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Sample records for dsp-based voice signal

  1. DSP Based System for Real time Voice Synthesis Applications Development

    OpenAIRE

    Arsinte, Radu; Ferencz, Attila; Miron, Costin

    2008-01-01

    This paper describes an experimental system designed for development of real time voice synthesis applications. The system is composed from a DSP coprocessor card, equipped with an TMS320C25 or TMS320C50 chip, voice acquisition module (ADDA2),host computer (IBM-PC compatible), software specific tools.

  2. Monitoring of electric-cardio signals based on DSP

    Science.gov (United States)

    Yan, Yi-xin; Sun, Hui-nan; Lv, Shuang

    2008-10-01

    Monitoring of electric-cardio signals is the most direct method of discovering heart diseases. This article presents an electric-cardio signal acquisition and processing system based on DSP. According to the features of electric-cardio signals, the proposed system uses the AgCl electrode as electric-cardio signals sensor, and acquires analog signals with AD620 as the prepositional amplifier, and the digital system equipped is with TMS320LF2407A DSP. The design of digital filter and the analysis of heart rate variation are realized by programming in the DSP. Finally the ECG is obtained with P and T waves along with obvious QRS multi-wave characteristics. The system has low power dissipation, low cost and high precision, which meets the requirements for medical instruments.

  3. Intelligent Shutter Speech Control System Based on DSP

    Directory of Open Access Journals (Sweden)

    Yonghong Deng

    2017-01-01

    Full Text Available Based on TMS320F28035 DSP, this paper designed a smart shutters voice control system, which realized the functions of opening and closing shutters, intelligent switching of lighting mode and solar power supply through voice control. The traditional control mode is converted to voice control at the same time with automatic lighting and solar power supply function. In the convenience of people’s lives at the same time more satisfied with today’s people on the intelligent and environmental protection of the two concepts of the pursuit. The whole system is simple, low cost, safe and reliable.

  4. An Application of PSV-S in Fast Development of a Real-Time DSP System

    Directory of Open Access Journals (Sweden)

    Armein Z.R. Langi

    2016-09-01

    Full Text Available Virtual prototyping is natural in developing digital signal processing (DSP systems using a product-service-value system (PSV-S approach. Our DSP virtual prototyping approach consists of four development phases: (1 a generic DSP system, (2 a functional DSP system, (3 an architectural DSP system, and (4 a real-time DSP system. Such an approach results in a more comprehensive approach in the DSP system development. This paper shows an example of prototyping a voice codec on a single-chip DSP processor.

  5. The design and implementation of signal decomposition system of CL multi-wavelet transform based on DSP builder

    Science.gov (United States)

    Huang, Yan; Wang, Zhihui

    2015-12-01

    With the development of FPGA, DSP Builder is widely applied to design system-level algorithms. The algorithm of CL multi-wavelet is more advanced and effective than scalar wavelets in processing signal decomposition. Thus, a system of CL multi-wavelet based on DSP Builder is designed for the first time in this paper. The system mainly contains three parts: a pre-filtering subsystem, a one-level decomposition subsystem and a two-level decomposition subsystem. It can be converted into hardware language VHDL by the Signal Complier block that can be used in Quartus II. After analyzing the energy indicator, it shows that this system outperforms Daubenchies wavelet in signal decomposition. Furthermore, it has proved to be suitable for the implementation of signal fusion based on SoPC hardware, and it will become a solid foundation in this new field.

  6. SIM-DSP: A DSP-Enhanced CAD Platform for Signal Integrity Macromodeling and Simulation

    Directory of Open Access Journals (Sweden)

    Chi-Un Lei

    2014-12-01

    Full Text Available Macromodeling-Simulation process for signal integrity verifications has become necessary for the high speed circuit system design. This paper aims to introduce a “VLSI Signal Integrity Macromodeling and Simulation via Digital Signal Processing Techniques” framework (known as SIM-DSP framework, which applies digital signal processing techniques to facilitate the SI verification process in the pre-layout design phase. Core identification modules and peripheral (pre-/post-processing modules have been developed and assembled to form a verification flow. In particular, a single-step discrete cosine transform truncation (DCTT module has been developed for modeling-simulation process. In DCTT, the response modeling problem is classified as a signal compression problem, wherein the system response can be represented by a truncated set of non-pole based DCT bases, and error can be analyzed through Parseval’s theorem. Practical examples are given to show the applicability of our proposed framework.

  7. Low Power Systolic Array Based Digital Filter for DSP Applications

    Directory of Open Access Journals (Sweden)

    S. Karthick

    2015-01-01

    Full Text Available Main concepts in DSP include filtering, averaging, modulating, and correlating the signals in digital form to estimate characteristic parameter of a signal into a desirable form. This paper presents a brief concept of low power datapath impact for Digital Signal Processing (DSP based biomedical application. Systolic array based digital filter used in signal processing of electrocardiogram analysis is presented with datapath architectural innovations in low power consumption perspective. Implementation was done with ASIC design methodology using TSMC 65 nm technological library node. The proposed systolic array filter has reduced leakage power up to 8.5% than the existing filter architectures.

  8. Programming a DSP card for generating an ECG signal with possibility of anomalies

    International Nuclear Information System (INIS)

    Hamrouni, Sayma

    2013-01-01

    This project consists of programming a DSP designed to generate an ECG signal with a probability of anomaly. To begin with, we get to know the characteristics of a DSP card and its architecture. As a second step, we programmed the DSP32C using the compiler D3CC associated with Textpad in order to obtain an analog signal in the respective outputs. And then finally, we developed a graphical user interface using the programming software LabVIEW that aims controlling the good operation of DSP. The tests previously made have proved the good operation of the application.

  9. Designing on ICT reconstruction software based on DSP techniques

    International Nuclear Information System (INIS)

    Liu Jinhui; Xiang Xincheng

    2006-01-01

    The convolution back project (CBP) algorithm is used to realize the CT image's reconstruction in ICT generally, which is finished by using PC or workstation. In order to add the ability of multi-platform operation of CT reconstruction software, a CT reconstruction method based on modern digital signal processor (DSP) technique is proposed and realized in this paper. The hardware system based on TI's C6701 DSP processor is selected to support the CT software construction. The CT reconstruction software is compiled only using assembly language related to the DSP hardware. The CT software can be run on TI's C6701 EVM board by inputting the CT data, and can get the CT Images that satisfy the real demands. (authors)

  10. A LabVIEW based Remote DSP Laboratory

    Directory of Open Access Journals (Sweden)

    Athanasios Kalantzopoulos

    2008-07-01

    Full Text Available Remote laboratories provide the students with the capability to perform laboratory exercises exploiting the relevant equipment any time of the day without their physical presence. Furthermore, providing the ability to use a single workstation by more than one student, they contribute to the reduction of the laboratory cost. Turning to advantage the above and according to the needs of post graduate modules in the fields of DSP Systems Design and Signal Processing Systems with DSPs, we designed and developed a Remote DSP Laboratory. A student using a Web Browser has the ability via internet to turn to account the R-DSP Lab and perform experiments using DSPs (Digital Signal Processors. For now, there is the opportunity to carry out laboratory exercises such as FIR, IIR digital filters and FFT as well as run any executable file developed by the user. In any case the observation of the results is carried out through the use of specially designed Graphical User Interfaces (GUIs.

  11. New methods to get valid signals at high temperature conditions by using DSP tools of the ASSA (Abnormal Signal Simulation Analyzer)

    International Nuclear Information System (INIS)

    Koo, Kil-Mo; Hong, Seong-Wan; Song, Jin-Ho; Baek, Won-Pil; Jung, Myung-Kwan

    2012-01-01

    A new method to get valid signals under high temperature conditions using DSP (Digital Signal Processing) tools of an ASSA (Abnormal Signal Simulation Analyzer) module through a signal analysis of important circuit modeling under severe accident conditions has been suggested. Already exist, such kinds of DSP technique operated by LabVIEW or MatLab code linked with PSpice code, which have convenient tools as a special function of the ASSA module including a signal reconstruction method. If we can obtain a shift data of the transient parameters such as the time constant of the R-L-C circuit affected by high temperature under a severe accident condition, it will be possible to reconstruct an abnormal signal using a trained deconvolution algorithm as a sort of DSP technique. (author)

  12. Design of overload vehicle monitoring and response system based on DSP

    Science.gov (United States)

    Yu, Yan; Liu, Yiheng; Zhao, Xuefeng

    2014-03-01

    The overload vehicles are making much more damage to the road surface than the regular ones. Many roads and bridges are equipped with structural health monitoring system (SHM) to provide early-warning to these damage and evaluate the safety of road and bridge. However, because of the complex nature of SHM system, it's expensive to manufacture, difficult to install and not well-suited for the regular bridges and roads. Based on this application background, this paper designs a compact structural health monitoring system based on DSP, which is highly integrated, low-power, easy to install and inexpensive to manufacture. The designed system is made up of sensor arrays, the charge amplifier module, the DSP processing unit, the alarm system for overload, and the estimate for damage of the road and bridge structure. The signals coming from sensor arrays go through the charge amplifier. DSP processing unit will receive the amplified signals, estimate whether it is an overload signal or not, and convert analog variables into digital ones so that they are compatible with the back-end digital circuit for further processing. The system will also restrict certain vehicles that are overweight, by taking image of the car brand, sending the alarm, and transferring the collected pressure data to remote data center for further monitoring analysis by rain-flow counting method.

  13. A fast DSP-based calorimeter hit scanning system

    International Nuclear Information System (INIS)

    Sekikawa, S.; Arai, I.; Suzuki, A.; Watanabe, A.; Marlow, D.R.; Mindas, C.R.; Wixted, R.L.

    1997-01-01

    A custom made digital signal processor (DSP) based system has been developed to scan calorimeter hits read by a 32-channel FASTBUS waveform recorder board. The scanner system identifies hit calorimeter elements by surveying their discriminated outputs. This information is used to generate a list of addresses, which guides the read-out process. The system is described and measurements of the scan times are given. (orig.)

  14. Design of a system based on DSP and FPGA for video recording and replaying

    Science.gov (United States)

    Kang, Yan; Wang, Heng

    2013-08-01

    This paper brings forward a video recording and replaying system with the architecture of Digital Signal Processor (DSP) and Field Programmable Gate Array (FPGA). The system achieved encoding, recording, decoding and replaying of Video Graphics Array (VGA) signals which are displayed on a monitor during airplanes and ships' navigating. In the architecture, the DSP is a main processor which is used for a large amount of complicated calculation during digital signal processing. The FPGA is a coprocessor for preprocessing video signals and implementing logic control in the system. In the hardware design of the system, Peripheral Device Transfer (PDT) function of the External Memory Interface (EMIF) is utilized to implement seamless interface among the DSP, the synchronous dynamic RAM (SDRAM) and the First-In-First-Out (FIFO) in the system. This transfer mode can avoid the bottle-neck of the data transfer and simplify the circuit between the DSP and its peripheral chips. The DSP's EMIF and two level matching chips are used to implement Advanced Technology Attachment (ATA) protocol on physical layer of the interface of an Integrated Drive Electronics (IDE) Hard Disk (HD), which has a high speed in data access and does not rely on a computer. Main functions of the logic on the FPGA are described and the screenshots of the behavioral simulation are provided in this paper. In the design of program on the DSP, Enhanced Direct Memory Access (EDMA) channels are used to transfer data between the FIFO and the SDRAM to exert the CPU's high performance on computing without intervention by the CPU and save its time spending. JPEG2000 is implemented to obtain high fidelity in video recording and replaying. Ways and means of acquiring high performance for code are briefly present. The ability of data processing of the system is desirable. And smoothness of the replayed video is acceptable. By right of its design flexibility and reliable operation, the system based on DSP and FPGA

  15. A VXI-GPIB protocol converter based on DSP

    International Nuclear Information System (INIS)

    Hu Yuanfeng; Yu Xiaoqi; Lu Jingping

    2006-01-01

    A VXI-GPIB protocol converter based on DSP is introduced. The word-serial protocol with the message-based interface is implemented by EPLD and DSP. The GPIB functions are implemented by programming to the GPIB control chip. The transfer from VXI messages to GPIB functions is implemented by DSP. As an example of application, the control functions for oscilloscopes have been implemented in a VXI-GPIB heterogeneous system using such modules. (authors)

  16. Integrated optical 3D digital imaging based on DSP scheme

    Science.gov (United States)

    Wang, Xiaodong; Peng, Xiang; Gao, Bruce Z.

    2008-03-01

    We present a scheme of integrated optical 3-D digital imaging (IO3DI) based on digital signal processor (DSP), which can acquire range images independently without PC support. This scheme is based on a parallel hardware structure with aid of DSP and field programmable gate array (FPGA) to realize 3-D imaging. In this integrated scheme of 3-D imaging, the phase measurement profilometry is adopted. To realize the pipeline processing of the fringe projection, image acquisition and fringe pattern analysis, we present a multi-threads application program that is developed under the environment of DSP/BIOS RTOS (real-time operating system). Since RTOS provides a preemptive kernel and powerful configuration tool, with which we are able to achieve a real-time scheduling and synchronization. To accelerate automatic fringe analysis and phase unwrapping, we make use of the technique of software optimization. The proposed scheme can reach a performance of 39.5 f/s (frames per second), so it may well fit into real-time fringe-pattern analysis and can implement fast 3-D imaging. Experiment results are also presented to show the validity of proposed scheme.

  17. Low-Power Embedded DSP Core for Communication Systems

    Science.gov (United States)

    Tsao, Ya-Lan; Chen, Wei-Hao; Tan, Ming Hsuan; Lin, Maw-Ching; Jou, Shyh-Jye

    2003-12-01

    This paper proposes a parameterized digital signal processor (DSP) core for an embedded digital signal processing system designed to achieve demodulation/synchronization with better performance and flexibility. The features of this DSP core include parameterized data path, dual MAC unit, subword MAC, and optional function-specific blocks for accelerating communication system modulation operations. This DSP core also has a low-power structure, which includes the gray-code addressing mode, pipeline sharing, and advanced hardware looping. Users can select the parameters and special functional blocks based on the character of their applications and then generating a DSP core. The DSP core has been implemented via a cell-based design method using a synthesizable Verilog code with TSMC 0.35[InlineEquation not available: see fulltext.]m SPQM and 0.25[InlineEquation not available: see fulltext.]m 1P5M library. The equivalent gate count of the core area without memory is approximately 50 k. Moreover, the maximum operating frequency of a[InlineEquation not available: see fulltext.] version is 100 MHz (0.35[InlineEquation not available: see fulltext.]m) and 140 MHz (0.25[InlineEquation not available: see fulltext.]m).

  18. METHODS FOR QUALITY ENHANCEMENT OF USER VOICE SIGNAL IN VOICE AUTHENTICATION SYSTEMS

    Directory of Open Access Journals (Sweden)

    O. N. Faizulaieva

    2014-03-01

    Full Text Available The reasonability for the usage of computer systems user voice in the authentication process is proved. The scientific task for improving the signal/noise ratio of the user voice signal in the authentication system is considered. The object of study is the process of input and output of the voice signal of authentication system user in computer systems and networks. Methods and means for input and extraction of voice signal against external interference signals are researched. Methods for quality enhancement of user voice signal in voice authentication systems are suggested. As modern computer facilities, including mobile ones, have two-channel audio card, the usage of two microphones is proposed in the voice signal input system of authentication system. Meanwhile, the task of forming a lobe of microphone array in a desired area of voice signal registration (100 Hz to 8 kHz is solved. The usage of directional properties of the proposed microphone array gives the possibility to have the influence of external interference signals two or three times less in the frequency range from 4 to 8 kHz. The possibilities for implementation of space-time processing of the recorded signals using constant and adaptive weighting factors are investigated. The simulation results of the proposed system for input and extraction of signals during digital processing of narrowband signals are presented. The proposed solutions make it possible to improve the value of the signal/noise ratio of the useful signals recorded up to 10, ..., 20 dB under the influence of external interference signals in the frequency range from 4 to 8 kHz. The results may be useful to specialists working in the field of voice recognition and speaker’s discrimination.

  19. Updating signal typing in voice: addition of type 4 signals.

    Science.gov (United States)

    Sprecher, Alicia; Olszewski, Aleksandra; Jiang, Jack J; Zhang, Yu

    2010-06-01

    The addition of a fourth type of voice to Titze's voice classification scheme is proposed. This fourth voice type is characterized by primarily stochastic noise behavior and is therefore unsuitable for both perturbation and correlation dimension analysis. Forty voice samples were classified into the proposed four types using narrowband spectrograms. Acoustic, perceptual, and correlation dimension analyses were completed for all voice samples. Perturbation measures tended to increase with voice type. Based on reliability cutoffs, the type 1 and type 2 voices were considered suitable for perturbation analysis. Measures of unreliability were higher for type 3 and 4 voices. Correlation dimension analyses increased significantly with signal type as indicated by a one-way analysis of variance. Notably, correlation dimension analysis could not quantify the type 4 voices. The proposed fourth voice type represents a subset of voices dominated by noise behavior. Current measures capable of evaluating type 4 voices provide only qualitative data (spectrograms, perceptual analysis, and an infinite correlation dimension). Type 4 voices are highly complex and the development of objective measures capable of analyzing these voices remains a topic of future investigation.

  20. Car audio using DSP for active sound control. DSP ni yoru active seigyo wo mochiita audio

    Energy Technology Data Exchange (ETDEWEB)

    Yamada, K.; Asano, S.; Furukawa, N. (Mitsubishi Motor Corp., Tokyo (Japan))

    1993-06-01

    In the automobile cabin, there are some unique problems which spoil the quality of sound reproduction from audio equipment, such as the narrow space and/or the background noise. The audio signal processing by using DSP (digital signal processor) makes enable a solution to these problems. A car audio with a high amenity has been successfully made by the active sound control using DSP. The DSP consists of an adder, coefficient multiplier, delay unit, and connections. For the actual processing by DSP, are used functions, such as sound field correction, response and processing of noises during driving, surround reproduction, graphic equalizer processing, etc. High effectiveness of the method was confirmed through the actual driving evaluation test. The present paper describes the actual method of sound control technology using DSP. Especially, the dynamic processing of the noise during driving is discussed in detail. 1 ref., 12 figs., 1 tab.

  1. Digital Signal Processing. A review of DSP formalism, algorithms and networks for the beam instrumentation workshop, Vancouver, Canada, October 4, 1994

    International Nuclear Information System (INIS)

    Linscott, I.

    1995-01-01

    The formalism of Digital Signal Processing (DSP), is reviewed with the objective of providing a framework for understanding the utility of DSP techniques for Beam Instrumentation and developiong criteria for assessing the merits of DSP applications. copyright 1995 American Institute of Physics

  2. Arithmetic circuits for DSP applications

    CERN Document Server

    Stouraitis, Thanos

    2017-01-01

    Arithmetic Circuits for DSP Applications is a complete resource on arithmetic circuits for digital signal processing (DSP). It covers the key concepts, designs and developments of different types of arithmetic circuits, which can be used for improving the efficiency of implementation of a multitude of DSP applications. Each chapter includes various applications of the respective class of arithmetic circuits along with information on the future scope of research. Written for students, engineers, and researchers in electrical and computer engineering, this comprehensive text offers a clear understanding of different types of arithmetic circuits used for digital signal processing applications. The text includes contributions from noted researchers on a wide range of topics, including a review o circuits used in implementing basic operations like additions and multiplications; distributed arithmetic as a technique for the multiplier-less implementation of inner products for DSP applications; discussions on look ...

  3. Design and implementation of DSP based solar converter for photovoltaic systems

    Energy Technology Data Exchange (ETDEWEB)

    Caliskan, Eser [TUBITAK - MRC, Kocaeli (Turkey). Energy Inst.; Ustun, Ozgur [Istanbul Technical Univ., Maslak (Turkey). Electrical Engineering Dept.

    2012-07-01

    This study discusses the design and implementation of a DSP controlled converter for photovoltaic system that can track the maximum power point, charge and discharge the battery. In the designed system, the boost converter operates the photovoltaic panels at the maximum power point and the bi-directional battery charger charges and discharges the battery bank as demanded. All required switching and control signals for these converters provided by the high performance C2000 series DSP produced by the Texas Instruments. The current, voltage and temperature data are collected by sensors from power stages by using DSP algorithms and the control signals are generated by the embedded software. The load bus is kept at constant voltage by the bi-directional battery charger. The boost converter is controlled by MPPT algorithms and the current sharing or battery charge modes are implemented depending on the radiation value. The designed photovoltaic system performance is verified by simulation and some experiments. (orig.)

  4. Inspector-2000. A DSP-based, portable, multi-purpose MCA

    International Nuclear Information System (INIS)

    Koskelo, M.J.; Sielaff, W.A.; Hall, D.L.; Kastner, M.H.; Jordanov, V.T.

    2001-01-01

    Various in-situ gamma-spectroscopy applications need a versatile, multi-purpose, portable multi-channel analyzer (MCA). Recently, Canberra has introduced the Inspector-2000 for this purpose. It uses digital signal processing (DSP) technology and weighs only about 1.2 kg. It also supports CdTe, NaI and Ge detectors. Due to its use of DSP technology, the Inspector-2000 also provides a longer battery life, a better detector resolution and a better temperature stability than most portable MCAs. A short description of the Inspector-2000 MCA is included and its performance characteristics compared to an analog MCA. (author)

  5. A DSP Based POD Implementation for High Speed Multimedia Communications

    Directory of Open Access Journals (Sweden)

    Chang Nian Zhang

    2002-09-01

    Full Text Available In the cable network services, the audio/video entertainment contents should be protected from unauthorized copying, intercepting, and tampering. Point-of-deployment (POD security module, proposed by OpenCableTM, allows viewers to receive secure cable services such as premium subscription channels, impulse pay-per-view, video-on-demand as well as other interactive services. In this paper, we present a digital signal processor (DSP (TMS320C6211 based POD implementation for the real-time applications which include elliptic curve digital signature algorithm (ECDSA, elliptic curve Diffie Hellman (ECDH key exchange, elliptic curve key derivation function (ECKDF, cellular automata (CA cryptography, communication processes between POD and Host, and Host authentication. In order to get different security levels and different rates of encryption/decryption, a CA based symmetric key cryptography algorithm is used whose encryption/decryption rate can be up to 75 Mbps. The experiment results indicate that the DSP based POD implementation provides high speed and flexibility, and satisfies the requirements of real-time video data transmission.

  6. A DSP controlled data acquisition system for CELSIUS

    International Nuclear Information System (INIS)

    Bengtsson, M.; Lofnes, T.; Ziemann, V.

    2000-01-01

    We describe a data acquisition system based on two 10 MHz A/D-converters, a SHARC Digital Signal Processor (DSP), and a digital synthesizer used for triggering the A/D-converters. The temporal macrostructure of the data acquisition can be determined by external triggers or by timer interrupts from the DSP. In this way up to two million samples can be stored in DSP external memory. The samples are analyzed by directly fast Fourier transforming blocks of samples. In another mode we use software-based downmixing and filtering techniques to increase the resolution and zoom in on a small frequency band. Spectra of up to 5 MHz can be manipulated and displayed as waterfall plots or spectral maps on the host computer directly. Moreover, signals of up to 70 MHz can be analyzed by undersampling techniques. We use this system to analyze Schottky spectra from electron-cooled ion beams in CELSIUS and report drag rate measurements and observations of instabilities

  7. A DSP controlled data acquisition system for CELSIUS

    CERN Document Server

    Bengtsson, M; Ziemann, Volker

    2000-01-01

    We describe a data acquisition system based on two 10 MHz A/D-converters, a SHARC Digital Signal Processor (DSP), and a digital synthesizer used for triggering the A/D-converters. The temporal macrostructure of the data acquisition can be determined by external triggers or by timer interrupts from the DSP. In this way up to two million samples can be stored in DSP external memory. The samples are analyzed by directly fast Fourier transforming blocks of samples. In another mode we use software-based downmixing and filtering techniques to increase the resolution and zoom in on a small frequency band. Spectra of up to 5 MHz can be manipulated and displayed as waterfall plots or spectral maps on the host computer directly. Moreover, signals of up to 70 MHz can be analyzed by undersampling techniques. We use this system to analyze Schottky spectra from electron-cooled ion beams in CELSIUS and report drag rate measurements and observations of instabilities.

  8. Methodology and Implementation on DSP of Heuristic Multiuser DS/CDMA Detectors

    Directory of Open Access Journals (Sweden)

    Alex Miyamoto Mussi

    2010-12-01

    Full Text Available The growing number of users of mobile communications networks and the scarcity of the electromagnetic spectrum make the use of diversity techniques and detection/decoding efficient, such as the use of multiple antennas at the transmitter and/or receiver, multiuser detection (MuD – Multiuser Detection, among others, have an increasingly prominent role in the telecommunications landscape. This paper presents a design methodology based on digital signal processors (DSP – Digital Signal Processor with a view to the implementation of multiuser heuristics detectors in systems DS/CDMA (Direct Sequence Code Division Multiple Access. Heuristics detection techniques result in near-optimal performance in order to approach the performance of maximum-likelihood (ML. In this work, was employed the DSP development platform called the C6713 DSK, which is based in Texas TMS320C6713 processor. The heuristics techniques proposed are based on well established algorithms in the literature. The efficiency of the algorithms implemented in DSP has been evaluated numerically by computing the measure of bit error rate (BER. Finally, the feasibility of implementation in DSP could then be verified by comparing results from multiple Monte-Carlo simulation in Matlab, with those obtained from implementation on DSP. It also demonstrates the effective increase in performance and system capacity of DS/CDMA with the use of heuristic multiuser detection techniques, implemented directly in the DSP.

  9. The study of interferometer spectrometer based on DSP and linear CCD

    Science.gov (United States)

    Kang, Hua; Peng, Yuexiang; Xu, Xinchen; Xing, Xiaoqiao

    2010-11-01

    In this paper, general theory of Fourier-transform spectrometer and polarization interferometer is presented. A new design is proposed for Fourier-transform spectrometer based on polarization interferometer with Wollaston prisms and linear CCD. Firstly, measured light is changed into linear polarization light by polarization plate. And then the light can be split into ordinary and extraordinary lights by going through one Wollaston prism. At last, after going through another Wollaston prism and analyzer, interfering fringes can be formed on linear CCD behind the analyzer. The linear CCD is driven by CPLD to output amplitude of interfering fringes and synchronous signals of frames and pixels respectively. DSP is used to collect interference pattern signals from CCD and the digital data of interfering fringes are processed by using 2048-point-FFT. Finally, optical spectrum of measured light can be display on LCD connected to DSP with RS232. The spectrometer will possess the features of firmness, portability and the ability of real-time analyzing. The work will provide a convenient and significant foundation for application of more high accuracy of Fourier-transform spectrometer.

  10. High-speed ultra-wideband wireless signals over fiber systems: photonic generation and DSP detection

    DEFF Research Database (Denmark)

    Yu, Xianbin; Gibbon, Timothy Braidwood; Tafur Monroy, Idelfonso

    2009-01-01

    We firstly review the efforts in the literature on ultra-wideband (UWB)-over-fiber systems. Secondly, we present experimental results on photonic generation of high-speed UWB signals by both direct modulation and external optical injecting an uncooled semiconductor laser. Furthermore, we introduce...... the use of digital signal processing (DSP) technology to receive the generated UWB signal at 781.25 Mbit/s. Error-free transmission is achieved....

  11. The design of multi-core DSP parallel model based on message passing and multi-level pipeline

    Science.gov (United States)

    Niu, Jingyu; Hu, Jian; He, Wenjing; Meng, Fanrong; Li, Chuanrong

    2017-10-01

    Currently, the design of embedded signal processing system is often based on a specific application, but this idea is not conducive to the rapid development of signal processing technology. In this paper, a parallel processing model architecture based on multi-core DSP platform is designed, and it is mainly suitable for the complex algorithms which are composed of different modules. This model combines the ideas of multi-level pipeline parallelism and message passing, and summarizes the advantages of the mainstream model of multi-core DSP (the Master-Slave model and the Data Flow model), so that it has better performance. This paper uses three-dimensional image generation algorithm to validate the efficiency of the proposed model by comparing with the effectiveness of the Master-Slave and the Data Flow model.

  12. Triangle bipolar pulse shaping and pileup correction based on DSP

    International Nuclear Information System (INIS)

    Esmaeili-sani, Vahid; Moussavi-zarandi, Ali; Akbar-ashrafi, Nafiseh; Boghrati, Behzad

    2011-01-01

    Programmable Digital Signal Processing (DSP) microprocessors are capable of doing complex discrete signal processing algorithms with clock rates above 50 MHz. This combined with their low expense, ease of use and selected dedicated hardware make them an ideal option for spectrometer data acquisition systems. For this generation of spectrometers, functions that are typically performed in dedicated circuits, or offline, are being migrated to the field programmable gate array (FPGA). This will not only reduce the electronics, but the features of modern FPGAs can be utilized to add considerable signal processing power to produce higher resolution spectra. In this paper we report on an all-digital triangle bipolar pulse shaping and pileup correction algorithm that is being developed for the DSP. The pileup mitigation algorithm will allow the spectrometers to run at higher count rates or with multiple sources without imposing large data losses due to the overlapping of scintillation signals. This correction technique utilizes a very narrow bipolar triangle digital pulse shaping algorithm to extract energy information for most pileup events.

  13. Triangle bipolar pulse shaping and pileup correction based on DSP

    Energy Technology Data Exchange (ETDEWEB)

    Esmaeili-sani, Vahid, E-mail: vaheed_esmaeely80@yahoo.com [Department of Nuclear Engineering and Physics, Amirkabir University of Technology, P.O. Box 4155-4494, Tehran (Iran, Islamic Republic of); Moussavi-zarandi, Ali; Akbar-ashrafi, Nafiseh; Boghrati, Behzad [Department of Nuclear Engineering and Physics, Amirkabir University of Technology, P.O. Box 4155-4494, Tehran (Iran, Islamic Republic of)

    2011-02-11

    Programmable Digital Signal Processing (DSP) microprocessors are capable of doing complex discrete signal processing algorithms with clock rates above 50 MHz. This combined with their low expense, ease of use and selected dedicated hardware make them an ideal option for spectrometer data acquisition systems. For this generation of spectrometers, functions that are typically performed in dedicated circuits, or offline, are being migrated to the field programmable gate array (FPGA). This will not only reduce the electronics, but the features of modern FPGAs can be utilized to add considerable signal processing power to produce higher resolution spectra. In this paper we report on an all-digital triangle bipolar pulse shaping and pileup correction algorithm that is being developed for the DSP. The pileup mitigation algorithm will allow the spectrometers to run at higher count rates or with multiple sources without imposing large data losses due to the overlapping of scintillation signals. This correction technique utilizes a very narrow bipolar triangle digital pulse shaping algorithm to extract energy information for most pileup events.

  14. A DSP-based neural network non-uniformity correction algorithm for IRFPA

    Science.gov (United States)

    Liu, Chong-liang; Jin, Wei-qi; Cao, Yang; Liu, Xiu

    2009-07-01

    An effective neural network non-uniformity correction (NUC) algorithm based on DSP is proposed in this paper. The non-uniform response in infrared focal plane array (IRFPA) detectors produces corrupted images with a fixed-pattern noise(FPN).We introduced and analyzed the artificial neural network scene-based non-uniformity correction (SBNUC) algorithm. A design of DSP-based NUC development platform for IRFPA is described. The DSP hardware platform designed is of low power consumption, with 32-bit fixed point DSP TMS320DM643 as the kernel processor. The dependability and expansibility of the software have been improved by DSP/BIOS real-time operating system and Reference Framework 5. In order to realize real-time performance, the calibration parameters update is set at a lower task priority then video input and output in DSP/BIOS. In this way, calibration parameters updating will not affect video streams. The work flow of the system and the strategy of real-time realization are introduced. Experiments on real infrared imaging sequences demonstrate that this algorithm requires only a few frames to obtain high quality corrections. It is computationally efficient and suitable for all kinds of non-uniformity.

  15. Criteria for the use of digital signal processors in the control technique of the COSY particle accelerator using the example of the MOTOROLA DSP56000

    International Nuclear Information System (INIS)

    Rath, U.

    1989-11-01

    On the Cooler Synchrotron project (COSY), the beam measurement data and their processing are collected digitally. From the requirements for quick computing time (real time operation) and exact results, the use of digital signal processors is intended. The digital signal processor DSP 56000 from MOTOROLA was selected as the test object. The DSP 56000 has a development environment which makes it possible to test it on an IBM-PC AT. Tests are carried out which show that the simulation program corresponds to the functions and processes of the DSP 56000. The above-mentioned applications program calculates a 'fast Fourier transform' (FFT). This program is used to judge the speed of calculation and the accuracy of calculation of the signal processor. The algorithm used by the FFT program is explained. In order to judge the results of the DSP 56000, a comparison is made with the equivalent FORTRAN FFT. The results which the DSP gives on the ADM and the Fortran program are compared and assessed. The speed of calculation of the DSP 56000 is determined and is judged in comparison with the manufacturer's data for other digital signal processors. (orig./HP) [de

  16. DSP for Matlab and Labview I fundamentals of discrete signal processing

    CERN Document Server

    Isen, Forester W

    2009-01-01

    This book is Volume I of the series DSP for MATLAB™ and LabVIEW™. The entire series consists of four volumes that collectively cover basic digital signal processing in a practical and accessible manner, but which nonetheless include all essential foundation mathematics. As the series title implies, the scripts (of which there are more than 200) described in the text and supplied in code form here will run on both MATLAB and LabVIEW. Volume I consists of four chapters. The first chapter gives a brief overview of the field of digital signal processing. This is followed by a chapter detailing man

  17. EXPERIMENTAL STUDY OF FIRMWARE FOR INPUT AND EXTRACTION OF USER’S VOICE SIGNAL IN VOICE AUTHENTICATION SYSTEMS

    Directory of Open Access Journals (Sweden)

    O. N. Faizulaieva

    2014-09-01

    Full Text Available Scientific task for improving the signal-to-noise ratio for user’s voice signal in computer systems and networks during the process of user’s voice authentication is considered. The object of study is the process of input and extraction of the voice signal of authentication system user in computer systems and networks. Methods and means for input and extraction of the voice signal on the background of external interference signals are investigated. Ways for quality improving of the user’s voice signal in systems of voice authentication are investigated experimentally. Firmware means for experimental unit of input and extraction of the user’s voice signal against external interference influence are considered. As modern computer means, including mobile, have two-channel audio card, two microphones are used in the voice signal input. The distance between sonic-wave sensors is 20 mm and it provides forming one direction pattern lobe of microphone array in a desired area of voice signal registration (from 100 Hz to 8 kHz. According to the results of experimental studies, the usage of directional properties of the proposed microphone array and space-time processing of the recorded signals with implementation of constant and adaptive weighting factors has made it possible to reduce considerably the influence of interference signals. The results of firmware experimental studies for input and extraction of the user’s voice signal against external interference influence are shown. The proposed solutions will give the possibility to improve the value of the signal/noise ratio of the useful signals recorded up to 20 dB under the influence of external interference signals in the frequency range from 4 to 8 kHz. The results may be useful to specialists working in the field of voice recognition and speaker discrimination.

  18. Hamming Weight Counters and Comparators based on Embedded DSP Blocks for Implementation in FPGA

    Directory of Open Access Journals (Sweden)

    SKLYAROV, V.

    2014-05-01

    Full Text Available This paper is dedicated to the design, implementation and evaluation of fast FPGA-based circuits that compute Hamming weights for binary vectors and compare the results with fixed thresholds and variable bounds. It is shown that digital signal processing (DSP slices that are widely available in contemporary FPGAs may be used efficiently and they frequently provide the fastest and least resource consuming solutions. A thorough analysis and comparison of these with the best known alternatives both in hardware and in software is presented. The results are supported by numerous experiments in recent prototyping boards. A fully synthesizable hardware description language (VHDL specification for one of the proposed core components is given that is ready to be synthesized, implemented, tested and compared in any FPGA that contains embedded DSP48E1 slices (or alternatively DSP48A1 slices from previous generations. Finally, the results of comparisons are provided that include discussions of designs in an ARM processor combined with reconfigurable logic for very long vectors.

  19. Method of power self-regulation of CFBR-II reactor based on DSP

    International Nuclear Information System (INIS)

    Bai Zhongxiong; Zhou Wenxiang

    2007-01-01

    To the control system of Power Self-regulation of CFBR-II Reactor, a new digital control scheme based on DSP has been brought forward. The TMS320F2812 DSP chip is adopted as the core controller to realize Power self-regulation of CFBR-II Reactor. In this paper, the successful program of DSP control system is introduced in both hardware and software technology in detail. (authors)

  20. The development of special equipment amplitude detection instrument based on DSP

    International Nuclear Information System (INIS)

    Dai Sidan; Chen Ligang; Lan Peng; Wang Huiting; Zhang Liangxu; Wang Lin

    2014-01-01

    Development and industrial application of special equipment plays an important role in the development of nuclear energy process. Equipment development process need to do a lot of tests, amplitude detection is a key test,it can analysis the device's electromechanical and physical properties. In the industrial application, the amplitude detection can effectively reflect the operational status of the current equipment, the equipment can also be a certain degree of fault diagnosis, identify problems in a timely manner. The main development target in this article is amplitude detection of special equipment. This article describes the development of special equipment amplitude detection instrument. The instrument uses a digital signal processor (DSP) as the central processing unit, and uses the DSP + CPLD + high-speed AD technology to build a complete set of high-precision signal acquisition and analysis processing systems, rechargeable lithium battery as the powered device. It can do a online monitoring of special equipment amplitude, speed parameters by acquiring and analysing the tachometer signal in the special equipment, and locally display through the LCD screen. (authors)

  1. Design And Implementation of Dsp-Based Intelligent Controller For Automobile Braking System

    Directory of Open Access Journals (Sweden)

    S.N. Sidek and M.J.E. Salami

    2012-08-01

    Full Text Available An intelligent braking system has great potential applications especially, in developed countries where research on smart vehicle and intelligent highways are receiving ample attention. The system when integrated with other subsystems like automatic traction control, intelligent throttle, and auto cruise systems, etc will result in smart vehicle maneuver. The driver at the end of the day will become the passenger, safety accorded the highest priority and the journey optimized in term of time duration, cost, efficiency and comfortability. The impact of such design and development will cater for the need of contemporary society that aspires to a quality drive as well as to accommodate the advancement of technology especially in the area of smart sensors and actuators.  The emergence of digital signal processor enhances the capacity and features of universal microcontroller.  This paper introduces the use of TI DSP, TMS320LF2407 as an engine of the system. The overall system is designed so that the value of inter-vehicle distance from infrared laser sensor and speed of follower car from speedometer are fed into the DSP for processing, resulting in the DSP issuing commands to the actuator to function appropriately.Key words:  Smart Vehicle, Digital Signal Processor, Fuzzy Controller, and Infra Red Laser Sensor

  2. DSP Architecture Design Essentials

    CERN Document Server

    Marković, Dejan

    2012-01-01

    In DSP Architecture Design Essentials, authors Dejan Marković and Robert W. Brodersen cover a key subject for the successful realization of DSP algorithms for communications, multimedia, and healthcare applications. The book addresses the need for DSP architecture design that maps advanced DSP algorithms to hardware in the most power- and area-efficient way. The key feature of this text is a design methodology based on a high-level design model that leads to hardware implementation with minimum power and area. The methodology includes algorithm-level considerations such as automated word-length reduction and intrinsic data properties that can be leveraged to reduce hardware complexity. From a high-level data-flow graph model, an architecture exploration methodology based on linear programming is used to create an array of architectural solutions tailored to the underlying hardware technology. The book is supplemented with online material: bibliography, design examples, CAD tutorials and custom software.

  3. 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)

  4. Design and Implementation of a FPGA and DSP Based MIMO Radar Imaging System

    Directory of Open Access Journals (Sweden)

    Wei Wang

    2015-06-01

    Full Text Available The work presented in this paper is aimed at the implementation of a real-time multiple-input multiple-output (MIMO imaging radar used for area surveillance. In this radar, the equivalent virtual array method and time-division technique are applied to make 16 virtual elements synthesized from the MIMO antenna array. The chirp signal generater is based on a combination of direct digital synthesizer (DDS and phase locked loop (PLL. A signal conditioning circuit is used to deal with the coupling effect within the array. The signal processing platform is based on an efficient field programmable gates array (FPGA and digital signal processor (DSP pipeline where a robust beamforming imaging algorithm is running on. The radar system was evaluated through a real field experiment. Imaging capability and real-time performance shown in the results demonstrate the practical feasibility of the implementation.

  5. Perception SoC Based on an Ultrasonic Array of Sensors: Efficient DSP Core Implementation and Subsequent Experimental Results

    Directory of Open Access Journals (Sweden)

    A. Haidar

    2005-05-01

    Full Text Available We are concerned with the design, implementation, and validation of a perception SoC based on an ultrasonic array of sensors. The proposed SoC is dedicated to ultrasonic echography applications. A rapid prototyping platform is used to implement and validate the new architecture of the digital signal processing (DSP core. The proposed DSP core efficiently integrates all of the necessary ultrasonic B-mode processing modules. It includes digital beamforming, quadrature demodulation of RF signals, digital filtering, and envelope detection of the received signals. This system handles 128 scan lines and 6400 samples per scan line with a 90° angle of view span. The design uses a minimum size lookup memory to store the initial scan information. Rapid prototyping using an ARM/FPGA combination is used to validate the operation of the described system. This system offers significant advantages of portability and a rapid time to market.

  6. Perception SoC Based on an Ultrasonic Array of Sensors: Efficient DSP Core Implementation and Subsequent Experimental Results

    Science.gov (United States)

    Kassem, A.; Sawan, M.; Boukadoum, M.; Haidar, A.

    2005-12-01

    We are concerned with the design, implementation, and validation of a perception SoC based on an ultrasonic array of sensors. The proposed SoC is dedicated to ultrasonic echography applications. A rapid prototyping platform is used to implement and validate the new architecture of the digital signal processing (DSP) core. The proposed DSP core efficiently integrates all of the necessary ultrasonic B-mode processing modules. It includes digital beamforming, quadrature demodulation of RF signals, digital filtering, and envelope detection of the received signals. This system handles 128 scan lines and 6400 samples per scan line with a[InlineEquation not available: see fulltext.] angle of view span. The design uses a minimum size lookup memory to store the initial scan information. Rapid prototyping using an ARM/FPGA combination is used to validate the operation of the described system. This system offers significant advantages of portability and a rapid time to market.

  7. A DSP based data acquisition module for colliding beam accelerators

    International Nuclear Information System (INIS)

    Mead, J.A.; Shea, T.J.

    1995-10-01

    In 1999, the Relativistic Heavy Ion Collider (RHIC) at Brookhaven National Laboratory will accelerate and store two beams of gold ions. The ions will then collide head on at a total energy of nearly 40 trillion electron volts. Attaining these conditions necessitates real-time monitoring of beam parameters and for this purpose a flexible data acquisition platform has been developed. By incorporating a floating point digital signal processor (DSP) and standard input/output modules, this system can acquire and process data from a variety of beam diagnostic devices. The DSP performs real time corrections, filtering, and data buffering to greatly reduce control system computation and bandwidth requirements. We will describe the existing hardware and software while emphasizing the compromises required to achieve a flexible yet cost effective system. Applications in several instrumentation systems currently construction will also be presented

  8. Design of adaptive filter amplifier in UV communication based on DSP

    Science.gov (United States)

    Lv, Zhaoshun; Wu, Hanping; Li, Junyu

    2016-10-01

    According to the problem of the weak signal at receiving end in UV communication, we design a high gain, continuously adjustable adaptive filter amplifier. Based on proposing overall technical indicators and analyzing its working principle of the signal amplifier, we use chip LMH6629MF and two chips of AD797BN to achieve three-level cascade amplification. And apply hardware of DSP TMS320VC5509A to implement digital filtering. Design and verification by Multisim, Protel 99SE and CCS, the results show that: the amplifier can realize continuously adjustable amplification from 1000 to 10000 times without distortion. Magnification error is <=%4@1000 10000. And equivalent input noise voltage of amplification circuit is <=6 nV/ √Hz @30KHz 45KHz, and realizing function of adaptive filtering. The design provides theoretical reference and technical support for the UV weak signal processing.

  9. Development of a system based in a digital signal processor (DSP) for a simulator of power regulation in a reactor: first stage

    International Nuclear Information System (INIS)

    Benitez R, J.S.; Perez C, B.

