognized.

  14. Developmental language and speech disability.

    Science.gov (United States)

    Spiel, G; Brunner, E; Allmayer, B; Pletz, A

    2001-09-01

    Speech disabilities (articulation deficits) and language disorders--expressive (vocabulary) receptive (language comprehension) are not uncommon in children. An overview of these along with a global description of the impairment of communication as well as clinical characteristics of language developmental disorders are presented in this article. The diagnostic tables, which are applied in the European and Anglo-American speech areas, ICD-10 and DSM-IV, have been explained and compared. Because of their strengths and weaknesses an alternative classification of language and speech developmental disorders is proposed, which allows a differentiation between expressive and receptive language capabilities with regard to the semantic and the morphological/syntax domains. Prevalence and comorbidity rates, psychosocial influences, biological factors and the biological social interaction have been discussed. The necessity of the use of standardized examinations is emphasised. General logopaedic treatment paradigms, specific therapy concepts and an overview of prognosis have been described.

  15. Monte Carlo codes and Monte Carlo simulator program

    International Nuclear Information System (INIS)

    Higuchi, Kenji; Asai, Kiyoshi; Suganuma, Masayuki.

    1990-03-01

    Four typical Monte Carlo codes KENO-IV, MORSE, MCNP and VIM have been vectorized on VP-100 at Computing Center, JAERI. The problems in vector processing of Monte Carlo codes on vector processors have become clear through the work. As the result, it is recognized that these are difficulties to obtain good performance in vector processing of Monte Carlo codes. A Monte Carlo computing machine, which processes the Monte Carlo codes with high performances is being developed at our Computing Center since 1987. The concept of Monte Carlo computing machine and its performance have been investigated and estimated by using a software simulator. In this report the problems in vectorization of Monte Carlo codes, Monte Carlo pipelines proposed to mitigate these difficulties and the results of the performance estimation of the Monte Carlo computing machine by the simulator are described. (author)

  16. Motor Speech Phenotypes of Frontotemporal Dementia, Primary Progressive Aphasia, and Progressive Apraxia of Speech

    Science.gov (United States)

    Poole, Matthew L.; Brodtmann, Amy; Darby, David; Vogel, Adam P.

    2017-01-01

    Purpose: Our purpose was to create a comprehensive review of speech impairment in frontotemporal dementia (FTD), primary progressive aphasia (PPA), and progressive apraxia of speech in order to identify the most effective measures for diagnosis and monitoring, and to elucidate associations between speech and neuroimaging. Method: Speech and…

  17. Visual context enhanced. The joint contribution of iconic gestures and visible speech to degraded speech comprehension.

    NARCIS (Netherlands)

    Drijvers, L.; Özyürek, A.

    2017-01-01

    Purpose: This study investigated whether and to what extent iconic co-speech gestures contribute to information from visible speech to enhance degraded speech comprehension at different levels of noise-vocoding. Previous studies of the contributions of these 2 visual articulators to speech

  18. Listeners Experience Linguistic Masking Release in Noise-Vocoded Speech-in-Speech Recognition

    Science.gov (United States)

    Viswanathan, Navin; Kokkinakis, Kostas; Williams, Brittany T.

    2018-01-01

    Purpose: The purpose of this study was to evaluate whether listeners with normal hearing perceiving noise-vocoded speech-in-speech demonstrate better intelligibility of target speech when the background speech was mismatched in language (linguistic release from masking [LRM]) and/or location (spatial release from masking [SRM]) relative to the…

  19. Predicting Speech Intelligibility with a Multiple Speech Subsystems Approach in Children with Cerebral Palsy

    Science.gov (United States)

    Lee, Jimin; Hustad, Katherine C.; Weismer, Gary

    2014-01-01

    Purpose: Speech acoustic characteristics of children with cerebral palsy (CP) were examined with a multiple speech subsystems approach; speech intelligibility was evaluated using a prediction model in which acoustic measures were selected to represent three speech subsystems. Method: Nine acoustic variables reflecting different subsystems, and…