    2002-01-01

    The first stage of the development of a digital system based on a DSP is presented which forms part of an hybrid simulator for the power regulation in am model of the punctual kinetics of a TRIGA reactor type. The DSP performs the regulation, using a Mandami type algorithm of diffuse control. In the algorithm, the universe of the output variable is discretized for performing in an unique stage the aggregation functions and dis-diffusization. (Author)

  10. MCNP-DSP, Monte Carlo Neutron-Particle Transport Code with Digital Signal Processing

    International Nuclear Information System (INIS)

    2002-01-01

    1 - Description of program or function: MCNP-DSP is recommended only for experienced MCNP users working with subcritical measurements. It is a modification of the Los Alamos National Laboratory's Monte Carlo code MCNP4a that is used to simulate a variety of subcritical measurements. The DSP version was developed to simulate frequency analysis measurements, correlation (Rossi-) measurements, pulsed neutron measurements, Feynman variance measurements, and multiplicity measurements. CCC-700/MCNP4C is recommended for general purpose calculations. 2 - Methods:MCNP-DSP performs calculations very similarly to MCNP and uses the same generalized geometry capabilities of MCNP. MCNP-DSP can only be used with the continuous-energy cross-section data. A variety of source and detector options are available. However, unlike standard MCNP, the source and detector options are limited to those described in the manual because these options are specified in the MCNP-DSP extra data file. MCNP-DSP is used to obtain the time-dependent response of detectors that are modeled in the simulation geometry. The detectors represent actual detectors used in measurements. These time-dependent detector responses are used to compute a variety of quantities such as frequency analysis signatures, correlation signatures, multiplicity signatures, etc., between detectors or sources and detectors. Energy ranges are 0-60 MeV for neutrons (data generally only available up to 20 MeV) and 1 keV - 1 GeV for photons and electrons. 3 - Restrictions on the complexity of the problem: None noted

  11. DSP accelerator for the wavelet compression/decompression of high- resolution images

    Energy Technology Data Exchange (ETDEWEB)

    Hunt, M.A.; Gleason, S.S.; Jatko, W.B.

    1993-07-23

    A Texas Instruments (TI) TMS320C30-based S-Bus digital signal processing (DSP) module was used to accelerate a wavelet-based compression and decompression algorithm applied to high-resolution fingerprint images. The law enforcement community, together with the National Institute of Standards and Technology (NISI), is adopting a standard based on the wavelet transform for the compression, transmission, and decompression of scanned fingerprint images. A two-dimensional wavelet transform of the input image is computed. Then spatial/frequency regions are automatically analyzed for information content and quantized for subsequent Huffman encoding. Compression ratios range from 10:1 to 30:1 while maintaining the level of image quality necessary for identification. Several prototype systems were developed using SUN SPARCstation 2 with a 1280 {times} 1024 8-bit display, 64-Mbyte random access memory (RAM), Tiber distributed data interface (FDDI), and Spirit-30 S-Bus DSP-accelerators from Sonitech. The final implementation of the DSP-accelerated algorithm performed the compression or decompression operation in 3.5 s per print. Further increases in system throughput were obtained by adding several DSP accelerators operating in parallel.

  12. Servo Platform Circuit Design of Pendulous Gyroscope Based on DSP

    Science.gov (United States)

    Tan, Lilong; Wang, Pengcheng; Zhong, Qiyuan; Zhang, Cui; Liu, Yunfei

    2018-03-01

    In order to solve the problem when a certain type of pendulous gyroscope in the initial installation deviation more than 40 degrees, that the servo platform can not be up to the speed of the gyroscope in the rough north seeking phase. This paper takes the digital signal processor TMS320F28027 as the core, uses incremental digital PID algorithm, carries out the circuit design of the servo platform. Firstly, the hardware circuit is divided into three parts: DSP minimum system, motor driving circuit and signal processing circuit, then the mathematical model of incremental digital PID algorithm is established, based on the model, writes the PID control program in CCS3.3, finally, the servo motor tracking control experiment is carried out, it shows that the design can significantly improve the tracking ability of the servo platform, and the design has good engineering practice.

  13. A Novel Fast and Secure Approach for Voice Encryption Based on DNA Computing

    Science.gov (United States)

    Kakaei Kate, Hamidreza; Razmara, Jafar; Isazadeh, Ayaz

    2018-06-01

    Today, in the world of information communication, voice information has a particular importance. One way to preserve voice data from attacks is voice encryption. The encryption algorithms use various techniques such as hashing, chaotic, mixing, and many others. In this paper, an algorithm is proposed for voice encryption based on three different schemes to increase flexibility and strength of the algorithm. The proposed algorithm uses an innovative encoding scheme, the DNA encryption technique and a permutation function to provide a secure and fast solution for voice encryption. The algorithm is evaluated based on various measures including signal to noise ratio, peak signal to noise ratio, correlation coefficient, signal similarity and signal frequency content. The results demonstrate applicability of the proposed method in secure and fast encryption of voice files

  14. Customized DSP-based vibration measurement for wind turbines

    Energy Technology Data Exchange (ETDEWEB)

    LaWhite, N.E.; Cohn, K.E. [Second Wind Inc., Somerville, MA (United States)

    1996-12-31

    As part of its Advanced Distributed Monitoring System (ADMS) project funded by NREL, Second Wind Inc. is developing a new vibration measurement system for use with wind turbines. The system uses low-cost accelerometers originally designed for automobile airbag crash-detection coupled with new software executed on a Digital Signal Processor (DSP) device. The system is envisioned as a means to monitor the mechanical {open_quotes}health{close_quotes} of the wind turbine over its lifetime. In addition the system holds promise as a customized emergency vibration detector. The two goals are very different and it is expected that different software programs will be executed for each function. While a fast Fourier transform (FFT) signature under given operating conditions can yield much information regarding turbine condition, the sampling period and processing requirements make it inappropriate for emergency condition monitoring. This paper briefly reviews the development of prototype DSP and accelerometer hardware. More importantly, it reviews our work to design prototype vibration alarm filters. Two-axis accelerometer test data from the experimental FloWind vertical axis wind turbine is analyzed and used as a development guide. Two levels of signal processing are considered. The first uses narrow band pre-processing filters at key fundamental frequencies such as the 1P, 2P and 3P. The total vibration energy in each frequency band is calculated and evaluated as a possible alarm trigger. In the second level of signal processing, the total vibration energy in each frequency band is further decomposed using the two-axis directional information. Directional statistics are calculated to differentiate between linear translations and circular translations. After analyzing the acceleration statistics for normal and unusual operating conditions, the acceleration processing system described could be used in automatic early detection of fault conditions. 9 figs.

  15. Multiformat decoder for a DSP-based IP set-top box

    Science.gov (United States)

    Pescador, F.; Garrido, M. J.; Sanz, C.; Juárez, E.; Samper, D.; Antoniello, R.

    2007-05-01

    Internet Protocol Set-Top Boxes (IP STBs) based on single-processor architectures have been recently introduced in the market. In this paper, the implementation of an MPEG-4 SP/ASP video decoder for a multi-format IP STB based on a TMS320DM641 DSP is presented. An initial decoder for PC platform was fully tested and ported to the DSP. Using this code an optimization process was started achieving a 90% speedup. This process allows real-time MPEG-4 SP/ASP decoding. The MPEG-4 decoder has been integrated in an IP STB and tested in a real environment using DVD movies and TV channels with excellent results.

  16. DSP-based CSO cancellation technique for RoF transmission system implemented by using directly modulated laser.

    Science.gov (United States)

    Kim, Byung Gon; Bae, Sung Hyun; Kim, Hoon; Chung, Yun C

    2017-05-29

    We propose and demonstrate a simple composite second-order (CSO) cancellation technique based on the digital signal processing (DSP) for the radio-over-fiber (RoF) transmission system implemented by using directly modulated lasers (DMLs). When the RoF transmission system is implemented by using DMLs, its performance could be limited by the CSO distortions caused by the interplay between the DML's chirp and fiber's chromatic dispersion. We present the theoretical analysis of these nonlinear distortions and show that they can be suppressed at the receiver by using a simple DSP. To verify the effectiveness of the proposed technique, we demonstrate the transmission of twenty-four 100-MHz filtered orthogonal frequency-division multiplexing (f-OFDM) signals in 64 quadrature amplitude modulation (QAM) format over 20 km of the standard single-mode fiber (SSMF). The results show that, by using the proposed technique, we can suppress the CSO distortion components by >10 dB and achieve the error-vector magnitude performance better than 6% even after the 20-km long SSMF transmission.

  17. Start/End Delays of Voiced and Unvoiced Speech Signals

    Energy Technology Data Exchange (ETDEWEB)

    Herrnstein, A

    1999-09-24

    Recent experiments using low power EM-radar like sensors (e.g, GEMs) have demonstrated a new method for measuring vocal fold activity and the onset times of voiced speech, as vocal fold contact begins to take place. Similarly the end time of a voiced speech segment can be measured. Secondly it appears that in most normal uses of American English speech, unvoiced-speech segments directly precede or directly follow voiced-speech segments. For many applications, it is useful to know typical duration times of these unvoiced speech segments. A corpus, assembled earlier of spoken ''Timit'' words, phrases, and sentences and recorded using simultaneously measured acoustic and EM-sensor glottal signals, from 16 male speakers, was used for this study. By inspecting the onset (or end) of unvoiced speech, using the acoustic signal, and the onset (or end) of voiced speech using the EM sensor signal, the average duration times for unvoiced segments preceding onset of vocalization were found to be 300ms, and for following segments, 500ms. An unvoiced speech period is then defined in time, first by using the onset of the EM-sensed glottal signal, as the onset-time marker for the voiced speech segment and end marker for the unvoiced segment. Then, by subtracting 300ms from the onset time mark of voicing, the unvoiced speech segment start time is found. Similarly, the times for a following unvoiced speech segment can be found. While data of this nature have proven to be useful for work in our laboratory, a great deal of additional work remains to validate such data for use with general populations of users. These procedures have been useful for applying optimal processing algorithms over time segments of unvoiced, voiced, and non-speech acoustic signals. For example, these data appear to be of use in speaker validation, in vocoding, and in denoising algorithms.

  18. Illustration of decimation in digital signal processing (DSP) systems ...

    African Journals Online (AJOL)

    ... and engineering, especially in the areas of communication and medicine. ... This multirate DSP had been found useful in application like digital audio, video and even GSM technology. The work is implemented using MATLABTM software.

  19. Multi-DSP and FPGA based Multi-channel Direct IF/RF Digital receiver for atmospheric radar

    Science.gov (United States)

    Yasodha, Polisetti; Jayaraman, Achuthan; Kamaraj, Pandian; Durga rao, Meka; Thriveni, A.

    2016-07-01

    to DDC block, which down converts the data to base-band. The DDC block has NCO, mixer and two chains of Bessel filters (fifth order cascaded integration comb filter, two FIR filters, two half band filters and programmable FIR filters) for in-phase (I) and Quadrature phase (Q) channels. The NCO has 32 bits and is set to match the output frequency of ADC. Further, DDC down samples (decimation) the data and reduces the data rate to 16 MSPS. This data is further decimated and the data rate is reduced down to 4/2/1/0.5/0.25/0.125/0.0625 MSPS for baud lengths 0.25/0.5/1/2/4/8/16 μs respectively. The down sampled data is then fed to decoding block, which performs cross correlation to achieve pulse compression of the binary-phase coded data to obtain better range resolution with maximum possible height coverage. This step improves the signal power by a factor equal to the length of the code. Coherent integration block integrates the decoded data coherently for successive pulses, which improves the signal to noise ratio and reduces the data volume. DDC, decoding and coherent integration blocks are implemented in Xilinx vertex5 FPGA. Till this point, function of all six channels is same for DBS mode and multi-receiver modes. Data from vertex5 FPGA is transferred to PC via GbE-1 interface for multi-modes or to two Analog devices make ADSP-TS201 DSP chips (A and B), via link port for DBS mode. ADSP-TS201 chips perform the normalization, DC removal, windowing, FFT computation and spectral averaging on the data, which is transferred to storage/display PC via GbE-2 interface for real-time data display and data storing. Physical layer of GbE interface is implemented in an external chip (Marvel 88E1111) and MAC layer is implemented internal to vertex5 FPGA. The MCDRx has total 4 GB of DDR2 memory for data storage. Spartan6 FPGA is used for generating timing signals, required for basic operation of the radar and testing of the MCDRx.

  20. DSP Based Direct Torque Control of Permanent Magnet Synchronous Motor (PMSM) using Space Vector Modulation (DTC-SVM)

    DEFF Research Database (Denmark)

    Swierczynski, Dariusz; Kazmierkowski, Marian P.; Blaabjerg, Frede

    2002-01-01

    DSP Based Direct Torque Control of Permanent Magnet Synchronous Motor (PMSM) using Space Vector Modulation (DTC-SVM)......DSP Based Direct Torque Control of Permanent Magnet Synchronous Motor (PMSM) using Space Vector Modulation (DTC-SVM)...

  1. High performance 3D adaptive filtering for DSP based portable medical imaging systems

    Science.gov (United States)

    Bockenbach, Olivier; Ali, Murtaza; Wainwright, Ian; Nadeski, Mark

    2015-03-01

    Portable medical imaging devices have proven valuable for emergency medical services both in the field and hospital environments and are becoming more prevalent in clinical settings where the use of larger imaging machines is impractical. Despite their constraints on power, size and cost, portable imaging devices must still deliver high quality images. 3D adaptive filtering is one of the most advanced techniques aimed at noise reduction and feature enhancement, but is computationally very demanding and hence often cannot be run with sufficient performance on a portable platform. In recent years, advanced multicore digital signal processors (DSP) have been developed that attain high processing performance while maintaining low levels of power dissipation. These processors enable the implementation of complex algorithms on a portable platform. In this study, the performance of a 3D adaptive filtering algorithm on a DSP is investigated. The performance is assessed by filtering a volume of size 512x256x128 voxels sampled at a pace of 10 MVoxels/sec with an Ultrasound 3D probe. Relative performance and power is addressed between a reference PC (Quad Core CPU) and a TMS320C6678 DSP from Texas Instruments.

  2. GSM Channel Equalization Algorithm - Modern DSP Coprocessor Approach

    Directory of Open Access Journals (Sweden)

    M. Drutarovsky

    1999-12-01

    Full Text Available The paper presents basic equations of efficient GSM Viterbi equalizer algorithm based on approximation of GMSK modulation by linear superposition of amplitude modulated pulses. This approximation allows to use Ungerboeck form of channel equalizer with significantly reduced arithmetic complexity. Proposed algorithm can be effectively implemented on the Viterbi and Filter coprocessors of new Motorola DSP56305 digital signal processor. Short overview of coprocessor features related to the proposed algorithm is included.

  3. Feasibility Study on a Portable Field Pest Classification System Design Based on DSP and 3G Wireless Communication Technology

    Directory of Open Access Journals (Sweden)

    Fei Liu

    2012-03-01

    Full Text Available This paper presents a feasibility study on a real-time in field pest classification system design based on Blackfin DSP and 3G wireless communication technology. This prototype system is composed of remote on-line classification platform (ROCP, which uses a digital signal processor (DSP as a core CPU, and a host control platform (HCP. The ROCP is in charge of acquiring the pest image, extracting image features and detecting the class of pest using an Artificial Neural Network (ANN classifier. It sends the image data, which is encoded using JPEG 2000 in DSP, to the HCP through the 3G network at the same time for further identification. The image transmission and communication are accomplished using 3G technology. Our system transmits the data via a commercial base station. The system can work properly based on the effective coverage of base stations, no matter the distance from the ROCP to the HCP. In the HCP, the image data is decoded and the pest image displayed in real-time for further identification. Authentication and performance tests of the prototype system were conducted. The authentication test showed that the image data were transmitted correctly. Based on the performance test results on six classes of pests, the average accuracy is 82%. Considering the different live pests’ pose and different field lighting conditions, the result is satisfactory. The proposed technique is well suited for implementation in field pest classification on-line for precision agriculture.

  4. Feasibility study on a portable field pest classification system design based on DSP and 3G wireless communication technology.

    Science.gov (United States)

    Han, Ruizhen; He, Yong; Liu, Fei

    2012-01-01

    This paper presents a feasibility study on a real-time in field pest classification system design based on Blackfin DSP and 3G wireless communication technology. This prototype system is composed of remote on-line classification platform (ROCP), which uses a digital signal processor (DSP) as a core CPU, and a host control platform (HCP). The ROCP is in charge of acquiring the pest image, extracting image features and detecting the class of pest using an Artificial Neural Network (ANN) classifier. It sends the image data, which is encoded using JPEG 2000 in DSP, to the HCP through the 3G network at the same time for further identification. The image transmission and communication are accomplished using 3G technology. Our system transmits the data via a commercial base station. The system can work properly based on the effective coverage of base stations, no matter the distance from the ROCP to the HCP. In the HCP, the image data is decoded and the pest image displayed in real-time for further identification. Authentication and performance tests of the prototype system were conducted. The authentication test showed that the image data were transmitted correctly. Based on the performance test results on six classes of pests, the average accuracy is 82%. Considering the different live pests' pose and different field lighting conditions, the result is satisfactory. The proposed technique is well suited for implementation in field pest classification on-line for precision agriculture.

  5. Modular uncooled video engines based on a DSP processor

    Science.gov (United States)

    Schapiro, F.; Milstain, Y.; Aharon, A.; Neboshchik, A.; Ben-Simon, Y.; Kogan, I.; Lerman, I.; Mizrahi, U.; Maayani, S.; Amsterdam, A.; Vaserman, I.; Duman, O.; Gazit, R.

    2011-06-01

    The market demand for low SWaP (Size, Weight and Power) uncooled engines keeps growing. Low SWaP is especially critical in battery-operated applications such as goggles and Thermal Weapon Sights. A new approach for the design of the engines was implemented by SCD to optimize size and power consumption at system level. The new approach described in the paper, consists of: 1. A modular hardware design that allows the user to define the exact level of integration needed for his system 2. An "open architecture" based on the OMAPTM530 DSP that allows the integrator to take advantage of unused hardware (FPGA) and software (DSP) resources, for implementation of additional algorithms or functionality. The approach was successfully implemented on the first generation of 25μm pitch BIRD detectors, and more recently on the new, 640 x480, 17 μm pitch detector.

  6. A High-Precision Counter Using the DSP Technique

    National Research Council Canada - National Science Library

    Chen, Shang-Shian

    2004-01-01

    .... We use an analog-to-digital converter (ADC) to sample the device under test (DUT). Once the signal is digitized, the DSP will be used to run the phase correlation and obtain the necessary information...

  7. A Computer- Based Digital Signal Processing for Nuclear Scintillator Detectors

    International Nuclear Information System (INIS)

    Ashour, M.A.; Abo Shosha, A.M.

    2000-01-01

    In this paper, a Digital Signal Processing (DSP) Computer-based system for the nuclear scintillation signals with exponential decay is presented. The main objective of this work is to identify the characteristics of the acquired signals smoothly, this can be done by transferring the signal environment from random signal domain to deterministic domain using digital manipulation techniques. The proposed system consists of two major parts. The first part is the high performance data acquisition system (DAQ) that depends on a multi-channel Logic Scope. Which is interfaced with the host computer through the General Purpose Interface Board (GPIB) Ver. IEEE 488.2. Also, a Graphical User Interface (GUI) has been designed for this purpose using the graphical programming facilities. The second of the system is the DSP software Algorithm which analyses, demonstrates, monitoring these data to obtain the main characteristics of the acquired signals; the amplitude, the pulse count, the pulse width, decay factor, and the arrival time

  8. Real-time video compressing under DSP/BIOS

    Science.gov (United States)

    Chen, Qiu-ping; Li, Gui-ju

    2009-10-01

    This paper presents real-time MPEG-4 Simple Profile video compressing based on the DSP processor. The programming framework of video compressing is constructed using TMS320C6416 Microprocessor, TDS510 simulator and PC. It uses embedded real-time operating system DSP/BIOS and the API functions to build periodic function, tasks and interruptions etcs. Realize real-time video compressing. To the questions of data transferring among the system. Based on the architecture of the C64x DSP, utilized double buffer switched and EDMA data transfer controller to transit data from external memory to internal, and realize data transition and processing at the same time; the architecture level optimizations are used to improve software pipeline. The system used DSP/BIOS to realize multi-thread scheduling. The whole system realizes high speed transition of a great deal of data. Experimental results show the encoder can realize real-time encoding of 768*576, 25 frame/s video images.

  9. The research of period measuring instruments on zero power assembly based on DSP

    International Nuclear Information System (INIS)

    Bai Zhongxiong

    2007-12-01

    In order to improving measure precision and anti-interference capacity, and respond to the digital trend, a new technique to measure reactor period is promoted, which is based on the DSP technique, calculate period with least-squares-fitting method. The systematic design is promoted, in which TMS320F2812 chip is chosen as the Central Processing/Controlling unit and software design is based on DSP/BIOS embedded operating system. Testing of both a simulation of the lab environment and an experiment shows that, as expected, the new TMS320F2812 based reactor period inspection equipment has excellent anti-interference capacity, high precision and fast response time, all of which prove that it has good prospective. (authors)

  10. Development of a system based in a digital signal processor (DSP) for a simulator of power regulation in a reactor: first stage; Desarrollo de un sistema basado en un DSP para un simulador de regulacion de potencia en un reactor: 1. etapa

    Energy Technology Data Exchange (ETDEWEB)

    Benitez R, J.S.; Perez C, B. [Instituto Nacional de Investigaciones Nucleares, Km. 36.5 Carretera Mexico-Toluca, Municipio de Ocoyoacac, 52045 Estado de Mexico (Mexico)

    2002-07-01

    The first stage of the development of a digital system based on a DSP is presented which forms part of an hybrid simulator for the power regulation in am model of the punctual kinetics of a TRIGA reactor type. The DSP performs the regulation, using a Mandami type algorithm of diffuse control. In the algorithm, the universe of the output variable is discretized for performing in an unique stage the aggregation functions and dis-diffusization. (Author)

  11. Optimized design of embedded DSP system hardware supporting complex algorithms

    Science.gov (United States)

    Li, Yanhua; Wang, Xiangjun; Zhou, Xinling

    2003-09-01

    The paper presents an optimized design method for a flexible and economical embedded DSP system that can implement complex processing algorithms as biometric recognition, real-time image processing, etc. It consists of a floating-point DSP, 512 Kbytes data RAM, 1 Mbytes FLASH program memory, a CPLD for achieving flexible logic control of input channel and a RS-485 transceiver for local network communication. Because of employing a high performance-price ratio DSP TMS320C6712 and a large FLASH in the design, this system permits loading and performing complex algorithms with little algorithm optimization and code reduction. The CPLD provides flexible logic control for the whole DSP board, especially in input channel, and allows convenient interface between different sensors and DSP system. The transceiver circuit can transfer data between DSP and host computer. In the paper, some key technologies are also introduced which make the whole system work efficiently. Because of the characters referred above, the hardware is a perfect flat for multi-channel data collection, image processing, and other signal processing with high performance and adaptability. The application section of this paper presents how this hardware is adapted for the biometric identification system with high identification precision. The result reveals that this hardware is easy to interface with a CMOS imager and is capable of carrying out complex biometric identification algorithms, which require real-time process.

  12. Research of Data Acquisition and Analysis System for Internal Combustion Engine Based on DSP

    International Nuclear Information System (INIS)

    Gao, Y H; Tian, X L; Cheng, P; Chang, X; Dou, W J

    2006-01-01

    In the paper, the structure, working principle, functions and characteristics of an data acquisition and analysis system for internal combustion engines (I.C. engine) based on DSP is introduced. The DSP can not only acquire and analyze the data alone, also can work with the PC together to form data acquisition and analysis system with high speed and large memory. The system takes advantages of TMS320F2812's plenty of peripherals on chip, becomes small and easy for installation. USB technique is used to translate data between DSP and PC in high speed, so the system's real time processing is proved very much. It is proved that the designed system can acquire and analyze the steady and transient parameters of the I.C. engine very well

  13. The Relationship Between Acoustic Signal Typing and Perceptual Evaluation of Tracheoesophageal Voice Quality for Sustained Vowels.

    Science.gov (United States)

    Clapham, Renee P; van As-Brooks, Corina J; van Son, Rob J J H; Hilgers, Frans J M; van den Brekel, Michiel W M

    2015-07-01

    To investigate the relationship between acoustic signal typing and perceptual evaluation of sustained vowels produced by tracheoesophageal (TE) speakers and the use of signal typing in the clinical setting. Two evaluators independently categorized 1.75-second segments of narrow-band spectrograms according to acoustic signal typing and independently evaluated the recording of the same segments on a visual analog scale according to overall perceptual acoustic voice quality. The relationship between acoustic signal typing and overall voice quality (as a continuous scale and as a four-point ordinal scale) was investigated and the proportion of inter-rater agreement as well as the reliability between the two measures is reported. The agreement between signal type (I-IV) and ordinal voice quality (four-point scale) was low but significant, and there was a significant linear relationship between the variables. Signal type correctly predicted less than half of the voice quality data. There was a significant main effect of signal type on continuous voice quality scores with significant differences in median quality scores between signal types I-IV, I-III, and I-II. Signal typing can be used as an adjunct to perceptual and acoustic evaluation of the same stimuli for TE speech as part of a multidimensional evaluation protocol. Signal typing in its current form provides limited predictive information on voice quality, and there is significant overlap between signal types II and III and perceptual categories. Future work should consider whether the current four signal types could be refined. Copyright © 2015 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

  14. Adaptive Signal Processing Testbed: VME-based DSP board market survey

    Science.gov (United States)

    Ingram, Rick E.

    1992-04-01

    The Adaptive Signal Processing Testbed (ASPT) is a real-time multiprocessor system utilizing digital signal processor technology on VMEbus based printed circuit boards installed on a Sun workstation. The ASPT has specific requirements, particularly as regards to the signal excision application, with respect to interfacing with current and planned data generation equipment, processing of the data, storage to disk of final and intermediate results, and the development tools for applications development and integration into the overall EW/COM computing environment. A prototype ASPT was implemented using three VME-C-30 boards from Applied Silicon. Experience gained during the prototype development led to the conclusions that interprocessor communications capability is the most significant contributor to overall ASPT performance. In addition, the host involvement should be minimized. Boards using different processors were evaluated with respect to the ASPT system requirements, pricing, and availability. Specific recommendations based on various priorities are made as well as recommendations concerning the integration and interaction of various tools developed during the prototype implementation.

  15. A Novel Voice Sensor for the Detection of Speech Signals

    Directory of Open Access Journals (Sweden)

    Kun-Ching Wang

    2013-12-01

    Full Text Available In order to develop a novel voice sensor to detect human voices, the use of features which are more robust to noise is an important issue. Voice sensor is also called voice activity detection (VAD. Due to that the inherent nature of the formant structure only occurred on the speech spectrogram (well-known as voiceprint, Wu et al. were the first to use band-spectral entropy (BSE to describe the characteristics of voiceprints. However, the performance of VAD based on BSE feature was degraded in colored noise (or voiceprint-like noise environments. In order to solve this problem, we propose the two-dimensional part-band energy entropy (TD-PBEE parameter based on two variables: part-band partition number upon frequency index and long-term window size upon time index to further improve the BSE-based VAD algorithm. The two variables can efficiently represent the characteristics of voiceprints on each critical frequency band and use long-term information for noisy speech spectrograms, respectively. The TD-PBEE parameter can be regarded as a PBEE parameter over time. First, the strength of voiceprints can be partly enhanced by using four entropies applied to four part-bands. We can use the four part-band energy entropies for describing the voiceprints in detail. Due to the characteristics of non-stationary for speech and various noises, we will then use long-term information processing to refine the PBEE, so the voice-like noise can be distinguished from noisy speech through the concept of PBEE with long-term information. Our experiments show that the proposed feature extraction with the TD-PBEE parameter is quite insensitive to background noise. The proposed TD-PBEE-based VAD algorithm is evaluated for four types of noises and five signal-to-noise ratio (SNR levels. We find that the accuracy of the proposed TD-PBEE-based VAD algorithm averaged over all noises and all SNR levels is better than that of other considered VAD algorithms.

  16. Enhancement of speech signals - with a focus on voiced speech models

    DEFF Research Database (Denmark)

    Nørholm, Sidsel Marie

    This thesis deals with speech enhancement, i.e., noise reduction in speech signals. This has applications in, e.g., hearing aids and teleconference systems. We consider a signal-driven approach to speech enhancement where a model of the speech is assumed and filters are generated based...... on this model. The basic model used in this thesis is the harmonic model which is a commonly used model for describing the voiced part of the speech signal. We show that it can be beneficial to extend the model to take inharmonicities or the non-stationarity of speech into account. Extending the model...

  17. A new FPGA architecture suitable for DSP applications

    Energy Technology Data Exchange (ETDEWEB)

    Wang Liyun; Lai Jinmei; Tong Jiarong; Tang Pushan; Chen Xing; Duan Xueyan; Chen Liguang; Wang Jian; Wang Yuan, E-mail: 071021037@fudan.edu.cn [ASIC and System State Key Laboratory, Fudan University, Shanghai 201203 (China)

    2011-05-15

    A new FPGA architecture suitable for digital signal processing applications is presented. DSP modules can be inserted into FPGA conveniently with the proposed architecture, which is much faster when used in the field of digital signal processing compared with traditional FPGAs. An advanced 2-level MUX (multiplexer) is also proposed. With the added SLEEP MODE PASS to traditional 2-level MUX, static leakage is reduced. Furthermore, buffers are inserted at early returns of long lines. With this kind of buffer, the delay of the long line is improved by 9.8% while the area increases by 4.37%. The layout of this architecture has been taped out in standard 0.13 {mu}m CMOS technology successfully. The die size is 6.3 x 4.5 mm{sup 2} with the QFP208 package. Test results show that performances of presented classical DSP cases are improved by 28.6%-302% compared with traditional FPGAs. (semiconductor integrated circuits)

  18. A new FPGA architecture suitable for DSP applications

    International Nuclear Information System (INIS)

    Wang Liyun; Lai Jinmei; Tong Jiarong; Tang Pushan; Chen Xing; Duan Xueyan; Chen Liguang; Wang Jian; Wang Yuan

    2011-01-01

    A new FPGA architecture suitable for digital signal processing applications is presented. DSP modules can be inserted into FPGA conveniently with the proposed architecture, which is much faster when used in the field of digital signal processing compared with traditional FPGAs. An advanced 2-level MUX (multiplexer) is also proposed. With the added SLEEP MODE PASS to traditional 2-level MUX, static leakage is reduced. Furthermore, buffers are inserted at early returns of long lines. With this kind of buffer, the delay of the long line is improved by 9.8% while the area increases by 4.37%. The layout of this architecture has been taped out in standard 0.13 μm CMOS technology successfully. The die size is 6.3 x 4.5 mm 2 with the QFP208 package. Test results show that performances of presented classical DSP cases are improved by 28.6%-302% compared with traditional FPGAs. (semiconductor integrated circuits)

  19. A long distance voice transmission system based on the white light LED

    Science.gov (United States)

    Tian, Chunyu; Wei, Chang; Wang, Yulian; Wang, Dachi; Yu, Benli; Xu, Feng

    2017-10-01

    A long distance voice transmission system based on a visible light communication technology (VLCT) is proposed in the paper. Our proposed system includes transmitter, receiver and the voice signal processing of single chip microcomputer. In the compact-sized LED transmitter, we use on-off-keying and not-return-to-zero (OOK-NRZ) to easily realize high speed modulation, and then systematic complexity is reduced. A voice transmission system, which possesses the properties of the low-noise and wide modulation band, is achieved by the design of high efficiency receiving optical path and using filters to reduce noise from the surrounding light. To improve the speed of the signal processing, we use single chip microcomputer to code and decode voice signal. Furthermore, serial peripheral interface (SPI) is adopted to accurately transmit voice signal data. The test results of our proposed system show that the transmission distance of this system is more than100 meters with the maximum data rate of 1.5 Mbit/s and a SNR of 30dB. This system has many advantages, such as simple construction, low cost and strong practicality. Therefore, it has extensive application prospect in the fields of the emergency communication and indoor wireless communication, etc.

  20. Design and DSP implementation of star image acquisition and star point fast acquiring and tracking

    Science.gov (United States)

    Zhou, Guohui; Wang, Xiaodong; Hao, Zhihang

    2006-02-01

    Star sensor is a special high accuracy photoelectric sensor. Attitude acquisition time is an important function index of star sensor. In this paper, the design target is to acquire 10 samples per second dynamic performance. On the basis of analyzing CCD signals timing and star image processing, a new design and a special parallel architecture for improving star image processing are presented in this paper. In the design, the operation moving the data in expanded windows including the star to the on-chip memory of DSP is arranged in the invalid period of CCD frame signal. During the CCD saving the star image to memory, DSP processes the data in the on-chip memory. This parallelism greatly improves the efficiency of processing. The scheme proposed here results in enormous savings of memory normally required. In the scheme, DSP HOLD mode and CPLD technology are used to make a shared memory between CCD and DSP. The efficiency of processing is discussed in numerical tests. Only in 3.5ms is acquired the five lightest stars in the star acquisition stage. In 43us, the data in five expanded windows including stars are moved into the internal memory of DSP, and in 1.6ms, five star coordinates are achieved in the star tracking stage.

  1. Design of double DC motor control system based on DSP

    Directory of Open Access Journals (Sweden)

    Suo WANG

    2017-10-01

    Full Text Available Aiming at the problems of speed control, commutation and so on in the multi-motor synchronous control system, based on automatic control technology, a control system with PC as principal computer and DSP as slave computer is designed, which can change dual DC motor speed and steering, as well as select work drive motors. Related hardware and software design of the control system are given. Through serial communication between DSP and PC using PC serial port software, digital control command is sent to the slave computer for controlling dual DC motor to do a series of preset functions. PWM pulse width modulation is used for motor speed regulation, photoelectric encoder is used to measure motor speed by T method, and the motor speed is displayed by the actual waveform. Experimental results show that the system can not only realize the synchronization of dual DC motor speed and steering adjustment, but also select the motor and achieve the dual DC motors synchronization control effect. The control system has certain reliability and effectiveness.

  2. A fast continuous magnetic field measurement system based on digital signal processors

    International Nuclear Information System (INIS)

    Velev, G.V.; Carcagno, R.; DiMarco, J.; Kotelnikov, S.; Lamm, M.; Makulski, A.; Maroussov, V.; Nehring, R.; Nogiec, J.; Orris, D.; Poukhov, O.; Prakoshyn, F.; Schlabach, P.; Tompkins, J.C.

    2005-01-01

    In order to study dynamic effects in accelerator magnets, such as the decay of the magnetic field during the dwell at injection and the rapid so-called ''snapback'' during the first few seconds of the resumption of the energy ramp, a fast continuous harmonics measurement system was required. A new magnetic field measurement system, based on the use of digital signal processors (DSP) and Analog to Digital (A/D) converters, was developed and prototyped at Fermilab. This system uses Pentek 6102 16 bit A/D converters and the Pentek 4288 DSP board with the SHARC ADSP-2106 family digital signal processor. It was designed to acquire multiple channels of data with a wide dynamic range of input signals, which are typically generated by a rotating coil probe. Data acquisition is performed under a RTOS, whereas processing and visualization are performed under a host computer. Firmware code was developed for the DSP to perform fast continuous readout of the A/D FIFO memory and integration over specified intervals, synchronized to the probe's rotation in the magnetic field. C, C++ and Java code was written to control the data acquisition devices and to process a continuous stream of data. The paper summarizes the characteristics of the system and presents the results of initial tests and measurements

  3. Applied Chaos Level Test for Validation of Signal Conditions Underlying Optimal Performance of Voice Classification Methods

    Science.gov (United States)

    Liu, Boquan; Polce, Evan; Sprott, Julien C.; Jiang, Jack J.

    2018-01-01

    Purpose: The purpose of this study is to introduce a chaos level test to evaluate linear and nonlinear voice type classification method performances under varying signal chaos conditions without subjective impression. Study Design: Voice signals were constructed with differing degrees of noise to model signal chaos. Within each noise power, 100…

  4. A digital signal processing system for coherent laser radar

    Science.gov (United States)

    Hampton, Diana M.; Jones, William D.; Rothermel, Jeffry

    1991-01-01

    A data processing system for use with continuous-wave lidar is described in terms of its configuration and performance during the second survey mission of NASA'a Global Backscatter Experiment. The system is designed to estimate a complete lidar spectrum in real time, record the data from two lidars, and monitor variables related to the lidar operating environment. The PC-based system includes a transient capture board, a digital-signal processing (DSP) board, and a low-speed data-acquisition board. Both unprocessed and processed lidar spectrum data are monitored in real time, and the results are compared to those of a previous non-DSP-based system. Because the DSP-based system is digital it is slower than the surface-acoustic-wave signal processor and collects 2500 spectra/s. However, the DSP-based system provides complete data sets at two wavelengths from the continuous-wave lidars.

  5. Digital signal processing theory and practice

    CERN Document Server

    Rao, K Deergha

    2018-01-01

    The book provides a comprehensive exposition of all major topics in digital signal processing (DSP). With numerous illustrative examples for easy understanding of the topics, it also includes MATLAB-based examples with codes in order to encourage the readers to become more confident of the fundamentals and to gain insights into DSP. Further, it presents real-world signal processing design problems using MATLAB and programmable DSP processors. In addition to problems that require analytical solutions, it discusses problems that require solutions using MATLAB at the end of each chapter. Divided into 13 chapters, it addresses many emerging topics, which are not typically found in advanced texts on DSP. It includes a chapter on adaptive digital filters used in the signal processing problems for faster acceptable results in the presence of changing environments and changing system requirements. Moreover, it offers an overview of wavelets, enabling readers to easily understand the basics and applications of this po...

  6. Real-time co-registered ultrasound and photoacoustic imaging system based on FPGA and DSP architecture

    Science.gov (United States)

    Alqasemi, Umar; Li, Hai; Aguirre, Andres; Zhu, Quing

    2011-03-01

    Co-registering ultrasound (US) and photoacoustic (PA) imaging is a logical extension to conventional ultrasound because both modalities provide complementary information of tumor morphology, tumor vasculature and hypoxia for cancer detection and characterization. In addition, both modalities are capable of providing real-time images for clinical applications. In this paper, a Field Programmable Gate Array (FPGA) and Digital Signal Processor (DSP) module-based real-time US/PA imaging system is presented. The system provides real-time US/PA data acquisition and image display for up to 5 fps* using the currently implemented DSP board. It can be upgraded to 15 fps, which is the maximum pulse repetition rate of the used laser, by implementing an advanced DSP module. Additionally, the photoacoustic RF data for each frame is saved for further off-line processing. The system frontend consists of eight 16-channel modules made of commercial and customized circuits. Each 16-channel module consists of two commercial 8-channel receiving circuitry boards and one FPGA board from Analog Devices. Each receiving board contains an IC† that combines. 8-channel low-noise amplifiers, variable-gain amplifiers, anti-aliasing filters, and ADC's‡ in a single chip with sampling frequency of 40MHz. The FPGA board captures the LVDSξ Double Data Rate (DDR) digital output of the receiving board and performs data conditioning and subbeamforming. A customized 16-channel transmission circuitry is connected to the two receiving boards for US pulseecho (PE) mode data acquisition. A DSP module uses External Memory Interface (EMIF) to interface with the eight 16-channel modules through a customized adaptor board. The DSP transfers either sub-beamformed data (US pulse-echo mode or PAI imaging mode) or raw data from FPGA boards to its DDR-2 memory through the EMIF link, then it performs additional processing, after that, it transfer the data to the PC** for further image processing. The PC code

  7. Assessments of Voice Use and Voice Quality among College/University Singing Students Ages 18–24 through Ambulatory Monitoring with a Full Accelerometer Signal

    Science.gov (United States)

    Schloneger, Matthew; Hunter, Eric

    2016-01-01

    The multiple social and performance demands placed on college/university singers could put their still developing voices at risk. Previous ambulatory monitoring studies have analyzed the duration, intensity, and frequency (in Hz) of voice use among such students. Nevertheless, no studies to date have incorporated the simultaneous acoustic voice quality measures into the acquisition of these measures to allow for direct comparison during the same voicing period. Such data could provide greater insight into how young singers use their voices, as well as identify potential correlations between vocal dose and acoustic changes in voice quality. The purpose of this study was to assess the voice use and estimated voice quality of college/university singing students (18–24 y/o, N = 19). Ambulatory monitoring was conducted over three full, consecutive weekdays measuring voice from an unprocessed accelerometer signal measured at the neck. From this signal were analyzed traditional vocal dose metrics such as phonation percentage, dose time, cycle dose, and distance dose. Additional acoustic measures included perceived pitch, pitch strength, LTAS slope, alpha ratio, dB SPL 1–3 kHz, and harmonic-to-noise ratio. Major findings from more than 800 hours of recording indicated that among these students (a) higher vocal doses correlated significantly with greater voice intensity, more vocal clarity and less perturbation; and (b) there were significant differences in some acoustic voice quality metrics between non-singing, solo singing and choral singing. PMID:26897545

  8. Implementation of KRoC on Analog Devices' "SHARC" DSP

    NARCIS (Netherlands)

    Otten, G.W.; Schwirtz, M.H.; Schwirtz, Marcellinus H.; Bruis, R.; Bruis, R.; Broenink, Johannes F.; Bakkers, André; O'Neill, Brian C.

    1996-01-01

    This paper summarises the experiences gained at the Control Laboratory of the University of Twente in porting the Kent Retargetable occam Compiler -KroC -to the Analog Devices' ADSP21060 SHARC Digital Signal Processor. The choice of porting the KRoC to the DSP processor was in our view both a

  9. Design of dual DC motor control system based on DSP

    Science.gov (United States)

    Shi, Peicheng; Wang, Suo; Xu, Zengwei; Xiao, Ping

    2017-08-01

    Multi-motor control systems are widely used in actual production and life, such as lifting stages, robots, printing systems. This paper through serial communication between PC and DSP, dual DC motor control system consisting of PC as the host computer, DSP as the lower computer with synchronous PWM speed regulation, commutation and selection functions is designed. It sends digital control instructions with host computer serial debugger to lower computer, to instruct the motor to complete corresponding actions. The hardware and software design of the control system are given, and feasibility and validity of the control system are verified by experiments. The expected design goal is achieved.

  10. Blind I/Q Signal Separation-Based Solutions for Receiver Signal Processing

    Directory of Open Access Journals (Sweden)

    Visa Koivunen

    2005-09-01

    Full Text Available This paper introduces some novel digital signal processing (DSP-based approaches to some of the most fundamental tasks of radio receivers, namely, channel equalization, carrier synchronization, and I/Q mismatch compensation. The leading principle is to show that all these problems can be solved blindly (i.e., without training signals by forcing the I and Q components of the observed data as independent as possible. Blind signal separation (BSS is then introduced as an efficient tool to carry out these tasks, and simulation examples are used to illustrate the performance of the proposed approaches. The main application area of the presented carrier synchronization and I/Q mismatch compensation techniques is in direct-conversion type receivers, while the proposed channel equalization principles basically apply to any radio architecture.

  11. A fast continuous magnetic field measurement system based on digital signal processors

    Energy Technology Data Exchange (ETDEWEB)

    Velev, G.V.; Carcagno, R.; DiMarco, J.; Kotelnikov, S.; Lamm, M.; Makulski, A.; /Fermilab; Maroussov, V.; /Purdue U.; Nehring, R.; Nogiec, J.; Orris, D.; /Fermilab; Poukhov,; Prakoshyn, F.; /Dubna, JINR; Schlabach, P.; Tompkins, J.C.; /Fermilab

    2005-09-01

    In order to study dynamic effects in accelerator magnets, such as the decay of the magnetic field during the dwell at injection and the rapid so-called ''snapback'' during the first few seconds of the resumption of the energy ramp, a fast continuous harmonics measurement system was required. A new magnetic field measurement system, based on the use of digital signal processors (DSP) and Analog to Digital (A/D) converters, was developed and prototyped at Fermilab. This system uses Pentek 6102 16 bit A/D converters and the Pentek 4288 DSP board with the SHARC ADSP-2106 family digital signal processor. It was designed to acquire multiple channels of data with a wide dynamic range of input signals, which are typically generated by a rotating coil probe. Data acquisition is performed under a RTOS, whereas processing and visualization are performed under a host computer. Firmware code was developed for the DSP to perform fast continuous readout of the A/D FIFO memory and integration over specified intervals, synchronized to the probe's rotation in the magnetic field. C, C++ and Java code was written to control the data acquisition devices and to process a continuous stream of data. The paper summarizes the characteristics of the system and presents the results of initial tests and measurements.

  12. Hardware description ADSP-21020 40-bit floating point DSP as designed in a remotely controlled digital CW Doppler radar

    Science.gov (United States)

    Morrison, R. E.; Robinson, S. H.

    A continuous wave Doppler radar system has been designed which is portable, easily deployed, and remotely controlled. The heart of this system is a DSP/control board using Analog Devices ADSP-21020 40-bit floating point digital signal processor (DSP) microprocessor. Two 18-bit audio A/D converters provide digital input to the DSP/controller board for near real time target detection. Program memory for the DSP is dual ported with an Intel 87C51 microcontroller allowing DSP code to be up-loaded or down-loaded from a central controlling computer. The 87C51 provides overall system control for the remote radar and includes a time-of-day/day-of-year real time clock, system identification (ID) switches, and input/output (I/O) expansion by an Intel 82C55 I/O expander.

  13. DSP+FPGA-based real-time histogram equalization system of infrared image

    Science.gov (United States)

    Gu, Dongsheng; Yang, Nansheng; Pi, Defu; Hua, Min; Shen, Xiaoyan; Zhang, Ruolan

    2001-10-01

    Histogram Modification is a simple but effective method to enhance an infrared image. There are several methods to equalize an infrared image's histogram due to the different characteristics of the different infrared images, such as the traditional HE (Histogram Equalization) method, and the improved HP (Histogram Projection) and PE (Plateau Equalization) method and so on. If to realize these methods in a single system, the system must have a mass of memory and extremely fast speed. In our system, we introduce a DSP + FPGA based real-time procession technology to do these things together. FPGA is used to realize the common part of these methods while DSP is to do the different part. The choice of methods and the parameter can be input by a keyboard or a computer. By this means, the function of the system is powerful while it is easy to operate and maintain. In this article, we give out the diagram of the system and the soft flow chart of the methods. And at the end of it, we give out the infrared image and its histogram before and after the process of HE method.

  14. The study of image processing of parallel digital signal processor

    International Nuclear Information System (INIS)

    Liu Jie

    2000-01-01

    The author analyzes the basic characteristic of parallel DSP (digital signal processor) TMS320C80 and proposes related optimized image algorithm and the parallel processing method based on parallel DSP. The realtime for many image processing can be achieved in this way

  15. Design of DSP-based high-power digital solar array simulator

    Science.gov (United States)

    Zhang, Yang; Liu, Zhilong; Tong, Weichao; Feng, Jian; Ji, Yibo

    2013-12-01

    To satisfy rigid performance specifications, a feedback control was presented for zoom optical lens plants. With the increasing of global energy consumption, research of the photovoltaic(PV) systems get more and more attention. Research of the digital high-power solar array simulator provides technical support for high-power grid-connected PV systems research.This paper introduces a design scheme of the high-power digital solar array simulator based on TMS320F28335. A DC-DC full-bridge topology was used in the system's main circuit. The switching frequency of IGBT is 25kHz.Maximum output voltage is 900V. Maximum output current is 20A. Simulator can be pre-stored solar panel IV curves.The curve is composed of 128 discrete points .When the system was running, the main circuit voltage and current values was feedback to the DSP by the voltage and current sensors in real-time. Through incremental PI,DSP control the simulator in the closed-loop control system. Experimental data show that Simulator output voltage and current follow a preset solar panels IV curve. In connection with the formation of high-power inverter, the system becomes gridconnected PV system. The inverter can find the simulator's maximum power point and the output power can be stabilized at the maximum power point (MPP).

  16. Digital control card based on digital signal processor

    International Nuclear Information System (INIS)

    Hou Shigang; Yin Zhiguo; Xia Le

    2008-01-01

    A digital control card based on digital signal processor was developed. Two Freescale DSP-56303 processors were utilized to achieve 3 channels proportional- integral-differential regulations. The card offers high flexibility for 100 MeV cyclotron RF system development. It was used as feedback controller in low level radio frequency control prototype, with the feedback gain parameters continuously adjustable. By using high precision analog to digital converter with 500 kHz sampling rate, a regulation bandwidth of 20 kHz was achieved. (authors)

  17. TC9447F, single-chip DSP (digital signal processor) for audio; 1 chip audio yo DSP LSI TC9447F

    Energy Technology Data Exchange (ETDEWEB)

    NONE

    1999-03-01

    TC9447F is a single-chip DSP for audio which builds in 2-channel AD converter/4-channel DA converter. It can build various application programs such as the sound field control like hall simulation, digital filter like equalizer, and dynamic range control, in the program memory (ROM). Further, it builds in {+-}10dB trim use electronic volume for two channels. It also builds data delay use RAM (64K-bit) in, so no RAM to be separately attached is necessary. (translated by NEDO)

  18. The novel programmable riometer for in-depth ionospheric and magnetospheric observations (PRIAMOS) using direct sampling DSP techniques

    OpenAIRE

    Dekoulis, G.; Honary, F.

    2005-01-01

    This paper describes the feasibility study and simulation results for the unique multi-frequency, multi-bandwidth, Programmable Riometer for in-depth Ionospheric And Magnetospheric ObservationS (PRIAMOS) based on direct sampling digital signal processing (DSP) techniques. This novel architecture is based on sampling the cosmic noise wavefront at the antenna. It eliminates the usage of any intermediate frequency (IF) mixer stages (-6 dB) and the noise balancing technique (-3 dB), providing a m...

  19. Hints About Some Baseful but Indispensable Elements in Speech Recognition and Reconstruction

    Directory of Open Access Journals (Sweden)

    Mihaela Costin

    2002-07-01

    Full Text Available The cochlear implant (CI is a device used to reconstruct the hearing capabilities of a person diagnosed with total cophosis. This impairment may occur after some accidents, chemotherapy etc., the person still having an intact hearing nerve. The cochlear implant has two parts: a programmable, external part, the Digital Signal Processing (DSP device which process and transform the speech signal, and another surgically implanted part, with a certain number of electrodes (depending on brand used to stimulate the hearing nerve. The speech signal is fully processed in the DSP external device resulting the ``coded'' information on speech. This is modulated with the support of the fundamental frequency F0 and the energy impulses are inductively sent to the hearing nerve. The correct detection of this frequency is very important, determining the manner of hearing and making the difference between a "computer'' voice and a natural one. The results are applicable not only in the medical domain, but also in the Romanian speech synthesis.

  20. MCNP-DSP users manual

    International Nuclear Information System (INIS)

    Valentine, T.E.

    1997-01-01

    The Monte Carlo code MCNP-DSP was developed from the Los Alamos MCNP4a code to calculate the time and frequency response statistics obtained from the 252 Cf-source-driven frequency analysis measurements. This code can be used to validate calculational methods and cross section data sets from subcritical experiments. This code provides a more general model for interpretation and planning of experiments for nuclear criticality safety, nuclear safeguards, and nuclear weapons identification and replaces the use of point kinetics models for interpreting the measurements. The use of MCNP-DSP extends the usefulness of this measurement method to systems with much lower neutron multiplication factors

  1. The Chameleon Architecture for Streaming DSP Applications

    NARCIS (Netherlands)

    Bergmann, N.; Smit, Gerardus Johannes Maria; Kokkeler, Andre B.J.; Platzner, M.; Wolkotte, P.T.; Teich, J.; Holzenspies, P.K.F.; van de Burgwal, M.D.; Heysters, P.M.

    2007-01-01

    We focus on architectures for streaming DSP applications such as wireless baseband processing and image processing. We aim at a single generic architecture that is capable of dealing with different DSP applications. This architecture has to be energy efficient and fault tolerant. We introduce a

  2. Cultural and language differences in voice quality perception: a preliminary investigation using synthesized signals.

    Science.gov (United States)

    Yiu, Edwin M-L; Murdoch, Bruce; Hird, Kathryn; Lau, Polly; Ho, Elaine Mandy

    2008-01-01

    Perceptual voice evaluation is a common clinical tool. However, to date, there is no consensus yet as to which common quality should be measured. Some available evidence shows that voice quality is a language-specific property which may be different across different languages. The familiarity of a language may affect the perception and reliability in rating voice quality. The present study set out to investigate the effects of listeners' cultural and language backgrounds on the perception of voice qualities. Forty speech pathology students from Australia and Hong Kong were asked to rate the breathy and rough qualities of synthesized voice signals in Cantonese and English. Results showed that the English stimulus sets as a whole were rated less severely than the Cantonese stimuli by both groups of listeners. In addition, the male Cantonese and English breathy stimuli were rated differently by the Australian and Hong Kong listeners. These results provided some evidence to support the claim that cultural and language backgrounds of the listeners would affect the perception for some voice quality types. Thus, the cultural and language backgrounds of judges should be taken into consideration in clinical voice evaluation. 2008 S. Karger AG, Basel.

  3. Real-time digital signal processing fundamentals, implementations and applications

    CERN Document Server

    Kuo, Sen M; Tian, Wenshun

    2013-01-01

    Combines both the DSP principles and real-time implementations and applications, and now updated with the new eZdsp USB Stick, which is very low cost, portable and widely employed at many DSP labs. Real-Time Digital Signal Processing introduces fundamental digital signal processing (DSP) principles and will be updated to include the latest DSP applications, introduce new software development tools and adjust the software design process to reflect the latest advances in the field. In the 3rd edition of the book, the key aspect of hands-on experiments will be enhanced to make the DSP principle

  4. Performance of Phonatory Deviation Diagrams in Synthesized Voice Analysis.

    Science.gov (United States)

    Lopes, Leonardo Wanderley; da Silva, Karoline Evangelista; da Silva Evangelista, Deyverson; Almeida, Anna Alice; Silva, Priscila Oliveira Costa; Lucero, Jorge; Behlau, Mara

    2018-05-02

    To analyze the performance of a phonatory deviation diagram (PDD) in discriminating the presence and severity of voice deviation and the predominant voice quality of synthesized voices. A speech-language pathologist performed the auditory-perceptual analysis of the synthesized voice (n = 871). The PDD distribution of voice signals was analyzed according to area, quadrant, shape, and density. Differences in signal distribution regarding the PDD area and quadrant were detected when differentiating the signals with and without voice deviation and with different predominant voice quality. Differences in signal distribution were found in all PDD parameters as a function of the severity of voice disorder. The PDD area and quadrant can differentiate normal voices from deviant synthesized voices. There are differences in signal distribution in PDD area and quadrant as a function of the severity of voice disorder and the predominant voice quality. However, the PDD area and quadrant do not differentiate the signals as a function of severity of voice disorder and differentiated only the breathy and rough voices from the normal and strained voices. PDD density is able to differentiate only signals with moderate and severe deviation. PDD shape shows differences between signals with different severities of voice deviation. © 2018 S. Karger AG, Basel.

  5. Optical voice encryption based on digital holography.

    Science.gov (United States)

    Rajput, Sudheesh K; Matoba, Osamu

    2017-11-15

    We propose an optical voice encryption scheme based on digital holography (DH). An off-axis DH is employed to acquire voice information by obtaining phase retardation occurring in the object wave due to sound wave propagation. The acquired hologram, including voice information, is encrypted using optical image encryption. The DH reconstruction and decryption with all the correct parameters can retrieve an original voice. The scheme has the capability to record the human voice in holograms and encrypt it directly. These aspects make the scheme suitable for other security applications and help to use the voice as a potential security tool. We present experimental and some part of simulation results.

  6. Natural asynchronies in audiovisual communication signals regulate neuronal multisensory interactions in voice-sensitive cortex.

    Science.gov (United States)

    Perrodin, Catherine; Kayser, Christoph; Logothetis, Nikos K; Petkov, Christopher I

    2015-01-06

    When social animals communicate, the onset of informative content in one modality varies considerably relative to the other, such as when visual orofacial movements precede a vocalization. These naturally occurring asynchronies do not disrupt intelligibility or perceptual coherence. However, they occur on time scales where they likely affect integrative neuronal activity in ways that have remained unclear, especially for hierarchically downstream regions in which neurons exhibit temporally imprecise but highly selective responses to communication signals. To address this, we exploited naturally occurring face- and voice-onset asynchronies in primate vocalizations. Using these as stimuli we recorded cortical oscillations and neuronal spiking responses from functional MRI (fMRI)-localized voice-sensitive cortex in the anterior temporal lobe of macaques. We show that the onset of the visual face stimulus resets the phase of low-frequency oscillations, and that the face-voice asynchrony affects the prominence of two key types of neuronal multisensory responses: enhancement or suppression. Our findings show a three-way association between temporal delays in audiovisual communication signals, phase-resetting of ongoing oscillations, and the sign of multisensory responses. The results reveal how natural onset asynchronies in cross-sensory inputs regulate network oscillations and neuronal excitability in the voice-sensitive cortex of macaques, a suggested animal model for human voice areas. These findings also advance predictions on the impact of multisensory input on neuronal processes in face areas and other brain regions.

  7. Hidden Markov Model-based Packet Loss Concealment for Voice over IP

    DEFF Research Database (Denmark)

    Rødbro, Christoffer A.; Murthi, Manohar N.; Andersen, Søren Vang

    2006-01-01

    As voice over IP proliferates, packet loss concealment (PLC) at the receiver has emerged as an important factor in determining voice quality of service. Through the use of heuristic variations of signal and parameter repetition and overlap-add interpolation to handle packet loss, conventional PLC...

  8. Streamlining digital signal processing a tricks of the trade guidebook

    CERN Document Server

    2012-01-01

    Streamlining Digital Signal Processing, Second Edition, presents recent advances in DSP that simplify or increase the computational speed of common signal processing operations and provides practical, real-world tips and tricks not covered in conventional DSP textbooks. It offers new implementations of digital filter design, spectrum analysis, signal generation, high-speed function approximation, and various other DSP functions. It provides:Great tips, tricks of the trade, secrets, practical shortcuts, and clever engineering solutions from seasoned signal processing professionalsAn assortment.

  9. Voice Disorder Classification Based on Multitaper Mel Frequency Cepstral Coefficients Features

    Directory of Open Access Journals (Sweden)

    Ömer Eskidere

    2015-01-01

    Full Text Available The Mel Frequency Cepstral Coefficients (MFCCs are widely used in order to extract essential information from a voice signal and became a popular feature extractor used in audio processing. However, MFCC features are usually calculated from a single window (taper characterized by large variance. This study shows investigations on reducing variance for the classification of two different voice qualities (normal voice and disordered voice using multitaper MFCC features. We also compare their performance by newly proposed windowing techniques and conventional single-taper technique. The results demonstrate that adapted weighted Thomson multitaper method could distinguish between normal voice and disordered voice better than the results done by the conventional single-taper (Hamming window technique and two newly proposed windowing methods. The multitaper MFCC features may be helpful in identifying voices at risk for a real pathology that has to be proven later.

  10. Anti-voice adaptation suggests prototype-based coding of voice identity

    Directory of Open Access Journals (Sweden)

    Marianne eLatinus

    2011-07-01

    Full Text Available We used perceptual aftereffects induced by adaptation with anti-voice stimuli to investigate voice identity representations. Participants learned a set of voices then were tested on a voice identification task with vowel stimuli morphed between identities, after different conditions of adaptation. In Experiment 1, participants chose the identity opposite to the adapting anti-voice significantly more often than the other two identities (e.g., after being adapted to anti-A, they identified the average voice as A. In Experiment 2, participants showed a bias for identities opposite to the adaptor specifically for anti-voice, but not for non anti-voice adaptors. These results are strikingly similar to adaptation aftereffects observed for facial identity. They are compatible with a representation of individual voice identities in a multidimensional perceptual voice space referenced on a voice prototype.

  11. A Novel Dual Separate Paths (DSP) Algorithm Providing Fault-Tolerant Communication for Wireless Sensor Networks.

    Science.gov (United States)

    Tien, Nguyen Xuan; Kim, Semog; Rhee, Jong Myung; Park, Sang Yoon

    2017-07-25

    Fault tolerance has long been a major concern for sensor communications in fault-tolerant cyber physical systems (CPSs). Network failure problems often occur in wireless sensor networks (WSNs) due to various factors such as the insufficient power of sensor nodes, the dislocation of sensor nodes, the unstable state of wireless links, and unpredictable environmental interference. Fault tolerance is thus one of the key requirements for data communications in WSN applications. This paper proposes a novel path redundancy-based algorithm, called dual separate paths (DSP), that provides fault-tolerant communication with the improvement of the network traffic performance for WSN applications, such as fault-tolerant CPSs. The proposed DSP algorithm establishes two separate paths between a source and a destination in a network based on the network topology information. These paths are node-disjoint paths and have optimal path distances. Unicast frames are delivered from the source to the destination in the network through the dual paths, providing fault-tolerant communication and reducing redundant unicast traffic for the network. The DSP algorithm can be applied to wired and wireless networks, such as WSNs, to provide seamless fault-tolerant communication for mission-critical and life-critical applications such as fault-tolerant CPSs. The analyzed and simulated results show that the DSP-based approach not only provides fault-tolerant communication, but also improves network traffic performance. For the case study in this paper, when the DSP algorithm was applied to high-availability seamless redundancy (HSR) networks, the proposed DSP-based approach reduced the network traffic by 80% to 88% compared with the standard HSR protocol, thus improving network traffic performance.

  12. Novel ultra-wideband photonic signal generation and transmission featuring digital signal processing bit error rate measurements

    DEFF Research Database (Denmark)

    Gibbon, Timothy Braidwood; Yu, Xianbin; Tafur Monroy, Idelfonso

    2009-01-01

    We propose the novel generation of photonic ultra-wideband signals using an uncooled DFB laser. For the first time we experimentally demonstrate bit-for-bit DSP BER measurements for transmission of a 781.25 Mbit/s photonic UWB signal.......We propose the novel generation of photonic ultra-wideband signals using an uncooled DFB laser. For the first time we experimentally demonstrate bit-for-bit DSP BER measurements for transmission of a 781.25 Mbit/s photonic UWB signal....

  13. Generation and coherent detection of QPSK signal using a novel method of digital signal processing

    Science.gov (United States)

    Zhao, Yuan; Hu, Bingliang; He, Zhen-An; Xie, Wenjia; Gao, Xiaohui

    2018-02-01

    We demonstrate an optical quadrature phase-shift keying (QPSK) signal transmitter and an optical receiver for demodulating optical QPSK signal with homodyne detection and digital signal processing (DSP). DSP on the homodyne detection scheme is employed without locking the phase of the local oscillator (LO). In this paper, we present an extracting one-dimensional array of down-sampling method for reducing unwanted samples of constellation diagram measurement. Such a novel scheme embodies the following major advantages over the other conventional optical QPSK signal detection methods. First, this homodyne detection scheme does not need strict requirement on LO in comparison with linear optical sampling, such as having a flat spectral density and phase over the spectral support of the source under test. Second, the LabVIEW software is directly used for recovering the QPSK signal constellation without employing complex DSP circuit. Third, this scheme is applicable to multilevel modulation formats such as M-ary PSK and quadrature amplitude modulation (QAM) or higher speed signals by making minor changes.

  14. Dominant de novo DSP mutations cause erythrokeratodermia-cardiomyopathy syndrome.

    Science.gov (United States)

    Boyden, Lynn M; Kam, Chen Y; Hernández-Martín, Angela; Zhou, Jing; Craiglow, Brittany G; Sidbury, Robert; Mathes, Erin F; Maguiness, Sheilagh M; Crumrine, Debra A; Williams, Mary L; Hu, Ronghua; Lifton, Richard P; Elias, Peter M; Green, Kathleen J; Choate, Keith A

    2016-01-15

    Disorders of keratinization (DOK) show marked genotypic and phenotypic heterogeneity. In most cases, disease is primarily cutaneous, and further clinical evaluation is therefore rarely pursued. We have identified subjects with a novel DOK featuring erythrokeratodermia and initially-asymptomatic, progressive, potentially fatal cardiomyopathy, a finding not previously associated with erythrokeratodermia. We show that de novo missense mutations clustered tightly within a single spectrin repeat of DSP cause this novel cardio-cutaneous disorder, which we term erythrokeratodermia-cardiomyopathy (EKC) syndrome. We demonstrate that DSP mutations in our EKC syndrome subjects affect localization of desmosomal proteins and connexin 43 in the skin, and result in desmosome aggregation, widening of intercellular spaces, and lipid secretory defects. DSP encodes desmoplakin, a primary component of desmosomes, intercellular adhesion junctions most abundant in the epidermis and heart. Though mutations in DSP are known to cause other disorders, our cohort features the unique clinical finding of severe whole-body erythrokeratodermia, with distinct effects on localization of desmosomal proteins and connexin 43. These findings add a severe, previously undescribed syndrome featuring erythrokeratodermia and cardiomyopathy to the spectrum of disease caused by mutation in DSP, and identify a specific region of the protein critical to the pathobiology of EKC syndrome and to DSP function in the heart and skin. © The Author 2015. Published by Oxford University Press. All rights reserved. For Permissions, please email: journals.permissions@oup.com.

  15. DSP implementation of a PV system with GA-MLP-NN based MPPT controller supplying BLDC motor drive

    International Nuclear Information System (INIS)

    Akkaya, R.; Kulaksiz, A.A.; Aydogdu, O.

    2007-01-01

    This paper presents a brushless dc motor drive for heating, ventilating and air conditioning fans, which is utilized as the load of a photovoltaic system with a maximum power point tracking (MPPT) controller. The MPPT controller is based on a genetic assisted, multi-layer perceptron neural network (GA-MLP-NN) structure and includes a DC-DC boost converter. Genetic assistance in the neural network is used to optimize the size of the hidden layer. Also, for training the network, a genetic assisted, Levenberg-Marquardt (GA-LM) algorithm is utilized. The off line GA-MLP-NN, trained by this hybrid algorithm, is utilized for online estimation of the voltage and current values in the maximum power point. A brushless dc (BLDC) motor drive system that incorporates a motor controller with proportional integral (PI) speed control loop is successfully implemented to operate the fans. The digital signal processor (DSP) based unit provides rapid achievement of the MPPT and current control of the BLDC motor drive. The performance results of the system are given, and experimental results are presented for a laboratory prototype of 120 W

  16. Optical network and FPGA/DSP based control system for free electron laser

    International Nuclear Information System (INIS)

    Romaniuk, R.S.; Pozniak, K.T.; Czarski, T.; Czuba, K.; Giergusiewicz, W.; Kasprowicz, G.; Koprek, W.

    2005-01-01

    The work presents a structural and functional model of a distributed low level radio frequency (LLRF) control, diagnostic and telemetric system for a large industrial object. An example of system implementation is the European TESLA-XFEL accelerator. The free electron laser is expected to work in the VUV region now and in the range of X-rays in the future. The design of a system based on the FPGA circuits and multi-gigabit optical network is discussed. The system design approach is fully parametric. The major emphasis is put on the methods of the functional and hardware concentration to use fully both: a very big transmission capacity of the optical fiber telemetric channels and very big processing power of the latest series of DSP/PC enhanced and optical I/O equipped, FPGA chips. The subject of the work is the design of a universal, laboratory module of the LLRF sub-system. The current parameters of the system model, under the design, are presented. The considerations are shown on the background of the system application in the hostile industrial environment. The work is a digest of a few development threads of the hybrid, optoelectronic, telemetric networks (HOTN). In particular, the outline of construction theory of HOTN node was presented as well as the technology of complex, modular, multilayer HOTN system PCBs. The PCBs contain critical sub-systems of the node and the network. The presented exemplary sub-systems are: fast optical data transmission of 2.5 Gbit/s, 3.125 Gbit/s and 10 Gbit/s; fast A/C and C/A multichannel data conversion managed by FPGA chip (40 MHz, 65 MHz, 105 MHz), data and functionality concentration, integration of floating point calculations in the DSP units of FPGA circuit, using now discrete and next integrated PC chip with embedded OS; optical distributed timing system of phase reference; and 1GbEth video interface (over UTP or FX) for CCD telemetry and monitoring. The data and functions concentration in the HOTN node is necessary to

  17. Student Voices in School-Based Assessment

    Science.gov (United States)

    Tong, Siu Yin Annie; Adamson, Bob

    2015-01-01

    The value of student voices in dialogues about learning improvement is acknowledged in the literature. This paper examines how the views of students regarding School-based Assessment (SBA), a significant shift in examination policy and practice in secondary schools in Hong Kong, have largely been ignored. The study captures student voices through…

  18. BOREAS Follow-On DSP-05 Process-Modeled Net Primary Productivity

    Data.gov (United States)

    National Aeronautics and Space Administration — ABSTRACT: The BOREAS DSP-5 team generated a NPP image over the BOREAS region from a process-based ecosystem model, the Boreal Ecosystem Productivity Simulator...

  19. Research on the adaptive optical control technology based on DSP

    Science.gov (United States)

    Zhang, Xiaolu; Xue, Qiao; Zeng, Fa; Zhao, Junpu; Zheng, Kuixing; Su, Jingqin; Dai, Wanjun

    2018-02-01

    Adaptive optics is a real-time compensation technique using high speed support system for wavefront errors caused by atmospheric turbulence. However, the randomness and instantaneity of atmospheric changing introduce great difficulties to the design of adaptive optical systems. A large number of complex real-time operations lead to large delay, which is an insurmountable problem. To solve this problem, hardware operation and parallel processing strategy are proposed, and a high-speed adaptive optical control system based on DSP is developed. The hardware counter is used to check the system. The results show that the system can complete a closed loop control in 7.1ms, and improve the controlling bandwidth of the adaptive optical system. Using this system, the wavefront measurement and closed loop experiment are carried out, and obtain the good results.

  20. Study on UPF Harmonic Current Detection Method Based on DSP

    Energy Technology Data Exchange (ETDEWEB)

    Zhao, H J [Northwestern Polytechnical University, Xi' an 710072 (China); Pang, Y F [Xi' an University of Technology, Xi' an 710048 (China); Qiu, Z M [Xi' an University of Technology, Xi' an 710048 (China); Chen, M [Northwestern Polytechnical University, Xi' an 710072 (China)

    2006-10-15

    Unity power factor (UPF) harmonic current detection method applied to active power filter (APF) is presented in this paper. The intention of this method is to make nonlinear loads and active power filter in parallel to be an equivalent resistance. So after compensation, source current is sinusoidal, and has the same shape of source voltage. Meanwhile, there is no harmonic in source current, and the power factor becomes one. The mathematic model of proposed method and the optimum project for equivalent low pass filter in measurement are presented. Finally, the proposed detection method applied to a shunt active power filter experimental prototype based on DSP TMS320F2812 is developed. Simulation and experiment results indicate the method is simple and easy to implement, and can obtain the real-time calculation of harmonic current exactly.

  1. A simple approach to digital signal processing

    CERN Document Server

    Marven, Craig

    1996-01-01

    A readable, understandable introduction to DSP for professionals and students alike . . . This practical guide is a welcome alternative to more complicated introductions to DSP. It assumes no prior DSP experience and takes the reader step-by-step through the most basic signal processing concepts to more complex functions and devices, including sampling, filtering, frequency transforms, data compression, and even DSP design decisions. The guide provides clear, concise explanations and examples, while keeping mathematics to a minimum, to help develop a fundamental understanding of DSP. Other features include: * An extensive resource bibliography of more advanced DSP books. * An example of a typical DSP system development cycle, including tool descriptions. * A complete glossary of DSP-related acronyms Whether you're a working engineer looking into DSP for the first time or an undergraduate struggling to comprehend the subject, this engaging introduction provides easy access to the basic knowledge that will l...

  2. The Sound of Voice: Voice-Based Categorization of Speakers' Sexual Orientation within and across Languages.

    Directory of Open Access Journals (Sweden)

    Simone Sulpizio

    Full Text Available Empirical research had initially shown that English listeners are able to identify the speakers' sexual orientation based on voice cues alone. However, the accuracy of this voice-based categorization, as well as its generalizability to other languages (language-dependency and to non-native speakers (language-specificity, has been questioned recently. Consequently, we address these open issues in 5 experiments: First, we tested whether Italian and German listeners are able to correctly identify sexual orientation of same-language male speakers. Then, participants of both nationalities listened to voice samples and rated the sexual orientation of both Italian and German male speakers. We found that listeners were unable to identify the speakers' sexual orientation correctly. However, speakers were consistently categorized as either heterosexual or gay on the basis of how they sounded. Moreover, a similar pattern of results emerged when listeners judged the sexual orientation of speakers of their own and of the foreign language. Overall, this research suggests that voice-based categorization of sexual orientation reflects the listeners' expectations of how gay voices sound rather than being an accurate detector of the speakers' actual sexual identity. Results are discussed with regard to accuracy, acoustic features of voices, language dependency and language specificity.

  3. 47 CFR 90.233 - Base/mobile non-voice operations.

    Science.gov (United States)

    2010-10-01

    ... 47 Telecommunication 5 2010-10-01 2010-10-01 false Base/mobile non-voice operations. 90.233... SERVICES PRIVATE LAND MOBILE RADIO SERVICES Non-Voice and Other Specialized Operations § 90.233 Base/mobile non-voice operations. The use of A1D, A2D, F1D, F2D, G1D, or G2D emission may be authorized to base...

  4. BOREAS Follow-On DSP-05 Process-Modeled Net Primary Productivity

    Data.gov (United States)

    National Aeronautics and Space Administration — The BOREAS DSP-5 team generated a NPP image over the BOREAS region from a process-based ecosystem model, the Boreal Ecosystem Productivity Simulator (BEPS). The NPP...

  5. An opto-electronic joint detection system based on DSP aiming at early cervical cancer screening

    Science.gov (United States)

    Wang, Weiya; Jia, Mengyu; Gao, Feng; Yang, Lihong; Qu, Pengpeng; Zou, Changping; Liu, Pengxi; Zhao, Huijuan

    2015-02-01

    The cervical cancer screening at a pre-cancer stage is beneficial to reduce the mortality of women. An opto-electronic joint detection system based on DSP aiming at early cervical cancer screening is introduced in this paper. In this system, three electrodes alternately discharge to the cervical tissue and three light emitting diodes in different wavelengths alternately irradiate the cervical tissue. Then the relative optical reflectance and electrical voltage attenuation curve are obtained by optical and electrical detection, respectively. The system is based on DSP to attain the portable and cheap instrument. By adopting the relative reflectance and the voltage attenuation constant, the classification algorithm based on Support Vector Machine (SVM) discriminates abnormal cervical tissue from normal. We use particle swarm optimization to optimize the two key parameters of SVM, i.e. nuclear factor and cost factor. The clinical data were collected on 313 patients to build a clinical database of tissue responses under optical and electrical stimulations with the histopathologic examination as the gold standard. The classification result shows that the opto-electronic joint detection has higher total coincidence rate than separate optical detection or separate electrical detection. The sensitivity, specificity, and total coincidence rate increase with the increasing of sample numbers in the training set. The average total coincidence rate of the system can reach 85.1% compared with the histopathologic examination.

  6. IMAGE TYPE WATER METER CHARACTER RECOGNITION BASED ON EMBEDDED DSP

    OpenAIRE

    LIU Ying; HAN Yan-bin; ZHANG Yu-lin

    2015-01-01

    In the paper, we combined DSP processor with image processing algorithm and studied the method of water meter character recognition. We collected water meter image through camera at a fixed angle, and the projection method is used to recognize those digital images. The experiment results show that the method can recognize the meter characters accurately and artificial meter reading is replaced by automatic digital recognition, which improves working efficiency.

  7. RISC & DSP System Application Design using VHDL

    OpenAIRE

    Rachana Solanki; Vinay Gupta

    2014-01-01

    The Reduced Instruction Set Computer (RISC) processor use fewer instructions with simple constructs, therefore they can be executed much faster within the CPU without having to use memory as often. It reduce execution time by simplifying the instruction set of the computer. The DSP processors are perform the operation such as Discrete Cosine transform (DCT), Inverse DCT, Discrete Fourier Transform (DFT) and Fast Fourier Transform (FFT) are performed by DSP system. This paper represent the des...

  8. Double Fourier analysis for Emotion Identification in Voiced Speech

    International Nuclear Information System (INIS)

    Sierra-Sosa, D.; Bastidas, M.; Ortiz P, D.; Quintero, O.L.

    2016-01-01

    We propose a novel analysis alternative, based on two Fourier Transforms for emotion recognition from speech. Fourier analysis allows for display and synthesizes different signals, in terms of power spectral density distributions. A spectrogram of the voice signal is obtained performing a short time Fourier Transform with Gaussian windows, this spectrogram portraits frequency related features, such as vocal tract resonances and quasi-periodic excitations during voiced sounds. Emotions induce such characteristics in speech, which become apparent in spectrogram time-frequency distributions. Later, the signal time-frequency representation from spectrogram is considered an image, and processed through a 2-dimensional Fourier Transform in order to perform the spatial Fourier analysis from it. Finally features related with emotions in voiced speech are extracted and presented. (paper)

  9. DSP-enabled reconfigurable and transparent spectral converters for converging optical and mobile fronthaul/backhaul networks.

    Science.gov (United States)

    Mao, M Z; Giddings, R P; Cao, B Y; Xu, Y T; Wang, M; Tang, J M

    2017-06-12

    Dynamically reconfigurable and transparent signal spectral conversion is expected to play a vital role in seamlessly integrating traditional metropolitan optical networks and mobile fronthaul/backhaul networks. In this paper, a simple digital signal processing (DSP)-enabled spectral converter is proposed and extensively investigated, for the first time, which just utilizes a single standard dual-parallel Mach-Zehnder modulator (DP-MZM) driven by SDN-controllable RF signals and DC bias currents. As an important thrust of the paper, optimum operating conditions of the proposed converter are analytically identified, statistically examined and experimentally verified. Optimum operating condition-supported spectral converter performances in IMDD-based network nodes are explored both theoretically and experimentally in terms of frequency detuning range-dependent conversion efficiency, spectral conversion-induced OSNR/power penalty and transparency to input signal characteristics. The proposed spectral converter has unique advantages including low configuration complexity, strict transparency, SDN-controllable performance reconfigurability and flexibility, as well as negligible spectral conversion-induced latency.

  10. Simulation of reactor noise analysis measurement for light-water critical assembly TCA using MCNP-DSP

    International Nuclear Information System (INIS)

    Yamamoto, Toshihiro; Sakurai, Kiyoshi; Tonoike, Kotaro; Miyoshi, Yoshinori

    2001-01-01

    Reactor noise analysis methods using Monte Carlo technique have been proposed and developed in the field of nuclear criticality safety. The Monte Carlo simulation for noise analysis can be made by simulating physical phenomena in the course of neutron transport in a nuclear fuel as practically as possible. MCNP-DSP was developed by T. Valentine of ORNL for this purpose and it is a modified version of MCNP-4A. The authors applied this code to frequency analysis measurements performed in light-water critical assembly TCA. Prompt neutron generation times for critical and subcritical cores were measured by doing the frequency analysis of detector signals. The Monte Carlo simulations for these experiments were carried out using MCNP-DSP, and prompt neutron generation times were calculated. (author)

  11. Control electronic platform based on floating-point DSP and FPGA for a NPC multilevel back-to-back converter

    Energy Technology Data Exchange (ETDEWEB)

    Rodriguez, Francisco J.; Cobreces, Santiago; Bueno, Emilio J.; Hernandez, Alvaro; Mateos, Raul; Espinosa, Felipe [Department of Electronics, University of Alcala, Alcala de Henares, Madrid (Spain)

    2008-09-15

    Modern energy concepts as Distributed Power Generation are changing the appearance of electric distribution and transmission and challenging power electronics researchers, which try to develop new solutions of electronic controllers. The aim is to enable the implementation of new and more complex control algorithms to verify the last standards related to the grid energy quality for new power converters, and, also, for equipments which nowadays are operating. This paper presents the design, implementation and test of a novel real-time controller for a Neutral Point Clamped (NPC) (three-level) multilevel converter based on a floating-point Digital Signal Processor (DSP) and on a Field-Programmable Gate Array (FPGA), by operating in a cooperative way. Although the proposed system can be readily applied to any power electronic application, in this work, it is focused on the next system: a 150 kVA back-to-back three-level NPC Voltage Source Converter (VSC) for wind power applications. (author)

  12. High-speed parallel implementation of a modified PBR algorithm on DSP-based EH topology

    Science.gov (United States)

    Rajan, K.; Patnaik, L. M.; Ramakrishna, J.

    1997-08-01

    Algebraic Reconstruction Technique (ART) is an age-old method used for solving the problem of three-dimensional (3-D) reconstruction from projections in electron microscopy and radiology. In medical applications, direct 3-D reconstruction is at the forefront of investigation. The simultaneous iterative reconstruction technique (SIRT) is an ART-type algorithm with the potential of generating in a few iterations tomographic images of a quality comparable to that of convolution backprojection (CBP) methods. Pixel-based reconstruction (PBR) is similar to SIRT reconstruction, and it has been shown that PBR algorithms give better quality pictures compared to those produced by SIRT algorithms. In this work, we propose a few modifications to the PBR algorithms. The modified algorithms are shown to give better quality pictures compared to PBR algorithms. The PBR algorithm and the modified PBR algorithms are highly compute intensive, Not many attempts have been made to reconstruct objects in the true 3-D sense because of the high computational overhead. In this study, we have developed parallel two-dimensional (2-D) and 3-D reconstruction algorithms based on modified PBR. We attempt to solve the two problems encountered by the PBR and modified PBR algorithms, i.e., the long computational time and the large memory requirements, by parallelizing the algorithm on a multiprocessor system. We investigate the possible task and data partitioning schemes by exploiting the potential parallelism in the PBR algorithm subject to minimizing the memory requirement. We have implemented an extended hypercube (EH) architecture for the high-speed execution of the 3-D reconstruction algorithm using the commercially available fast floating point digital signal processor (DSP) chips as the processing elements (PEs) and dual-port random access memories (DPR) as channels between the PEs. We discuss and compare the performances of the PBR algorithm on an IBM 6000 RISC workstation, on a Silicon

  13. Familiarity and Voice Representation: From Acoustic-Based Representation to Voice Averages

    Directory of Open Access Journals (Sweden)

    Maureen Fontaine

    2017-07-01

    Full Text Available The ability to recognize an individual from their voice is a widespread ability with a long evolutionary history. Yet, the perceptual representation of familiar voices is ill-defined. In two experiments, we explored the neuropsychological processes involved in the perception of voice identity. We specifically explored the hypothesis that familiar voices (trained-to-familiar (Experiment 1, and famous voices (Experiment 2 are represented as a whole complex pattern, well approximated by the average of multiple utterances produced by a single speaker. In experiment 1, participants learned three voices over several sessions, and performed a three-alternative forced-choice identification task on original voice samples and several “speaker averages,” created by morphing across varying numbers of different vowels (e.g., [a] and [i] produced by the same speaker. In experiment 2, the same participants performed the same task on voice samples produced by familiar speakers. The two experiments showed that for famous voices, but not for trained-to-familiar voices, identification performance increased and response times decreased as a function of the number of utterances in the averages. This study sheds light on the perceptual representation of familiar voices, and demonstrates the power of average in recognizing familiar voices. The speaker average captures the unique characteristics of a speaker, and thus retains the information essential for recognition; it acts as a prototype of the speaker.

  14. Fixed-point signal processing

    CERN Document Server

    Padgett, Wayne T

    2009-01-01

    This book is intended to fill the gap between the ""ideal precision"" digital signal processing (DSP) that is widely taught, and the limited precision implementation skills that are commonly required in fixed-point processors and field programmable gate arrays (FPGAs). These skills are often neglected at the university level, particularly for undergraduates. We have attempted to create a resource both for a DSP elective course and for the practicing engineer with a need to understand fixed-point implementation. Although we assume a background in DSP, Chapter 2 contains a review of basic theory

  15. Vehicle recognition by using acoustic signature and classic DSP techniques

    Directory of Open Access Journals (Sweden)

    María Fernanda Díaz Velásquez

    2016-06-01

    Full Text Available This paper shows the application of the classic technique of digital signal processing (DSP, the cross-correlation, used for the detection of acoustic signatures of road traffic in Cali city, Colombia. Future goal is to build a detection software that through real time measures allows us estimate the levels of acoustic pollution in the city by using simulation models of road traffic, in the framework of environmentally-friendly smart cities. Final results of the experimental tests showed an accuracy of 71.43% for specific vehicle detection.

  16. Voice Morphing Using 3D Waveform Interpolation Surfaces and Lossless Tube Area Functions

    Directory of Open Access Journals (Sweden)

    Lavner Yizhar

    2005-01-01

    Full Text Available Voice morphing is the process of producing intermediate or hybrid voices between the utterances of two speakers. It can also be defined as the process of gradually transforming the voice of one speaker to that of another. The ability to change the speaker's individual characteristics and to produce high-quality voices can be used in many applications. Examples include multimedia and video entertainment, as well as enrichment of speech databases in text-to-speech systems. In this study we present a new technique which enables production of a given number of intermediate voices or of utterances which gradually change from one voice to another. This technique is based on two components: (1 creation of a 3D prototype waveform interpolation (PWI surface from the LPC residual signal, to produce an intermediate excitation signal; (2 a representation of the vocal tract by a lossless tube area function, and an interpolation of the parameters of the two speakers. The resulting synthesized signal sounds like a natural voice lying between the two original voices.

  17. Genomic signal processing methods for computation of alignment-free distances from DNA sequences.

    Science.gov (United States)

    Borrayo, Ernesto; Mendizabal-Ruiz, E Gerardo; Vélez-Pérez, Hugo; Romo-Vázquez, Rebeca; Mendizabal, Adriana P; Morales, J Alejandro

    2014-01-01

    Genomic signal processing (GSP) refers to the use of digital signal processing (DSP) tools for analyzing genomic data such as DNA sequences. A possible application of GSP that has not been fully explored is the computation of the distance between a pair of sequences. In this work we present GAFD, a novel GSP alignment-free distance computation method. We introduce a DNA sequence-to-signal mapping function based on the employment of doublet values, which increases the number of possible amplitude values for the generated signal. Additionally, we explore the use of three DSP distance metrics as descriptors for categorizing DNA signal fragments. Our results indicate the feasibility of employing GAFD for computing sequence distances and the use of descriptors for characterizing DNA fragments.

  18. Advanced Time-Frequency Representation in Voice Signal Analysis

    Directory of Open Access Journals (Sweden)

    Dariusz Mika

    2018-03-01

    Full Text Available The most commonly used time-frequency representation of the analysis in voice signal is spectrogram. This representation belongs in general to Cohen's class, the class of time-frequency energy distributions. From the standpoint of properties of the resolution spectrogram representation is not optimal. In Cohen class representations are known which have a better resolution properties. All of them are created by smoothing the Wigner-Ville'a (WVD distribution characterized by the best resolution, however, the biggest harmful interference. Used smoothing functions decide about a compromise between the properties of resolution and eliminating harmful interference term. Another class of time-frequency energy distributions is the affine class of distributions. From the point of view of readability of analysis the best properties are known so called Redistribution of energy caused by the use of a general methodology referred to as reassignment to any time-frequency representation. Reassigned distributions efficiently combine a reduction of the interference terms provided by a well adapted smoothing kernel and an increased concentration of the signal components.

  19. Study of Fourier transform spectrometer based on Michelson interferometer wave-meter

    Science.gov (United States)

    Peng, Yuexiang; Wang, Liqiang; Lin, Li

    2008-03-01

    A wave-meter based on Michelson interferometer consists of a reference and a measurement channel. The voice-coiled motor using PID means can realize to move in stable motion. The wavelength of a measurement laser can be obtained by counting interference fringes of reference and measurement laser. Reference laser with frequency stabilization creates a cosine interferogram signal whose frequency is proportional to velocity of the moving motor. The interferogram of the reference laser is converted to pulse signal, and it is subdivided into 16 times. In order to get optical spectrum, the analog signal of measurement channel should be collected. The Analog-to-Digital Converter (ADC) for measurement channel is triggered by the 16-times pulse signal of reference laser. So the sampling rate is constant only depending on frequency of reference laser and irrelative to the motor velocity. This means the sampling rate of measurement channel signals is on a uniform time-scale. The optical spectrum of measurement channel can be processed with Fast Fourier Transform (FFT) method by DSP and displayed on LCD.