  20. Visual Context Enhanced: The Joint Contribution of Iconic Gestures and Visible Speech to Degraded Speech Comprehension

    Science.gov (United States)

    Drijvers, Linda; Ozyurek, Asli

    2017-01-01

    Purpose: This study investigated whether and to what extent iconic co-speech gestures contribute to information from visible speech to enhance degraded speech comprehension at different levels of noise-vocoding. Previous studies of the contributions of these 2 visual articulators to speech comprehension have only been performed separately. Method:…

  1. An experimental Dutch keyboard-to-speech system for the speech impaired

    NARCIS (Netherlands)

    Deliege, R.J.H.

    1989-01-01

    An experimental Dutch keyboard-to-speech system has been developed to explor the possibilities and limitations of Dutch speech synthesis in a communication aid for the speech impaired. The system uses diphones and a formant synthesizer chip for speech synthesis. Input to the system is in

  2. Perceived Liveliness and Speech Comprehensibility in Aphasia: The Effects of Direct Speech in Auditory Narratives

    Science.gov (United States)

    Groenewold, Rimke; Bastiaanse, Roelien; Nickels, Lyndsey; Huiskes, Mike

    2014-01-01

    Background: Previous studies have shown that in semi-spontaneous speech, individuals with Broca's and anomic aphasia produce relatively many direct speech constructions. It has been claimed that in "healthy" communication direct speech constructions contribute to the liveliness, and indirectly to the comprehensibility, of speech.…

  3. Poor Speech Perception Is Not a Core Deficit of Childhood Apraxia of Speech: Preliminary Findings

    Science.gov (United States)

    Zuk, Jennifer; Iuzzini-Seigel, Jenya; Cabbage, Kathryn; Green, Jordan R.; Hogan, Tiffany P.

    2018-01-01

    Purpose: Childhood apraxia of speech (CAS) is hypothesized to arise from deficits in speech motor planning and programming, but the influence of abnormal speech perception in CAS on these processes is debated. This study examined speech perception abilities among children with CAS with and without language impairment compared to those with…

  4. Common neural substrates support speech and non-speech vocal tract gestures.

    Science.gov (United States)

    Chang, Soo-Eun; Kenney, Mary Kay; Loucks, Torrey M J; Poletto, Christopher J; Ludlow, Christy L

    2009-08-01

    The issue of whether speech is supported by the same neural substrates as non-speech vocal tract gestures has been contentious. In this fMRI study we tested whether producing non-speech vocal tract gestures in humans shares the same functional neuroanatomy as non-sense speech syllables. Production of non-speech vocal tract gestures, devoid of phonological content but similar to speech in that they had familiar acoustic and somatosensory targets, was compared to the production of speech syllables without meaning. Brain activation related to overt production was captured with BOLD fMRI using a sparse sampling design for both conditions. Speech and non-speech were compared using voxel-wise whole brain analyses, and ROI analyses focused on frontal and temporoparietal structures previously reported to support speech production. Results showed substantial activation overlap between speech and non-speech function in regions. Although non-speech gesture production showed greater extent and amplitude of activation in the regions examined, both speech and non-speech showed comparable left laterality in activation for both target perception and production. These findings posit a more general role of the previously proposed "auditory dorsal stream" in the left hemisphere--to support the production of vocal tract gestures that are not limited to speech processing.

  5. The treatment of apraxia of speech : Speech and music therapy, an innovative joint effort

    NARCIS (Netherlands)

    Hurkmans, Josephus Johannes Stephanus

    2016-01-01

    Apraxia of Speech (AoS) is a neurogenic speech disorder. A wide variety of behavioural methods have been developed to treat AoS. Various therapy programmes use musical elements to improve speech production. A unique therapy programme combining elements of speech therapy and music therapy is called

  6. Speech Perception and Short-Term Memory Deficits in Persistent Developmental Speech Disorder

    Science.gov (United States)

    Kenney, Mary Kay; Barac-Cikoja, Dragana; Finnegan, Kimberly; Jeffries, Neal; Ludlow, Christy L.