  20. A Two-Level Task Scheduler on Multiple DSP System for OpenCL

    Directory of Open Access Journals (Sweden)

    Li Tian

    2014-04-01

    Full Text Available This paper addresses the problem that multiple DSP system does not support OpenCL programming. With the compiler, runtime, and the kernel scheduler proposed, an OpenCL application becomes portable not only between multiple CPU and GPU, but also between embedded multiple DSP systems. Firstly, the LLVM compiler was imported for source-to-source translation in which the translated source was supported by CCS. Secondly, two-level schedulers were proposed to support efficient OpenCL kernel execution. The DSP/BIOS is used to schedule system level tasks such as interrupts and drivers; however, the synchronization mechanism resulted in heavy overhead during task switching. So we designed an efficient second level scheduler especially for OpenCL kernel work-item scheduling. The context switch process utilizes the 8 functional units and cross path links which was superior to DSP/BIOS in the aspect of task switching. Finally, dynamic loading and software managed CACHE were redesigned for OpenCL running on multiple DSP system. We evaluated the performance using some common OpenCL kernels from NVIDIA, AMD, NAS, and Parboil benchmarks. Experimental results show that the DSP OpenCL can efficiently exploit the computing resource of multiple cores.

  1. Foundations of digital signal processing theory, algorithms and hardware design

    CERN Document Server

    Gaydecki, Patrick

    2005-01-01

    An excellent introductory text, this book covers the basic theoretical, algorithmic and real-time aspects of digital signal processing (DSP). Detailed information is provided on off-line, real-time and DSP programming and the reader is effortlessly guided through advanced topics such as DSP hardware design, FIR and IIR filter design and difference equation manipulation.

  2. PC-based digital feedback control for scanning force microscope

    International Nuclear Information System (INIS)

    Mohd Ashhar Khalid

    2002-01-01

    In the past, most digital feedback implementation for scanned-probe microscope were based on a digital signal processor (DSP). At present DSP plug-in card with the input-output interface module is still expensive compared to a fast pentium PC motherboard. For a magnetic force microscope (MFM) digital feedback has an advantage where the magnetic signal can be easily separated from the topographic signal. In this paper, a simple low-cost PC-based digital feedback and imaging system for Scanning Force Microscope (SFM) is presented. (Author)

  3. Design and test of 4πβ-γ coincidence measurement device based on DSP technology

    International Nuclear Information System (INIS)

    Zeng Herong; Feng Qijie; Leng Jun; Qian Dazhi; Bai Lixin; Zhang Yiyun

    2012-01-01

    The paper illustrates the hardware and software of the 4πβ-γ coincidence measurement device based on DSP technology in detail. In such device, the single-channel analyzer, gate generator, coincidence circuit and scalar in the traditional coincidence measurement device are replaced by the digital coincidence acquirer which is researched and manufactured by ourselves. Doing so, the measurement efficiency will be respectively improved, and the hardware cost will be lowered. The comparison experiment shows that the design of such device is a success. (authors)

  4. Power systems signal processing for smart grids

    NARCIS (Netherlands)

    Ribeiro, P.F.; Duque, C.A.; Da Silveira, P.M.; Cerqueira, A.S.

    2013-01-01

    With special relation to smart grids, this book provides clear and comprehensive explanation of how Digital Signal Processing (DSP) and Computational Intelligence (CI) techniques can be applied to solve problems in the power system. Its unique coverage bridges the gap between DSP, electrical power

  5. A Dual Digital Signal Processor VME Board for Instrumentation and Control Applications

    International Nuclear Information System (INIS)

    H. Dong; R. Flood; C. Hovater; J. Musson

    2001-01-01

    A Dual Digital Signal Processing VME Board is being developed for the CEBAF Beam Current Monitor system at Jefferson Lab. It is a versatile general-purpose digital signal processing board using an open architecture, which allows for adaptation to various applications. The base design uses two independent Texas Instrument (TI) TMS320C6711, which are 900 MFLOPS floating-point digital signal processors (DSP). Applications that require a fixed point DSP can be implemented by replacing the baseline DSP with the pin-for-pin compatible TMS320C6211. Both parallel and serial protocols have been implemented for communicating with off board devices. The initial implementation makes use of TI Multi-channel Serial protocol and VME bus protocol. Other communication protocols can be implemented by reprogramming the FPGA. Each DSP is equipped with FLASH PROM and SDRAM for program and data storage. Additionally, each DSP has 16 bits of digital I/O, two digital analog converters, and two analog to digital converters. Dual 160 pins mezzanine connectors provide expansion capability without design modifications. The mezzanine interface conforms to the TI Expansion Daughter Card Interface standard. The design can be manufactured with a reduced chip set without redesigning the printed circuit board. For example, it can be implemented as a single-channel DSP with no analog I/O. The board supports JTAG 1149 boundary scan to facilitate testing, debugging, and programming. It is fully programmable using software development tools such as TI Code Composer Studio and a JTAG emulator such as Spectrum Digital DS510PP-PLUS. Using these tools allows one program the flash memory and FPGA through the JTAG ports, thus eliminating the need for a separate ROM/FPGA programmer. This work supported by U.S. DOE Contract No. DE-AC05-84ER40150

  6. REVIEW ARTICLE: Spectrophotometric applications of digital signal processing

    Science.gov (United States)

    Morawski, Roman Z.

    2006-09-01

    Spectrophotometry is more and more often the method of choice not only in analysis of (bio)chemical substances, but also in the identification of physical properties of various objects and their classification. The applications of spectrophotometry include such diversified tasks as monitoring of optical telecommunications links, assessment of eating quality of food, forensic classification of papers, biometric identification of individuals, detection of insect infestation of seeds and classification of textiles. In all those applications, large numbers of data, generated by spectrophotometers, are processed by various digital means in order to extract measurement information. The main objective of this paper is to review the state-of-the-art methodology for digital signal processing (DSP) when applied to data provided by spectrophotometric transducers and spectrophotometers. First, a general methodology of DSP applications in spectrophotometry, based on DSP-oriented models of spectrophotometric data, is outlined. Then, the most important classes of DSP methods for processing spectrophotometric data—the methods for DSP-aided calibration of spectrophotometric instrumentation, the methods for the estimation of spectra on the basis of spectrophotometric data, the methods for the estimation of spectrum-related measurands on the basis of spectrophotometric data—are presented. Finally, the methods for preprocessing and postprocessing of spectrophotometric data are overviewed. Throughout the review, the applications of DSP are illustrated with numerous examples related to broadly understood spectrophotometry.

  7. High mobility group protein DSP1 negatively regulates HSP70 transcription in Crassostrea hongkongensis

    Energy Technology Data Exchange (ETDEWEB)

    Miao, Zongyu; Xu, Delin; Cui, Miao; Zhang, Qizhong, E-mail: zhangqzdr@126.com

    2016-06-10

    HSP70 acts mostly as a molecular chaperone and plays important roles in facilitating the folding of nascent peptides as well as the refolding or degradation of the denatured proteins. Under stressed conditions, the expression level of HSP70 is upregulated significantly and rapidly, as is known to be achieved by various regulatory factors controlling the transcriptional level. In this study, a high mobility group protein DSP1 was identified by DNA-affinity purification from the nuclear extracts of Crassostrea hongkongensis using the ChHSP70 promoter as a bait. The specific interaction between the prokaryotically expressed ChDSP1 and the FITC-labeled ChHSP70 promoter was confirmed by EMSA analysis. ChDSP1 was shown to negatively regulate ChHSP70 promoter expression by Luciferase Reporter Assay in the heterologous HEK293T cells. Both ChHSP70 and ChDSP1 transcriptions were induced by either thermal or CdCl{sub 2} stress, while the accumulated expression peaks of ChDSP1 were always slightly delayed when compared with that of ChHSP70. This indicates that ChDSP1 is involved, very likely to exert its suppressive role, in the recovery of the ChHSP70 expression from the induced level to its original state. This study is the first to report negative regulator of HSP70 gene transcription, and provides novel insights into the mechanisms controlling heat shock protein expression. -- Highlights: •HMG protein ChDSP1 shows affinity to ChHSP70 promoter in Crassostrea hongkongensis. •ChDSP1 negatively regulates ChHSP70 transcription. •ChHSP70 and ChDSP1 transcriptions were coordinately induced by thermal/Cd stress. •ChDSP1 may contribute to the recovery of the induced ChHSP70 to its original state. •This is the first report regarding negative regulator of HSP70 transcription.

  8. Evidence-Based Clinical Voice Assessment: A Systematic Review

    Science.gov (United States)

    Roy, Nelson; Barkmeier-Kraemer, Julie; Eadie, Tanya; Sivasankar, M. Preeti; Mehta, Daryush; Paul, Diane; Hillman, Robert

    2013-01-01

    Purpose: To determine what research evidence exists to support the use of voice measures in the clinical assessment of patients with voice disorders. Method: The American Speech-Language-Hearing Association (ASHA) National Center for Evidence-Based Practice in Communication Disorders staff searched 29 databases for peer-reviewed English-language…

  9. PSpice for digital signal processing

    CERN Document Server

    Tobin, Paul

    2007-01-01

    PSpice for Digital Signal Processing is the last in a series of five books using Cadence Orcad PSpice version 10.5 and introduces a very novel approach to learning digital signal processing (DSP). DSP is traditionally taught using Matlab/Simulink software but has some inherent weaknesses for students particularly at the introductory level. The 'plug in variables and play' nature of these software packages can lure the student into thinking they possess an understanding they don't actually have because these systems produce results quicklywithout revealing what is going on. However, it must be

  10. Digital signal processors for cryogenic high-resolution x-ray detector readout

    International Nuclear Information System (INIS)

    Friedrich, Stephan; Drury, Owen B.; Bechstein, Sylke; Hennig, Wolfgang; Momayezi, Michael

    2003-01-01

    We are developing fast digital signal processors (DSPs) to read out superconducting high-resolution X-ray detectors with on-line pulse processing. For superconducting tunnel junction (STJ) detector read-out, the DSPs offer online filtering, rise time discrimination and pile-up rejection. Compared to analog pulse processing, DSP readout somewhat degrades the detector resolution, but improves the spectral purity of the detector response. We discuss DSP performance with our 9-channel STJ array for synchrotron-based high-resolution X-ray spectroscopy. (author)

  11. Digital Signal Processing for Optical Coherent Communication Systems

    DEFF Research Database (Denmark)

    Zhang, Xu

    spectrum narrowing tolerance 112-Gb/s DP-QPSK optical coherent systems using digital adaptive equalizer. The demonstrated results show that off-line DSP algorithms are able to reduce the bit error rate (BER) penalty induced by signal spectrum narrowing. Third, we also investigate bi...... wavelength division multiplex (U-DWDM) optical coherent systems based on 10-Gbaud QPSK. We report U-DWDM 1.2-Tb/s QPSK coherent system achieving spectral efficiency of 4.0-bit/s/Hz. In the experimental demonstration, digital decision feed back equalizer (DFE) algorithms and a finite impulse response (FIR......In this thesis, digital signal processing (DSP) algorithms are studied to compensate for physical layer impairments in optical fiber coherent communication systems. The physical layer impairments investigated in this thesis include optical fiber chromatic dispersion, polarization demultiplexing...

  12. Comparison report of open calculations for ATLAS Domestic Standard Problem (DSP 02)

    International Nuclear Information System (INIS)

    Choi, Ki Yong; Kim, Y. S.; Kang, K. H.; Cho, S.; Park, H. S.; Choi, N. H.; Kim, B. D.; Min, K. H.; Park, J. K.; Chun, H. G.; Yu, Xin Guo; Kim, H. T.; Song, C. H.; Sim, S. K.; Jeon, S. S.; Kim, S. Y.; Kang, D. G.; Choi, T. S.; Kim, Y. M.; Lim, S. G.; Kim, H. S.; Kang, D. H.; Lee, G. H.; Jang, M. J.

    2012-09-01

    KAERI (Korea Atomic Energy Research Institute) has been operating an integral effect test facility, the Advanced Thermal Hydraulic Test Loop for Accident Simulation (ATLAS) for transient and accident simulations of advanced pressurized water reactors (PWRs). By using the ATLAS, a high quality integral effect test database has been established for major design basis accidents of the APR1400. A Domestic Standard Problem (DSP) exercise using the ATLAS database was promoted in order to transfer the database to domestic nuclear industries and to contribute to improving safety analysis methodology for PWRs. This 2nd ATLAS DSP exercise was led by KAERI in collaboration with KINS since the successful completion of the 1st ATLAS DSP in 2009. This exercise aims at effective utilization of integral effect database obtained from the ATLAS, establishment of cooperation framework among the domestic nuclear industry, better understanding of thermal hydraulic phenomena, and investigation of the possible limitation of the existing best estimate safety analysis codes. A small break loss of coolant accident of 6 inch break at the cold leg was determined as a target scenario by considering its technical importance and by incorporating with interests from participants. Twelve domestic organizations joined this DSP 02 exercise. Finally, eleven out of the joined organizations submitted their calculation results, including universities, government, and nuclear industries. This DSP exercise was performed in an open calculation environment where the integral effect test data was open to participants prior to code calculations. This report includes all information of the 2nd ATLAS DSP (DSP 02) exercise as well as comparison results between the calculations and the experimental data

  13. EVIDENCE-BASED ASSESSMENT OF VOICE DISORDERS: A THEORETICAL OVERVIEW AND MODEL

    Directory of Open Access Journals (Sweden)

    Dobrinka GEORGIEVA

    2011-04-01

    Full Text Available This article deals with the current paradigm of evidence-based practices of the Speech Therapy (Speech language pathology, especially diagnosing based on evidences of voice disorders. One of the main goals of this article is to define voice disorders according to the World Health Organization’s ICF multidimensional concept. Using a comparative method, this study attempts to prove that traditionally, the assessment outcomes of voice disorders in the Speech Therapy have been largely based on the speech therapist’s point of view and never on the client’s position. The research insists on establishing and adopting definitive gold standards, with respect to voice assessment and therapy in Bulgaria.

  14. Ontario Hydro's DSP update

    International Nuclear Information System (INIS)

    Anon.

    1992-01-01

    Ontario Hydro's Demand/Supply Plan (DSP), the 25 year plan which was submitted in December 1989, is currently being reviewed by the Environmental Assessment Board (EAB). Since 1989 there have been several changes which have led Ontario Hydro to update the original Demand/Supply Plan. This information sheet gives a quick overview of what has changed and how Ontario Hydro is adapting to that change

  15. Analysis of the Auditory Feedback and Phonation in Normal Voices.

    Science.gov (United States)

    Arbeiter, Mareike; Petermann, Simon; Hoppe, Ulrich; Bohr, Christopher; Doellinger, Michael; Ziethe, Anke

    2018-02-01

    The aim of this study was to investigate the auditory feedback mechanisms and voice quality during phonation in response to a spontaneous pitch change in the auditory feedback. Does the pitch shift reflex (PSR) change voice pitch and voice quality? Quantitative and qualitative voice characteristics were analyzed during the PSR. Twenty-eight healthy subjects underwent transnasal high-speed video endoscopy (HSV) at 8000 fps during sustained phonation [a]. While phonating, the subjects heard their sound pitched up for 700 cents (interval of a fifth), lasting 300 milliseconds in their auditory feedback. The electroencephalography (EEG), acoustic voice signal, electroglottography (EGG), and high-speed-videoendoscopy (HSV) were analyzed to compare feedback mechanisms for the pitched and unpitched condition of the phonation paradigm statistically. Furthermore, quantitative and qualitative voice characteristics were analyzed. The PSR was successfully detected within all signals of the experimental tools (EEG, EGG, acoustic voice signal, HSV). A significant increase of the perturbation measures and an increase of the values of the acoustic parameters during the PSR were observed, especially for the audio signal. The auditory feedback mechanism seems not only to control for voice pitch but also for voice quality aspects.

  16. Representing Voices from the Life-World in Evidence-Based Practice

    Science.gov (United States)

    Kovarsky, Dana

    2008-01-01

    Background: Current models of evidence-based practice marginalize and even silence the voices of those who are the potential beneficiaries of assessment and intervention. These missing voices can be found in the reflections of clients on their own life-world experiences. Aims: This paper examines how voices from the life-world are silenced in…

  17. Complexity Analysis and DSP Implementation of the Fractional-Order Lorenz Hyperchaotic System

    Directory of Open Access Journals (Sweden)

    Shaobo He

    2015-12-01

    Full Text Available The fractional-order hyperchaotic Lorenz system is solved as a discrete map by applying the Adomian decomposition method (ADM. Lyapunov Characteristic Exponents (LCEs of this system are calculated according to this deduced discrete map. Complexity of this system versus parameters are analyzed by LCEs, bifurcation diagrams, phase portraits, complexity algorithms. Results show that this system has rich dynamical behaviors. Chaos and hyperchaos can be generated by decreasing fractional order q in this system. It also shows that the system is more complex when q takes smaller values. SE and C 0 complexity algorithms provide a parameter choice criteria for practice applications of fractional-order chaotic systems. The fractional-order system is implemented by digital signal processor (DSP, and a pseudo-random bit generator is designed based on the implemented system, which passes the NIST test successfully.

  18. Realizing Ternary Logic in FPGAs for SWL DSP Systems

    Directory of Open Access Journals (Sweden)

    Tayeb Din

    2013-07-01

    Full Text Available Recently SWL (Short Word Length DSP (Digital Signal Processing applications has been proposed to overcome multiplier complexity that is evident in most of the digital applications. These SWL applications have been processed through sigma-delta modulation as a key element. For such applications, adder design plays vital role and can impact upon the chip area and its performance. In this paper, a ternary approach for adder tree has been proposed instead of binary that can accommodate more data with less chip-area at the cost of extra pin. The proposed ternary adder tree has been designed and developed in Quartus-II using three different design strategies namely T-gate (Ternary gate, LUT (Look Up Table and algebraic equations. Through rigorous simulation it was found that T-gate technique results in superior performance, an average of 23.5 and 33% improvement compared to the same adder structure based on Boolean Algebraic Equation and LUT, respectively. The proposed adder design would benefit the efficient implementation of SWL applications.

  19. The recognition of female voice based on voice registers in singing techniques in real-time using hankel transform method and macdonald function

    Science.gov (United States)

    Meiyanti, R.; Subandi, A.; Fuqara, N.; Budiman, M. A.; Siahaan, A. P. U.

    2018-03-01

    A singer doesn’t just recite the lyrics of a song, but also with the use of particular sound techniques to make it more beautiful. In the singing technique, more female have a diverse sound registers than male. There are so many registers of the human voice, but the voice registers used while singing, among others, Chest Voice, Head Voice, Falsetto, and Vocal fry. Research of speech recognition based on the female’s voice registers in singing technique is built using Borland Delphi 7.0. Speech recognition process performed by the input recorded voice samples and also in real time. Voice input will result in weight energy values based on calculations using Hankel Transformation method and Macdonald Functions. The results showed that the accuracy of the system depends on the accuracy of sound engineering that trained and tested, and obtained an average percentage of the successful introduction of the voice registers record reached 48.75 percent, while the average percentage of the successful introduction of the voice registers in real time to reach 57 percent.

  20. The Study of MSADQ/CDMA Protocol in Voice/Data Integration Packet Networks

    Institute of Scientific and Technical Information of China (English)

    2001-01-01

    A new packet medium access protocol, namely, minislot signalingaccess based on distributed queues(MSADQ/CDMA), is proposed in voice and data intergration CDMA networks. The MSADQ protocol is based on distributed queues and collision resolution algorithm. Through proper management of the PN codes, the number of random competition collision reduces greatly, the multiple access interference (MAI) decreases. It has several special access signaling channels to carry the voice and data access request. Each slot is devided into several control minislots (CMSs), in which the Data Terminals (DT) or Voice Terminals (VT) transmit their request. According to the voice and data traffic character, the signaling access structure is proposed. The code assign rules and queue managing rules are also proposed to ensure the QoS requirement of each traffic. Comparisions with other three protocol are developed by simulation, which shows that MSADQ/CDMA protocol occupies less PN codes, but still has very good performance.

  1. Sex determination using the Probabilistic Sex Diagnosis (DSP: Diagnose Sexuelle Probabiliste) tool in a virtual environment.

    Science.gov (United States)

    Chapman, Tara; Lefevre, Philippe; Semal, Patrick; Moiseev, Fedor; Sholukha, Victor; Louryan, Stéphane; Rooze, Marcel; Van Sint Jan, Serge

    2014-01-01

    The hip bone is one of the most reliable indicators of sex in the human body due to the fact it is the most dimorphic bone. Probabilistic Sex Diagnosis (DSP: Diagnose Sexuelle Probabiliste) developed by Murail et al., in 2005, is a sex determination method based on a worldwide hip bone metrical database. Sex is determined by comparing specific measurements taken from each specimen using sliding callipers and computing the probability of specimens being female or male. In forensic science it is sometimes not possible to sex a body due to corpse decay or injury. Skeletalization and dissection of a body is a laborious process and desecrates the body. There were two aims to this study. The first aim was to examine the accuracy of the DSP method in comparison with a current visual sexing method on sex determination. A further aim was to see if it was possible to virtually utilise the DSP method on both the hip bone and the pelvic girdle in order to utilise this method for forensic sciences. For the first part of the study, forty-nine dry hip bones of unknown sex were obtained from the Body Donation Programme of the Université Libre de Bruxelles (ULB). A comparison was made between DSP analysis and visual sexing on dry bone by two researchers. CT scans of bones were then analysed to obtain three-dimensional (3D) virtual models and the method of DSP was analysed virtually by importing the models into a customised software programme called lhpFusionBox which was developed at ULB. The software enables DSP distances to be measured via virtually-palpated bony landmarks. There was found to be 100% agreement of sex between the manual and virtual DSP method. The second part of the study aimed to further validate the method by analysing thirty-nine supplementary pelvic girdles of known sex blind. There was found to be a 100% accuracy rate further demonstrating that the virtual DSP method is robust. Statistically significant differences were found in the identification of sex

  2. Second ATLAS Domestic Standard Problem (DSP-02) For A Code Assessment

    International Nuclear Information System (INIS)

    Kim, Yeonsik; Choi, Kiyong; Cho, Seok; Park, Hyunsik; Kang, Kyungho; Song, Chulhwa; Baek, Wonpil

    2013-01-01

    KAERI (Korea Atomic Energy Research Institute) has been operating an integral effect test facility, the Advanced Thermal-Hydraulic Test Loop for Accident Simulation (ATLAS), for transient and accident simulations of advanced pressurized water reactors (PWRs). Using ATLAS, a high-quality integral effect test database has been established for major design basis accidents of the APR1400 plant. A Domestic Standard Problem (DSP) exercise using the ATLAS database was promoted to transfer the database to domestic nuclear industries and contribute to improving a safety analysis methodology for PWRs. This 2 nd ATLAS DSP (DSP-02) exercise aims at an effective utilization of an integral effect database obtained from ATLAS, the establishment of a cooperation framework among the domestic nuclear industry, a better understanding of the thermal hydraulic phenomena, and an investigation into the possible limitation of the existing best-estimate safety analysis codes. A small break loss of coolant accident with a 6-inch break at the cold leg was determined as a target scenario by considering its technical importance and by incorporating interests from participants. This DSP exercise was performed in an open calculation environment where the integral effect test data was open to participants prior to the code calculations. This paper includes major information of the DSP-02 exercise as well as comparison results between the calculations and the experimental data

  3. SECOND ATLAS DOMESTIC STANDARD PROBLEM (DSP-02 FOR A CODE ASSESSMENT

    Directory of Open Access Journals (Sweden)

    YEON-SIK KIM

    2013-12-01

    Full Text Available KAERI (Korea Atomic Energy Research Institute has been operating an integral effect test facility, the Advanced Thermal-Hydraulic Test Loop for Accident Simulation (ATLAS, for transient and accident simulations of advanced pressurized water reactors (PWRs. Using ATLAS, a high-quality integral effect test database has been established for major design basis accidents of the APR1400 plant. A Domestic Standard Problem (DSP exercise using the ATLAS database was promoted to transfer the database to domestic nuclear industries and contribute to improving a safety analysis methodology for PWRs. This 2nd ATLAS DSP (DSP-02 exercise aims at an effective utilization of an integral effect database obtained from ATLAS, the establishment of a cooperation framework among the domestic nuclear industry, a better understanding of the thermal hydraulic phenomena, and an investigation into the possible limitation of the existing best-estimate safety analysis codes. A small break loss of coolant accident with a 6-inch break at the cold leg was determined as a target scenario by considering its technical importance and by incorporating interests from participants. This DSP exercise was performed in an open calculation environment where the integral effect test data was open to participants prior to the code calculations. This paper includes major information of the DSP-02 exercise as well as comparison results between the calculations and the experimental data.

  4. CAS - Great success for the DSP course

    CERN Multimedia

    2007-01-01

    The CERN Accelerator School (CAS) and the Uppsala University jointly organized a specialized school on "Digital Signal Processing" in Sigtuna, Sweden from 1-9 June, 2007. This course was a "première" in many ways: firstly the topic had never been addressed by CAS, and secondly the structure of the course differed from the usual specialized courses in the sense that it was composed of 32 hours of theoretical lectures in the mornings and 16 hours "hands-on" courses in the afternoons. The latter, which have been designed by CERN experts, had some logistic implications in transporting computers and circuit boards (DSP and FPGA) to Sweden. The principle of this new approach was extremely well received by the accelerator community and 97 participants representing 23 different nationalities (80% of the participants originating from the CERN Member States) attended the course. As illustrated by the very positive feedback received from th...

  5. Power systems signal processing for smart grids

    CERN Document Server

    Ribeiro, Paulo Fernando; Ribeiro, Paulo Márcio; Cerqueira, Augusto Santiago

    2013-01-01

    With special relation to smart grids, this book provides clear and comprehensive explanation of how Digital Signal Processing (DSP) and Computational Intelligence (CI) techniques can be applied to solve problems in the power system. Its unique coverage bridges the gap between DSP, electrical power and energy engineering systems, showing many different techniques applied to typical and expected system conditions with practical power system examples. Surveying all recent advances on DSP for power systems, this book enables engineers and researchers to understand the current state of the art a

  6. Feedback control and beam diagnostic algorithms for a multiprocessor DSP system

    International Nuclear Information System (INIS)

    Teytelman, D.; Claus, R.; Fox, J.; Hindi, H.; Linscott, I.; Prabhakar, S.

    1996-09-01

    The multibunch longitudinal feedback system developed for use by PEP-II, ALS and DAΦNE uses a parallel array of digital signal processors to calculate the feedback signals from measurements of beam motion. The system is designed with general-purpose programmable elements which allow many feedback operating modes as well as system diagnostics, calibrations and accelerator measurements. The overall signal processing architecture of the system is illustrated. The real-time DSP algorithms and off-line postprocessing tools are presented. The problems in managing 320 K samples of data collected in one beam transient measurement are discussed and the solutions are presented. Example software structures are presented showing the beam feedback process, techniques for modal analysis of beam motion(used to quantify growth and damping rates of instabilities) and diagnostic functions (such as timing adjustment of beam pick-up and kicker components). These operating techniques are illustrated with example results obtained from the system installed at the Advanced Light Source at LBL

  7. VLSI signal processing technology

    CERN Document Server

    Swartzlander, Earl

    1994-01-01

    This book is the first in a set of forthcoming books focussed on state-of-the-art development in the VLSI Signal Processing area. It is a response to the tremendous research activities taking place in that field. These activities have been driven by two factors: the dramatic increase in demand for high speed signal processing, especially in consumer elec­ tronics, and the evolving microelectronic technologies. The available technology has always been one of the main factors in determining al­ gorithms, architectures, and design strategies to be followed. With every new technology, signal processing systems go through many changes in concepts, design methods, and implementation. The goal of this book is to introduce the reader to the main features of VLSI Signal Processing and the ongoing developments in this area. The focus of this book is on: • Current developments in Digital Signal Processing (DSP) pro­ cessors and architectures - several examples and case studies of existing DSP chips are discussed in...

  8. AVSynDEx: A Rapid Prototyping Process Dedicated to the Implementation of Digital Image Processing Applications on Multi-DSP and FPGA Architectures

    Directory of Open Access Journals (Sweden)

    Fresse Virginie

    2002-01-01

    Full Text Available We present AVSynDEx (concatenation of AVS SynDEx, a rapid prototyping process aiming to the implementation of digital signal processing applications on mixed architectures (multi-DSP FPGA. This process is based on the use of widely available and efficient CAD tools established along the design process so that most of the implementation tasks become automatic. These tools and architectures are judiciously selected and integrated during the implementation process to help a signal processing specialist without relevant hardware experience. We have automated the translation between the different levels of the process to increase and secure it. One main advantage is that only a signal processing designer is needed, all the other specialized manual tasks being transparent in this prototyping methodology, hereby reducing the implementation time.

  9. Seamless integration of 57.2-Gb/s signal wireline transmission and 100-GHz wireless delivery.

    Science.gov (United States)

    Li, Xinying; Yu, Jianjun; Dong, Ze; Cao, Zizheng; Chi, Nan; Zhang, Junwen; Shao, Yufeng; Tao, Li

    2012-10-22

    We experimentally demonstrated the seamless integration of 57.2-Gb/s signal wireline transmission and 100-GHz wireless delivery adopting polarization-division-multiplexing quadrature-phase-shift-keying (PDM-QPSK) modulation with 400-km single-mode fiber-28 (SMF-28) transmission and 1-m wireless delivery. The X- and Y-polarization components of optical PDM-QPSK baseband signal are simultaneously up-converted to 100 GHz by optical polarization-diversity heterodyne beating, and then independently transmitted and received by two pairs of transmitter and receiver antennas, which make up a 2x2 multiple-input multiple-output (MIMO) wireless link based on microwave polarization multiplexing. At the wireless receiver, a two-stage down conversion is firstly done in analog domain based on balanced mixer and sinusoidal radio frequency (RF) signal, and then in digital domain based on digital signal processing (DSP). Polarization de-multiplexing is realized by constant modulus algorithm (CMA) based on DSP in heterodyne coherent detection. Our experimental results show that more taps are required for CMA when the X- and Y-polarization antennas have different wireless distance.

  10. Reliability in perceptual analysis of voice quality.

    Science.gov (United States)

    Bele, Irene Velsvik

    2005-12-01

    This study focuses on speaking voice quality in male teachers (n = 35) and male actors (n = 36), who represent untrained and trained voice users, because we wanted to investigate normal and supranormal voices. In this study, both substantial and methodologic aspects were considered. It includes a method for perceptual voice evaluation, and a basic issue was rater reliability. A listening group of 10 listeners, 7 experienced speech-language therapists, and 3 speech-language therapist students evaluated the voices by 15 vocal characteristics using VA scales. Two sets of voice signals were investigated: text reading (2 loudness levels) and sustained vowel (3 levels). The results indicated a high interrater reliability for most perceptual characteristics. Connected speech was evaluated more reliably, especially at the normal level, but both types of voice signals were evaluated reliably, although the reliability for connected speech was somewhat higher than for vowels. Experienced listeners tended to be more consistent in their ratings than did the student raters. Some vocal characteristics achieved acceptable reliability even with a smaller panel of listeners. The perceptual characteristics grouped in 4 factors reflected perceptual dimensions.

  11. Digital signal processing for He3 proportional counter

    International Nuclear Information System (INIS)

    Zeynalov, Sh.S.; Ahmadov, Q.S.

    2010-01-01

    Full text : Data acquisition systems for nuclear spectroscopy have traditionally been based on systems with analog shaping amplifiers followed by analog-to-digital converters. Recently, however, new systems based on digital signal processing make possible to replace the analog shaping and timing circuitry the numerical algorithms to derive properties of the pulse such as its amplitude. DSP is a fully numerical analysis of the detector pulse signals and this technique demonstrates significant advantages over analog systems in some circumstances. From a mathematical point of view, one can consider the signal evolution from the detector to the ADC as a sequence of transformations that can be described by precisely defined mathematical expressions. Digital signal processing with ADCs has the possibility to utilize further information on the signal pulses from radiation detectors. In the experiment each step of the signal generation in the 3He filled proportional counter was described using digital signal processing techniques (DSP). The electronic system has consisted of a detector, a preamplifier and a digital oscilloscope. The pulses from the detector were digitized using a digital storage oscilloscope. This oscilloscope allowed signal digitization with accuracy of 8 bit (256 levels) and with frequency of up to 5 * 10 8 samples/s. As a neutron source was used Cf-252. To obtain detector output current pulse I(t) created by the motions of the ions/electrons pairs was written an algorithm which can easily be programmed using modern computer programming languages.

  12. Voice and gesture-based 3D multimedia presentation tool

    Science.gov (United States)

    Fukutake, Hiromichi; Akazawa, Yoshiaki; Okada, Yoshihiro

    2007-09-01

    This paper proposes a 3D multimedia presentation tool that allows the user to manipulate intuitively only through the voice input and the gesture input without using a standard keyboard or a mouse device. The authors developed this system as a presentation tool to be used in a presentation room equipped a large screen like an exhibition room in a museum because, in such a presentation environment, it is better to use voice commands and the gesture pointing input rather than using a keyboard or a mouse device. This system was developed using IntelligentBox, which is a component-based 3D graphics software development system. IntelligentBox has already provided various types of 3D visible, reactive functional components called boxes, e.g., a voice input component and various multimedia handling components. IntelligentBox also provides a dynamic data linkage mechanism called slot-connection that allows the user to develop 3D graphics applications by combining already existing boxes through direct manipulations on a computer screen. Using IntelligentBox, the 3D multimedia presentation tool proposed in this paper was also developed as combined components only through direct manipulations on a computer screen. The authors have already proposed a 3D multimedia presentation tool using a stage metaphor and its voice input interface. This time, we extended the system to make it accept the user gesture input besides voice commands. This paper explains details of the proposed 3D multimedia presentation tool and especially describes its component-based voice and gesture input interfaces.

  13. Photonic Ultra-Wideband 781.25-Mb/s Signal Generation and Transmission Incorporating Digital Signal Processing Detection

    DEFF Research Database (Denmark)

    Gibbon, Timothy Braidwood; Yu, Xianbin; Tafur Monroy, Idelfonso

    2009-01-01

    The generation of photonic ultra-wideband (UWB) impulse signals using an uncooled distributed-feedback laser is proposed. For the first time, we experimentally demonstrate bit-for-bit digital signal processing (DSP) bit-error-rate measurements for transmission of a 781.25-Mb/s photonic UWB signal...

  14. Multirate Digital Filters Based on FPGA and Its Applications

    International Nuclear Information System (INIS)

    Sharaf El-Din, R.M.A.

    2013-01-01

    Digital Signal Processing (DSP) is one of the fastest growing techniques in the electronics industry. It is used in a wide range of application fields such as, telecommunications, data communications, image enhancement and processing, video signals, digital TV broadcasting, and voice synthesis and recognition. Field Programmable Gate Array (FPGA) offers good solution for addressing the needs of high performance DSP systems. The focus of this thesis is on one of the basic DSP functions, namely filtering signals to remove unwanted frequency bands. Multi rate Digital Filters (MDFs) are the main theme here. Theory and implementation of MDF, as a special class of digital filters, will be discussed. Multi rate digital filters represent a class of digital filters having a number of attractive features like, low requirements for the coefficient word lengths, significant saving in computation and storage requirements results in a significant reduction in its dynamic power consumption. This thesis introduces an efficient FPGA realization of a multi rate decimation filter with narrow pass-band and narrow transition band to reduce the frequency sample rate by factor of 64 for noise thermometer applications. The proposed multi rate decimation filter is composed of three stages; the first stage is a Cascaded Integrator Comb (CIC) decimation filter, the second stage is a two-coefficient Half-Band (HB) filter and the last stage is a sharper transition HB filter. The frequency responses of individual stages as well as the overall filter response have been demonstrated with full simulation using MATLAB. The design and implementation of the proposed MDF on FPGA (XILINX Virtex XCV800 BG432-4), using VHSIC Hardware Description Language (VHDL), has been introduced. The implementation areas of the proposed filter stages are compared. Using CIC-HB technique saves 18% of the design area, compared to using six stages HB decimation filters.

  15. Signal Processing in Medical Ultrasound B-mode Imaging

    International Nuclear Information System (INIS)

    Song, Tai Kyong

    2000-01-01

    Ultrasonic imaging is the most widely used modality among modern imaging device for medical diagnosis and the system performance has been improved dramatically since early 90's due to the rapid advances in DSP performance and VLSI technology that made it possible to employ more sophisticated algorithms. This paper describes 'main stream' digital signal processing functions along with the associated implementation considerations in modern medical ultrasound imaging systems. Topics covered include signal processing methods for resolution improvement, ultrasound imaging system architectures, roles and necessity of the applications of DSP and VLSI technology in the development of the medical ultrasound imaging systems, and array signal processing techniques for ultrasound focusing

  16. Enhanced Living by Assessing Voice Pathology Using a Co-Occurrence Matrix.

    Science.gov (United States)

    Muhammad, Ghulam; Alhamid, Mohammed F; Hossain, M Shamim; Almogren, Ahmad S; Vasilakos, Athanasios V

    2017-01-29

    A large number of the population around the world suffers from various disabilities. Disabilities affect not only children but also adults of different professions. Smart technology can assist the disabled population and lead to a comfortable life in an enhanced living environment (ELE). In this paper, we propose an effective voice pathology assessment system that works in a smart home framework. The proposed system takes input from various sensors, and processes the acquired voice signals and electroglottography (EGG) signals. Co-occurrence matrices in different directions and neighborhoods from the spectrograms of these signals were obtained. Several features such as energy, entropy, contrast, and homogeneity from these matrices were calculated and fed into a Gaussian mixture model-based classifier. Experiments were performed with a publicly available database, namely, the Saarbrucken voice database. The results demonstrate the feasibility of the proposed system in light of its high accuracy and speed. The proposed system can be extended to assess other disabilities in an ELE.

  17. Fast, multi-channel real-time processing of signals with microsecond latency using graphics processing units

    Energy Technology Data Exchange (ETDEWEB)

    Rath, N., E-mail: Nikolaus@rath.org; Levesque, J. P.; Mauel, M. E.; Navratil, G. A.; Peng, Q. [Department of Applied Physics and Applied Mathematics, Columbia University, 500 W 120th St, New York, New York 10027 (United States); Kato, S. [Department of Information Engineering, Nagoya University, Nagoya (Japan)

    2014-04-15

    Fast, digital signal processing (DSP) has many applications. Typical hardware options for performing DSP are field-programmable gate arrays (FPGAs), application-specific integrated DSP chips, or general purpose personal computer systems. This paper presents a novel DSP platform that has been developed for feedback control on the HBT-EP tokamak device. The system runs all signal processing exclusively on a Graphics Processing Unit (GPU) to achieve real-time performance with latencies below 8 μs. Signals are transferred into and out of the GPU using PCI Express peer-to-peer direct-memory-access transfers without involvement of the central processing unit or host memory. Tests were performed on the feedback control system of the HBT-EP tokamak using forty 16-bit floating point inputs and outputs each and a sampling rate of up to 250 kHz. Signals were digitized by a D-TACQ ACQ196 module, processing done on an NVIDIA GTX 580 GPU programmed in CUDA, and analog output was generated by D-TACQ AO32CPCI modules.

  18. Fast, multi-channel real-time processing of signals with microsecond latency using graphics processing units

    International Nuclear Information System (INIS)

    Rath, N.; Levesque, J. P.; Mauel, M. E.; Navratil, G. A.; Peng, Q.; Kato, S.

    2014-01-01

    Fast, digital signal processing (DSP) has many applications. Typical hardware options for performing DSP are field-programmable gate arrays (FPGAs), application-specific integrated DSP chips, or general purpose personal computer systems. This paper presents a novel DSP platform that has been developed for feedback control on the HBT-EP tokamak device. The system runs all signal processing exclusively on a Graphics Processing Unit (GPU) to achieve real-time performance with latencies below 8 μs. Signals are transferred into and out of the GPU using PCI Express peer-to-peer direct-memory-access transfers without involvement of the central processing unit or host memory. Tests were performed on the feedback control system of the HBT-EP tokamak using forty 16-bit floating point inputs and outputs each and a sampling rate of up to 250 kHz. Signals were digitized by a D-TACQ ACQ196 module, processing done on an NVIDIA GTX 580 GPU programmed in CUDA, and analog output was generated by D-TACQ AO32CPCI modules

  19. Digital signal processing using MATLAB

    CERN Document Server

    Schilling, Robert L

    2016-01-01

    Focus on the development, implementation, and application of modern DSP techniques with DIGITAL SIGNAL PROCESSING USING MATLAB(R), 3E. Written in an engaging, informal style, this edition immediately captures your attention and encourages you to explore each critical topic. Every chapter starts with a motivational section that highlights practical examples and challenges that you can solve using techniques covered in the chapter. Each chapter concludes with a detailed case study example, a chapter summary with learning outcomes, and practical homework problems cross-referenced to specific chapter sections for your convenience. DSP Companion software accompanies each book to enable further investigation. The DSP Companion software operates with MATLAB(R) and provides intriguing demonstrations as well as interactive explorations of analysis and design concepts.

  20. Digital signal processing for He3 proportional counter

    International Nuclear Information System (INIS)

    Ahmadov, Q.S.; Institute of Radiation Problems, ANAS, Baku

    2011-01-01

    Full text: Data acquisition systems for nuclear spectroscopy have traditionally been based on systems with analog shaping amplifiers followed by analog-to-digital converters. Recently, however, new systems based on digital signal processing allow us to replace the analog shaping and timing circuitry the numerical algorithms to derive properties of the pulse such as its amplitude. DSP is a fully numerical analysis of the detector pulse signals and this technique demonstrates significant advantages over analog systems in some circumstances. From a mathematical point of view, one can consider the signal evolution from the detector to the ADC as a sequence of transformations that can be described by precisely defined mathematical expressions.Digital signal processing with ADCs has the possibility to utilize further information on the signal pulses from radiation detectors [1] [2]. In the experiment each step of the signal generation in the 3He filled proportional counter was described using digital signal processing techniques (DSP). The electronic system has consisted of a detector, a preamplifier and a digital oscilloscope. The pulses from the detector were digitized using a OTSZS-02 (250USB)-4 digital storage oscilloscope from ZAO R UDNEV-SHILYAYEV . This oscilloscope allowed signal digitization with accuracy of 8 bit(256 levels) and with frequency of up to 5.10''8 samples/s. As a neutron source was used Cf-252.To obtain detector output current pulse I(t) created by the motions of the ions/electrons pairs was written an algorithm which can easily be programmed using modern computer programming languages

  1. Expression of the dspA/E gene of Erwinia amylovora in non-host plant Arabidopsis thaliana

    Directory of Open Access Journals (Sweden)

    Hasan Murat Aksoy

    2017-01-01

    Full Text Available In the Erwinia amylovora genome, the hrp gene cluster containing the dspA/E/EB/F operon plays a crucial role in mediating the pathogenicity and the hypersensitive response (HR in the host plant. The role of the dspA/E gene derived from E. amylovora was investigated by monitoring the expression of the β-glucuronidase (GUS reporter system in transgenic Arabidopsis thaliana cv. Pri-Gus seedlings. A mutant ΔdspA/E strain of E. amylovora was generated to contain a deletion of the dspA/E gene for the purpose of this study. Two-week-old seedlings of GUS transgenic Arabidopsis were vacuum-infiltrated with the wild-type and the mutant (ΔdspA/E E. amylovora strains. The Arabidopsis seedlings were fixed and stained for GUS activity after 3–5 days following infiltration. The appearance of dense spots with blue staining on the Arabidopsis leaves indicated the typical characteristic of GUS activity. This observation indicated that the wild-type E. amylovora strain had induced a successful and efficient infection on the A. thaliana Pri-Gus leaves. In contrast, there was no visible GUS expression on leaf tissues which were inoculated with the ΔdspA/E mutant E. amylovora strain. These results indicate that the dspA/E gene is required by the bacterial cells to induce HR in non-host plants.

  2. AT89S52 Microcontroller Based Digital Compass With Voice Output

    Directory of Open Access Journals (Sweden)

    Fahmi Fardiyan Arief

    2008-04-01

    Full Text Available In this paper, the design of digital compass with voice output is described, so that the blind can also use it. The digital compass is designed based on up-graded conventional compass. In the axis direction of conventional compass be added a disc as source of wind direction information, and phototransistor as sensor. The digital compass system is designed, based on AT89S52 microcontroller, as control of all interfaces and read sensor. The LCD component is used as display and ISD 2590 IC as voice recorder. The IC can record with maximum capacity 90 seconds. The voices output of compass is divided into 8 direction from the north, southwest, west and the next. The result showed that the design of digital compass work as like conventional compass completely by voice feature.

  3. A Wireless LAN and Voice Information System for Underground Coal Mine

    Directory of Open Access Journals (Sweden)

    Yu Zhang

    2014-06-01

    Full Text Available In this paper we constructed a wireless information system, and developed a wireless voice communication subsystem based on Wireless Local Area Networks (WLAN for underground coal mine, which employs Voice over IP (VoIP technology and Session Initiation Protocol (SIP to achieve wireless voice dispatching communications. The master control voice dispatching interface and call terminal software are also developed on the WLAN ground server side to manage and implement the voice dispatching communication. A testing system for voice communication was constructed in tunnels of an underground coal mine, which was used to actually test the wireless voice communication subsystem via a network analysis tool, named Clear Sight Analyzer. In tests, the actual flow charts of registration, call establishment and call removal were analyzed by capturing call signaling of SIP terminals, and the key performance indicators were evaluated in coal mine, including average subjective value of voice quality, packet loss rate, delay jitter, disorder packet transmission and end-to- end delay. Experimental results and analysis demonstrate that the wireless voice communication subsystem developed communicates well in underground coal mine environment, achieving the designed function of voice dispatching communication.

  4. Efficacy of DSP30-IL2/TPA for detection of cytogenetic abnormalities in chronic lymphocytic leukaemia/small lymphocytic lymphoma.

    Science.gov (United States)

    Holmes, P J; Peiper, S C; Uppal, G K; Gong, J Z; Wang, Z-X; Bajaj, R

    2016-10-01

    Chronic lymphocytic leukaemia (CLL) is the most prevalent leukaemia in the Western Hemisphere. Cytogenetic abnormalities in CLL are used for diagnosis, prognosis and treatment. However, detecting these is difficult because mature B cells do not readily divide in culture. Here, we present data on two mitogen cocktails: CpG-oligonucleotide DSP30/Interleukin-2 (IL-2) and DSP30/IL-2 in combination with 12-O-tetradecanoylphorbol-13-acetate (TPA). We analysed 165 cases of CLL with FISH and cytogenetics from January 2011 to June 2013. In 2011, three cultures were set-up: unstimulated, DSP30/IL-2-stimulated and TPA-stimulated. In 2012-2013, two cultures were set-up: unstimulated and stimulated with TPA/DSP30/IL-2. In 2011, FISH had a detection rate of 91% and cytogenetics using DSP30/IL2 had a detection rate of 91% (n = 22). In 2012-2013, FISH had a detection rate of 79% and cytogenetics using TPA/DSP30/IL-2 had a detection rate of 98% (n = 40). The percentage of cases with normal FISH but abnormal cytogenetics increased from 9% in 2011 to 21% in 2012-2013. The TPA/DSP30/IL-2 cultures in 2012-2013 detected more novel abnormalities (n = 5) as compared to DSP30/IL-2 alone (n = 3). TPA/DSP30/IL2 was as good as or better than DSP30/IL2 alone. TPA/DSP30/IL-2 offers a high detection rate for CLL abnormalities with a single stimulated culture and may increase detection of clinically significant abnormalities. © 2016 John Wiley & Sons Ltd.

  5. Digital signals processing using non-linear orthogonal transformation in frequency domain

    Directory of Open Access Journals (Sweden)

    Ivanichenko E.V.

    2017-12-01

    Full Text Available The rapid progress of computer technology in recent decades led to a wide introduction of methods of digital information processing practically in all fields of scientific research. In this case, among various applications of computing one of the most important places is occupied by digital processing systems signals (DSP that are used in data processing remote solution tasks of navigation of aerospace and marine objects, communications, radiophysics, digital optics and in a number of other applications. Digital Signal Processing (DSP is a dynamically developing an area that covers both technical and software tools. Related areas for digital signal processing are theory information, in particular, the theory of optimal signal reception and theory pattern recognition. In the first case, the main problem is signal extraction against a background of noise and interference of a different physical nature, and in the second - automatic recognition, i.e. classification and signal identification. In the digital processing of signals under a signal, we mean its mathematical description, i.e. a certain real function, containing information on the state or behavior of a physical system under an event that can be defined on a continuous or discrete space of time variation or spatial coordinates. In the broad sense, DSP systems mean a complex algorithmic, hardware and software. As a rule, systems contain specialized technical means of preliminary (or primary signal processing and special technical means for secondary processing of signals. Means of pretreatment are designed to process the original signals observed in general case against a background of random noise and interference of a different physical nature and represented in the form of discrete digital samples, for the purpose of detecting and selection (selection of the useful signal and evaluation characteristics of the detected signal. A new method of digital signal processing in the frequency

  6. An FPGA-based rapid prototyping platform for wavelet coprocessors

    Science.gov (United States)

    Vera, Alonzo; Meyer-Baese, Uwe; Pattichis, Marios

    2007-04-01

    MatLab/Simulink-based design flows are being used by DSP designers to improve time-to-market of FPGA implementations. 1 Commonly, digital signal processing cores are integrated in an embedded system as coprocessors. Existing CAD tools do not fully address the integration of a DSP coprocessor into an embedded system design. This integration might prove to be time consuming and error prone. It also requires that the DSP designer has an excellent knowledge of embedded systems and computer architecture details. We present a prototyping platform and design flow that allows rapid integration of embedded systems with a wavelet coprocessor. The platform comprises of software and hardware modules that allow a DSP designer a painless integration of a coprocessor with a PowerPC-based embedded system. The platform has a wide range of applications, from industrial to educational environments.

  7. Programmable optical processor chips: toward photonic RF filters with DSP-level flexibility and MHz-band selectivity

    Directory of Open Access Journals (Sweden)

    Xie Yiwei

    2017-12-01

    Full Text Available Integrated optical signal processors have been identified as a powerful engine for optical processing of microwave signals. They enable wideband and stable signal processing operations on miniaturized chips with ultimate control precision. As a promising application, such processors enables photonic implementations of reconfigurable radio frequency (RF filters with wide design flexibility, large bandwidth, and high-frequency selectivity. This is a key technology for photonic-assisted RF front ends that opens a path to overcoming the bandwidth limitation of current digital electronics. Here, the recent progress of integrated optical signal processors for implementing such RF filters is reviewed. We highlight the use of a low-loss, high-index-contrast stoichiometric silicon nitride waveguide which promises to serve as a practical material platform for realizing high-performance optical signal processors and points toward photonic RF filters with digital signal processing (DSP-level flexibility, hundreds-GHz bandwidth, MHz-band frequency selectivity, and full system integration on a chip scale.

  8. Performance of the phonatory deviation diagram in the evaluation of rough and breathy synthesized voices.

    Science.gov (United States)

    Lopes, Leonardo Wanderley; Freitas, Jonas Almeida de; Almeida, Anna Alice; Silva, Priscila Oliveira Costa; Alves, Giorvan Ânderson Dos Santos

    2017-07-05

    Voice disorders alter the sound signal in several ways, combining several types of vocal emission disturbances and noise. The Phonatory Deviation Diagram (PDD) is a two-dimensional chart that allows the evaluation of the vocal signal based on the combination of periodicity (jitter, shimmer, and correlation coefficient) and noise (Glottal to Noise Excitation - GNE) measurements. The use of synthesized signals, where one has a greater control and knowledge of the production conditions, may allow a better understanding of the physiological and acoustic mechanisms underlying the vocal emission and its main perceptual-auditory correlates regarding the intensity of the deviation and types of vocal quality. To analyze the performance of the PDD in the discrimination of the presence and degree of roughness and breathiness in synthesized voices. 871 synthesized vocal signals were used corresponding to the vowel /ɛ/. The perceptual-auditory analysis of the degree of roughness and breathiness of the synthesized signals was performed using Visual Analogue Scale (VAS). Subsequently, the signals were categorized regarding the presence/absence of these parameters based on the VAS cutoff values. Acoustic analysis was performed by assessing the distribution of vocal signals according to the PDD area, quadrant, shape, and density. The equality of proportions and the chi-square tests were performed to compare the variables. Rough and breathy vocal signals were located predominantly outside the normal range and in the lower right quadrant of the PDD. Voices with higher degrees of roughness and breathiness were located outside the area of normality in the lower right quadrant and had concentrated density. The normality area and the PDD quadrant can discriminate healthy voices from rough and breathy ones. Voices with higher degrees of roughness and breathiness are proportionally located outside the area of normality, in the lower right quadrant and with concentrated density. Copyright

  9. Types for DSP Assembler Programs

    DEFF Research Database (Denmark)

    Larsen, Ken

    2006-01-01

    for reuse, and a procedure that computes point-wise vector multiplication. The latter uses a common idiom of prefetching memory resulting in out-of-bounds reading from memory. I present two extensions to the baseline type system: The first extension is a simple modification of some type rules to allow out......-ofbounds reading from memory. The second extension is based on two major modifications of the baseline type system: • Abandoning the type-invariance principle of memory locations and using a variation of alias types instead. • Introducing aggregate types, making it possible to have different views of a block...... of memory, thus enabling type checking of programs that directly manage and reuse memory. I show that both the baseline type system and the extended type system can be used to give type annotations to handwritten DSP assembler code, and that these annotations precisely and succinctly describe...

  10. Using a digital signal processor as a data stream controller for digital subtraction angiography

    International Nuclear Information System (INIS)

    Meng, J.D.; Katz, J.E.

    1991-10-01

    High speed, flexibility, and good arithmetic abilities make digital signal processors (DSP) a good choice as input/output controllers for real time applications. The DSP can be made to pre-process data in real time to reduce data volume, to open early windows on what is being acquired and to implement local servo loops. We present an example of a DSP as an input/output controller for a digital subtraction angiographic imaging system. The DSP pre-processes the raw data, reducing data volume by a factor of two, and is potentially capable of producing real-time subtracted images for immediate display

  11. Signal processing for molecular and cellular biological physics: an emerging field.

    Science.gov (United States)

    Little, Max A; Jones, Nick S

    2013-02-13

    Recent advances in our ability to watch the molecular and cellular processes of life in action--such as atomic force microscopy, optical tweezers and Forster fluorescence resonance energy transfer--raise challenges for digital signal processing (DSP) of the resulting experimental data. This article explores the unique properties of such biophysical time series that set them apart from other signals, such as the prevalence of abrupt jumps and steps, multi-modal distributions and autocorrelated noise. It exposes the problems with classical linear DSP algorithms applied to this kind of data, and describes new nonlinear and non-Gaussian algorithms that are able to extract information that is of direct relevance to biological physicists. It is argued that these new methods applied in this context typify the nascent field of biophysical DSP. Practical experimental examples are supplied.

  12. Mapping Phonetic Features for Voice-Driven Sound Synthesis

    Science.gov (United States)

    Janer, Jordi; Maestre, Esteban

    In applications where the human voice controls the synthesis of musical instruments sounds, phonetics convey musical information that might be related to the sound of the imitated musical instrument. Our initial hypothesis is that phonetics are user- and instrument-dependent, but they remain constant for a single subject and instrument. We propose a user-adapted system, where mappings from voice features to synthesis parameters depend on how subjects sing musical articulations, i.e. note to note transitions. The system consists of two components. First, a voice signal segmentation module that automatically determines note-to-note transitions. Second, a classifier that determines the type of musical articulation for each transition based on a set of phonetic features. For validating our hypothesis, we run an experiment where subjects imitated real instrument recordings with their voice. Performance recordings consisted of short phrases of saxophone and violin performed in three grades of musical articulation labeled as: staccato, normal, legato. The results of a supervised training classifier (user-dependent) are compared to a classifier based on heuristic rules (user-independent). Finally, from the previous results we show how to control the articulation in a sample-concatenation synthesizer by selecting the most appropriate samples.

  13. Modern Methods of Voice Authentication in Mobile Devices

    Directory of Open Access Journals (Sweden)

    Vladimir Leonovich Evseev

    2016-03-01

    Full Text Available Modern methods of voice authentication in mobile devices.The proposed evaluation of the probability errors of the first and second kind for multi-modal methods of voice authentication. The advantages of multimodal multivariate methods before, when authentication takes place in several stages – this is the one-stage, which means convenience for customers. Further development of multimodal methods of authentication will be based on the significantly increased computing power of mobile devices, the growing number and improved accuracy built-in mobile device sensors, as well as to improve the algorithms of signal processing.

  14. Single Event Upset Analysis: On-orbit performance of the Alpha Magnetic Spectrometer Digital Signal Processor Memory aboard the International Space Station

    Science.gov (United States)

    Li, Jiaqiang; Choutko, Vitaly; Xiao, Liyi

    2018-03-01

    Based on the collection of error data from the Alpha Magnetic Spectrometer (AMS) Digital Signal Processors (DSP), on-orbit Single Event Upsets (SEUs) of the DSP program memory are analyzed. The daily error distribution and time intervals between errors are calculated to evaluate the reliability of the system. The particle density distribution of International Space Station (ISS) orbit is presented and the effects from the South Atlantic Anomaly (SAA) and the geomagnetic poles are analyzed. The impact of solar events on the DSP program memory is carried out combining data analysis and Monte Carlo simulation (MC). From the analysis and simulation results, it is concluded that the area corresponding to the SAA is the main source of errors on the ISS orbit. Solar events can also cause errors on DSP program memory, but the effect depends on the on-orbit particle density.

  15. Removing the Influence of Shimmer in the Calculation of Harmonics-To-Noise Ratios Using Ensemble-Averages in Voice Signals

    OpenAIRE

    Carlos Ferrer; Eduardo González; María E. Hernández-Díaz; Diana Torres; Anesto del Toro

    2009-01-01

    Harmonics-to-noise ratios (HNRs) are affected by general aperiodicity in voiced speech signals. To specifically reflect a signal-to-additive-noise ratio, the measurement should be insensitive to other periodicity perturbations, like jitter, shimmer, and waveform variability. The ensemble averaging technique is a time-domain method which has been gradually refined in terms of its sensitivity to jitter and waveform variability and required number of pulses. In this paper, shimmer is introduced ...

  16. Mobile voice health monitoring using a wearable accelerometer sensor and a smartphone platform.

    Science.gov (United States)

    Mehta, Daryush D; Zañartu, Matías; Feng, Shengran W; Cheyne, Harold A; Hillman, Robert E

    2012-11-01

    Many common voice disorders are chronic or recurring conditions that are likely to result from faulty and/or abusive patterns of vocal behavior, referred to generically as vocal hyperfunction. An ongoing goal in clinical voice assessment is the development and use of noninvasively derived measures to quantify and track the daily status of vocal hyperfunction so that the diagnosis and treatment of such behaviorally based voice disorders can be improved. This paper reports on the development of a new, versatile, and cost-effective clinical tool for mobile voice monitoring that acquires the high-bandwidth signal from an accelerometer sensor placed on the neck skin above the collarbone. Using a smartphone as the data acquisition platform, the prototype device provides a user-friendly interface for voice use monitoring, daily sensor calibration, and periodic alert capabilities. Pilot data are reported from three vocally normal speakers and three subjects with voice disorders to demonstrate the potential of the device to yield standard measures of fundamental frequency and sound pressure level and model-based glottal airflow properties. The smartphone-based platform enables future clinical studies for the identification of the best set of measures for differentiating between normal and hyperfunctional patterns of voice use.

  17. The sound of trustworthiness: Acoustic-based modulation of perceived voice personality.

    Directory of Open Access Journals (Sweden)

    Pascal Belin

    Full Text Available When we hear a new voice we automatically form a "first impression" of the voice owner's personality; a single word is sufficient to yield ratings highly consistent across listeners. Past studies have shown correlations between personality ratings and acoustical parameters of voice, suggesting a potential acoustical basis for voice personality impressions, but its nature and extent remain unclear. Here we used data-driven voice computational modelling to investigate the link between acoustics and perceived trustworthiness in the single word "hello". Two prototypical voice stimuli were generated based on the acoustical features of voices rated low or high in perceived trustworthiness, respectively, as well as a continuum of stimuli inter- and extrapolated between these two prototypes. Five hundred listeners provided trustworthiness ratings on the stimuli via an online interface. We observed an extremely tight relationship between trustworthiness ratings and position along the trustworthiness continuum (r = 0.99. Not only were trustworthiness ratings higher for the high- than the low-prototypes, but the difference could be modulated quasi-linearly by reducing or exaggerating the acoustical difference between the prototypes, resulting in a strong caricaturing effect. The f0 trajectory, or intonation, appeared a parameter of particular relevance: hellos rated high in trustworthiness were characterized by a high starting f0 then a marked decrease at mid-utterance to finish on a strong rise. These results demonstrate a strong acoustical basis for voice personality impressions, opening the door to multiple potential applications.

  18. Motorcycle Start-stop System based on Intelligent Biometric Voice Recognition

    Science.gov (United States)

    Winda, A.; E Byan, W. R.; Sofyan; Armansyah; Zariantin, D. L.; Josep, B. G.

    2017-03-01

    Current mechanical key in the motorcycle is prone to bulgary, being stolen or misplaced. Intelligent biometric voice recognition as means to replace this mechanism is proposed as an alternative. The proposed system will decide whether the voice is belong to the user or not and the word utter by the user is ‘On’ or ‘Off’. The decision voice will be sent to Arduino in order to start or stop the engine. The recorded voice is processed in order to get some features which later be used as input to the proposed system. The Mel-Frequency Ceptral Coefficient (MFCC) is adopted as a feature extraction technique. The extracted feature is the used as input to the SVM-based identifier. Experimental results confirm the effectiveness of the proposed intelligent voice recognition and word recognition system. It show that the proposed method produces a good training and testing accuracy, 99.31% and 99.43%, respectively. Moreover, the proposed system shows the performance of false rejection rate (FRR) and false acceptance rate (FAR) accuracy of 0.18% and 17.58%, respectively. In the intelligent word recognition shows that the training and testing accuracy are 100% and 96.3%, respectively.

  19. A method of incident angle estimation for high resolution spectral recovery in filter-array-based spectrometers

    Science.gov (United States)

    Kim, Cheolsun; Lee, Woong-Bi; Ju, Gun Wu; Cho, Jeonghoon; Kim, Seongmin; Oh, Jinkyung; Lim, Dongsung; Lee, Yong Tak; Lee, Heung-No

    2017-02-01

    In recent years, there has been an increasing interest in miniature spectrometers for research and development. Especially, filter-array-based spectrometers have advantages of low cost and portability, and can be applied in various fields such as biology, chemistry and food industry. Miniaturization in optical filters causes degradation of spectral resolution due to limitations on spectral responses and the number of filters. Nowadays, many studies have been reported that the filter-array-based spectrometers have achieved resolution improvements by using digital signal processing (DSP) techniques. The performance of the DSP-based spectral recovery highly depends on the prior information of transmission functions (TFs) of the filters. The TFs vary with respect to an incident angle of light onto the filter-array. Conventionally, it is assumed that the incident angle of light on the filters is fixed and the TFs are known to the DSP. However, the incident angle is inconstant according to various environments and applications, and thus TFs also vary, which leads to performance degradation of spectral recovery. In this paper, we propose a method of incident angle estimation (IAE) for high resolution spectral recovery in the filter-array-based spectrometers. By exploiting sparse signal reconstruction of the L1- norm minimization, IAE estimates an incident angle among all possible incident angles which minimizes the error of the reconstructed signal. Based on IAE, DSP effectively provides a high resolution spectral recovery in the filter-array-based spectrometers.

  20. Design and evaluation of online arithmetic for signal processing applications on FPGAs

    Science.gov (United States)

    Galli, Reto; Tenca, Alexandre F.

    2001-11-01

    This paper shows the design and the evaluation of on-line arithmetic modules for the most common operators used in DSP applications, using FPGAs as the target technology. The designs are highly optimized for the target technology and the common range of precision in DSP. The results are based on experimental data collected using CAD tools. All designs are synthesized for the same type of devices (Xilinx XC4000) for comparison, avoiding rough estimates of the system performance, and generating a more reliable and detailed comparison of on-line signal processing solutions with other state of the art approaches, such as distributed arithmetic. We show that on-line designs have a hard stand for basic DSP applications that use only addition and multiplication. However, we also show that on-line designs are able to overtake other approaches as the applications become more sophisticated, e.g. when data dependencies exist, or when non constant multiplicands restrict the use of other approaches.

  1. Voice Based City Panic Button System

    Science.gov (United States)

    Febriansyah; Zainuddin, Zahir; Bachtiar Nappu, M.

    2018-03-01

    The development of voice activated panic button application aims to design faster early notification of hazardous condition in community to the nearest police by using speech as the detector where the current application still applies touch-combination on screen and use coordination of orders from control center then the early notification still takes longer time. The method used in this research was by using voice recognition as the user voice detection and haversine formula for the comparison of closest distance between the user and the police. This research was equipped with auto sms, which sent notification to the victim’s relatives, that was also integrated with Google Maps application (GMaps) as the map to the victim’s location. The results show that voice registration on the application reaches 100%, incident detection using speech recognition while the application is running is 94.67% in average, and the auto sms to the victim relatives reaches 100%.

  2. Chromosomal aberrations in chronic lymphocytic leukemia detected by conventional cytogenetics with DSP30 as a single agent: comparison with FISH.

    Science.gov (United States)

    Kotkowska, Aleksandra; Wawrzyniak, Ewa; Blonski, Jerzy Z; Robak, Tadeusz; Korycka-Wolowiec, Anna

    2011-08-01

    The aim of our study was to estimate the usefulness for conventional cytogenetics (CC) of DSP30 as a single agent (CC-DSP30) for detecting the most important chromosomal aberrations revealed in CLL by FISH and to find other abnormalities possibly existing but undetected by FISH with standard probes. Using CC-DSP30, the metaphases suitable for analysis were obtained in 90% of patients. CC-DSP30 and FISH were similarly efficacious for detecting del(11)(q22) and trisomy 12, whereas FISH was more sensitive for del(13)(q14). Sole del(13)(q14) detected by FISH, in 50% of patients was associated with other aberrations revealed by CC-DSP30. Additionally, the most recurrent anomaly detected by CC-DSP30 were structural aberrations of chromosome 2. Copyright © 2011 Elsevier Ltd. All rights reserved.

  3. Practical considerations for the implantation of a fuzzy control algorithm in a DSP; Consideraciones practicas para la implantacion de un algoritmo de control difuso en un DSP

    Energy Technology Data Exchange (ETDEWEB)

    Perez C, B.; Benitez R, J.S.; Pacheco S, J.O. [ININ, 52045 Ocoyoacac, Estado de Mexico (Mexico)

    2003-07-01

    The development of a digital system based on a DSP to implant a Mamdani type algorithm of fuzzy control whose objective is to regulate the neutron power in a nuclear research reactor Type TRIGA Mark III is presented. Its are simultaneously carried out the aggregation des fuzzy stages discreeting the universe of the output variable. The format MPF for the handling of the floating point in the arithmetic operations is used. (Author)

  4. Intra-oral pressure-based voicing control of electrolaryngeal speech with intra-oral vibrator.

    Science.gov (United States)

    Takahashi, Hirokazu; Nakao, Masayuki; Kikuchi, Yataro; Kaga, Kimitaka

    2008-07-01

    In normal speech, coordinated activities of intrinsic laryngeal muscles suspend a glottal sound at utterance of voiceless consonants, automatically realizing a voicing control. In electrolaryngeal speech, however, the lack of voicing control is one of the causes of unclear voice, voiceless consonants tending to be misheard as the corresponding voiced consonants. In the present work, we developed an intra-oral vibrator with an intra-oral pressure sensor that detected utterance of voiceless phonemes during the intra-oral electrolaryngeal speech, and demonstrated that an intra-oral pressure-based voicing control could improve the intelligibility of the speech. The test voices were obtained from one electrolaryngeal speaker and one normal speaker. We first investigated on the speech analysis software how a voice onset time (VOT) and first formant (F1) transition of the test consonant-vowel syllables contributed to voiceless/voiced contrasts, and developed an adequate voicing control strategy. We then compared the intelligibility of consonant-vowel syllables among the intra-oral electrolaryngeal speech with and without online voicing control. The increase of intra-oral pressure, typically with a peak ranging from 10 to 50 gf/cm2, could reliably identify utterance of voiceless consonants. The speech analysis and intelligibility test then demonstrated that a short VOT caused the misidentification of the voiced consonants due to a clear F1 transition. Finally, taking these results together, the online voicing control, which suspended the prosthetic tone while the intra-oral pressure exceeded 2.5 gf/cm2 and during the 35 milliseconds that followed, proved efficient to improve the voiceless/voiced contrast.

  5. Interference and protection of electromagnetic pulse to digital signal processor

    International Nuclear Information System (INIS)

    Wang Yan; Jiao Hongling; He Shanhong; Pan Chao; Feng Deren; Che Wenquan; Xiong Ying

    2013-01-01

    The effective electromagnetic pulse protection is studied in this paper, first the interference of electromagnetic pulse simulator path is analyzed, including the digital signal processor (DSP) and the discharge circuit of coupling interference and net electricity coupling interference. Using the structure optimization design, the hardware block reinforcement measurement and the setting of open software trap, and the watchdog anti-jamming measures, the interference test is completed such as the central processor core voltage of DSP, input/output (I/O) ports of DSP and the display screen. The experimental results show that the combination of hardware and software protection reinforcement technology is effective, and the interference pulse amplitude of DSP board I/O port and the kernel work voltage are reduced, and the interference duration is reduced from 2 μs to 400 ns. The interference pulse is effectively restrained. (authors)

  6. High precision locating control system based on VCM for Talbot lithography

    Science.gov (United States)

    Yao, Jingwei; Zhao, Lixin; Deng, Qian; Hu, Song

    2016-10-01

    Aiming at the high precision and efficiency requirements of Z-direction locating in Talbot lithography, a control system based on Voice Coil Motor (VCM) was designed. In this paper, we built a math model of VCM and its moving characteristic was analyzed. A double-closed loop control strategy including position loop and current loop were accomplished. The current loop was implemented by driver, in order to achieve the rapid follow of the system current. The position loop was completed by the digital signal processor (DSP) and the position feedback was achieved by high precision linear scales. Feed forward control and position feedback Proportion Integration Differentiation (PID) control were applied in order to compensate for dynamic lag and improve the response speed of the system. And the high precision and efficiency of the system were verified by simulation and experiments. The results demonstrated that the performance of Z-direction gantry was obviously improved, having high precision, quick responses, strong real-time and easily to expend for higher precision.

  7. TMS320C25 Digital Signal Processor For 2-Dimensional Fast Fourier Transform Computation

    International Nuclear Information System (INIS)

    Ardisasmita, M. Syamsa

    1996-01-01

    The Fourier transform is one of the most important mathematical tool in signal processing and analysis, which converts information from the time/spatial domain into the frequency domain. Even with implementation of the Fast Fourier Transform algorithms in imaging data, the discrete Fourier transform execution consume a lot of time. Digital signal processors are designed specifically to perform computation intensive digital signal processing algorithms. By taking advantage of the advanced architecture. parallel processing, and dedicated digital signal processing (DSP) instruction sets. This device can execute million of DSP operations per second. The device architecture, characteristics and feature suitable for fast Fourier transform application and speed-up are discussed

  8. A posteriori error estimates in voice source recovery

    Science.gov (United States)

    Leonov, A. S.; Sorokin, V. N.

    2017-12-01

    The inverse problem of voice source pulse recovery from a segment of a speech signal is under consideration. A special mathematical model is used for the solution that relates these quantities. A variational method of solving inverse problem of voice source recovery for a new parametric class of sources, that is for piecewise-linear sources (PWL-sources), is proposed. Also, a technique for a posteriori numerical error estimation for obtained solutions is presented. A computer study of the adequacy of adopted speech production model with PWL-sources is performed in solving the inverse problems for various types of voice signals, as well as corresponding study of a posteriori error estimates. Numerical experiments for speech signals show satisfactory properties of proposed a posteriori error estimates, which represent the upper bounds of possible errors in solving the inverse problem. The estimate of the most probable error in determining the source-pulse shapes is about 7-8% for the investigated speech material. It is noted that a posteriori error estimates can be used as a criterion of the quality for obtained voice source pulses in application to speaker recognition.

  9. Voice application development for Android

    CERN Document Server

    McTear, Michael

    2013-01-01

    This book will give beginners an introduction to building voice-based applications on Android. It will begin by covering the basic concepts and will build up to creating a voice-based personal assistant. By the end of this book, you should be in a position to create your own voice-based applications on Android from scratch in next to no time.Voice Application Development for Android is for all those who are interested in speech technology and for those who, as owners of Android devices, are keen to experiment with developing voice apps for their devices. It will also be useful as a starting po

  10. Writing with Voice

    Science.gov (United States)

    Kesler, Ted

    2012-01-01

    In this Teaching Tips article, the author argues for a dialogic conception of voice, based in the work of Mikhail Bakhtin. He demonstrates a dialogic view of voice in action, using two writing examples about the same topic from his daughter, a fifth-grade student. He then provides five practical tips for teaching a dialogic conception of voice in…

  11. Evolving Spiking Neural Networks for Recognition of Aged Voices.

    Science.gov (United States)

    Silva, Marco; Vellasco, Marley M B R; Cataldo, Edson

    2017-01-01

    The aging of the voice, known as presbyphonia, is a natural process that can cause great change in vocal quality of the individual. This is a relevant problem to those people who use their voices professionally, and its early identification can help determine a suitable treatment to avoid its progress or even to eliminate the problem. This work focuses on the development of a new model for the identification of aging voices (independently of their chronological age), using as input attributes parameters extracted from the voice and glottal signals. The proposed model, named Quantum binary-real evolving Spiking Neural Network (QbrSNN), is based on spiking neural networks (SNNs), with an unsupervised training algorithm, and a Quantum-Inspired Evolutionary Algorithm that automatically determines the most relevant attributes and the optimal parameters that configure the SNN. The QbrSNN model was evaluated in a database composed of 120 records, containing samples from three groups of speakers. The results obtained indicate that the proposed model provides better accuracy than other approaches, with fewer input attributes. Copyright © 2017 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

  12. Comparison of two different high performance mixed signal controllers for DC/DC converters

    DEFF Research Database (Denmark)

    Jakobsen, Lars Tønnes; Andersen, Michael Andreas E.

    2006-01-01

    This paper describes how mixed signal controllers combining a cheap microcontroller with a simple analogue circuit can offer high performance digital control for DC/DC converters. Mixed signal controllers have the same versatility and performance as DSP based controllers. It is important to have...... an engineer experienced in microcontroller programming write the software algorithms to achieve optimal performance. Two mixed signal controller designs based on the same 8-bit microcontroller are compared both theoretically and experimentally. A 16-bit PID compensator with a sampling frequency of 200 k......Hz implemented in the 16 MIPS, 8-bit ATTiny26 microcontroller is demonstrated....

  13. Low power digital signal processing

    DEFF Research Database (Denmark)

    Paker, Ozgun

    2003-01-01

    hardwired ASICs and more than 6 21 times lower than current state of the art low-power DSP processors. An orthogonal but practical contribution of this thesis is the test bench implementation. A PCI-based FPGA board has been used to equip a standard desktop PC with tester facilities. The test bench proved...... to be a viable alternative to conventional expensive test equipment. Finally, the work presented in this thesis has been published at several IEEE workshops and conferences, and in the Journal of VLSI Signal Processing....

  14. The Chameleon Architecture for Streaming DSP Applications

    Directory of Open Access Journals (Sweden)

    André B. J. Kokkeler

    2007-02-01

    Full Text Available We focus on architectures for streaming DSP applications such as wireless baseband processing and image processing. We aim at a single generic architecture that is capable of dealing with different DSP applications. This architecture has to be energy efficient and fault tolerant. We introduce a heterogeneous tiled architecture and present the details of a domain-specific reconfigurable tile processor called Montium. This reconfigurable processor has a small footprint (1.8 mm2 in a 130 nm process, is power efficient and exploits the locality of reference principle. Reconfiguring the device is very fast, for example, loading the coefficients for a 200 tap FIR filter is done within 80 clock cycles. The tiles on the tiled architecture are connected to a Network-on-Chip (NoC via a network interface (NI. Two NoCs have been developed: a packet-switched and a circuit-switched version. Both provide two types of services: guaranteed throughput (GT and best effort (BE. For both NoCs estimates of power consumption are presented. The NI synchronizes data transfers, configures and starts/stops the tile processor. For dynamically mapping applications onto the tiled architecture, we introduce a run-time mapping tool.

  15. The Chameleon Architecture for Streaming DSP Applications

    Directory of Open Access Journals (Sweden)

    Heysters PaulM

    2007-01-01

    Full Text Available We focus on architectures for streaming DSP applications such as wireless baseband processing and image processing. We aim at a single generic architecture that is capable of dealing with different DSP applications. This architecture has to be energy efficient and fault tolerant. We introduce a heterogeneous tiled architecture and present the details of a domain-specific reconfigurable tile processor called Montium. This reconfigurable processor has a small footprint (1.8 mm2 in a 130 nm process, is power efficient and exploits the locality of reference principle. Reconfiguring the device is very fast, for example, loading the coefficients for a 200 tap FIR filter is done within 80 clock cycles. The tiles on the tiled architecture are connected to a Network-on-Chip (NoC via a network interface (NI. Two NoCs have been developed: a packet-switched and a circuit-switched version. Both provide two types of services: guaranteed throughput (GT and best effort (BE. For both NoCs estimates of power consumption are presented. The NI synchronizes data transfers, configures and starts/stops the tile processor. For dynamically mapping applications onto the tiled architecture, we introduce a run-time mapping tool.

  16. FGF-2 potently induces both proliferation and DSP expression in collagen type I gel cultures of adult incisor immature pulp cells

    International Nuclear Information System (INIS)

    Nakao, Kazuhisa; Itoh, Makoto; Tomita, Yusuke; Tomooka, Yasuhiro; Tsuji, Takashi

    2004-01-01

    We investigated the effects of both cytokines and extracellular matrices on the proliferation and differentiation of immature adult rat incisor dental pulp cells. These immature cells, which have a high-proliferative potency in vitro and do not express mRNAs for dentin non-collagenous proteins such as dentin sialoprotein (DSP), bone sialoprotein (BSP), and osteocalcin, exist in the root regions of adult rat incisors. Fibroblast growth factor-2 (FGF-2) stimulated the proliferation of these immature cells and the subsequent production of mineralized calcium was induced by β-glycerophosphate treatment. Additionally, FGF-2 dramatically induced the expression of DSP and BSP mRNAs, but only in collagen type I gel cultures, whereas neither plate-coated collagen type I nor fibronectin, laminin or collagen type IV cultures could produce this effect and generate sufficient physiological levels of these transcripts. Although bone morphogenetic protein-4 could not induce the proliferation of immature dental pulp cells nor upregulate DSP mRNA expression, it had a synergistic effect upon DSP transcript levels in conjunction with FGF-2. These results suggest that both the presence of FGF-2 and the three-dimensional formation of immature dental pulp cells in collagen type I gel cultures are essential for both DSP expression and odontoblast differentiation. These observations provide valuable information concerning the study of the commitment and differentiation of odontoblast lineages, and also provide a basis for the rational design of cytokine and extracellular matrix based compounds for regenerative therapies in new dental treatments

  17. 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.

  18. Face the voice

    DEFF Research Database (Denmark)

    Lønstrup, Ansa

    2014-01-01

    will be based on a reception aesthetic and phenomenological approach, the latter as presented by Don Ihde in his book Listening and Voice. Phenomenologies of Sound , and my analytical sketches will be related to theoretical statements concerning the understanding of voice and media (Cavarero, Dolar, La......Belle, Neumark). Finally, the article will discuss the specific artistic combination and our auditory experience of mediated human voices and sculpturally projected faces in an art museum context under the general conditions of the societal panophonia of disembodied and mediated voices, as promoted by Steven...

  19. Singing voice outcomes following singing voice therapy.

    Science.gov (United States)

    Dastolfo-Hromack, Christina; Thomas, Tracey L; Rosen, Clark A; Gartner-Schmidt, Jackie

    2016-11-01

    The objectives of this study were to describe singing voice therapy (SVT), describe referred patient characteristics, and document the outcomes of SVT. Retrospective. Records of patients receiving SVT between June 2008 and June 2013 were reviewed (n = 51). All diagnoses were included. Demographic information, number of SVT sessions, and symptom severity were retrieved from the medical record. Symptom severity was measured via the 10-item Singing Voice Handicap Index (SVHI-10). Treatment outcome was analyzed by diagnosis, history of previous training, and SVHI-10. SVHI-10 scores decreased following SVT (mean change = 11, 40% decrease) (P singing lessons (n = 10) also completed an average of three SVT sessions. Primary muscle tension dysphonia (MTD1) and benign vocal fold lesion (lesion) were the most common diagnoses. Most patients (60%) had previous vocal training. SVHI-10 decrease was not significantly different between MTD and lesion. This is the first outcome-based study of SVT in a disordered population. Diagnosis of MTD or lesion did not influence treatment outcomes. Duration of SVT was short (approximately three sessions). Voice care providers are encouraged to partner with a singing voice therapist to provide optimal care for the singing voice. This study supports the use of SVT as a tool for the treatment of singing voice disorders. 4 Laryngoscope, 126:2546-2551, 2016. © 2016 The American Laryngological, Rhinological and Otological Society, Inc.

  20. Individual versus Interactive Task-Based Performance through Voice-Based Computer-Mediated Communication

    Science.gov (United States)

    Granena, Gisela

    2016-01-01

    Interaction is a necessary condition for second language (L2) learning (Long, 1980, 1996). Research in computer-mediated communication has shown that interaction opportunities make learners pay attention to form in a variety of ways that promote L2 learning. This research has mostly investigated text-based rather than voice-based interaction. The…

  1. A Voice-Based E-Examination Framework for Visually Impaired Students in Open and Distance Learning

    Science.gov (United States)

    Azeta, Ambrose A.; Inam, Itorobong A.; Daramola, Olawande

    2018-01-01

    Voice-based systems allow users access to information on the internet over a voice interface. Prior studies on Open and Distance Learning (ODL) e-examination systems that make use of voice interface do not sufficiently exhibit intelligent form of assessment, which diminishes the rigor of examination. The objective of this paper is to improve on…

  2. Comparison of Medical and Voice Therapy for reflux Laryngitis Based on Acoustic and Laryngeal Characteristics

    Directory of Open Access Journals (Sweden)

    Abbas Dehestani Ardakani

    2011-12-01

    Full Text Available Background and Aim: Reflux laryngitis is extremely common among patients with voice disorder. Medical therapy approaches are not efficient enough. The main goal of this study is to assess the acoustic and laryngeal characteristics of patients with dysphonia before and after medical or voice therapy, and to evaluate the effectiveness of each.Methods: In this retrospective study, 16 reflux laryngitis patients were assessed. Five received complete voice therapy, tow ceased voice therapy and nine received medication. Perceptual voice evaluation was performed by a speech-language pathologist, the severity of voice problem was calculated, based on the affected acoustic and laryngeal characteristics pre- and post-treatment.Results: Post-treatment evaluation in patients who received complete voice therapy indicates 80 percent improvement in the severity of disorder and 100 percent improvement in the perceptual voice evaluation. After medical therapy, we observed that voice disorder and perceptual voice evaluation are improved 44 and 66 percent respectively. The improvement was statistically significant in both treatment approaches: complete voice therapy (P=0.039 and medical therapy (p=0.017.Conclusion: In patients with reflux laryngitis, most acoustic and laryngeal characteristics were normal and satisfying after the treatment. It can be concluded that the proficiency of voice therapy in improving the acoustic and laryngeal characteristics is comparable to medical therapy

  3. Input/output Buffer based Vedic Multiplier Design for Thermal Aware Energy Efficient Digital Signal Processing on 28nm FPGA

    DEFF Research Database (Denmark)

    Goswami, Kavita; Pandey, Bishwajeet; Hussain, Dil muhammed Akbar

    2016-01-01

    Multiplier is used for multiplication of a signal and a constant in digital signal processing (DSP). 28nm technology based Vedic multiplier is implemented with use of VHDL HDL, Xilinx ISE, Kintex-7 FPGA and XPower Analyzer. Vedic multiplier gain speed improvements by parallelizing the generation...... Programmable Gate Array (FPGA) in order to reduce the development cost. The development cost for Application Specific Integrated Circuits (ASICs) are high in compare to FPGA. Selection of the most energy efficient IO standards in place of signal gating is the main design methodology for design of energy...... efficient Vedic multiplier.There is 68.51%, 69.86%, 74.65%, and 78.39% contraction in total power of Vedic multiplier on 28nm Kintex-7 FPGA, when we use HSTL_II in place of HSTL_II_DCI_18 at 56.7oC, 53.5oC, 40oC and 21oC respectively....

  4. Design and realization of intelligent tourism service system based on voice interaction

    Science.gov (United States)

    Hu, Lei-di; Long, Yi; Qian, Cheng-yang; Zhang, Ling; Lv, Guo-nian

    2008-10-01

    Voice technology is one of the important contents to improve the intelligence and humanization of tourism service system. Combining voice technology, the paper concentrates on application needs and the composition of system to present an overall intelligent tourism service system's framework consisting of presentation layer, Web services layer, and tourism application service layer. On the basis, the paper further elaborated the implementation of the system and its key technologies, including intelligent voice interactive technology, seamless integration technology of multiple data sources, location-perception-based guides' services technology, and tourism safety control technology. Finally, according to the situation of Nanjing tourism, a prototype of Tourism Services System is realized.

  5. DspA/E contributes to apoplastic accumulation of ROS in nonhost A. thaliana

    Directory of Open Access Journals (Sweden)

    Alban eLaunay

    2016-04-01

    Full Text Available The bacterium Erwinia amylovora is responsible for the fire blight disease of Maleae, which provokes necrotic symptoms on aerial parts. The pathogenicity of this bacterium in hosts relies on its type three-secretion system (T3SS, a molecular syringe that allows the bacterium to inject effectors into the plant cell. E. amylovora-triggered disease in host plants is associated with the T3SS-dependent production of reactive oxygen species (ROS, although ROS are generally associated with resistance in other pathosystems. We showed previously that E. amylovora can multiply transiently in the nonhost plant Arabidopsis thaliana and that a T3SS-dependent production of intracellular ROS occurs during this interaction. In the present work we characterize the localization and source of hydrogen peroxide accumulation following E. amylovora infection. Transmission electron microscope (TEM analysis of infected tissues showed that hydrogen peroxide accumulation occurs in the cytosol, plastids, peroxisomes, and mitochondria as well as in the apoplast. Furthermore, TEM analysis showed that an E. amylovora dspA/E-deficient strain does not induce hydrogen peroxide accumulation in the apoplast. Consistently, a transgenic line expressing DspA/E accumulated ROS in the apoplast. The NADPH oxidase-deficient rbohD mutant showed a very strong reduction in hydrogen peroxide accumulation in response to E. amylovora inoculation. However, we did not find an increase in bacterial titers of E. amylovora in the rbohD mutant and the rbohD mutation did not suppress the toxicity of DspA/E when introgressed into a DspA/E-expressing transgenic line. Co-inoculation of E. amylovora with cycloheximide (CHX, which we found previously to suppress callose deposition and allow strong multiplication of E. amylovora in A. thaliana leaves, led to a strong reduction of apoplastic ROS accumulation but did not affect intracellular ROS. Our data strongly suggest that apoplastic ROS accumulation is

  6. A digital-signal-processor-based optical tomographic system for dynamic imaging of joint diseases

    Science.gov (United States)

    Lasker, Joseph M.