    2006-01-01

    Children with developmental speech disorders may have additional deficits in speech perception and/or short-term memory. To determine whether these are only transient developmental delays that can accompany the disorder in childhood or persist as part of the speech disorder, adults with a persistent familial speech disorder were tested on speech…

  7. A machine-hearing system exploiting head movements for binaural sound localisation in reverberant conditions

    DEFF Research Database (Denmark)

    May, Tobias; Ma, Ning; Wierstorf, Hagen

    2015-01-01

    This paper is concerned with machine localisation of multiple active speech sources in reverberant environments using two (binaural) microphones. Such conditions typically present a problem for ‘classical’ binaural models. Inspired by the human ability to utilise head movements, the current study...

  8. Parallel processing Monte Carlo radiation transport codes

    International Nuclear Information System (INIS)

    McKinney, G.W.

    1994-01-01

    Issues related to distributed-memory multiprocessing as applied to Monte Carlo radiation transport are discussed. Measurements of communication overhead are presented for the radiation transport code MCNP which employs the communication software package PVM, and average efficiency curves are provided for a homogeneous virtual machine

  9. THE ONTOGENESIS OF SPEECH DEVELOPMENT

    Directory of Open Access Journals (Sweden)

    T. E. Braudo

    2017-01-01

    Full Text Available The purpose of this article is to acquaint the specialists, working with children having developmental disorders, with age-related norms for speech development. Many well-known linguists and psychologists studied speech ontogenesis (logogenesis. Speech is a higher mental function, which integrates many functional systems. Speech development in infants during the first months after birth is ensured by the innate hearing and emerging ability to fix the gaze on the face of an adult. Innate emotional reactions are also being developed during this period, turning into nonverbal forms of communication. At about 6 months a baby starts to pronounce some syllables; at 7–9 months – repeats various sounds combinations, pronounced by adults. At 10–11 months a baby begins to react on the words, referred to him/her. The first words usually appear at an age of 1 year; this is the start of the stage of active speech development. At this time it is acceptable, if a child confuses or rearranges sounds, distorts or misses them. By the age of 1.5 years a child begins to understand abstract explanations of adults. Significant vocabulary enlargement occurs between 2 and 3 years; grammatical structures of the language are being formed during this period (a child starts to use phrases and sentences. Preschool age (3–7 y. o. is characterized by incorrect, but steadily improving pronunciation of sounds and phonemic perception. The vocabulary increases; abstract speech and retelling are being formed. Children over 7 y. o. continue to improve grammar, writing and reading skills. The described stages may not have strict age boundaries, as soon as they are dependent not only on environment, but also on the child’s mental constitution, heredity and character.

  10. Integrating speech in time depends on temporal expectancies and attention.

    Science.gov (United States)

    Scharinger, Mathias; Steinberg, Johanna; Tavano, Alessandro

    2017-08-01

    Sensory information that unfolds in time, such as in speech perception, relies on efficient chunking mechanisms in order to yield optimally-sized units for further processing. Whether or not two successive acoustic events receive a one-unit or a two-unit interpretation seems to depend on the fit between their temporal extent and a stipulated temporal window of integration. However, there is ongoing debate on how flexible this temporal window of integration should be, especially for the processing of speech sounds. Furthermore, there is no direct evidence of whether attention may modulate the temporal constraints on the integration window. For this reason, we here examine how different word durations, which lead to different temporal separations of sound onsets, interact with attention. In an Electroencephalography (EEG) study, participants actively and passively listened to words where word-final consonants were occasionally omitted. Words had either a natural duration or were artificially prolonged in order to increase the separation of speech sound onsets. Omission responses to incomplete speech input, originating in left temporal cortex, decreased when the critical speech sound was separated from previous sounds by more than 250 msec, i.e., when the separation was larger than the stipulated temporal window of integration (125-150 msec). Attention, on the other hand, only increased omission responses for stimuli with natural durations. We complemented the event-related potential (ERP) analyses by a frequency-domain analysis on the stimulus presentation rate. Notably, the power of stimulation frequency showed the same duration and attention effects than the omission responses. We interpret these findings on the background of existing research on temporal integration windows and further suggest that our findings may be accounted for within the framework of predictive coding. Copyright © 2017 Elsevier Ltd. All rights reserved.