    Over the last decade, optical tomography (OT) has emerged as viable biomedical imaging modality. Various imaging systems have been developed that are employed in preclinical as well as clinical studies, mostly targeting breast imaging, brain imaging, and cancer related studies. Of particular interest are so-called dynamic imaging studies where one attempts to image changes in optical properties and/or physiological parameters as they occur during a system perturbation. To successfully perform dynamic imaging studies, great effort is put towards system development that offers increasingly enhanced signal-to-noise performance at ever shorter data acquisition times, thus capturing high fidelity tomographic data within narrower time periods. Towards this goal, I have developed in this thesis a dynamic optical tomography system that is, unlike currently available analog instrumentation, based on digital data acquisition and filtering techniques. At the core of this instrument is a digital signal processor (DSP) that collects, collates, and processes the digitized data set. Complementary protocols between the DSP and a complex programmable logic device synchronizes the sampling process and organizes data flow. Instrument control is implemented through a comprehensive graphical user interface which integrates automated calibration, data acquisition, and signal post-processing. Real-time data is generated at frame rates as high as 140 Hz. An extensive dynamic range (˜190 dB) accommodates a wide scope of measurement geometries and tissue types. Performance analysis demonstrates very low system noise (˜1 pW rms noise equivalent power), excellent signal precision (˜0.04%--0.2%) and long term system stability (˜1% over 40 min). Experiments on tissue phantoms validate spatial and temporal accuracy of the system. As a potential new application of dynamic optical imaging I present the first application of this method to use vascular hemodynamics as a means of characterizing

  7. Optimizing optical pre-dispersion using transmit DSP for mitigation of Kerr nonlinearities in dispersion managed cables

    Science.gov (United States)

    Hopkins, James; Gaudette, Jamie; Mehta, Priyanth

    2013-10-01

    With the advent of digital signal processing (DSP) in optical transmitters and receivers, the ability to finely tune the ratio of pre and post dispersion compensation can be exploited to best mitigate the nonlinear penalties caused by the Kerr effect. A portion of the nonlinear penalty in optical communication channels has been explained by an increase in peak to average power ratio (PAPR) inherent in highly dispersed signals. The standard approach for minimizing these impairments applies 50% pre dispersion compensation and 50% post dispersion compensation, thereby decreasing average PAPR along the length of the cable, as compared with either 100% pre or post dispersion compensation. In this paper we demonstrate that simply considering the net accumulated dispersion, and applying 50/50 pre/post dispersion is not necessarily the best way to minimize PAPR and subsequent Kerr nonlinearities. Instead, we consider the cumulative dispersion along the entire length of the cable, and, taking into account this additional information, derive an analytic formula for the minimization of PAPR. Alignment with simulation and experimental measurements is presented using a commercially available 100Gb/s dual-polarization binary phase-shift-keying (DP-BPSK) coherent modem, with transmitter and receiver DSP. Measurements are provided from two different 5000km dispersion managed Submarine test-beds, as well as a 3800km terrestrial test-bed with a mixture of SMF-28 and TWRS optical fiber. This method is shown to deviate significantly from the conventional 50/50 method described above, in dispersion managed communications systems, and more closely aligns with results obtained from simulation and data collected from laboratory test-beds.

  8. Practical considerations for the implantation of a fuzzy control algorithm in a DSP

    International Nuclear Information System (INIS)

    Perez C, B.; Benitez R, J.S.; Pacheco S, J.O.

    2003-01-01

    The development of a digital system based on a DSP to implant a Mamdani type algorithm of fuzzy control whose objective is to regulate the neutron power in a nuclear research reactor Type TRIGA Mark III is presented. Its are simultaneously carried out the aggregation des fuzzy stages discreeting the universe of the output variable. The format MPF for the handling of the floating point in the arithmetic operations is used. (Author)

  9. Pin level neutronic - thermal hydraulic two-way-coupling using DYN3D-SP3 and SUBCHANFLOW

    International Nuclear Information System (INIS)

    Torres, Armando Gomez; Espinoza, Victor Sanchez; Imke, Uwe; Juan, Rafael Macian

    2011-01-01

    Nowadays several Reactor Dynamic Codes, (RDC) are able to solve the diffusion equation or even the transport equation (SP3 approximation) considering feedback parameters coming from the thermalhydraulic (TH) core behavior. These kinds of codes (DYN3D, PARCS, among others) usually contain a 1D two phase flow thermalhydraulic model capable to pass them assembly averaged feedback parameters. At fuel assembly base this nodal coupling is completely a two way coupling. The Neutronic part calculates the mean power of the whole assembly and passes it to the TH part in order to actualize the heat source. In turn, the TH model passes the assembly-based feedback parameters to the neutronic code for actualizing the nodal cross sections. The process will be repeated until convergence. At pin level, the current situation is somehow different. Although the neutronic solver can pass the pin power distribution in every sub - node (pin distribution), the 1-D TH model will average the pin power distribution to assembly-based scale and will give back assembly averaged feedbacks to the neutronic part for cross sections up-date (one and a half way coupling), leading to information loss in the calculation. A new coupled program system DYNSUB was developed by coupling DYN3D-SP3 and SUBCHANFLOW at pin level. DYNSUB was used to analyze stationary PWR minicore problems at pin-level. The comparison of the Keff predicted by DYNSUB with the one calculated by DYN3D-SP3 (coarse TH solution) shows small differences of up to 26 pcm. Differences up to 4.5% were found in the radial distribution of the pin power. The local safety parameters such as cladding and fuel temperature predicted with DYNSUB shows larger deviations compared with the ones obtained with DYN3D-SP3. These differences may increase when analyzing transients. (author)

  10. Linear and Nonlinear Impairment Compensation in Coherent Optical Transmission with Digital Signal Processing

    DEFF Research Database (Denmark)

    Porto da Silva, Edson

    Digital signal processing (DSP) has become one of the main enabling technologies for the physical layer of coherent optical communication networks. The DSP subsystems are used to implement several functionalities in the digital domain, from synchronization to channel equalization. Flexibility...... nonlinearity compensation, (II) spectral shaping, and (III) adaptive equalization. For (I), original contributions are presented to the study of the nonlinearity compensation (NLC) with digital backpropagation (DBP). Numerical and experimental performance investigations are shown for different application...... scenarios. Concerning (II), it is demonstrated how optical and electrical (digital) pulse shaping can be allied to improve the spectral confinement of a particular class of optical time-division multiplexing (OTDM) signals that can be used as a building block for fast signaling single-carrier transceivers...

  11. A self-teaching image processing and voice-recognition-based, intelligent and interactive system to educate visually impaired children

    Science.gov (United States)

    Iqbal, Asim; Farooq, Umar; Mahmood, Hassan; Asad, Muhammad Usman; Khan, Akrama; Atiq, Hafiz Muhammad

    2010-02-01

    A self teaching image processing and voice recognition based system is developed to educate visually impaired children, chiefly in their primary education. System comprises of a computer, a vision camera, an ear speaker and a microphone. Camera, attached with the computer system is mounted on the ceiling opposite (on the required angle) to the desk on which the book is placed. Sample images and voices in the form of instructions and commands of English, Urdu alphabets, Numeric Digits, Operators and Shapes are already stored in the database. A blind child first reads the embossed character (object) with the help of fingers than he speaks the answer, name of the character, shape etc into the microphone. With the voice command of a blind child received by the microphone, image is taken by the camera which is processed by MATLAB® program developed with the help of Image Acquisition and Image processing toolbox and generates a response or required set of instructions to child via ear speaker, resulting in self education of a visually impaired child. Speech recognition program is also developed in MATLAB® with the help of Data Acquisition and Signal Processing toolbox which records and process the command of the blind child.

  12. Enhancement and Noise Statistics Estimation for Non-Stationary Voiced Speech

    DEFF Research Database (Denmark)

    Nørholm, Sidsel Marie; Jensen, Jesper Rindom; Christensen, Mads Græsbøll

    2016-01-01

    In this paper, single channel speech enhancement in the time domain is considered. We address the problem of modelling non-stationary speech by describing the voiced speech parts by a harmonic linear chirp model instead of using the traditional harmonic model. This means that the speech signal...... through simulations on synthetic and speech signals, that the chirp versions of the filters perform better than their harmonic counterparts in terms of output signal-to-noise ratio (SNR) and signal reduction factor. For synthetic signals, the output SNR for the harmonic chirp APES based filter...... is increased 3 dB compared to the harmonic APES based filter at an input SNR of 10 dB, and at the same time the signal reduction factor is decreased. For speech signals, the increase is 1.5 dB along with a decrease in the signal reduction factor of 0.7. As an implicit part of the APES filter, a noise...

  13. Connections between voice ergonomic risk factors and voice symptoms, voice handicap, and respiratory tract diseases.

    Science.gov (United States)

    Rantala, Leena M; Hakala, Suvi J; Holmqvist, Sofia; Sala, Eeva

    2012-11-01

    The aim of the study was to investigate the connections between voice ergonomic risk factors found in classrooms and voice-related problems in teachers. Voice ergonomic assessment was performed in 39 classrooms in 14 elementary schools by means of a Voice Ergonomic Assessment in Work Environment--Handbook and Checklist. The voice ergonomic risk factors assessed included working culture, noise, indoor air quality, working posture, stress, and access to a sound amplifier. Teachers from the above-mentioned classrooms reported their voice symptoms, respiratory tract diseases, and completed a Voice Handicap Index (VHI). The more voice ergonomic risk factors found in the classroom the higher were the teachers' total scores on voice symptoms and VHI. Stress was the factor that correlated most strongly with voice symptoms. Poor indoor air quality increased the occurrence of laryngitis. Voice ergonomics were poor in the classrooms studied and voice ergonomic risk factors affected the voice. It is important to convey information on voice ergonomics to education administrators and those responsible for school planning and taking care of school buildings. Copyright © 2012 The Voice Foundation. Published by Mosby, Inc. All rights reserved.

  14. Effects of methylphenidate on attention in Wistar rats treated with the neurotoxin N-(2-chloroethyl)-N-ethyl-2-bromobenzylamine (DSP4).

    Science.gov (United States)

    Hauser, Joachim; Reissmann, Andreas; Sontag, Thomas-A; Tucha, Oliver; Lange, Klaus W

    2017-05-01

    The aim of this study was to assess the effects of the neurotoxin N-(2-chloroethyl)-N-ethyl-2-bromobenzylamine (DSP4) on attention in rats as measured using the 5-choice-serial-reaction-time task (5CSRTT) and to investigate whether methylphenidate has effects on DSP4-treated rats. Methylphenidate is a noradrenaline and dopamine reuptake inhibitor and commonly used in the pharmacological treatment of individuals with attention deficit/hyperactivity disorder (ADHD). Wistar rats were trained in the 5CSRTT and treated with one of three doses of DSP4 or saline. Following the DSP4 treatment rats were injected with three doses of methylphenidate or saline and again tested in the 5CSRTT. The treatment with DSP4 caused a significant decline of performance in the number of correct responses and a decrease in response accuracy. A reduction in activity could also be observed. Whether or not the cognitive impairments are due to attention deficits or changes in explorative behaviour or activity remains to be investigated. The treatment with methylphenidate had no beneficial effect on the rats' performance regardless of the DSP4 treatment. In the group without DSP4 treatment, methylphenidate led to a reduction in response accuracy and bidirectional effects in regard to parameters related to attention. These findings support the role of noradrenaline in modulating attention and call for further investigations concerning the effects of methylphenidate on attentional processes in rats.

  15. Multi-Frame Rate Based Multiple-Model Training for Robust Speaker Identification of Disguised Voice

    DEFF Research Database (Denmark)

    Prasad, Swati; Tan, Zheng-Hua; Prasad, Ramjee

    2013-01-01

    Speaker identification systems are prone to attack when voice disguise is adopted by the user. To address this issue,our paper studies the effect of using different frame rates on the accuracy of the speaker identification system for disguised voice.In addition, a multi-frame rate based multiple......-model training method is proposed. The experimental results show the superior performance of the proposed method compared to the commonly used single frame rate method for three types of disguised voice taken from the CHAINS corpus....

  16. Advanced digital signal processing for short haul optical fiber transmission beyond 100G

    Science.gov (United States)

    Kikuchi, Nobuhiko

    2017-01-01

    Significant increase of intra and inter data center traffic has been expected by the rapid spread of various network applications like SNS, IoT, mobile and cloud computing, and the needs for ultra-high speed and cost-effective short- to medium-reach optical fiber links beyond 100-Gbit/s is becoming larger and larger. Such high-speed links typically use multilevel modulation to lower signaling speed, which in turn face serious challenges in limited loss budget and waveform distortion tolerance. One of the promising techniques to overcome them is the use of advanced digital signal processing (DSP) and we review various DSP applications for short-to-medium reach applications.

  17. Validation of the MCNP-DSP Monte Carlo code for calculating source-driven noise parameters of subcritical systems

    International Nuclear Information System (INIS)

    Valentine, T.E.; Mihalczo, J.T.

    1995-01-01

    This paper describes calculations performed to validate the modified version of the MCNP code, the MCNP-DSP, used for: the neutron and photon spectra of the spontaneous fission of californium 252; the representation of the detection processes for scattering detectors; the timing of the detection process; and the calculation of the frequency analysis parameters for the MCNP-DSP code

  18. Digital signal processing an experimental approach

    CERN Document Server

    Engelberg, Shlomo

    2008-01-01

    Digital Signal Processing is a mathematically rigorous but accessible treatment of digital signal processing that intertwines basic theoretical techniques with hands-on laboratory instruction. Divided into three parts, the book covers various aspects of the digital signal processing (DSP) ""problem."" It begins with the analysis of discrete-time signals and explains sampling and the use of the discrete and fast Fourier transforms. The second part of the book???covering digital to analog and analog to digital conversion???provides a practical interlude in the mathematical content before Part II

  19. Multivariate sensitivity to voice during auditory categorization.

    Science.gov (United States)

    Lee, Yune Sang; Peelle, Jonathan E; Kraemer, David; Lloyd, Samuel; Granger, Richard

    2015-09-01

    Past neuroimaging studies have documented discrete regions of human temporal cortex that are more strongly activated by conspecific voice sounds than by nonvoice sounds. However, the mechanisms underlying this voice sensitivity remain unclear. In the present functional MRI study, we took a novel approach to examining voice sensitivity, in which we applied a signal detection paradigm to the assessment of multivariate pattern classification among several living and nonliving categories of auditory stimuli. Within this framework, voice sensitivity can be interpreted as a distinct neural representation of brain activity that correctly distinguishes human vocalizations from other auditory object categories. Across a series of auditory categorization tests, we found that bilateral superior and middle temporal cortex consistently exhibited robust sensitivity to human vocal sounds. Although the strongest categorization was in distinguishing human voice from other categories, subsets of these regions were also able to distinguish reliably between nonhuman categories, suggesting a general role in auditory object categorization. Our findings complement the current evidence of cortical sensitivity to human vocal sounds by revealing that the greatest sensitivity during categorization tasks is devoted to distinguishing voice from nonvoice categories within human temporal cortex. Copyright © 2015 the American Physiological Society.

  20. A Novel Chaos-Based Voice Controlled FTP Tool

    Directory of Open Access Journals (Sweden)

    Muhammed Maruf Ozturk

    2015-08-01

    Full Text Available To manage file transfer operation various tools have been developed so far. However these tools can not respond adequately for conduct a secure transfer. Also few works have been done using encrypted voice controlled system yet. By regarding this lack we investigate how to built a useful and secure tool. This work presents a novel improved voice controlled FTP tool Wb-CFTP using chaotic system. A chaotic system called as logistic map is associated with Wb-FTP designed on the basis of Asp.Net and C. Here we depict the prominence of encryption in voice controlled systems.

  1. Modular version of SIMCON, FPGA based, DSP integrated, LLRF control system for TESLA FEL part II: measurement of SIMCON 3.0 DSP daughterboard

    Science.gov (United States)

    Giergusiewicz, Wojciech; Koprek, Waldemar; Jalmuzna, Wojciech; Pozniak, Krzysztof T.; Romaniuk, Ryszard S.

    2006-02-01

    The paper describes design, construction and initial measurements of an eight channel electronic LLRF device predicted for building of the control system for the W-FEL accelerator at DESY (Hamburg). The device, referred in the paper to as the SIMCON 3.0 (from the SC cavity simulator and controller) consists of a 16 layer, VME size, PCB, a large FPGA chip (VirtexII-4000 by Xilinx), eight fast ADCs and four DACs (by Analog Devices). To our knowledge, the proposed device is the first of this kind for the accelerator technology in which there was achieved (the FPGA based) DSP latency below 200 ns. With the optimized data transmission system, the overall LLRF system latency can be as low as 500 ns. The SIMCON 3.0 sub-system was applied for initial tests with the ACCl module of the VUV FEL accelerator (eight channels) and with the CHECHIA test stand (single channel), both at the DESY. The promising results with the SIMCON 3.0. encouraged us to enter the design of SIMCON 3.1. possessing 10 measurement and control channels and some additional features to be reported in the next technical note. SIMCON 3.0. is a modular solution, while SIMCON 3.1. will be an integrated board of the all-in-one type. Two design approaches - modular and all-in-one - after branching off in this version of the Simcon, will be continued.

  2. ANN-based wavelet analysis for predicting electrical signal from photovoltaic power supply system

    Energy Technology Data Exchange (ETDEWEB)

    Mellit, A. [Medea Univ., Medea (Algeria). Inst. of Science Engineering, Dept. of Electronics

    2007-07-01

    This study was conducted to predict different electrical signals from a photovoltaic power supply system (PVPS) using an artificial neural networks (ANN) with wavelet analysis. It involved the creation of a database of electrical signals (PV-generator current, voltage, battery current voltage, regulator current and voltage) obtained from an experimental PVPS system installed in the south of Algeria. The potential applications were for sizing and analyzing the performance of PVPS systems; control of maximum power point tracker (MPPT) in order to deliver the maximum energy from the PV-array; prediction of the optimal configuration (PV-array and battery sizing) of PVPS systems; expert configuration of PV-systems; faults diagnosis; supervision; and, control and monitoring. First, based on the wavelet analysis each electrical signal was mapped in several time frequency domains. The PV-system was then divided into 3-subsystems corresponding to ANN-PV generator model, ANN-battery model, and ANN-regulator model. An example of day-by-day prediction for each electrical signal was presented. The results of the proposed approach were in good agreement with experimental results. In addition, the accuracy of the proposed approach was more satisfactory when only ANN was used. It was concluded that this methodology offers the possibility of developing a new expert configuration of PVPS by implementing the soft computing ANN-wavelet program with a digital signal processing (DSP) circuit. 26 refs., 1 tab., 5 figs.

  3. Human vocal attractiveness as signaled by body size projection.

    Directory of Open Access Journals (Sweden)

    Yi Xu

    Full Text Available Voice, as a secondary sexual characteristic, is known to affect the perceived attractiveness of human individuals. But the underlying mechanism of vocal attractiveness has remained unclear. Here, we presented human listeners with acoustically altered natural sentences and fully synthetic sentences with systematically manipulated pitch, formants and voice quality based on a principle of body size projection reported for animal calls and emotional human vocal expressions. The results show that male listeners preferred a female voice that signals a small body size, with relatively high pitch, wide formant dispersion and breathy voice, while female listeners preferred a male voice that signals a large body size with low pitch and narrow formant dispersion. Interestingly, however, male vocal attractiveness was also enhanced by breathiness, which presumably softened the aggressiveness associated with a large body size. These results, together with the additional finding that the same vocal dimensions also affect emotion judgment, indicate that humans still employ a vocal interaction strategy used in animal calls despite the development of complex language.

  4. Removing the Influence of Shimmer in the Calculation of Harmonics-To-Noise Ratios Using Ensemble-Averages in Voice Signals

    Directory of Open Access Journals (Sweden)

    Carlos Ferrer

    2009-01-01

    Full Text Available Harmonics-to-noise ratios (HNRs are affected by general aperiodicity in voiced speech signals. To specifically reflect a signal-to-additive-noise ratio, the measurement should be insensitive to other periodicity perturbations, like jitter, shimmer, and waveform variability. The ensemble averaging technique is a time-domain method which has been gradually refined in terms of its sensitivity to jitter and waveform variability and required number of pulses. In this paper, shimmer is introduced in the model of the ensemble average, and a formula is derived which allows the reduction of shimmer effects in HNR calculation. The validity of the technique is evaluated using synthetically shimmered signals, and the prerequisites (glottal pulse positions and amplitudes are obtained by means of fully automated methods. The results demonstrate the feasibility and usefulness of the correction.

  5. Emotional voices in context: a neurobiological model of multimodal affective information processing.

    Science.gov (United States)

    Brück, Carolin; Kreifelts, Benjamin; Wildgruber, Dirk

    2011-12-01

    Just as eyes are often considered a gateway to the soul, the human voice offers a window through which we gain access to our fellow human beings' minds - their attitudes, intentions and feelings. Whether in talking or singing, crying or laughing, sighing or screaming, the sheer sound of a voice communicates a wealth of information that, in turn, may serve the observant listener as valuable guidepost in social interaction. But how do human beings extract information from the tone of a voice? In an attempt to answer this question, the present article reviews empirical evidence detailing the cerebral processes that underlie our ability to decode emotional information from vocal signals. The review will focus primarily on two prominent classes of vocal emotion cues: laughter and speech prosody (i.e. the tone of voice while speaking). Following a brief introduction, behavioral as well as neuroimaging data will be summarized that allows to outline cerebral mechanisms associated with the decoding of emotional voice cues, as well as the influence of various context variables (e.g. co-occurring facial and verbal emotional signals, attention focus, person-specific parameters such as gender and personality) on the respective processes. Building on the presented evidence, a cerebral network model will be introduced that proposes a differential contribution of various cortical and subcortical brain structures to the processing of emotional voice signals both in isolation and in context of accompanying (facial and verbal) emotional cues. Copyright © 2011 Elsevier B.V. All rights reserved.

  6. Emotional voices in context: A neurobiological model of multimodal affective information processing

    Science.gov (United States)

    Brück, Carolin; Kreifelts, Benjamin; Wildgruber, Dirk

    2011-12-01

    Just as eyes are often considered a gateway to the soul, the human voice offers a window through which we gain access to our fellow human beings' minds - their attitudes, intentions and feelings. Whether in talking or singing, crying or laughing, sighing or screaming, the sheer sound of a voice communicates a wealth of information that, in turn, may serve the observant listener as valuable guidepost in social interaction. But how do human beings extract information from the tone of a voice? In an attempt to answer this question, the present article reviews empirical evidence detailing the cerebral processes that underlie our ability to decode emotional information from vocal signals. The review will focus primarily on two prominent classes of vocal emotion cues: laughter and speech prosody (i.e. the tone of voice while speaking). Following a brief introduction, behavioral as well as neuroimaging data will be summarized that allows to outline cerebral mechanisms associated with the decoding of emotional voice cues, as well as the influence of various context variables (e.g. co-occurring facial and verbal emotional signals, attention focus, person-specific parameters such as gender and personality) on the respective processes. Building on the presented evidence, a cerebral network model will be introduced that proposes a differential contribution of various cortical and subcortical brain structures to the processing of emotional voice signals both in isolation and in context of accompanying (facial and verbal) emotional cues.

  7. Real time polarization sensor image processing on an embedded FPGA/multi-core DSP system

    Science.gov (United States)

    Bednara, Marcus; Chuchacz-Kowalczyk, Katarzyna

    2015-05-01

    Most embedded image processing SoCs available on the market are highly optimized for typical consumer applications like video encoding/decoding, motion estimation or several image enhancement processes as used in DSLR or digital video cameras. For non-consumer applications, on the other hand, optimized embedded hardware is rarely available, so often PC based image processing systems are used. We show how a real time capable image processing system for a non-consumer application - namely polarization image data processing - can be efficiently implemented on an FPGA and multi-core DSP based embedded hardware platform.

  8. 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.

  9. Speech enhancement on smartphone voice recording

    International Nuclear Information System (INIS)

    Atmaja, Bagus Tris; Farid, Mifta Nur; Arifianto, Dhany

    2016-01-01

    Speech enhancement is challenging task in audio signal processing to enhance the quality of targeted speech signal while suppress other noises. In the beginning, the speech enhancement algorithm growth rapidly from spectral subtraction, Wiener filtering, spectral amplitude MMSE estimator to Non-negative Matrix Factorization (NMF). Smartphone as revolutionary device now is being used in all aspect of life including journalism; personally and professionally. Although many smartphones have two microphones (main and rear) the only main microphone is widely used for voice recording. This is why the NMF algorithm widely used for this purpose of speech enhancement. This paper evaluate speech enhancement on smartphone voice recording by using some algorithms mentioned previously. We also extend the NMF algorithm to Kulback-Leibler NMF with supervised separation. The last algorithm shows improved result compared to others by spectrogram and PESQ score evaluation. (paper)

  10. [Diagnostics and therapy in professional voice-users].

    Science.gov (United States)

    Richter, B; Echternach, M

    2010-04-01

    Voice is one of the most important instruments for expression and communication in humans. Dysphonia remains very frequent. Generally people in voice-intensive professions, such as teachers, call center employees, singers and actors suffer from these complaints. In recent years methods have been developed which facilitate appropriate diagnosis and therapy, based on the criteria of evidence based medicine, in voice patients appropriate to their degree of disease. The basic protocol of the European Laryngological Society offers a standardized evaluation of multidimensional voice parameters. In our own patient collective there were statistically significant improvements in voice quality, according to a pre/post mean value comparison, in both phonomicrosurgical (n=45) and voice therapy (n=30) patients in relation to RBH, DSI and VHI.

  11. Implicit multisensory associations influence voice recognition.

    Directory of Open Access Journals (Sweden)

    Katharina von Kriegstein

    2006-10-01

    Full Text Available Natural objects provide partially redundant information to the brain through different sensory modalities. For example, voices and faces both give information about the speech content, age, and gender of a person. Thanks to this redundancy, multimodal recognition is fast, robust, and automatic. In unimodal perception, however, only part of the information about an object is available. Here, we addressed whether, even under conditions of unimodal sensory input, crossmodal neural circuits that have been shaped by previous associative learning become activated and underpin a performance benefit. We measured brain activity with functional magnetic resonance imaging before, while, and after participants learned to associate either sensory redundant stimuli, i.e. voices and faces, or arbitrary multimodal combinations, i.e. voices and written names, ring tones, and cell phones or brand names of these cell phones. After learning, participants were better at recognizing unimodal auditory voices that had been paired with faces than those paired with written names, and association of voices with faces resulted in an increased functional coupling between voice and face areas. No such effects were observed for ring tones that had been paired with cell phones or names. These findings demonstrate that brief exposure to ecologically valid and sensory redundant stimulus pairs, such as voices and faces, induces specific multisensory associations. Consistent with predictive coding theories, associative representations become thereafter available for unimodal perception and facilitate object recognition. These data suggest that for natural objects effective predictive signals can be generated across sensory systems and proceed by optimization of functional connectivity between specialized cortical sensory modules.

  12. Loss-Free Counting with Digital Signal Processors

    International Nuclear Information System (INIS)

    Markku Koskelo; Dave Hall; Martin Moslinger

    2000-01-01

    Loss-free-counting (LFC) techniques have frequently been used with traditional analog pulse processing systems to compensate for the time or pulses lost when a spectroscopy system is unavailable (busy) for processing an accepted pulse. With the availability of second-generation digital signal processing (DSP) electronics that offer a significantly improved performance for both high and low count rate applications, the LFC technique has been revisited. Specific attention was given to the high and ultra-high count rate behavior, using high-purity germanium (HPGe) detectors with both transistor reset preamplifiers (TRP) and conventional RC preamplifiers. The experiments conducted for this work show that the known LFC techniques further benefit when combined with modern DSP pulse shaping

  13. A simple clockless Network-on-Chip for a commercial audio DSP chip

    DEFF Research Database (Denmark)

    Stensgaard, Mikkel Bystrup; Bjerregaard, Tobias; Sparsø, Jens

    2006-01-01

    We design a very small, packet-switched, clockless Network-on-Chip (NoC) as a replacement for the existing crossbar-based communication infrastructure in a commercial audio DSP chip. Both solutions are laid out in a 0.18 um process, and compared in terms of area, power consumption and routing...... to the existing crossbar, it allows all blocks to communicate. The total wire length is decreased by 22% which eases the layout process and makes the design less prone to routing congestion. Not least, the communicating blocks are decoupled by means of the NoC, providing a Globally-Asynchronous, Locally...

  14. Compact lidar system using laser diode, binary continuous wave power modulation, and an avalanche photodiode-based receiver controlled by a digital signal processor

    Science.gov (United States)

    Ardanuy, Antoni; Comerón, Adolfo

    2018-04-01

    We analyze the practical limits of a lidar system based on the use of a laser diode, random binary continuous wave power modulation, and an avalanche photodiode (APD)-based photereceiver, combined with the control and computing power of the digital signal processors (DSP) currently available. The target is to design a compact portable lidar system made all in semiconductor technology, with a low-power demand and an easy configuration of the system, allowing change in some of its features through software. Unlike many prior works, we emphasize the use of APDs instead of photomultiplier tubes to detect the return signal and the application of the system to measure not only hard targets, but also medium-range aerosols and clouds. We have developed an experimental prototype to evaluate the behavior of the system under different environmental conditions. Experimental results provided by the prototype are presented and discussed.

  15. Reconfigurable multi-DSP parallel computing architecture based on DSM%基于DSM的可重构多DSP并行处理架构

    Institute of Scientific and Technical Information of China (English)

    程鑫; 吴华春

    2012-01-01

    提出一种基于DSM的可在线重构多DSP并行处理架构,采用基于自定义内部总线的信息传递服务,在分布式物理内存上实现了统一编址的共享内存模型,减小了DSP之间的数据传递开销;设计基于VME总线的在线重构来实现针对消息传递服务的重定义,增强了并行计算架构的通用性.实验表明,采用此DSM能减小了并行DSP对共享数据同步访问开销,满足多轴精密同步运动控制系统需求.%A design of reconfigurable multi-digital signal processor (DSP) parallel computing architecture based on distributed shared memory (DSM) was proposed. A message-passing communication based on the user-defined internal bus (IB) was designed to implement a shared memory model on physically distributed memory, which decreased the data transmission overhead. Online reconfiguration mechanism was designed to implement message-passing communication reconfiguration, which in-creasd the universality of parallel architecture. The experiment shows that adopting the DSM introduced can reduce simultaneous access overhead to shared data, which satisfies the requirements of ultra-precise multi-axis motion control system.

  16. The DVB Channel Coding Application Using the DSP Development Board MDS TM-13 IREF

    Directory of Open Access Journals (Sweden)

    M. Slanina

    2004-12-01

    Full Text Available The paper deals with the implementation of the channel codingaccording to DVB standard on DSP development board MDS TM-13 IREF andPC. The board is based on Philips Nexperia media processor andintegrates hardware video ADC and DAC. The program libraries featuresused for MPEG based video compression are outlined and then thealgorithms of channel decoding (FEC protection against errors arepresented including the flowchart diagrams. The paper presents thepartial hardware implementation of the simulation system that coversselected phenomena of DVB baseband processing and it is used for realtime interactive demonstration of error protection influence ontransmitted digital video in laboratory and education.

  17. Collaboration and conquest: MTD as viewed by voice teacher (singing voice specialist) and speech-language pathologist.

    Science.gov (United States)

    Goffi-Fynn, Jeanne C; Carroll, Linda M

    2013-05-01

    This study was designed as a qualitative case study to demonstrate the process of diagnosis and treatment between a voice team to manage a singer diagnosed with muscular tension dysphonia (MTD). Traditionally, literature suggests that MTD is challenging to treat and little in the literature directly addresses singers with MTD. Data collected included initial medical screening with laryngologist, referral to speech-language pathologist (SLP) specializing in voice disorders among singers, and adjunctive voice training with voice teacher trained in vocology (singing voice specialist or SVS). Initial target goals with SLP included reducing extrinsic laryngeal tension, using a relaxed laryngeal posture, and effective abdominal-diaphragmatic support for all phonation events. Balance of respiratory forces, laryngeal coordination, and use of optimum filtering of the source signal through resonance and articulatory awareness was emphasized. Further work with SVS included three main goals including a lowered breathing pattern to aid in decreasing subglottic air pressure, vertical laryngeal position to lower to allow for a relaxed laryngeal position, and a top-down singing approach to encourage an easier, more balanced registration, and better resonance. Initial results also emphasize the retraining of subject toward a sensory rather than auditory mode of monitoring. Other areas of consideration include singers' training and vocal use, the psychological effects of MTD, the personalities potentially associated with it, and its relationship with stress. Finally, the results emphasize that a positive rapport with the subject and collaboration between all professionals involved in a singer's care are essential for recovery. Copyright © 2013 The Voice Foundation. Published by Mosby, Inc. All rights reserved.

  18. Noise generator for tinnitus treatment based on look-up tables

    Science.gov (United States)

    Uriz, Alejandro J.; Agüero, Pablo; Tulli, Juan C.; Castiñeira Moreira, Jorge; González, Esteban; Hidalgo, Roberto; Casadei, Manuel

    2016-04-01

    Treatment of tinnitus by means of masking sounds allows to obtain a significant improve of the quality of life of the individual that suffer that condition. In view of that, it is possible to develop noise synthesizers based on random number generators in digital signal processors (DSP), which are used in almost any digital hearing aid devices. DSP architecture have limitations to implement a pseudo random number generator, due to it, the noise statistics can be not as good as expectations. In this paper, a technique to generate additive white gaussian noise (AWGN) or other types of filtered noise using coefficients stored in program memory of the DSP is proposed. Also, an implementation of the technique is carried out on a dsPIC from Microchip®. Objective experiments and experimental measurements are performed to analyze the proposed technique.

  19. Comparison of acoustic voice characteristics in smoking and nonsmoking teachers

    Directory of Open Access Journals (Sweden)

    Šehović Ivana

    2012-01-01

    Full Text Available Voice of vocal professionals is exposed to great temptations, i.e. there is a high probability of voice alterations. Smoking, allergies and respiratory infections greatly affect the voice, which can change its acoustic characteristics. In smokers, the vocal cords mass increases, resulting in changes in vocal fold vibratory cycle. Pathological changes of vocal folds deform the acoustic signal and affect voice production. As vocal professionals, teachers are much more affected by voice disorders than average speakers. The aim of this study was to examine the differences in acoustic parameters of voice between smoking and nonsmoking teachers, in a sample of vocal professionals. The sample consisted of 60 female subjects, aged from 25 to 59. For voice analysis we used Computer lab, model 4300, 'Kay Elemetrics Corporation'. The statistical significance of differences in the values of acoustic parameters between smokers and nonsmokers was tested by ANOVA. Results showed that in the sample of female teachers, professional use of voice combined with the smoking habit can be linked to the changes in voice parameters. Comparing smokers and nonsmokers, average values of the parameters in short-term and long-term disturbances of frequency and amplitude proved to be significantly different.

  20. SIP Signaling Implementations and Performance Enhancement over MANET: A Survey

    OpenAIRE

    Alshamrani, M; Cruickshank, Haitham; Sun, Zhili; Ansa, G; Alshahwan, F

    2016-01-01

    The implementation of the Session Initiation Protocol (SIP)-based Voice over Internet Protocol (VoIP) and multimedia over MANET is still a challenging issue. Many routing factors affect the performance of SIP signaling and the voice Quality of Service (QoS). Node mobility in MANET causes dynamic changes to route calculations, topology, hop numbers, and the connectivity status between the correspondent nodes. SIP-based VoIP depends on the caller’s registration, call initiation, and call termin...

  1. DSP30 and interleukin-2 as a mitotic stimulant in B-cell disorders including those with a low disease burden.

    Science.gov (United States)

    Dun, Karen A; Riley, Louise A; Diano, Giuseppe; Adams, Leanne B; Chiu, Eleanor; Sharma, Archna

    2018-05-01

    Chromosome abnormalities detected during cytogenetic investigations for B-cell malignancy offer prognostic information that can have wide ranging clinical impacts on patients. These impacts may include monitoring frequency, treatment type, and disease staging level. The use of the synthetic oligonucleotide DSP30 combined with interleukin 2 (IL2) has been described as an effective mitotic stimulant in B-cell disorders, not only in chronic lymphocytic leukemia (CLL) but also in a range of other B-cell malignancies. Here, we describe the comparison of two B-cell mitogens, lipopolysaccharide (LPS), and DSP30 combined with IL2 as mitogens in a range of common B-cell disorders excluding CLL. The results showed that DSP30/IL2 was an effective mitogen in mature B-cell disorders, revealing abnormal cytogenetic results in a range of B-cell malignancies. The abnormality rate increased when compared to the use of LPS to 64% (DSP30/IL2) from 14% (LPS). In a number of cases the disease burden was proportionally very low, less than 10% of white cells. In 37% of these cases, the DSP30 culture revealed abnormal results. Importantly, we also obtained abnormal conventional cytogenetics results in 3 bone marrow cases in which immunophenotyping showed an absence of an abnormal B-cell clone. In these cases, the cytogenetics results correlated with the provisional diagnosis and altered their staging level. The use of DSP30 and IL2 is recommended for use in many B-cell malignancies as an effective mitogen and their use has been shown to enable successful culture of the malignant clone, even at very low levels of disease. © 2018 Wiley Periodicals, Inc.

  2. Classification of voice disorder in children with cochlear implantation and hearing aid using multiple classifier fusion

    Directory of Open Access Journals (Sweden)

    Tayarani Hamid

    2011-01-01

    Full Text Available Abstract Background Speech production and speech phonetic features gradually improve in children by obtaining audio feedback after cochlear implantation or using hearing aids. The aim of this study was to develop and evaluate automated classification of voice disorder in children with cochlear implantation and hearing aids. Methods We considered 4 disorder categories in children's voice using the following definitions: Level_1: Children who produce spontaneous phonation and use words spontaneously and imitatively. Level_2: Children, who produce spontaneous phonation, use words spontaneously and make short sentences imitatively. Level_3: Children, who produce spontaneous phonations, use words and arbitrary sentences spontaneously. Level_4: Normal children without any hearing loss background. Thirty Persian children participated in the study, including six children in each level from one to three and 12 children in level four. Voice samples of five isolated Persian words "mashin", "mar", "moosh", "gav" and "mouz" were analyzed. Four levels of the voice quality were considered, the higher the level the less significant the speech disorder. "Frame-based" and "word-based" features were extracted from voice signals. The frame-based features include intensity, fundamental frequency, formants, nasality and approximate entropy and word-based features include phase space features and wavelet coefficients. For frame-based features, hidden Markov models were used as classifiers and for word-based features, neural network was used. Results After Classifiers fusion with three methods: Majority Voting Rule, Linear Combination and Stacked fusion, the best classification rates were obtained using frame-based and word-based features with MVR rule (level 1:100%, level 2: 93.75%, level 3: 100%, level 4: 94%. Conclusions Result of this study may help speech pathologists follow up voice disorder recovery in children with cochlear implantation or hearing aid who are

  3. Synergy analysis reveals association between insulin signaling and desmoplakin expression in palmitate treated HepG2 cells.

    Directory of Open Access Journals (Sweden)

    Xuewei Wang

    Full Text Available The regulation of complex cellular activities in palmitate treated HepG2 cells, and the ensuing cytotoxic phenotype, involves cooperative interactions between genes. While previous approaches have largely focused on identifying individual target genes, elucidating interacting genes has thus far remained elusive. We applied the concept of information synergy to reconstruct a "gene-cooperativity" network for palmititate-induced cytotoxicity in liver cells. Our approach integrated gene expression data with metabolic profiles to select a subset of genes for network reconstruction. Subsequent analysis of the network revealed insulin signaling as the most significantly enriched pathway, and desmoplakin (DSP as its top neighbor. We determined that palmitate significantly reduces DSP expression, and treatment with insulin restores the lost expression of DSP. Insulin resistance is a common pathological feature of fatty liver and related ailments, whereas loss of DSP has been noted in liver carcinoma. Reduced DSP expression can lead to loss of cell-cell adhesion via desmosomes, and disrupt the keratin intermediate filament network. Our findings suggest that DSP expression may be perturbed by palmitate and, along with insulin resistance, may play a role in palmitate induced cytotoxicity, and serve as potential targets for further studies on non-alcoholic fatty liver disease (NAFLD.

  4. Optical Doppler tomography based on a field programmable gate array

    DEFF Research Database (Denmark)

    Larsen, Henning Engelbrecht; Nilsson, Ronnie Thorup; Thrane, Lars

    2008-01-01

    We report the design of and results obtained by using a field programmable gate array (FPGA) to digitally process optical Doppler tomography signals. The processor fits into the analog signal path in an existing optical coherence tomography setup. We demonstrate both Doppler frequency and envelope...... extraction using the Hilbert transform, all in a single FPGA. An FPGA implementation has certain advantages over general purpose digital signal processor (DSP) due to the fact that the processing elements operate in parallel as opposed to the DSP. which is primarily a sequential processor....

  5. Erwinia amylovora expresses fast and simultaneously hrp/dsp virulence genes during flower infection on apple trees.

    Directory of Open Access Journals (Sweden)

    Doris Pester

    Full Text Available BACKGROUND: Pathogen entry through host blossoms is the predominant infection pathway of the gram-negative bacterium Erwinia amylovora leading to manifestation of the disease fire blight. Like in other economically important plant pathogens, E. amylovora pathogenicity depends on a type III secretion system encoded by hrp genes. However, timing and transcriptional order of hrp gene expression during flower infections are unknown. METHODOLOGY/PRINCIPAL FINDINGS: Using quantitative real-time PCR analyses, we addressed the questions of how fast, strong and uniform key hrp virulence genes and the effector dspA/E are expressed when bacteria enter flowers provided with the full defense mechanism of the apple plant. In non-invasive bacterial inoculations of apple flowers still attached to the tree, E. amylovora activated expression of key type III secretion genes in a narrow time window, mounting in a single expression peak of all investigated hrp/dspA/E genes around 24-48 h post inoculation (hpi. This single expression peak coincided with a single depression in the plant PR-1 expression at 24 hpi indicating transient manipulation of the salicylic acid pathway as one target of E. amylovora type III effectors. Expression of hrp/dspA/E genes was highly correlated to expression of the regulator hrpL and relative transcript abundances followed the ratio: hrpA>hrpN>hrpL>dspA/E. Acidic conditions (pH 4 in flower infections led to reduced virulence/effector gene expression without the typical expression peak observed under natural conditions (pH 7. CONCLUSION/SIGNIFICANCE: The simultaneous expression of hrpL, hrpA, hrpN, and the effector dspA/E during early floral infection indicates that speed and immediate effector transmission is important for successful plant invasion. When this delicate balance is disturbed, e.g., by acidic pH during infection, virulence gene expression is reduced, thus partly explaining the efficacy of acidification in fire blight

  6. Logic Foundry: Rapid Prototyping for FPGA-Based DSP Systems

    Directory of Open Access Journals (Sweden)

    Bhattacharyya Shuvra S

    2003-01-01

    Full Text Available We introduce the Logic Foundry, a system for the rapid creation and integration of FPGA-based digital signal processing systems. Recognizing that some of the greatest challenges in creating FPGA-based systems occur in the integration of the various components, we have proposed a system that targets the following four areas of integration: design flow integration, component integration, platform integration, and software integration. Using the Logic Foundry, a system can be easily specified, and then automatically constructed and integrated with system level software.

  7. Understanding the mechanisms of familiar voice-identity recognition in the human brain.

    Science.gov (United States)

    Maguinness, Corrina; Roswandowitz, Claudia; von Kriegstein, Katharina

    2018-03-31

    Humans have a remarkable skill for voice-identity recognition: most of us can remember many voices that surround us as 'unique'. In this review, we explore the computational and neural mechanisms which may support our ability to represent and recognise a unique voice-identity. We examine the functional architecture of voice-sensitive regions in the superior temporal gyrus/sulcus, and bring together findings on how these regions may interact with each other, and additional face-sensitive regions, to support voice-identity processing. We also contrast findings from studies on neurotypicals and clinical populations which have examined the processing of familiar and unfamiliar voices. Taken together, the findings suggest that representations of familiar and unfamiliar voices might dissociate in the human brain. Such an observation does not fit well with current models for voice-identity processing, which by-and-large assume a common sequential analysis of the incoming voice signal, regardless of voice familiarity. We provide a revised audio-visual integrative model of voice-identity processing which brings together traditional and prototype models of identity processing. This revised model includes a mechanism of how voice-identity representations are established and provides a novel framework for understanding and examining the potential differences in familiar and unfamiliar voice processing in the human brain. Copyright © 2018 Elsevier Ltd. All rights reserved.

  8. Robust signal selection for lineair prediction analysis of voiced speech

    NARCIS (Netherlands)

    Ma, C.; Kamp, Y.; Willems, L.F.

    1993-01-01

    This paper investigates a weighted LPC analysis of voiced speech. In view of the speech production model, the weighting function is either chosen to be the short-time energy function of the preemphasized speech sample sequence with certain delays or is obtained by thresholding the short-time energy

  9. Large Signal Model of a Four-quadrant AC to DC Converter for Accelerator Magnets

    CERN Document Server

    De la Calle, R; Rinaldi, L; Völker, F V

    2001-01-01

    This paper presents the large signal model of a four-quadrant AC to DC converter, which is expected to be used in the area of particle accelerators. The system’s first stage is composed of a three-phase boost PWM (Pulse Width Modulated) rectifier with DSP (Digital Signal Processing) based power factor correction (PFC) and output voltage regulation. The second stage is a full-bridge PWM inverter that allows fast four-quadrant operation. The structure is fully reversible, and an additional resistance (brake chopper) is not needed to dissipate the energy when the beam deflection magnet acts as generator.

  10. Digital Signal Processing for In-Vehicle Systems and Safety

    CERN Document Server

    Boyraz, Pinar; Takeda, Kazuya; Abut, Hüseyin

    2012-01-01

    Compiled from papers of the 4th Biennial Workshop on DSP (Digital Signal Processing) for In-Vehicle Systems and Safety this edited collection features world-class experts from diverse fields focusing on integrating smart in-vehicle systems with human factors to enhance safety in automobiles. Digital Signal Processing for In-Vehicle Systems and Safety presents new approaches on how to reduce driver inattention and prevent road accidents. The material addresses DSP technologies in adaptive automobiles, in-vehicle dialogue systems, human machine interfaces, video and audio processing, and in-vehicle speech systems. The volume also features: Recent advances in Smart-Car technology – vehicles that take into account and conform to the driver Driver-vehicle interfaces that take into account the driving task and cognitive load of the driver Best practices for In-Vehicle Corpus Development and distribution Information on multi-sensor analysis and fusion techniques for robust driver monitoring and driver recognition ...

  11. A rigorous analysis of digital pre-emphasis and DAC resolution for interleaved DAC Nyquist-WDM signal generation in high-speed coherent optical transmission systems

    Science.gov (United States)

    Weng, Yi; Wang, Junyi; He, Xuan; Pan, Zhongqi

    2018-02-01

    The Nyquist spectral shaping techniques facilitate a promising solution to enhance spectral efficiency (SE) and further reduce the cost-per-bit in high-speed wavelength-division multiplexing (WDM) transmission systems. Hypothetically, any Nyquist WDM signals with arbitrary shapes can be generated by the use of the digital signal processing (DSP) based electrical filters (E-filter). Nonetheless, in actual 100G/ 200G coherent systems, the performance as well as DSP complexity are increasingly restricted by cost and power consumption. Henceforward it is indispensable to optimize DSP to accomplish the preferred performance at the least complexity. In this paper, we systematically investigated the minimum requirements and challenges of Nyquist WDM signal generation, particularly for higher-order modulation formats, including 16 quadrature amplitude modulation (QAM) or 64QAM. A variety of interrelated parameters, such as channel spacing and roll-off factor, have been evaluated to optimize the requirements of the digital-to-analog converter (DAC) resolution and transmitter E-filter bandwidth. The impact of spectral pre-emphasis has been predominantly enhanced via the proposed interleaved DAC architecture by at least 4%, and hence reducing the required optical signal to noise ratio (OSNR) at a bit error rate (BER) of 10-3 by over 0.45 dB at a channel spacing of 1.05 symbol rate and an optimized roll-off factor of 0.1. Furthermore, the requirements of sampling rate for different types of super-Gaussian E-filters are discussed for 64QAM Nyquist WDM transmission systems. Finally, the impact of the non-50% duty cycle error between sub-DACs upon the quality of the generated signals for the interleaved DAC structure has been analyzed.

  12. Finding CreativeVoice: Applying Arts-Based Research in the Context of Biodiversity Conservation

    Directory of Open Access Journals (Sweden)

    Flor Rivera Lopez

    2018-05-01

    Full Text Available The integration of creative arts–based methods into scientific research offers a host of advantages, including the ability to capture the complex texture of lived experience, explore interconnections between nature and culture, support nonhierarchical relations, and communicate insights in engaging and empowering new ways. In this article, we describe a new method—CreativeVoice—integrating the creative arts and qualitative research, which we developed and applied in a context of pursuing community-based conservation of agricultural biodiversity. We developed CreativeVoice as an integrative method to help us understand the local contexts, cultures, and perspectives from community members of different ages and genders, in two contrasting farming communities in Oaxaca, Mexico. CreativeVoice effectively adapts and extends the Photovoice method so as to retain its benefits but address some of its limitations. This includes allowing participants to choose a genre of artistic expression connected to their own specific individual or cultural contexts and providing the capacity to move beyond capturing present-day realities to directly bring in connections to the past and visions for the future. This article describes both the CreativeVoice approach and the significant value of integrating arts-based methods into research for advancing sustainability.

  13. A DSP-based readout and online processing system for a new focal-plane polarimeter at AGOR

    Energy Technology Data Exchange (ETDEWEB)

    Hagemann, M.; Bassini, R.; Berg, A.M. van den; Ellinghaus, F.; Frekers, D.; Hannen, V.M.; Haeupke, T.; Heyse, J.; Jacobs, E.; Kirsch, M.; Kruesemann, B.; Rakers, S.; Sohlbach, H.; Woertche, H.J. E-mail: wortche@ikp.uni-muenster.de

    1999-11-21

    A Focal-Plane Polarimeter (FPP) for the large acceptance Big-Bite Spectrometer (BBS) at AGOR using a novel readout architecture has been commissioned at the KVI Groningen. The instrument is optimized for medium-energy polarized proton scattering near or at 0 deg. . For the handling of the high counting rates at extreme forward angles and for the suppression of small-angle scattering in the graphite analyzer, a high-performance data processing DSP system connecting to the LeCroy FERA and PCOS ECL bus architecture has been made operational and tested successfully. Details of the system and the functions of the various electronic components are described.

  14. Youth Voice in Nigerian School-based Management Committees

    Directory of Open Access Journals (Sweden)

    Bashiru Bako Umar

    2017-01-01

    Full Text Available In Nigeria, School-Based Management Committees (SBMCs aim to provide an opportunity for all stakeholders, particularly the vulnerable groups in the school’s host communities such as young people and women to partake in school governance. Research on the experiences of youth voice in the committees is scant, however, as much of the existing literature on SBMCs focuses on program outcomes. Using qualitative research interviews, observations, and document analysis, this study addressed this gap by exploring how youth participate and express themselves in two SBMCs in Niger State, Nigeria. The findings, which were derived from 19 youth and adult participants, were drawn from SBMC members out of which 12 were youth between the ages 13 and 25, while 7 were adults aged 40 and above. The participants revealed that youth committee members expressed their voice in the committees through participating in a number of committee activities. Specifically, the youth participated in decision-making during meetings, aided in the construction of committee projects, undertook administrative/managerial functions and monitored the committee’s projects. They also participated in revenue generation, planning, school visits and supervision, advocacy, and sensitization campaigns.

  15. The shouted voice: A pilot study of laryngeal physiology under extreme aerodynamic pressure.

    Science.gov (United States)

    Lagier, Aude; Legou, Thierry; Galant, Camille; Amy de La Bretèque, Benoit; Meynadier, Yohann; Giovanni, Antoine

    2017-12-01

    The objective was to study the behavior of the larynx during shouted voice production, when the larynx is exposed to extremely high subglottic pressure. The study involved electroglottographic, acoustic, and aerodynamic analyses of shouts produced at maximum effort by three male participants. Under a normal speaking voice, the voice sound pressure level (SPL) is proportional to the subglottic pressure. However, when the subglottic pressure reached high levels, the voice SPL reached a maximum value and then decreased as subglottic pressure increased further. Furthermore, the electroglottographic signal sometimes lost its periodicity during the shout, suggesting irregular vocal fold vibration.

  16. Some recent work on lattice structures for digital signal processing

    Indian Academy of Sciences (India)

    Digital signal processing (DSP); lattice structures; finite impulse ... fascinated this author for a long time, and for the known non-canonical ...... where M

  17. When the face fits: recognition of celebrities from matching and mismatching faces and voices.

    Science.gov (United States)

    Stevenage, Sarah V; Neil, Greg J; Hamlin, Iain

    2014-01-01

    The results of two experiments are presented in which participants engaged in a face-recognition or a voice-recognition task. The stimuli were face-voice pairs in which the face and voice were co-presented and were either "matched" (same person), "related" (two highly associated people), or "mismatched" (two unrelated people). Analysis in both experiments confirmed that accuracy and confidence in face recognition was consistently high regardless of the identity of the accompanying voice. However accuracy of voice recognition was increasingly affected as the relationship between voice and accompanying face declined. Moreover, when considering self-reported confidence in voice recognition, confidence remained high for correct responses despite the proportion of these responses declining across conditions. These results converged with existing evidence indicating the vulnerability of voice recognition as a relatively weak signaller of identity, and results are discussed in the context of a person-recognition framework.

  18. Voice Quality Estimation in Wireless Networks

    Directory of Open Access Journals (Sweden)

    Petr Zach

    2015-01-01

    Full Text Available This article deals with the impact of Wireless (Wi-Fi networks on the perceived quality of voice services. The Quality of Service (QoS metrics must be monitored in the computer network during the voice data transmission to ensure proper voice service quality the end-user has paid for, especially in the wireless networks. In addition to the QoS, research area called Quality of Experience (QoE provides metrics and methods for quality evaluation from the end-user’s perspective. This article focuses on a QoE estimation of Voice over IP (VoIP calls in the wireless networks using network simulator. Results contribute to voice quality estimation based on characteristics of the wireless network and location of a wireless client.

  19. Instantaneous Fundamental Frequency Estimation with Optimal Segmentation for Nonstationary Voiced Speech

    DEFF Research Database (Denmark)

    Nørholm, Sidsel Marie; Jensen, Jesper Rindom; Christensen, Mads Græsbøll

    2016-01-01

    In speech processing, the speech is often considered stationary within segments of 20–30 ms even though it is well known not to be true. In this paper, we take the non-stationarity of voiced speech into account by using a linear chirp model to describe the speech signal. We propose a maximum...... likelihood estimator of the fundamental frequency and chirp rate of this model, and show that it reaches the Cramer-Rao bound. Since the speech varies over time, a fixed segment length is not optimal, and we propose to make a segmentation of the signal based on the maximum a posteriori (MAP) criterion. Using...... of the chirp model than the harmonic model to the speech signal. The methods are based on an assumption of white Gaussian noise, and, therefore, two prewhitening filters are also proposed....

  20. Interband cascade laser-based ppbv-level mid-infrared methane detection using two digital lock-in amplifier schemes

    Science.gov (United States)

    Song, Fang; Zheng, Chuantao; Yu, Di; Zhou, Yanwen; Yan, Wanhong; Ye, Weilin; Zhang, Yu; Wang, Yiding; Tittel, Frank K.

    2018-03-01

    A parts-per-billion in volume (ppbv) level mid-infrared methane (CH4) sensor system was demonstrated using second-harmonic wavelength modulation spectroscopy (2 f-WMS). A 3291 nm interband cascade laser (ICL) and a multi-pass gas cell (MPGC) with a 16 m optical path length were adopted in the reported sensor system. Two digital lock-in amplifier (DLIA) schemes, a digital signal processor (DSP)-based DLIA and a LabVIEW-based DLIA, were used for harmonic signal extraction. A limit of detection (LoD) of 13.07 ppbv with an averaging time of 2 s was achieved using the DSP-based DLIA and a LoD of 5.84 ppbv was obtained using the LabVIEW-based DLIA with the same averaging time. A rise time of 0→2 parts-per-million in volume (ppmv) and fall time of 2→0 ppmv were observed. Outdoor atmospheric CH4 concentration measurements were carried out to evaluate the sensor performance using the two DLIA schemes.

  1. Voice stress analysis and evaluation

    Science.gov (United States)

    Haddad, Darren M.; Ratley, Roy J.

    2001-02-01

    Voice Stress Analysis (VSA) systems are marketed as computer-based systems capable of measuring stress in a person's voice as an indicator of deception. They are advertised as being less expensive, easier to use, less invasive in use, and less constrained in their operation then polygraph technology. The National Institute of Justice have asked the Air Force Research Laboratory for assistance in evaluating voice stress analysis technology. Law enforcement officials have also been asking questions about this technology. If VSA technology proves to be effective, its value for military and law enforcement application is tremendous.

  2. Recent progress on high-speed optical transmission

    Directory of Open Access Journals (Sweden)

    Jianjun Yu

    2016-05-01

    Full Text Available The recently reported high spectral efficiency (SE and high-baud-rate signal transmission are all based on digital coherent optical communications and digital signal processing (DSP. DSP simplifies the reception of advanced modulation formats and also enables the major electrical and optical impairments to be processed and compensated in the digital domain, at the transmitter or receiver side. In this paper, we summarize the research progress on high-speed signal generation and detection and also show the progress on DSP for high-speed signal detection. We also report the latest progress on multi-core and multi-mode multiplexing.

  3. Clinical Features of Psychogenic Voice Disorder and the Efficiency of Voice Therapy and Psychological Evaluation.

    Science.gov (United States)

    Tezcaner, Zahide Çiler; Gökmen, Muhammed Fatih; Yıldırım, Sibel; Dursun, Gürsel

    2017-11-06

    The aim of this study was to define the clinical features of psychogenic voice disorder (PVD) and explore the treatment efficiency of voice therapy and psychological evaluation. Fifty-eight patients who received treatment following the PVD diagnosis and had no organic or other functional voice disorders were assessed retrospectively based on laryngoscopic examinations and subjective and objective assessments. Epidemiological characteristics, accompanying organic and psychological disorders, preferred methods of treatment, and previous treatment outcomes were examined for each patient. A comparison was made based on voice disorders and responses to treatment between patients who received psychotherapy and patients who did not. Participants in this study comprised 58 patients, 10 male and 48 female. Voice therapy was applied in all patients, 54 (93.1%) of whom had improvement in their voice. Although all patients were advised to undergo psychological assessment, only 60.3% (35/58) of them underwent psychological assessment. No statistically significant difference was found between patients who did receive psychological support concerning their treatment responses and patients who did not. Relapse occurred in 14.7% (5/34) of the patients who applied for psychological assessment and in 50% (10/20) of those who did not. There was a statistically significant difference in relapse rates, which was higher among patients who did not receive psychological support (P therapy is an efficient treatment method for PVD. However, in the long-term follow-up, relapse of the disease is observed to be higher among patients who failed to follow up on the recommendation for psychological assessment. Copyright © 2017 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

  4. Exploring expressivity and emotion with artificial voice and speech technologies.

    Science.gov (United States)

    Pauletto, Sandra; Balentine, Bruce; Pidcock, Chris; Jones, Kevin; Bottaci, Leonardo; Aretoulaki, Maria; Wells, Jez; Mundy, Darren P; Balentine, James

    2013-10-01

    Emotion in audio-voice signals, as synthesized by text-to-speech (TTS) technologies, was investigated to formulate a theory of expression for user interface design. Emotional parameters were specified with markup tags, and the resulting audio was further modulated with post-processing techniques. Software was then developed to link a selected TTS synthesizer with an automatic speech recognition (ASR) engine, producing a chatbot that could speak and listen. Using these two artificial voice subsystems, investigators explored both artistic and psychological implications of artificial speech emotion. Goals of the investigation were interdisciplinary, with interest in musical composition, augmentative and alternative communication (AAC), commercial voice announcement applications, human-computer interaction (HCI), and artificial intelligence (AI). The work-in-progress points towards an emerging interdisciplinary ontology for artificial voices. As one study output, HCI tools are proposed for future collaboration.

  5. Application of the cyclic permutation for analysis of synthesized sinusoidal signal

    Czech Academy of Sciences Publication Activity Database

    Čížek, Václav; Švandová, Hana

    2002-01-01

    Roč. 2, č. 1 (2002), s. 69-72 ISSN 1335-8243. [Digital Signal Processing and Multimedia Communications DSP-MCOM 2001 /5./. Košice, 27.11.2001-29.11.2001] R&D Projects: GA ČR GA102/00/0958 Institutional research plan: CEZ:AV0Z2067918 Keywords : direct digital synthesis * quantisation-signal * number theory Subject RIV: JA - Electronics ; Optoelectronics, Electrical Engineering

  6. Can we respond mindfully to distressing voices? A systematic review of evidence for engagement, acceptability, effectiveness and mechanisms of change for mindfulness-based interventions for people distressed by hearing voices

    Directory of Open Access Journals (Sweden)

    Clara eStrauss

    2015-08-01

    Full Text Available Adapted mindfulness-based interventions (MBIs could be of benefit for people distressed by hearing voices. This paper presents a systematic review of studies exploring this possibility and we ask five questions: (1 Is trait mindfulness associated with reduced distress and disturbance in relation to hearing voices? (2 Are MBIs feasible for people distressed by hearing voices? (3 Are MBIs acceptable and safe for people distressed by hearing voices? (4 Are MBIs effective at reducing distress and disturbance in people distressed by hearing voices? (5 If effective, what are the mechanisms of change through which MBIs for distressing voices work?Fifteen studies were identified through a systematic search (n=479. In relation to the five review questions: (1 data from cross-sectional studies showed an association between trait mindfulness and distress and disturbance in relation to hearing voices; (2 evidence from qualitative studies suggested that people distressed by hearing voices could engage meaningfully in mindfulness practice; (3 MBIs were seen as acceptable and safe; (4 there were no adequately powered RCTs allowing conclusions about effectiveness to be drawn; and (5 it was not possible to draw on robust empirical data to comment on potential mechanisms of change although findings from the qualitative studies identified three potential change processes; (i reorientation of attention; (ii decentring; and (iii acceptance of voices. This review provided evidence that MBIs are engaging, acceptable and safe. Evidence for effectiveness in reducing distress and disturbance is lacking however. We call for funding for adequately powered RCTs that will allow questions of effectiveness, maintenance of effects, mechanisms of change and moderators of outcome to be definitively addressed.

  7. Real-time implementations of acoustic signal enhancement techniques for aerial based surveillance and rescue applications

    Science.gov (United States)

    Ramos, Antonio L. L.; Shao, Zhili; Holthe, Aleksander; Sandli, Mathias F.

    2017-05-01

    The introduction of the System-on-Chip (SoC) technology has brought exciting new opportunities for the development of smart low cost embedded systems spanning a wide range of applications. Currently available SoC devices are capable of performing high speed digital signal processing tasks in software while featuring relatively low development costs and reduced time-to-market. Unmanned aerial vehicles (UAV) are an application example that has shown tremendous potential in an increasing number of scenarios, ranging from leisure to surveillance as well as in search and rescue missions. Video capturing from UAV platforms is a relatively straightforward task that requires almost no preprocessing. However, that does not apply to audio signals, especially in cases where the data is to be used to support real-time decision making. In fact, the enormous amount of acoustic interference from the surroundings, including the noise from the UAVs propellers, becomes a huge problem. This paper discusses a real-time implementation of the NLMS adaptive filtering algorithm applied to enhancing acoustic signals captured from UAV platforms. The model relies on a combination of acoustic sensors and a computational inexpensive algorithm running on a digital signal processor. Given its simplicity, this solution can be incorporated into the main processing system of an UAV using the SoC technology, and run concurrently with other required tasks, such as flight control and communications. Simulations and real-time DSP-based implementations have shown significant signal enhancement results by efficiently mitigating the interference from the noise generated by the UAVs propellers as well as from other external noise sources.

  8. Voice preprocessing system incorporating a real-time spectrum analyzer with programmable switched-capacitor filters

    Science.gov (United States)

    Knapp, G.

    1984-01-01

    As part of a speaker verification program for BISS (Base Installation Security System), a test system is being designed with a flexible preprocessing system for the evaluation of voice spectrum/verification algorithm related problems. The main part of this report covers the design, construction, and testing of a voice analyzer with 16 integrating real-time frequency channels ranging from 300 Hz to 3 KHz. The bandpass filter response of each channel is programmable by NMOS switched capacitor quad filter arrays. Presently, the accuracy of these units is limited to a moderate precision by the finite steps of programming. However, repeatability of characteristics between filter units and sections seems to be excellent for the implemented fourth-order Butterworth bandpass responses. We obtained a 0.1 dB linearity error of signal detection and measured a signal-to-noise ratio of approximately 70 dB. The proprocessing system discussed includes preemphasis filter design, gain normalizer design, and data acquisition system design as well as test results.

  9. Artificially intelligent recognition of Arabic speaker using voice print-based local features

    Science.gov (United States)

    Mahmood, Awais; Alsulaiman, Mansour; Muhammad, Ghulam; Akram, Sheeraz

    2016-11-01

    Local features for any pattern recognition system are based on the information extracted locally. In this paper, a local feature extraction technique was developed. This feature was extracted in the time-frequency plain by taking the moving average on the diagonal directions of the time-frequency plane. This feature captured the time-frequency events producing a unique pattern for each speaker that can be viewed as a voice print of the speaker. Hence, we referred to this technique as voice print-based local feature. The proposed feature was compared to other features including mel-frequency cepstral coefficient (MFCC) for speaker recognition using two different databases. One of the databases used in the comparison is a subset of an LDC database that consisted of two short sentences uttered by 182 speakers. The proposed feature attained 98.35% recognition rate compared to 96.7% for MFCC using the LDC subset.

  10. Prior and posterior probabilistic models of uncertainties in a model for producing voice

    International Nuclear Information System (INIS)

    Cataldo, Edson; Sampaio, Rubens; Soize, Christian

    2010-01-01

    The aim of this paper is to use Bayesian statistics to update a probability density function related to the tension parameter, which is one of the main parameters responsible for the changing of the fundamental frequency of a voice signal, generated by a mechanical/mathematical model for producing voiced sounds. We follow a parametric approach for stochastic modeling, which requires the adoption of random variables to represent the uncertain parameters present in the cited model. For each random variable, a probability density function is constructed using the Maximum Entropy Principle and the Monte Carlo method is used to generate voice signals as the output of the model. Then, a probability density function of the voice fundamental frequency is constructed. The random variables are fit to experimental data so that the probability density function of the fundamental frequency obtained by the model can be as near as possible of a probability density function obtained from experimental data. New values are obtained experimentally for the fundamental frequency and they are used to update the probability density function of the tension parameter, via Bayes's Theorem.

  11. Mathematical pattern, smoothing and digital filtering of a speech signal

    International Nuclear Information System (INIS)

    Razzam, Mohamed Habib

    1979-01-01

    After presentation of speech synthesis methods, characterized by a treatment of pre-recorded natural signals, or by an analog simulation of vocal tract, we present a new synthesis method especially based on a mathematical pattern of the signal, as a development of M. RODET's method. For their physiological origin, these signals are partially or totally voiced, or aleatory. For the phoneme voiced parts, we compute the formant curves, the sum of which constitute the wave, directly in time-domain by applying a specific envelope (operating as a time-window analysis) to a sinusoidal wave, The sinusoidal wave computation is made at the beginning of each signal's pseudo-period. The transition from successive periods is assured by a polynomial smoothing followed by a digital filtering. For the aleatory parts, we present an aleatory computation method of formant curves. Each signal is subjected to a melodic diagrams computed in accordance with the nature of the phoneme (vowel or consonant) and its context (isolated or not). (author) [fr

  12. Monitoring of DSP toxins in small-sized plankton fraction of seawater collected in Mutsu Bay, Japan, by ELISA method: relation with toxin contamination of scallop.

    Science.gov (United States)

    Imai, Ichiro; Sugioka, Hikaru; Nishitani, Goh; Mitsuya, Tadashi; Hamano, Yonekazu

    2003-01-01

    Monitorings were conducted on DSP toxins in mid-gut gland of scallop (mouse assay), cell numbers of toxic dinoflagellate species of Dinophysis, and diarrhetic shellfish poisoning (DSP) toxins in small-sized (0.7-5 microm) plankton fraction of seawater collected from surface (0 m) and 20 m depth at a station in Mutsu Bay, Aomori Prefecture, Japan, in 2000. A specific enzyme-linked immunosorbent assay (ELISA) was employed for the analysis of DSP toxins in small-sized plankton fraction using a mouse monoclonal anti-okadaic acid antibody which recognizes okadaic acid, dinophysistoxin-1, and dinophysistoxin-3. DSP toxins were detected twice in the mid-gut gland of scallops at 1.1-2.3 MU (mouse units) g(-1) on 26 June and at 0.6-1.2 MU g(-1) on 3 July, respectively. Relatively high cell densities of D. fortii were observed on 26 June and 11 September, and may only contribute to the bivalve toxicity during late June to early July. D. acuminata did not appear to be responsible for the toxicity of scallops in Mutsu Bay in 2000. ELISA monitoring of small-sized plankton fraction in seawater could detect DSP toxins two weeks before the detection of the toxin in scallops, and could do so two weeks after the loss of the bivalve toxicity by mouse assay. On 17 July, toxic D. fortii was detected at only small number, <10 cells l(-1), but DSP toxins were detected by the ELISA assay, suggesting a presence of other toxic small-sized plankton in seawater. For the purpose of reducing negative impacts of DSP occurrences, monitorings have been carried out hitherto on DSP toxins of bivalve tissues by mouse assay and on cell densities of "toxic" species of Dinophysis. Here we propose a usefulness of ELISA monitoring of plankton toxicity, especially in small-sized fraction, which are possible foods of mixotrophic Dinophysis, as a practical tool for detecting and predicting DSPs in coastal areas of fisheries grounds of bivalve aquaculture.

  13. Emotional Prosody Measurement (EPM): a voice-based evaluation method for psychological therapy effectiveness.

    Science.gov (United States)

    van den Broek, Egon L

    2004-01-01

    The voice embodies three sources of information: speech, the identity, and the emotional state of the speaker (i.e., emotional prosody). The latter feature is resembled by the variability of the F0 (also named fundamental frequency of pitch) (SD F0). To extract this feature, Emotional Prosody Measurement (EPM) was developed, which consists of 1) speech recording, 2) removal of speckle noise, 3) a Fourier Transform to extract the F0-signal, and 4) the determination of SD F0. After a pilot study in which six participants mimicked emotions by their voice, the core experiment was conducted to see whether EPM is successful. Twenty-five patients suffering from a panic disorder with agoraphobia participated. Two methods (story-telling and reliving) were used to trigger anxiety and were compared with comparable but more relaxed conditions. This resulted in a unique database of speech samples that was used to compare the EPM with the Subjective Unit of Distress to validate it as measure for anxiety/stress. The experimental manipulation of anxiety proved to be successful and EPM proved to be a successful evaluation method for psychological therapy effectiveness.

  14. Finding your mate at a cocktail party: frequency separation promotes auditory stream segregation of concurrent voices in multi-species frog choruses.

    Directory of Open Access Journals (Sweden)

    Vivek Nityananda

    Full Text Available Vocal communication in crowded social environments is a difficult problem for both humans and nonhuman animals. Yet many important social behaviors require listeners to detect, recognize, and discriminate among signals in a complex acoustic milieu comprising the overlapping signals of multiple individuals, often of multiple species. Humans exploit a relatively small number of acoustic cues to segregate overlapping voices (as well as other mixtures of concurrent sounds, like polyphonic music. By comparison, we know little about how nonhuman animals are adapted to solve similar communication problems. One important cue enabling source segregation in human speech communication is that of frequency separation between concurrent voices: differences in frequency promote perceptual segregation of overlapping voices into separate "auditory streams" that can be followed through time. In this study, we show that frequency separation (ΔF also enables frogs to segregate concurrent vocalizations, such as those routinely encountered in mixed-species breeding choruses. We presented female gray treefrogs (Hyla chrysoscelis with a pulsed target signal (simulating an attractive conspecific call in the presence of a continuous stream of distractor pulses (simulating an overlapping, unattractive heterospecific call. When the ΔF between target and distractor was small (e.g., ≤3 semitones, females exhibited low levels of responsiveness, indicating a failure to recognize the target as an attractive signal when the distractor had a similar frequency. Subjects became increasingly more responsive to the target, as indicated by shorter latencies for phonotaxis, as the ΔF between target and distractor increased (e.g., ΔF = 6-12 semitones. These results support the conclusion that gray treefrogs, like humans, can exploit frequency separation as a perceptual cue to segregate concurrent voices in noisy social environments. The ability of these frogs to segregate

  15. [Voice disorders in female teachers assessed by Voice Handicap Index].

    Science.gov (United States)

    Niebudek-Bogusz, Ewa; Kuzańska, Anna; Woźnicka, Ewelina; Sliwińska-Kowalska, Mariola

    2007-01-01

    The aim of this study was to assess the application of Voice Handicap Index (VHI) in the diagnosis of occupational voice disorders in female teachers. The subjective assessment of voice by VHI was performed in fifty subjects with dysphonia diagnosed in laryngovideostroboscopic examination. The control group comprised 30 women whose jobs did not involve vocal effort. The results of the total VHI score and each of its subscales: functional, emotional and physical was significantly worse in the study group than in controls (p teachers estimated their own voice problems as a moderate disability, while 12% of them reported severe voice disability. However, all non-teachers assessed their voice problems as slight, their results ranged at the lowest level of VHI score. This study confirmed that VHI as a tool for self-assessment of voice can be a significant contribution to the diagnosis of occupational dysphonia.

  16. Obligatory and facultative brain regions for voice-identity recognition

    Science.gov (United States)

    Roswandowitz, Claudia; Kappes, Claudia; Obrig, Hellmuth; von Kriegstein, Katharina

    2018-01-01

    Abstract Recognizing the identity of others by their voice is an important skill for social interactions. To date, it remains controversial which parts of the brain are critical structures for this skill. Based on neuroimaging findings, standard models of person-identity recognition suggest that the right temporal lobe is the hub for voice-identity recognition. Neuropsychological case studies, however, reported selective deficits of voice-identity recognition in patients predominantly with right inferior parietal lobe lesions. Here, our aim was to work towards resolving the discrepancy between neuroimaging studies and neuropsychological case studies to find out which brain structures are critical for voice-identity recognition in humans. We performed a voxel-based lesion-behaviour mapping study in a cohort of patients (n = 58) with unilateral focal brain lesions. The study included a comprehensive behavioural test battery on voice-identity recognition of newly learned (voice-name, voice-face association learning) and familiar voices (famous voice recognition) as well as visual (face-identity recognition) and acoustic control tests (vocal-pitch and vocal-timbre discrimination). The study also comprised clinically established tests (neuropsychological assessment, audiometry) and high-resolution structural brain images. The three key findings were: (i) a strong association between voice-identity recognition performance and right posterior/mid temporal and right inferior parietal lobe lesions; (ii) a selective association between right posterior/mid temporal lobe lesions and voice-identity recognition performance when face-identity recognition performance was factored out; and (iii) an association of right inferior parietal lobe lesions with tasks requiring the association between voices and faces but not voices and names. The results imply that the right posterior/mid temporal lobe is an obligatory structure for voice-identity recognition, while the inferior parietal

  17. Investigation of a glottal related harmonics-to-noise ratio and spectral tilt as indicators of glottal noise in synthesized and human voice signals.

    LENUS (Irish Health Repository)

    Murphy, Peter J

    2008-03-01

    The harmonics-to-noise ratio (HNR) of the voiced speech signal has implicitly been used to infer information regarding the turbulent noise level at the glottis. However, two problems exist for inferring glottal noise attributes from the HNR of the speech wave form: (i) the measure is fundamental frequency (f0) dependent for equal levels of glottal noise, and (ii) any deviation from signal periodicity affects the ratio, not just turbulent noise. An alternative harmonics-to-noise ratio formulation [glottal related HNR (GHNR\\')] is proposed to overcome the former problem. In GHNR\\' a mean over the spectral range of interest of the HNRs at specific harmonic\\/between-harmonic frequencies (expressed in linear scale) is calculated. For the latter issue [(ii)] two spectral tilt measures are shown, using synthesis data, to be sensitive to glottal noise while at the same time being comparatively insensitive to other glottal aperiodicities. The theoretical development predicts that the spectral tilt measures reduce as noise levels increase. A conventional HNR estimator, GHNR\\' and two spectral tilt measures are applied to a data set of 13 pathological and 12 normal voice samples. One of the tilt measures and GHNR\\' are shown to provide statistically significant differentiating power over a conventional HNR estimator.

  18. FPGA based, DSP board for LLRF 8-Channel SIMCON 3.0 Part I: Hardware

    Science.gov (United States)

    Giergusiewicz, Wojciech; Koprek, Waldemar; Jalmuzna, Wojciech; Pozniak, Krzysztof T.; Romaniuk, Ryszard S.

    2005-09-01

    The paper describes design, construction and initial measurements of an eight channel electronic LLRF device predicted for building of the control system for the VUV-FEL accelerator at DESY (Hamburg). The device, referred in the paper to as the SIMCON 3.0 (from the SC cavity simulator and controller) consists of a 16 layers, VME size, PCB, a large FPGA chip (VirtexII-4000 by Xilinx), eight fast ADCs and four DACs (by Analog Devices). To our knowledge, the proposed device is the first of this kind for the accelerator technology in which there was achieved (the FPGA based) DSP latency below 200 ns. With the optimized data transmission system, the overall LLRF system latency can be as low as 500 ns. The SIMCON 3.0 sub-system was applied for initial tests with the ACC1 module of the VUV FEL accelerator (eight channels) and with the CHECHIA test stand (single channel), both at the DESY. The promising results with the SIMCON 3.0 encouraged us to enter the design of SIMCON 3.1 possessing 10 measurement and control channels and some additional features to be reported in the next technical note. SIMCON 3.0 is a modular solution, while SIMCON 3.1 will be an integrated board of the all-in-one type. Two design approaches - modular and all-in-one, after branching off in this version of the SIMCON, will be continued.

  19. Low-complexity camera digital signal imaging for video document projection system

    Science.gov (United States)

    Hsia, Shih-Chang; Tsai, Po-Shien

    2011-04-01

    We present high-performance and low-complexity algorithms for real-time camera imaging applications. The main functions of the proposed camera digital signal processing (DSP) involve color interpolation, white balance, adaptive binary processing, auto gain control, and edge and color enhancement for video projection systems. A series of simulations demonstrate that the proposed method can achieve good image quality while keeping computation cost and memory requirements low. On the basis of the proposed algorithms, the cost-effective hardware core is developed using Verilog HDL. The prototype chip has been verified with one low-cost programmable device. The real-time camera system can achieve 1270 × 792 resolution with the combination of extra components and can demonstrate each DSP function.

  20. Clinical evaluation of a membrane-based voice-producing element for female laryngectomized patients

    NARCIS (Netherlands)

    Tack, Johannes W.; Qiu, Qingjun; Schutte, Harm K.; Kooijman, Piet G.C.; Meeuwis, Cees A.; van der Houwen, Eduard B.; Mahieu, Hans F.; Verkerke, Gijsbertus Jacob

    2008-01-01

    Background: A newly developed artificial voice source was clinically evaluated in laryngectomized women for voice quality improvements. The prosthesis was placed in a commercially available, tracheoesophageal shunt valve. - Methods: In 17 subjects, voice-producing element (VPE) prototypes were

  1. A Voice Processing Technology for Rural Specific Context

    Science.gov (United States)

    He, Zhiyong; Zhang, Zhengguang; Zhao, Chunshen

    Durian the promotion and applications of rural information, different geographical dialect voice interaction is a very complex issue. Through in-depth analysis of TTS core technologies, this paper presents the methods of intelligent segmentation, word segmentation algorithm and intelligent voice thesaurus construction in the different dialects context. And then COM based development methodology for specific context voice processing system implementation and programming method. The method has a certain reference value for the rural dialect and voice processing applications.

  2. A BUNCH TO BUCKET PHASE DETECTOR USING DIGITAL RECEIVER TECHNOLOGY

    International Nuclear Information System (INIS)

    DELONG, J.; BRENNAN, J.M.; HAYES, T.; LE, T.N.; SMITH, K.

    2003-01-01

    Transferring high-speed digital signals to a Digital Signal Processor is limited by the IO bandwidth of the DSP. A digital receiver circuit is used to translate high frequency W signals to base-band. The translated output frequency is close to DC and the data rate can be reduced, by decimation, before transfer to the DSP. By translating both the longitudinal beam (bunch) and RF cavity pick-ups (bucket) to DC, a DSP can be used to measure their relative phase angle. The result can be used as an error signal in a beam control servo loop and any phase differences can be compensated

  3. Initial Progress Toward Development of a Voice-Based Computer-Delivered Motivational Intervention for Heavy Drinking College Students: An Experimental Study

    Science.gov (United States)

    Lechner, William J; MacGlashan, James; Wray, Tyler B; Littman, Michael L

    2017-01-01

    Background Computer-delivered interventions have been shown to be effective in reducing alcohol consumption in heavy drinking college students. However, these computer-delivered interventions rely on mouse, keyboard, or touchscreen responses for interactions between the users and the computer-delivered intervention. The principles of motivational interviewing suggest that in-person interventions may be effective, in part, because they encourage individuals to think through and speak aloud their motivations for changing a health behavior, which current computer-delivered interventions do not allow. Objective The objective of this study was to take the initial steps toward development of a voice-based computer-delivered intervention that can ask open-ended questions and respond appropriately to users’ verbal responses, more closely mirroring a human-delivered motivational intervention. Methods We developed (1) a voice-based computer-delivered intervention that was run by a human controller and that allowed participants to speak their responses to scripted prompts delivered by speech generation software and (2) a text-based computer-delivered intervention that relied on the mouse, keyboard, and computer screen for all interactions. We randomized 60 heavy drinking college students to interact with the voice-based computer-delivered intervention and 30 to interact with the text-based computer-delivered intervention and compared their ratings of the systems as well as their motivation to change drinking and their drinking behavior at 1-month follow-up. Results Participants reported that the voice-based computer-delivered intervention engaged positively with them in the session and delivered content in a manner consistent with motivational interviewing principles. At 1-month follow-up, participants in the voice-based computer-delivered intervention condition reported significant decreases in quantity, frequency, and problems associated with drinking, and increased

  4. The Study of Application System for Small and Medium CTI Based on Voice Card

    Directory of Open Access Journals (Sweden)

    Zhong Dong

    2016-01-01

    Full Text Available With the rapid development of computer telecommunications integration (CTI technology, the development of application system for small and medium CTI are updated constantly, but the study of application system for small and medium CTI, we are lack of a stability and unified model. In this paper, the author analyzes the unified structure platform of application system for small and medium CTI based on voice card. Meanwhile, the author introduces a suitable software architecture model and general procedural framework for application system for small and medium CTI based on voice card by using the idea of hierarchical design, which shows the versatility of the architecture. It provided an efficient channel for the development of small and medium CTI.

  5. Voice Quality and Gender Stereotypes: A Study of Lebanese Women With Reinke's Edema.

    Science.gov (United States)

    Matar, Nayla; Portes, Cristel; Lancia, Leonardo; Legou, Thierry; Baider, Fabienne

    2016-12-01

    Women with Reinke's edema (RW) report being mistaken for men during telephone conversations. For this reason, their masculine-sounding voices are interesting for the study of gender stereotypes. The study's objective is to verify their complaint and to understand the cues used in gender identification. Using a self-evaluation study, we verified RW's perception of their own voices. We compared the acoustic parameters of vowels produced by 10 RW to those produced by 10 men and 10 women with healthy voices (hereafter referred to as NW) in Lebanese Arabic. We conducted a perception study for the evaluation of RW, healthy men's, and NW voices by naïve listeners. RW self-evaluated their voices as masculine and their gender identities as feminine. The acoustic parameters that distinguish RW from NW voices concern fundamental frequency, spectral slope, harmonicity of the voicing signal, and complexity of the spectral envelope. Naïve listeners very often rate RW as surely masculine. Listeners may rate RW's gender incorrectly. These incorrect gender ratings are correlated with acoustic measures of fundamental frequency and voice quality. Further investigations will reveal the contribution of each of these parameters to gender perception and guide the treatment plan of patients complaining of a gender ambiguous voice.

  6. Voice-activated intelligent radiologic image display

    International Nuclear Information System (INIS)

    Fisher, P.

    1989-01-01

    The authors present a computer-based expert computer system called Mammo-Icon, which automatically assists the radiologist's case analysis by reviewing the trigger phrase output of a commercially available voice transcription system in he domain of mammography. A commercially available PC-based voice dictation system is coupled to an expert system implemented on a microcomputer. Software employs the LISP and C computer languages. Mammo-Icon responds to the trigger phrase output of a voice dictation system with a textual discussion of the potential significance of the findings that have been described and a display of reference images that may help the radiologist to confirm a suspected diagnosis or consider additional diagnoses. This results in automatic availability of potentially useful computer-based expert advice, making such systems much more likely to be used in routine clinical practice

  7. I like my voice better: self-enhancement bias in perceptions of voice attractiveness.

    Science.gov (United States)

    Hughes, Susan M; Harrison, Marissa A

    2013-01-01

    Previous research shows that the human voice can communicate a wealth of nonsemantic information; preferences for voices can predict health, fertility, and genetic quality of the speaker, and people often use voice attractiveness, in particular, to make these assessments of others. But it is not known what we think of the attractiveness of our own voices as others hear them. In this study eighty men and women rated the attractiveness of an array of voice recordings of different individuals and were not told that their own recorded voices were included in the presentation. Results showed that participants rated their own voices as sounding more attractive than others had rated their voices, and participants also rated their own voices as sounding more attractive than they had rated the voices of others. These findings suggest that people may engage in vocal implicit egotism, a form of self-enhancement.

  8. The STAFF-DWP wave instrument on the DSP equatorial spacecraft: description and first results

    Directory of Open Access Journals (Sweden)

    N. Cornilleau-Wehrlin

    2005-11-01

    Full Text Available The STAFF-DWP wave instrument on board the equatorial spacecraft (TC1 of the Double Star Project consists of a combination of 2 instruments which are a heritage of the Cluster mission: the Spatio-Temporal Analysis of Field Fluctuations (STAFF experiment and the Digital Wave-Processing experiment (DWP. On DSP-TC1 STAFF consists of a three-axis search coil magnetometer, used to measure magnetic fluctuations at frequencies up to 4 kHz and a waveform unit, up to 10 Hz, plus snapshots up to 180 Hz. DWP provides several onboard analysis tools: a complex FFT to fully characterise electromagnetic waves in the frequency range 10 Hz-4 kHz, a particle correlator linked to the PEACE electron experiment, and compression of the STAFF waveform data. The complementary Cluster and TC1 orbits, together with the similarity of the instruments, permits new multi-point studies. The first results show the capabilities of the experiment, with examples in the different regions of the magnetosphere-solar wind system that have been encountered by DSP-TC1 at the beginning of its operational phase. An overview of the different kinds of electromagnetic waves observed on the dayside from perigee to apogee is given, including the different whistler mode waves (hiss, chorus, lion roars and broad-band ULF emissions. The polarisation and propagation characteristics of intense waves in the vicinity of a bow shock crossing are analysed using the dedicated PRASSADCO tool, giving results compatible with previous studies: the broad-band ULF waves consist of a superimposition of different wave modes, whereas the magnetosheath lion roars are right-handed and propagate close to the magnetic field. An example of a combined Cluster DSP-TC1 magnetopause crossing is given. This first case study shows that the ULF wave power intensity is higher at low latitude (DSP than at high latitude (Cluster. On the nightside in the tail, a first wave event comparison - in a rather quiet time interval

  9. Associations between the Transsexual Voice Questionnaire (TVQMtF ) and self-report of voice femininity and acoustic voice measures.

    Science.gov (United States)

    Dacakis, Georgia; Oates, Jennifer; Douglas, Jacinta

    2017-11-01

    The Transsexual Voice Questionnaire (TVQ MtF ) was designed to capture the voice-related perceptions of individuals whose gender identity as female is the opposite of their birth-assigned gender (MtF women). Evaluation of the psychometric properties of the TVQ MtF is ongoing. To investigate associations between TVQ MtF scores and (1) self-perceptions of voice femininity and (2) acoustic parameters of voice pitch and voice quality in order to evaluate further the validity of the TVQ MtF . A strong correlation between TVQ MtF scores and self-ratings of voice femininity was predicted, but no association between TVQ MtF scores and acoustic measures of voice pitch and quality was proposed. Participants were 148 MtF women (mean age 48.14 years) recruited from the La Trobe Communication Clinic and the clinics of three doctors specializing in transgender health. All participants completed the TVQ MtF and 34 of these participants also provided a voice sample for acoustic analysis. Pearson product-moment correlation analysis was conducted to examine the associations between TVQ MtF scores and (1) self-perceptions of voice femininity and (2) acoustic measures of F0, jitter (%), shimmer (dB) and harmonic-to-noise ratio (HNR). Strong negative correlations between the participants' perceptions of their voice femininity and the TVQ MtF scores demonstrated that for this group of MtF women a low self-rating of voice femininity was associated with more frequent negative voice-related experiences. This association was strongest with the vocal-functioning component of the TVQ MtF . These strong correlations and high levels of shared variance between the TVQ MtF and a measure of a related construct provides evidence for the convergent validity of the TVQ MtF . The absence of significant correlations between the TVQ MtF and the acoustic data is consistent with the equivocal findings of earlier research. This finding indicates that these two measures assess different aspects of the voice

  10. Obligatory and facultative brain regions for voice-identity recognition.

    Science.gov (United States)

    Roswandowitz, Claudia; Kappes, Claudia; Obrig, Hellmuth; von Kriegstein, Katharina

    2018-01-01

    Recognizing the identity of others by their voice is an important skill for social interactions. To date, it remains controversial which parts of the brain are critical structures for this skill. Based on neuroimaging findings, standard models of person-identity recognition suggest that the right temporal lobe is the hub for voice-identity recognition. Neuropsychological case studies, however, reported selective deficits of voice-identity recognition in patients predominantly with right inferior parietal lobe lesions. Here, our aim was to work towards resolving the discrepancy between neuroimaging studies and neuropsychological case studies to find out which brain structures are critical for voice-identity recognition in humans. We performed a voxel-based lesion-behaviour mapping study in a cohort of patients (n = 58) with unilateral focal brain lesions. The study included a comprehensive behavioural test battery on voice-identity recognition of newly learned (voice-name, voice-face association learning) and familiar voices (famous voice recognition) as well as visual (face-identity recognition) and acoustic control tests (vocal-pitch and vocal-timbre discrimination). The study also comprised clinically established tests (neuropsychological assessment, audiometry) and high-resolution structural brain images. The three key findings were: (i) a strong association between voice-identity recognition performance and right posterior/mid temporal and right inferior parietal lobe lesions; (ii) a selective association between right posterior/mid temporal lobe lesions and voice-identity recognition performance when face-identity recognition performance was factored out; and (iii) an association of right inferior parietal lobe lesions with tasks requiring the association between voices and faces but not voices and names. The results imply that the right posterior/mid temporal lobe is an obligatory structure for voice-identity recognition, while the inferior parietal lobe is

  11. Enhanced Living by Assessing Voice Pathology Using a Co-Occurrence Matrix

    OpenAIRE

    Muhammad, Ghulam; Alhamid, Mohammed F.; Hossain, M. Shamim; Almogren, Ahmad S.; Vasilakos, Athanasios V.