  11. Common neural substrates support speech and non-speech vocal tract gestures

    OpenAIRE

    Chang, Soo-Eun; Kenney, Mary Kay; Loucks, Torrey M.J.; Poletto, Christopher J.; Ludlow, Christy L.

    2009-01-01

    The issue of whether speech is supported by the same neural substrates as non-speech vocal-tract gestures has been contentious. In this fMRI study we tested whether producing non-speech vocal tract gestures in humans shares the same functional neuroanatomy as non-sense speech syllables. Production of non-speech vocal tract gestures, devoid of phonological content but similar to speech in that they had familiar acoustic and somatosensory targets, were compared to the production of speech sylla...

  12. Analysing Afrikaans-English bilingual children's conversational code ...

    African Journals Online (AJOL)

    It has been observed that children mix languages more often if they have been exposed to mixed speech, especially if they are in bilingual company. Very little research, however, exists on the code switching (CS) of children brought up in multilingual contexts. The study discussed in this paper investigates the grammatical ...

  13. Design of man-machine-communication-systems

    International Nuclear Information System (INIS)

    Zimmermann, R.

    1975-04-01

    This paper shows some fundamentals of man-machine-communication and deduces demands and recommendations for the design of communication systems. The main points are the directives for the design of optic display systems with details for visual perception and resolution, luminance and contrast, as well as discernibility and coding of displayed information. The most important rules are recommendations for acoustic information systems, control devices and for design of consoles are also given. (orig.) [de

  14. Automatic analysis of slips of the tongue: Insights into the cognitive architecture of speech production.

    Science.gov (United States)

    Goldrick, Matthew; Keshet, Joseph; Gustafson, Erin; Heller, Jordana; Needle, Jeremy

    2016-04-01

    Traces of the cognitive mechanisms underlying speaking can be found within subtle variations in how we pronounce sounds. While speech errors have traditionally been seen as categorical substitutions of one sound for another, acoustic/articulatory analyses show they partially reflect the intended sound. When "pig" is mispronounced as "big," the resulting /b/ sound differs from correct productions of "big," moving towards intended "pig"-revealing the role of graded sound representations in speech production. Investigating the origins of such phenomena requires detailed estimation of speech sound distributions; this has been hampered by reliance on subjective, labor-intensive manual annotation. Computational methods can address these issues by providing for objective, automatic measurements. We develop a novel high-precision computational approach, based on a set of machine learning algorithms, for measurement of elicited speech. The algorithms are trained on existing manually labeled data to detect and locate linguistically relevant acoustic properties with high accuracy. Our approach is robust, is designed to handle mis-productions, and overall matches the performance of expert coders. It allows us to analyze a very large dataset of speech errors (containing far more errors than the total in the existing literature), illuminating properties of speech sound distributions previously impossible to reliably observe. We argue that this provides novel evidence that two sources both contribute to deviations in speech errors: planning processes specifying the targets of articulation and articulatory processes specifying the motor movements that execute this plan. These findings illustrate how a much richer picture of speech provides an opportunity to gain novel insights into language processing. Copyright © 2016 Elsevier B.V. All rights reserved.

  15. Multimicrophone Speech Dereverberation: Experimental Validation

    Directory of Open Access Journals (Sweden)

    Marc Moonen

    2007-05-01

    Full Text Available Dereverberation is required in various speech processing applications such as handsfree telephony and voice-controlled systems, especially when signals are applied that are recorded in a moderately or highly reverberant environment. In this paper, we compare a number of classical and more recently developed multimicrophone dereverberation algorithms, and validate the different algorithmic settings by means of two performance indices and a speech recognition system. It is found that some of the classical solutions obtain a moderate signal enhancement. More advanced subspace-based dereverberation techniques, on the other hand, fail to enhance the signals despite their high-computational load.

  16. Discriminative learning for speech recognition

    CERN Document Server

    He, Xiadong

    2008-01-01

    In this book, we introduce the background and mainstream methods of probabilistic modeling and discriminative parameter optimization for speech recognition. The specific models treated in depth include the widely used exponential-family distributions and the hidden Markov model. A detailed study is presented on unifying the common objective functions for discriminative learning in speech recognition, namely maximum mutual information (MMI), minimum classification error, and minimum phone/word error. The unification is presented, with rigorous mathematical analysis, in a common rational-functio

  17. Microsoft Azure machine learning

    CERN Document Server

    Mund, Sumit

    2015-01-01

    The book is intended for those who want to learn how to use Azure Machine Learning. Perhaps you already know a bit about Machine Learning, but have never used ML Studio in Azure; or perhaps you are an absolute newbie. In either case, this book will get you up-and-running quickly.

  18. The Hooey Machine.

    Science.gov (United States)

    Scarnati, James T.; Tice, Craig J.

    1992-01-01

    Describes how students can make and use Hooey Machines to learn how mechanical energy can be transferred from one object to another within a system. The Hooey Machine is made using a pencil, eight thumbtacks, one pushpin, tape, scissors, graph paper, and a plastic lid. (PR)

  19. Nanocomposites for Machining Tools

    DEFF Research Database (Denmark)

    Sidorenko, Daria; Loginov, Pavel; Mishnaevsky, Leon

    2017-01-01

    Machining tools are used in many areas of production. To a considerable extent, the performance characteristics of the tools determine the quality and cost of obtained products. The main materials used for producing machining tools are steel, cemented carbides, ceramics and superhard materials...

  20. A nucleonic weighing machine

    International Nuclear Information System (INIS)

    Anon.

    1978-01-01

    The design and operation of a nucleonic weighing machine fabricated for continuous weighing of material over conveyor belt are described. The machine uses a 40 mCi cesium-137 line source and a 10 litre capacity ionization chamber. It is easy to maintain as there are no moving parts. It can also be easily removed and reinstalled. (M.G.B.)

  1. An asymptotical machine

    Science.gov (United States)

    Cristallini, Achille

    2016-07-01

    A new and intriguing machine may be obtained replacing the moving pulley of a gun tackle with a fixed point in the rope. Its most important feature is the asymptotic efficiency. Here we obtain a satisfactory description of this machine by means of vector calculus and elementary trigonometry. The mathematical model has been compared with experimental data and briefly discussed.

  2. Machine learning with R

    CERN Document Server

    Lantz, Brett

    2015-01-01

    Perhaps you already know a bit about machine learning but have never used R, or perhaps you know a little R but are new to machine learning. In either case, this book will get you up and running quickly. It would be helpful to have a bit of familiarity with basic programming concepts, but no prior experience is required.

  3. The deleuzian abstract machines

    DEFF Research Database (Denmark)

    Werner Petersen, Erik

    2005-01-01

    To most people the concept of abstract machines is connected to the name of Alan Turing and the development of the modern computer. The Turing machine is universal, axiomatic and symbolic (E.g. operating on symbols). Inspired by Foucault, Deleuze and Guattari extended the concept of abstract...

  4. Human Machine Learning Symbiosis

    Science.gov (United States)

    Walsh, Kenneth R.; Hoque, Md Tamjidul; Williams, Kim H.

    2017-01-01

    Human Machine Learning Symbiosis is a cooperative system where both the human learner and the machine learner learn from each other to create an effective and efficient learning environment adapted to the needs of the human learner. Such a system can be used in online learning modules so that the modules adapt to each learner's learning state both…

  5. Contribution of auditory working memory to speech understanding in mandarin-speaking cochlear implant users.

    Science.gov (United States)

    Tao, Duoduo; Deng, Rui; Jiang, Ye; Galvin, John J; Fu, Qian-Jie; Chen, Bing

    2014-01-01

    of voice pitch cues (albeit poorly coded by the CI) did not influence the relationship between working memory and speech perception.

  6. Applications guide to the RSIC-distributed version of the MCNP code (coupled Monte Carlo neutron-photon Code)

    International Nuclear Information System (INIS)

    Cramer, S.N.

    1985-09-01

    An overview of the RSIC-distributed version of the MCNP code (a soupled Monte Carlo neutron-photon code) is presented. All general features of the code, from machine hardware requirements to theoretical details, are discussed. The current nuclide cross-section and other libraries available in the standard code package are specified, and a realistic example of the flexible geometry input is given. Standard and nonstandard source, estimator, and variance-reduction procedures are outlined. Examples of correct usage and possible misuse of certain code features are presented graphically and in standard output listings. Finally, itemized summaries of sample problems, various MCNP code documentation, and future work are given

  7. Relationship between Speech Intelligibility and Speech Comprehension in Babble Noise

    Science.gov (United States)

    Fontan, Lionel; Tardieu, Julien; Gaillard, Pascal; Woisard, Virginie; Ruiz, Robert

    2015-01-01

    Purpose: The authors investigated the relationship between the intelligibility and comprehension of speech presented in babble noise. Method: Forty participants listened to French imperative sentences (commands for moving objects) in a multitalker babble background for which intensity was experimentally controlled. Participants were instructed to…

  8. Spectral integration in speech and non-speech sounds

    Science.gov (United States)

    Jacewicz, Ewa

    2005-04-01

    Spectral integration (or formant averaging) was proposed in vowel perception research to account for the observation that a reduction of the intensity of one of two closely spaced formants (as in /u/) produced a predictable shift in vowel quality [Delattre et al., Word 8, 195-210 (1952)]. A related observation was reported in psychoacoustics, indicating that when the components of a two-tone periodic complex differ in amplitude and frequency, its perceived pitch is shifted toward that of the more intense tone [Helmholtz, App. XIV (1875/1948)]. Subsequent research in both fields focused on the frequency interval that separates these two spectral components, in an attempt to determine the size of the bandwidth for spectral integration to occur. This talk will review the accumulated evidence for and against spectral integration within the hypothesized limit of 3.5 Bark for static and dynamic signals in speech perception and psychoacoustics. Based on similarities in the processing of speech and non-speech sounds, it is suggested that spectral integration may reflect a general property of the auditory system. A larger frequency bandwidth, possibly close to 3.5 Bark, may be utilized in integrating acoustic information, including speech, complex signals, or sound quality of a violin.

  9. Intelligibility of synthetic speech in the presence of interfering speech

    NARCIS (Netherlands)

    Eggen, J.H.

    1989-01-01

    Standard articulation tests are not always sensitive enough to discriminate between speech samples which are of high intelligibility. One can increase the sensitivity of such tests by presenting the test materials in noise. In this way, small differences in intelligibility can be magnified into

  10. Multimedia with a speech track: searching spontaneous conversational speech

    NARCIS (Netherlands)

    Larson, Martha; Ordelman, Roeland J.F.; de Jong, Franciska M.G.; Kohler, Joachim; Kraaij, Wessel

    After two successful years at SIGIR in 2007 and 2008, the third workshop on Searching Spontaneous Conversational Speech (SSCS 2009) was held conjunction with the ACM Multimedia 2009. The goal of the SSCS series is to serve as a forum that brings together the disciplines that collaborate on spoken

  11. SPEECH ACT ANALYSIS: HOSNI MUBARAK'S SPEECHES IN PRE ...

    African Journals Online (AJOL)

    enerco

    from movements of certain organs with his (man‟s) throat and mouth…. By means ... In other words, government engages language; and how this affects the ... address the audience in a social gathering in order to have a new dawn. ..... Agbedo, C. U. Speech Act Analysis of Political discourse in the Nigerian Print Media in.

  12. Cognitive Functions in Childhood Apraxia of Speech

    Science.gov (United States)

    Nijland, Lian; Terband, Hayo; Maassen, Ben

    2015-01-01

    Purpose: Childhood apraxia of speech (CAS) is diagnosed on the basis of specific speech characteristics, in the absence of problems in hearing, intelligence, and language comprehension. This does not preclude the possibility that children with this speech disorder might demonstrate additional problems. Method: Cognitive functions were investigated…

  13. Phonetic recalibration of speech by text

    NARCIS (Netherlands)

    Keetels, M.N.; Schakel, L.; de Bonte, M.; Vroomen, J.

    2016-01-01

    Listeners adjust their phonetic categories to cope with variations in the speech signal (phonetic recalibration). Previous studies have shown that lipread speech (and word knowledge) can adjust the perception of ambiguous speech and can induce phonetic adjustments (Bertelson, Vroomen, & de Gelder in

  14. Speech and Debate as Civic Education

    Science.gov (United States)

    Hogan, J. Michael; Kurr, Jeffrey A.; Johnson, Jeremy D.; Bergmaier, Michael J.

    2016-01-01

    In light of the U.S. Senate's designation of March 15, 2016 as "National Speech and Debate Education Day" (S. Res. 398, 2016), it only seems fitting that "Communication Education" devote a special section to the role of speech and debate in civic education. Speech and debate have been at the heart of the communication…

  15. Speech Synthesis Applied to Language Teaching.

    Science.gov (United States)

    Sherwood, Bruce

    1981-01-01

    The experimental addition of speech output to computer-based Esperanto lessons using speech synthesized from text is described. Because of Esperanto's phonetic spelling and simple rhythm, it is particularly easy to describe the mechanisms of Esperanto synthesis. Attention is directed to how the text-to-speech conversion is performed and the ways…

  16. Epoch-based analysis of speech signals

    Indian Academy of Sciences (India)

    on speech production characteristics, but also helps in accurate analysis of speech. .... include time delay estimation, speech enhancement from single and multi- ...... log. (. E[k]. ∑K−1 l=0. E[l]. ) ,. (7) where K is the number of samples in the ...

  17. Normal Aspects of Speech, Hearing, and Language.

    Science.gov (United States)

    Minifie, Fred. D., Ed.; And Others

    This book is written as a guide to the understanding of the processes involved in human speech communication. Ten authorities contributed material to provide an introduction to the physiological aspects of speech production and reception, the acoustical aspects of speech production and transmission, the psychophysics of sound reception, the nature…

  18. Audiovisual Asynchrony Detection in Human Speech

    Science.gov (United States)

    Maier, Joost X.; Di Luca, Massimiliano; Noppeney, Uta

    2011-01-01

    Combining information from the visual and auditory senses can greatly enhance intelligibility of natural speech. Integration of audiovisual speech signals is robust even when temporal offsets are present between the component signals. In the present study, we characterized the temporal integration window for speech and nonspeech stimuli with…

  19. The interpersonal level in English: reported speech

    NARCIS (Netherlands)

    Keizer, E.

    2009-01-01

    The aim of this article is to describe and classify a number of different forms of English reported speech (or thought), and subsequently to analyze and represent them within the theory of FDG. First, the most prototypical forms of reported speech are discussed (direct and indirect speech);

  20. Cognitive functions in Childhood Apraxia of Speech

    NARCIS (Netherlands)

    Nijland, L.; Terband, H.; Maassen, B.

    2015-01-01

    Purpose: Childhood Apraxia of Speech (CAS) is diagnosed on the basis of specific speech characteristics, in the absence of problems in hearing, intelligence, and language comprehension. This does not preclude the possibility that children with this speech disorder might demonstrate additional