    2017-01-01

    A large number of the population around the world suffers from various disabilities. Disabilities affect not only children but also adults of different professions. Smart technology can assist the disabled population and lead to a comfortable life in an enhanced living environment (ELE). In this paper, we propose an effective voice pathology assessment system that works in a smart home framework. The proposed system takes input from various sensors, and processes the acquired voice signals an...

  12. Multidimensional assessment of strongly irregular voices such as in substitution voicing and spasmodic dysphonia: a compilation of own research.

    Science.gov (United States)

    Moerman, Mieke; Martens, Jean-Pierre; Dejonckere, Philippe

    2015-04-01

    This article is a compilation of own research performed during the European COoperation in Science and Technology (COST) action 2103: 'Advance Voice Function Assessment', an initiative of voice and speech processing teams consisting of physicists, engineers, and clinicians. This manuscript concerns analyzing largely irregular voicing types, namely substitution voicing (SV) and adductor spasmodic dysphonia (AdSD). A specific perceptual rating scale (IINFVo) was developed, and the Auditory Model Based Pitch Extractor (AMPEX), a piece of software that automatically analyses running speech and generates pitch values in background noise, was applied. The IINFVo perceptual rating scale has been shown to be useful in evaluating SV. The analysis of strongly irregular voices stimulated a modification of the European Laryngological Society's assessment protocol which was originally designed for the common types of (less severe) dysphonia. Acoustic analysis with AMPEX demonstrates that the most informative features are, for SV, the voicing-related acoustic features and, for AdSD, the perturbation measures. Poor correlations between self-assessment and acoustic and perceptual dimensions in the assessment of highly irregular voices argue for a multidimensional approach.

  13. Sparking Passion: Engaging Student Voice through Project-Based Learning in Learning Communities

    Science.gov (United States)

    Ball, Christy L.

    2016-01-01

    How do we confront entrenched educational practices in higher education that lead to student demotivation, poor retention, and low persistence? This article argues that project-based learning that situates student voice and capacity at the center of culturally-responsive curriculum has the potential to spark student passion for problem-solving…

  14. Voice coil based scanning probe microscopy

    Czech Academy of Sciences Publication Activity Database

    Klapetek, P.; Valtr, M.; Duchoň, V.; Sobota, Jaroslav

    2012-01-01

    Roč. 7, č. 6 (2012), 332:1-7 ISSN 1931-7573 R&D Projects: GA MPO FR-TI1/241; GA AV ČR KAN311610701; GA MŠk ED0017/01/01 Institutional support: RVO:68081731 Keywords : SPM * Voice coil * Interferometry Subject RIV: BM - Solid Matter Physics ; Magnetism Impact factor: 2.524, year: 2012

  15. Voice disorders in teachers. A review.

    Science.gov (United States)

    Martins, Regina Helena Garcia; Pereira, Eny Regina Bóia Neves; Hidalgo, Caio Bosque; Tavares, Elaine Lara Mendes

    2014-11-01

    Voice disorders are very prevalent among teachers and consequences are serious. Although the literature is extensive, there are differences in the concepts and methodology related to voice problems; most studies are restricted to analyzing the responses of teachers to questionnaires and only a few studies include vocal assessments and videolaryngoscopic examinations to obtain a definitive diagnosis. To review demographic studies related to vocal disorders in teachers to analyze the diverse methodologies, the prevalence rates pointed out by the authors, the main risk factors, the most prevalent laryngeal lesions, and the repercussions of dysphonias on professional activities. The available literature (from 1997 to 2013) was narratively reviewed based on Medline, PubMed, Lilacs, SciELO, and Cochrane library databases. Excluded were articles that specifically analyzed treatment modalities and those that did not make their abstracts available in those databases. The keywords included were teacher, dysphonia, voice disorders, professional voice. Copyright © 2014 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

  16. Connections between voice ergonomic risk factors in classrooms and teachers' voice production.

    Science.gov (United States)

    Rantala, Leena M; Hakala, Suvi; Holmqvist, Sofia; Sala, Eeva

    2012-01-01

    The aim of the study was to investigate if voice ergonomic risk factors in classrooms correlated with acoustic parameters of teachers' voice production. The voice ergonomic risk factors in the fields of working culture, working postures and indoor air quality were assessed in 40 classrooms using the Voice Ergonomic Assessment in Work Environment - Handbook and Checklist. Teachers (32 females, 8 males) from the above-mentioned classrooms recorded text readings before and after a working day. Fundamental frequency, sound pressure level (SPL) and the slope of the spectrum (alpha ratio) were analyzed. The higher the number of the risk factors in the classrooms, the higher SPL the teachers used and the more strained the males' voices (increased alpha ratio) were. The SPL was already higher before the working day in the teachers with higher risk than in those with lower risk. In the working environment with many voice ergonomic risk factors, speakers increase voice loudness and use more strained voice quality (males). A practical implication of the results is that voice ergonomic assessments are needed in schools. Copyright © 2013 S. Karger AG, Basel.

  17. Detection of Pathological Voice Using Cepstrum Vectors: A Deep Learning Approach.

    Science.gov (United States)

    Fang, Shih-Hau; Tsao, Yu; Hsiao, Min-Jing; Chen, Ji-Ying; Lai, Ying-Hui; Lin, Feng-Chuan; Wang, Chi-Te

    2018-03-19

    Computerized detection of voice disorders has attracted considerable academic and clinical interest in the hope of providing an effective screening method for voice diseases before endoscopic confirmation. This study proposes a deep-learning-based approach to detect pathological voice and examines its performance and utility compared with other automatic classification algorithms. This study retrospectively collected 60 normal voice samples and 402 pathological voice samples of 8 common clinical voice disorders in a voice clinic of a tertiary teaching hospital. We extracted Mel frequency cepstral coefficients from 3-second samples of a sustained vowel. The performances of three machine learning algorithms, namely, deep neural network (DNN), support vector machine, and Gaussian mixture model, were evaluated based on a fivefold cross-validation. Collective cases from the voice disorder database of MEEI (Massachusetts Eye and Ear Infirmary) were used to verify the performance of the classification mechanisms. The experimental results demonstrated that DNN outperforms Gaussian mixture model and support vector machine. Its accuracy in detecting voice pathologies reached 94.26% and 90.52% in male and female subjects, based on three representative Mel frequency cepstral coefficient features. When applied to the MEEI database for validation, the DNN also achieved a higher accuracy (99.32%) than the other two classification algorithms. By stacking several layers of neurons with optimized weights, the proposed DNN algorithm can fully utilize the acoustic features and efficiently differentiate between normal and pathological voice samples. Based on this pilot study, future research may proceed to explore more application of DNN from laboratory and clinical perspectives. Copyright © 2018 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

  18. Static Mapping of Functional Programs: An Example in Signal Processing

    Directory of Open Access Journals (Sweden)

    Jack B. Dennis

    1996-01-01

    Full Text Available Complex signal-processing problems are naturally described by compositions of program modules that process streams of data. In this article we discuss how such compositions may be analyzed and mapped onto multiprocessor computers to effectively exploit the massive parallelism of these applications. The methods are illustrated with an example of signal processing for an optical surveillance problem. Program transformation and analysis are used to construct a program description tree that represents the given computation as an acyclic interconnection of stream-processing modules. Each module may be mapped to a set of threads run on a group of processing elements of a target multiprocessor. Performance is considered for two forms of multiprocessor architecture, one based on conventional DSP technology and the other on a multithreaded-processing element design.

  19. Model based Binaural Enhancement of Voiced and Unvoiced Speech

    DEFF Research Database (Denmark)

    Kavalekalam, Mathew Shaji; Christensen, Mads Græsbøll; Boldt, Jesper B.

    2017-01-01

    This paper deals with the enhancement of speech in presence of non-stationary babble noise. A binaural speech enhancement framework is proposed which takes into account both the voiced and unvoiced speech production model. The usage of this model in enhancement requires the Short term predictor...... (STP) parameters and the pitch information to be estimated. This paper uses a codebook based approach for estimating the STP parameters and a parametric binaural method is proposed for estimating the pitch parameters. Improvements in objective score are shown when using the voicedunvoiced speech model...

  20. The effect of voice quality on hiring decisions

    Directory of Open Access Journals (Sweden)

    Lea Tylečková

    2017-09-01

    Full Text Available This paper examines the effect of voice quality on hiring decisions. Considering voice quality an important tool in an individual’s self-presentation in the job market, it may very well enhance his/her job prospects, while some voice qualities may affect employers’ judgments in a negative way. Five men and five women were recorded reading four different utterances representing answers to job interviewers’ questions in four different phonation guises: modal, breathy, creaky and pressed. 38 professional employment interviewers recorded the speakers’ hireability and personality ratings (likeability, self-confidence and trustworthiness on 7-point semantic differential scales based on the speakers’ voice. The results revealed a significant effect of the phonation guises on the speakers’ ratings with the modal voice being superior to the cluster of non-modal voices. Interestingly, the non-modal guises were evaluated in a very similar way, except for the self-confidence category with the breathy voice getting the lowest scores on the one hand and the pressed voice correlating with high self-confidence ratings on the other.

  1. Speech masking and cancelling and voice obscuration

    Science.gov (United States)

    Holzrichter, John F.

    2013-09-10

    A non-acoustic sensor is used to measure a user's speech and then broadcasts an obscuring acoustic signal diminishing the user's vocal acoustic output intensity and/or distorting the voice sounds making them unintelligible to persons nearby. The non-acoustic sensor is positioned proximate or contacting a user's neck or head skin tissue for sensing speech production information.

  2. Validation and reliability of the sex estimation of the human os coxae using freely available DSP2 software for bioarchaeology and forensic anthropology.

    Science.gov (United States)

    Brůžek, Jaroslav; Santos, Frédéric; Dutailly, Bruno; Murail, Pascal; Cunha, Eugenia

    2017-10-01

    A new tool for skeletal sex estimation based on measurements of the human os coxae is presented using skeletons from a metapopulation of identified adult individuals from twelve independent population samples. For reliable sex estimation, a posterior probability greater than 0.95 was considered to be the classification threshold: below this value, estimates are considered indeterminate. By providing free software, we aim to develop an even more disseminated method for sex estimation. Ten metric variables collected from 2,040 ossa coxa of adult subjects of known sex were recorded between 1986 and 2002 (reference sample). To test both the validity and reliability, a target sample consisting of two series of adult ossa coxa of known sex (n = 623) was used. The DSP2 software (Diagnose Sexuelle Probabiliste v2) is based on Linear Discriminant Analysis, and the posterior probabilities are calculated using an R script. For the reference sample, any combination of four dimensions provides a correct sex estimate in at least 99% of cases. The percentage of individuals for whom sex can be estimated depends on the number of dimensions; for all ten variables it is higher than 90%. Those results are confirmed in the target sample. Our posterior probability threshold of 0.95 for sex estimate corresponds to the traditional sectioning point used in osteological studies. DSP2 software is replacing the former version that should not be used anymore. DSP2 is a robust and reliable technique for sexing adult os coxae, and is also user friendly. © 2017 Wiley Periodicals, Inc.

  3. Initial Progress Toward Development of a Voice-Based Computer-Delivered Motivational Intervention for Heavy Drinking College Students: An Experimental Study.

    Science.gov (United States)

    Kahler, Christopher W; Lechner, William J; MacGlashan, James; Wray, Tyler B; Littman, Michael L

    2017-06-28

    Computer-delivered interventions have been shown to be effective in reducing alcohol consumption in heavy drinking college students. However, these computer-delivered interventions rely on mouse, keyboard, or touchscreen responses for interactions between the users and the computer-delivered intervention. The principles of motivational interviewing suggest that in-person interventions may be effective, in part, because they encourage individuals to think through and speak aloud their motivations for changing a health behavior, which current computer-delivered interventions do not allow. The objective of this study was to take the initial steps toward development of a voice-based computer-delivered intervention that can ask open-ended questions and respond appropriately to users' verbal responses, more closely mirroring a human-delivered motivational intervention. We developed (1) a voice-based computer-delivered intervention that was run by a human controller and that allowed participants to speak their responses to scripted prompts delivered by speech generation software and (2) a text-based computer-delivered intervention that relied on the mouse, keyboard, and computer screen for all interactions. We randomized 60 heavy drinking college students to interact with the voice-based computer-delivered intervention and 30 to interact with the text-based computer-delivered intervention and compared their ratings of the systems as well as their motivation to change drinking and their drinking behavior at 1-month follow-up. Participants reported that the voice-based computer-delivered intervention engaged positively with them in the session and delivered content in a manner consistent with motivational interviewing principles. At 1-month follow-up, participants in the voice-based computer-delivered intervention condition reported significant decreases in quantity, frequency, and problems associated with drinking, and increased perceived importance of changing drinking

  4. Voice Habits and Behaviors: Voice Care Among Flamenco Singers.

    Science.gov (United States)

    Garzón García, Marina; Muñoz López, Juana; Y Mendoza Lara, Elvira

    2017-03-01

    The purpose of this study is to analyze the vocal behavior of flamenco singers, as compared with classical music singers, to establish a differential vocal profile of voice habits and behaviors in flamenco music. Bibliographic review was conducted, and the Singer's Vocal Habits Questionnaire, an experimental tool designed by the authors to gather data regarding hygiene behavior, drinking and smoking habits, type of practice, voice care, and symptomatology perceived in both the singing and the speaking voice, was administered. We interviewed 94 singers, divided into two groups: the flamenco experimental group (FEG, n = 48) and the classical control group (CCG, n = 46). Frequency analysis, a Likert scale, and discriminant and exploratory factor analysis were used to obtain a differential profile for each group. The FEG scored higher than the CCG in speaking voice symptomatology. The FEG scored significantly higher than the CCG in use of "inadequate vocal technique" when singing. Regarding voice habits, the FEG scored higher in "lack of practice and warm-up" and "environmental habits." A total of 92.6% of the subjects classified themselves correctly in each group. The Singer's Vocal Habits Questionnaire has proven effective in differentiating flamenco and classical singers. Flamenco singers are exposed to numerous vocal risk factors that make them more prone to vocal fatigue, mucosa dehydration, phonotrauma, and muscle stiffness than classical singers. Further research is needed in voice training in flamenco music, as a means to strengthen the voice and enable it to meet the requirements of this musical genre. Copyright © 2017 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

  5. Experimental demonstration of adaptive digital monitoring and compensation of chromatic dispersion for coherent DP-QPSK receiver

    DEFF Research Database (Denmark)

    Borkowski, Robert; Zhang, Xu; Zibar, Darko

    2011-01-01

    We experimentally demonstrate a digital signal processing (DSP)-based optical performance monitoring (OPM) algorithm for inservice monitoring of chromatic dispersion (CD) in coherent transport networks. Dispersion accumulated in 40 Gbit/s QPSK signal after 80 km of fiber transmission is successfu...... drives an adaptive digital CD equalizer. © 2011 Optical Society of America.......We experimentally demonstrate a digital signal processing (DSP)-based optical performance monitoring (OPM) algorithm for inservice monitoring of chromatic dispersion (CD) in coherent transport networks. Dispersion accumulated in 40 Gbit/s QPSK signal after 80 km of fiber transmission...

  6. Suggestions for Layout and Functional Behavior of Software-Based Voice Switch Keysets

    Science.gov (United States)

    Scott, David W.

    2010-01-01

    Marshall Space Flight Center (MSFC) provides communication services for a number of real time environments, including Space Shuttle Propulsion support and International Space Station (ISS) payload operations. In such settings, control team members speak with each other via multiple voice circuits or loops. Each loop has a particular purpose and constituency, and users are assigned listen and/or talk capabilities for a given loop based on their role in fulfilling the purpose. A voice switch is a given facility's hardware and software that supports such communication, and may be interconnected with other facilities switches to create a large network that, from an end user perspective, acts like a single system. Since users typically monitor and/or respond to several voice loops concurrently for hours on end and real time operations can be very dynamic and intense, it s vital that a control panel or keyset for interfacing with the voice switch be a servant that reduces stress, not a master that adds it. Implementing the visual interface on a computer screen provides tremendous flexibility and configurability, but there s a very real risk of overcomplication. (Remember how office automation made life easier, which led to a deluge of documents that made life harder?) This paper a) discusses some basic human factors considerations related to keysets implemented as application software windows, b) suggests what to standardize at the facility level and what to leave to the user's preference, and c) provides screen shot mockups for a robust but reasonably simple user experience. Concepts apply to keyset needs in almost any type of operations control or support center.

  7. Sound induced activity in voice sensitive cortex predicts voice memory ability

    Directory of Open Access Journals (Sweden)

    Rebecca eWatson

    2012-04-01

    Full Text Available The ‘temporal voice areas’ (TVAs (Belin et al., 2000 of the human brain show greater neuronal activity in response to human voices than to other categories of nonvocal sounds. However, a direct link between TVA activity and voice perceptionbehaviour has not yet been established. Here we show that a functional magnetic resonance imaging (fMRI measure of activity in the TVAs predicts individual performance at a separately administered voice memory test. This relation holds whengeneral sound memory ability is taken into account. These findings provide the first evidence that the TVAs are specifically involved in voice cognition.

  8. Integrating cues of social interest and voice pitch in men's preferences for women's voices.

    Science.gov (United States)

    Jones, Benedict C; Feinberg, David R; Debruine, Lisa M; Little, Anthony C; Vukovic, Jovana

    2008-04-23

    Most previous studies of vocal attractiveness have focused on preferences for physical characteristics of voices such as pitch. Here we examine the content of vocalizations in interaction with such physical traits, finding that vocal cues of social interest modulate the strength of men's preferences for raised pitch in women's voices. Men showed stronger preferences for raised pitch when judging the voices of women who appeared interested in the listener than when judging the voices of women who appeared relatively disinterested in the listener. These findings show that voice preferences are not determined solely by physical properties of voices and that men integrate information about voice pitch and the degree of social interest expressed by women when forming voice preferences. Women's preferences for raised pitch in women's voices were not modulated by cues of social interest, suggesting that the integration of cues of social interest and voice pitch when men judge the attractiveness of women's voices may reflect adaptations that promote efficient allocation of men's mating effort.

  9. Voice Therapy Practices and Techniques: A Survey of Voice Clinicians.

    Science.gov (United States)

    Mueller, Peter B.; Larson, George W.

    1992-01-01

    Eighty-three voice disorder therapists' ratings of statements regarding voice therapy practices indicated that vocal nodules are the most frequent disorder treated; vocal abuse and hard glottal attack elimination, counseling, and relaxation were preferred treatment approaches; and voice therapy is more effective with adults than with children.…

  10. Digital Signal Processor System for AC Power Drivers

    Directory of Open Access Journals (Sweden)

    Ovidiu Neamtu

    2009-10-01

    Full Text Available DSP (Digital Signal Processor is the bestsolution for motor control systems to make possible thedevelopment of advanced motor drive systems. The motorcontrol processor calculates the required motor windingvoltage magnitude and frequency to operate the motor atthe desired speed. A PWM (Pulse Width Modulationcircuit controls the on and off duty cycle of the powerinverter switches to vary the magnitude of the motorvoltages.

  11. Software design and implementation of ship heave motion monitoring system based on MBD method

    Science.gov (United States)

    Yu, Yan; Li, Yuhan; Zhang, Chunwei; Kang, Won-Hee; Ou, Jinping

    2015-03-01

    Marine transportation plays a significant role in the modern transport sector due to its advantage of low cost, large capacity. It is being attached enormous importance to all over the world. Nowadays the related areas of product development have become an existing hot spot. DSP signal processors feature micro volume, low cost, high precision, fast processing speed, which has been widely used in all kinds of monitoring systems. But traditional DSP code development process is time-consuming, inefficiency, costly and difficult. MathWorks company proposed Model-based Design (MBD) to overcome these defects. By calling the target board modules in simulink library to compile and generate the corresponding code for the target processor. And then automatically call DSP integrated development environment CCS for algorithm validation on the target processor. This paper uses the MDB to design the algorithm for the ship heave motion monitoring system. It proves the effectiveness of the MBD run successfully on the processor.

  12. Noise-Canceling Helmet Audio System

    Science.gov (United States)

    Seibert, Marc A.; Culotta, Anthony J.

    2007-01-01

    A prototype helmet audio system has been developed to improve voice communication for the wearer in a noisy environment. The system was originally intended to be used in a space suit, wherein noise generated by airflow of the spacesuit life-support system can make it difficult for remote listeners to understand the astronaut s speech and can interfere with the astronaut s attempt to issue vocal commands to a voice-controlled robot. The system could be adapted to terrestrial use in helmets of protective suits that are typically worn in noisy settings: examples include biohazard, fire, rescue, and diving suits. The system (see figure) includes an array of microphones and small loudspeakers mounted at fixed positions in a helmet, amplifiers and signal-routing circuitry, and a commercial digital signal processor (DSP). Notwithstanding the fixed positions of the microphones and loudspeakers, the system can accommodate itself to any normal motion of the wearer s head within the helmet. The system operates in conjunction with a radio transceiver. An audio signal arriving via the transceiver intended to be heard by the wearer is adjusted in volume and otherwise conditioned and sent to the loudspeakers. The wearer s speech is collected by the microphones, the outputs of which are logically combined (phased) so as to form a microphone- array directional sensitivity pattern that discriminates in favor of sounds coming from vicinity of the wearer s mouth and against sounds coming from elsewhere. In the DSP, digitized samples of the microphone outputs are processed to filter out airflow noise and to eliminate feedback from the loudspeakers to the microphones. The resulting conditioned version of the wearer s speech signal is sent to the transceiver.

  13. Student voice: An emerging discourse in Irish education policy

    Directory of Open Access Journals (Sweden)

    Domnall Fleming

    2015-09-01

    Full Text Available In positioning student voice within the Irish education policy discourse it is imperative that this emergent and complex concept is explored and theorized in the context of its definition and motivation. Student voice can then be positioned and critiqued as it emerged within Irish education policy primarily following Ireland’s ratification of the United Nations Charter on the Rights of the Child (UNCRC in 1992. Initially emerging in policy from a rights-based and democratic citizenship perspective, the student council became the principal construct for student voice in Irish post-primary schools. While central to the policy discourse, the student council construct has become tokenistic and redundant in practice. School evaluation policy, both external and internal, became a further catalyst for student voice in Ireland. Both processes further challenge and contest the motivation for student voice and point to the concept as an instrument for school improvement and performativity that lacks any centrality for a person-centered, rights-based, dialogic and consultative student voice within an inclusive classroom and school culture.

  14. Large angle and high linearity two-dimensional laser scanner based on voice coil actuators

    Science.gov (United States)

    Wu, Xin; Chen, Sihai; Chen, Wei; Yang, Minghui; Fu, Wen

    2011-10-01

    A large angle and high linearity two-dimensional laser scanner with an in-house ingenious deflection angle detecting system is developed based on voice coil actuators direct driving mechanism. The specially designed voice coil actuators make the steering mirror moving at a sufficiently large angle. Frequency sweep method based on virtual instruments is employed to achieve the natural frequency of the laser scanner. The response shows that the performance of the laser scanner is limited by the mechanical resonances. The closed-loop controller based on mathematical model is used to reduce the oscillation of the laser scanner at resonance frequency. To design a qualified controller, the model of the laser scanner is set up. The transfer function of the model is identified with MATLAB according to the tested data. After introducing of the controller, the nonlinearity decreases from 13.75% to 2.67% at 50 Hz. The laser scanner also has other advantages such as large deflection mirror, small mechanical structure, and high scanning speed.

  15. Voice following radiotherapy

    International Nuclear Information System (INIS)

    Stoicheff, M.L.

    1975-01-01

    This study was undertaken to provide information on the voice of patients following radiotherapy for glottic cancer. Part I presents findings from questionnaires returned by 227 of 235 patients successfully irradiated for glottic cancer from 1960 through 1971. Part II presents preliminary findings on the speaking fundamental frequencies of 22 irradiated patients. Normal to near-normal voice was reported by 83 percent of the 227 patients; however, 80 percent did indicate persisting vocal difficulties such as fatiguing of voice with much usage, inability to sing, reduced loudness, hoarse voice quality and inability to shout. Amount of talking during treatments appeared to affect length of time for voice to recover following treatments in those cases where it took from nine to 26 weeks; also, with increasing years since treatment, patients rated their voices more favorably. Smoking habits following treatments improved significantly with only 27 percent smoking heavily as compared with 65 percent prior to radiation therapy. No correlation was found between smoking (during or after treatments) and vocal ratings or between smoking and length of time for voice to recover. There was no relationship found between reported vocal ratings and stage of the disease

  16. Voices Not Heard: Voice-Use Profiles of Elementary Music Teachers, the Effects of Voice Amplification on Vocal Load, and Perceptions of Issues Surrounding Voice Use

    Science.gov (United States)

    Morrow, Sharon L.

    2009-01-01

    Teachers represent the largest group of occupational voice users and have voice-related problems at a rate of over twice that found in the general population. Among teachers, music teachers are roughly four times more likely than classroom teachers to develop voice-related problems. Although it has been established that music teachers use their…

  17. Dimensionality in voice quality.

    Science.gov (United States)

    Bele, Irene Velsvik

    2007-05-01

    This study concerns speaking voice quality in a group of male teachers (n = 35) and male actors (n = 36), as the purpose was to investigate normal and supranormal voices. The goal was the development of a method of valid perceptual evaluation for normal to supranormal and resonant voices. The voices (text reading at two loudness levels) had been evaluated by 10 listeners, for 15 vocal characteristics using VA scales. In this investigation, the results of an exploratory factor analysis of the vocal characteristics used in this method are presented, reflecting four dimensions of major importance for normal and supranormal voices. Special emphasis is placed on the effects on voice quality of a change in the loudness variable, as two loudness levels are studied. Furthermore, the vocal characteristics Sonority and Ringing voice quality are paid special attention, as the essence of the term "resonant voice" was a basic issue throughout a doctoral dissertation where this study was included.

  18. [Assessment of voice acoustic parameters in female teachers with diagnosed occupational voice disorders].

    Science.gov (United States)

    Niebudek-Bogusz, Ewa; Fiszer, Marta; Sliwińska-Kowalska, Mariola

    2005-01-01

    Laryngovideostroboscopy is the method most frequently used in the assessment of voice disorders. However, the employment of quantitative methods, such as voice acoustic analysis, is essential for evaluating the effectiveness of prophylactic and therapeutic activities as well as for objective medical certification of larynx pathologies. The aim of this study was to examine voice acoustic parameters in female teachers with occupational voice diseases. Acoustic analysis (IRIS software) was performed in 66 female teachers, including 35 teachers with occupational voice diseases and 31 with functional dysphonia. The teachers with occupational voice diseases presented the lower average fundamental frequency (193 Hz) compared to the group with functional dysphonia (209 Hz) and to the normative value (236 Hz), whereas other acoustic parameters did not differ significantly in both groups. Voice acoustic analysis, when applied separately from vocal loading, cannot be used as a testing method to verify the diagnosis of occupational voice disorders.

  19. Partitioning and Scheduling DSP Applications with Maximal Memory Access Hiding

    Directory of Open Access Journals (Sweden)

    Sha Edwin Hsing-Mean

    2002-01-01

    Full Text Available This paper presents an iteration space partitioning scheme to reduce the CPU idle time due to the long memory access latency. We take into consideration both the data accesses of intermediate and initial data. An algorithm is proposed to find the largest overlap for initial data to reduce the entire memory traffic. In order to efficiently hide the memory latency, another algorithm is developed to balance the ALU and memory schedules. The experiments on DSP benchmarks show that the algorithms significantly outperform the known existing methods.

  20. Muscular tension and body posture in relation to voice handicap and voice quality in teachers with persistent voice complaints.

    Science.gov (United States)

    Kooijman, P G C; de Jong, F I C R S; Oudes, M J; Huinck, W; van Acht, H; Graamans, K

    2005-01-01

    The aim of this study was to investigate the relationship between extrinsic laryngeal muscular hypertonicity and deviant body posture on the one hand and voice handicap and voice quality on the other hand in teachers with persistent voice complaints and a history of voice-related absenteeism. The study group consisted of 25 female teachers. A voice therapist assessed extrinsic laryngeal muscular tension and a physical therapist assessed body posture. The assessed parameters were clustered in categories. The parameters in the different categories represent the same function. Further a tension/posture index was created, which is the summation of the different parameters. The different parameters and the index were related to the Voice Handicap Index (VHI) and the Dysphonia Severity Index (DSI). The scores of the VHI and the individual parameters differ significantly except for the posterior weight bearing and tension of the sternocleidomastoid muscle. There was also a significant difference between the individual parameters and the DSI, except for tension of the cricothyroid muscle and posterior weight bearing. The score of the tension/posture index correlates significantly with both the VHI and the DSI. In a linear regression analysis, the combination of hypertonicity of the sternocleidomastoid, the geniohyoid muscles and posterior weight bearing is the most important predictor for a high voice handicap. The combination of hypertonicity of the geniohyoid muscle, posterior weight bearing, high position of the hyoid bone, hypertonicity of the cricothyroid muscle and anteroposition of the head is the most important predictor for a low DSI score. The results of this study show the higher the score of the index, the higher the score of the voice handicap and the worse the voice quality is. Moreover, the results are indicative for the importance of assessment of muscular tension and body posture in the diagnosis of voice disorders.

  1. Mindfulness of voices, self-compassion, and secure attachment in relation to the experience of hearing voices.

    Science.gov (United States)

    Dudley, James; Eames, Catrin; Mulligan, John; Fisher, Naomi

    2018-03-01

    Developing compassion towards oneself has been linked to improvement in many areas of psychological well-being, including psychosis. Furthermore, developing a non-judgemental, accepting way of relating to voices is associated with lower levels of distress for people who hear voices. These factors have also been associated with secure attachment. This study explores associations between the constructs of mindfulness of voices, self-compassion, and distress from hearing voices and how secure attachment style related to each of these variables. Cross-sectional online. One hundred and twenty-eight people (73% female; M age  = 37.5; 87.5% Caucasian) who currently hear voices completed the Self-Compassion Scale, Southampton Mindfulness of Voices Questionnaire, Relationships Questionnaire, and Hamilton Programme for Schizophrenia Voices Questionnaire. Results showed that mindfulness of voices mediated the relationship between self-compassion and severity of voices, and self-compassion mediated the relationship between mindfulness of voices and severity of voices. Self-compassion and mindfulness of voices were significantly positively correlated with each other and negatively correlated with distress and severity of voices. Mindful relation to voices and self-compassion are associated with reduced distress and severity of voices, which supports the proposed potential benefits of mindful relating to voices and self-compassion as therapeutic skills for people experiencing distress by voice hearing. Greater self-compassion and mindfulness of voices were significantly associated with less distress from voices. These findings support theory underlining compassionate mind training. Mindfulness of voices mediated the relationship between self-compassion and distress from voices, indicating a synergistic relationship between the constructs. Although the current findings do not give a direction of causation, consideration is given to the potential impact of mindful and

  2. The Voice Pump: an Affectively Engaging Interface for Changing Attachments

    DEFF Research Database (Denmark)

    Fritsch, Jonas; Jacobsen, Mogens

    2017-01-01

    In this paper, we present the preliminary results from an ongoing interaction design experiment, the Voice Pump. The Voice Pump is an affectively engaging air-based interface for attuning to the differential qualities of voices in order to change attachments between native Danish speakers and non-native...

  3. A Wireless LAN and Voice Information System for Underground Coal Mine

    OpenAIRE

    Yu Zhang; Wei Yang; Dongsheng Han; Young-Il Kim

    2014-01-01

    In this paper we constructed a wireless information system, and developed a wireless voice communication subsystem based on Wireless Local Area Networks (WLAN) for underground coal mine, which employs Voice over IP (VoIP) technology and Session Initiation Protocol (SIP) to achieve wireless voice dispatching communications. The master control voice dispatching interface and call terminal software are also developed on the WLAN ground server side to manage and implement the voice dispatching co...

  4. Voice amplification for primary school teachers with voice disorders: a randomized clinical trial.

    Science.gov (United States)

    Bovo, Roberto; Trevisi, Patrizia; Emanuelli, Enzo; Martini, Alessandro

    2013-06-01

    Several studies have demonstrated a high prevalence of voice disorders in teachers, together with the personal, professional and economical consequences of the problem. Good primary prevention should be based on 3 aspects: 1) amelioration of classroom acoustics, 2) voice care programs for future professional voice users, including teachers and 3) classroom or portable amplification systems. The aim of the study was to assess the benefit obtained from the use of portable amplification systems by female primary school teachers in their occupational setting. Forty female primary school teachers attended a course about professional voice care, which comprised two theoretical lectures, each 60 min long. Thereafter, they were randomized into 2 groups: the teachers of the first group were asked to use a portable vocal amplifier for 3 months, till the end of school-year. The other 20 teachers were part of the control group, matched for age and years of employment. All subjects had a grade 1 of dysphonia with no significant organic lesion of the vocal folds. Most teachers of the experimental group used the amplifier consistently for the whole duration of the experiment and found it very useful in reducing the symptoms of vocal fatigue. In fact, after 3 months, Voice Handicap Index (VHI) scores in "course + amplifier" group demonstrated a significant amelioration (p = 0.003). The perceptual grade of dysphonia also improved significantly (p = 0.0005). The same parameters changed favourably also in the "course only" group, but the results were not statistically significant (p = 0.4 for VHI and p = 0.03 for perceptual grade). In teachers, and particularly in those with a constitutional weak voice and/or those who are prone to vocal fold pathology, vocal amplifiers may be an effective and low-cost intervention to decrease potentially damaging vocal loads and may represent a necessary form of prevention.

  5. "Voice Forum" The Human Voice as Primary Instrument in Music Therapy

    DEFF Research Database (Denmark)

    Pedersen, Inge Nygaard; Storm, Sanne

    2009-01-01

    Aspects will be drawn on the human voice as tool for embodying our psychological and physiological state, and attempting integration of feelings. Presentations and dialogues on different methods and techniques in "Therapy related body-and voice work.", as well as the human voice as a tool for non...

  6. A low-cost, high-performance, digital signal processor-based lock-in amplifier capable of measuring multiple frequency sweeps simultaneously

    International Nuclear Information System (INIS)

    Sonnaillon, Maximiliano Osvaldo; Bonetto, Fabian Jose

    2005-01-01

    A high-performance digital lock-in amplifier implemented in a low-cost digital signal processor (DSP) board is described. This lock in is capable of measuring simultaneously multiple frequencies that change in time as frequency sweeps (chirps). The used 32-bit DSP has enough computing power to generate N=3 simultaneous reference signals and accurately measure the N=3 responses, operating as three lock ins connected in parallel to a linear system. The lock in stores the measured values in memory until they are downloaded to the a personal computer (PC). The lock in works in stand-alone mode and can be programmed and configured through the PC serial port. Downsampling and multiple filter stages were used in order to obtain a sharp roll off and a long time constant in the filters. This makes measurements possible in presence of high-noise levels. Before each measurement, the lock in performs an autocalibration that measures the frequency response of analog output and input circuitry in order to compensate for the departure from ideal operation. Improvements from previous lock-in implementations allow measuring the frequency response of a system in a short time. Furthermore, the proposed implementation can measure how the frequency response changes with time, a characteristic that is very important in our biotechnological application. The number of simultaneous components that the lock in can generate and measure can be extended, without reprogramming, by only using other DSPs of the same family that are code compatible and work at higher clock frequencies

  7. A low-cost, high-performance, digital signal processor-based lock-in amplifier capable of measuring multiple frequency sweeps simultaneously

    Energy Technology Data Exchange (ETDEWEB)

    Sonnaillon, Maximiliano Osvaldo; Bonetto, Fabian Jose [Laboratorio de Cavitacion y Biotecnologia, San Carlos de Bariloche (8400) (Argentina)

    2005-02-01

    A high-performance digital lock-in amplifier implemented in a low-cost digital signal processor (DSP) board is described. This lock in is capable of measuring simultaneously multiple frequencies that change in time as frequency sweeps (chirps). The used 32-bit DSP has enough computing power to generate N=3 simultaneous reference signals and accurately measure the N=3 responses, operating as three lock ins connected in parallel to a linear system. The lock in stores the measured values in memory until they are downloaded to the a personal computer (PC). The lock in works in stand-alone mode and can be programmed and configured through the PC serial port. Downsampling and multiple filter stages were used in order to obtain a sharp roll off and a long time constant in the filters. This makes measurements possible in presence of high-noise levels. Before each measurement, the lock in performs an autocalibration that measures the frequency response of analog output and input circuitry in order to compensate for the departure from ideal operation. Improvements from previous lock-in implementations allow measuring the frequency response of a system in a short time. Furthermore, the proposed implementation can measure how the frequency response changes with time, a characteristic that is very important in our biotechnological application. The number of simultaneous components that the lock in can generate and measure can be extended, without reprogramming, by only using other DSPs of the same family that are code compatible and work at higher clock frequencies.

  8. Does the speaker's voice quality influence children's performance on a language comprehension test?

    Science.gov (United States)

    Lyberg-Åhlander, Viveka; Haake, Magnus; Brännström, Jonas; Schötz, Susanne; Sahlén, Birgitta

    2015-02-01

    A small number of studies have explored children's perception of speakers' voice quality and its possible influence on language comprehension. The aim of this explorative study was to investigate the relationship between the examiner's voice quality, the child's performance on a digital version of a language comprehension test, the Test for Reception of Grammar (TROG-2), and two measures of cognitive functioning. The participants were (n = 86) mainstreamed 8-year old children with typical language development. Two groups of children (n = 41/45) were presented with the TROG-2 through recordings of one female speaker: one group was presented with a typical voice and the other with a simulated dysphonic voice. Significant associations were found between executive functioning and language comprehension. The results also showed that children listening to the dysphonic voice achieved significantly lower scores for more difficult sentences ("the man but not the horse jumps") and used more self-corrections on simpler sentences ("the girl is sitting"). Findings suggest that a dysphonic speaker's voice may force the child to allocate capacity to the processing of the voice signal at the expense of comprehension. The findings have implications for clinical and research settings where standardized language tests are used.

  9. Rapid Prototyping of High Performance Signal Processing Applications

    Science.gov (United States)

    Sane, Nimish

    Advances in embedded systems for digital signal processing (DSP) are enabling many scientific projects and commercial applications. At the same time, these applications are key to driving advances in many important kinds of computing platforms. In this region of high performance DSP, rapid prototyping is critical for faster time-to-market (e.g., in the wireless communications industry) or time-to-science (e.g., in radio astronomy). DSP system architectures have evolved from being based on application specific integrated circuits (ASICs) to incorporate reconfigurable off-the-shelf field programmable gate arrays (FPGAs), the latest multiprocessors such as graphics processing units (GPUs), or heterogeneous combinations of such devices. We, thus, have a vast design space to explore based on performance trade-offs, and expanded by the multitude of possibilities for target platforms. In order to allow systematic design space exploration, and develop scalable and portable prototypes, model based design tools are increasingly used in design and implementation of embedded systems. These tools allow scalable high-level representations, model based semantics for analysis and optimization, and portable implementations that can be verified at higher levels of abstractions and targeted toward multiple platforms for implementation. The designer can experiment using such tools at an early stage in the design cycle, and employ the latest hardware at later stages. In this thesis, we have focused on dataflow-based approaches for rapid DSP system prototyping. This thesis contributes to various aspects of dataflow-based design flows and tools as follows: 1. We have introduced the concept of topological patterns, which exploits commonly found repetitive patterns in DSP algorithms to allow scalable, concise, and parameterizable representations of large scale dataflow graphs in high-level languages. We have shown how an underlying design tool can systematically exploit a high

  10. Short- and long-term performance of a tripolar down-sized single lead for implantable cardioverter defibrillator treatment: a randomized prospective European multicenter study. European Endotak DSP Investigator Group.

    Science.gov (United States)

    Sandstedt, B; Kennergren, C; Schaumann, A; Herse, B; Neuzner, J

    1998-11-01

    A new, thinner (10 Fr) and more flexible, single-pass transvenous endocardial ICD lead, Endotak DSP, was compared with a conventional lead, Endotak C, as a control in a prospective randomized multicenter study in combination with a nonactive can ICD. A total of 123 patients were enrolled, 55 of whom received a down-sized DSP lead. Lead-alone configuration was successfully implanted in 95% of the DSP patients vs 88% in the control group. The mean defibrillation threshold (DFT) was determined by means of a step-down protocol, and was identical in the two groups, 10.5 +/- 4.8 J in the DSP group versus 10.5 +/- 4.8 J in the control group. At implantation, the DSP mean pacing threshold was lower, 0.51 +/- 0.18 V versus 0.62 +/- 0.35 V (p < 0.05) in the control group, and the mean pacing impedance higher, 594 +/- 110 omega vs 523 +/- 135 omega (p < 0.05). During the follow-up period, the statistically significant difference in thresholds disappeared, while the difference in impedance remained. Tachyarrhythmia treatment by shock or antitachycardia pacing (ATP) was delivered in 53% and 41%, respectively, of the patients with a 100% success rate. In the DSP group, all 28 episodes of polymorphic ventricular tachycardia or ventricular fibrillation were converted by the first shock as compared to 57 of 69 episodes (83%) in the control group (p < 0.05). Monomorphic ventricular tachycardias were terminated by ATP alone in 96% versus 94%. Lead related problems were minor and observed in 5% and 7%, respectively. In summary, both leads were safe and efficacious in the detection and treatment of ventricular tachyarrhythmias. There were no differences between the DSP and control groups regarding short- or long-term lead related complications.

  11. Voice Use Among Music Theory Teachers: A Voice Dosimetry and Self-Assessment Study.

    Science.gov (United States)

    Schiller, Isabel S; Morsomme, Dominique; Remacle, Angélique

    2017-07-25

    This study aimed (1) to investigate music theory teachers' professional and extra-professional vocal loading and background noise exposure, (2) to determine the correlation between vocal loading and background noise, and (3) to determine the correlation between vocal loading and self-evaluation data. Using voice dosimetry, 13 music theory teachers were monitored for one workweek. The parameters analyzed were voice sound pressure level (SPL), fundamental frequency (F0), phonation time, vocal loading index (VLI), and noise SPL. Spearman correlation was used to correlate vocal loading parameters (voice SPL, F0, and phonation time) and noise SPL. Each day, the subjects self-assessed their voice using visual analog scales. VLI and self-evaluation data were correlated using Spearman correlation. Vocal loading parameters and noise SPL were significantly higher in the professional than in the extra-professional environment. Voice SPL, phonation time, and female subjects' F0 correlated positively with noise SPL. VLI correlated with self-assessed voice quality, vocal fatigue, and amount of singing and speaking voice produced. Teaching music theory is a profession with high vocal demands. More background noise is associated with increased vocal loading and may indirectly increase the risk for voice disorders. Correlations between VLI and self-assessments suggest that these teachers are well aware of their vocal demands and feel their effect on voice quality and vocal fatigue. Visual analog scales seem to represent a useful tool for subjective vocal loading assessment and associated symptoms in these professional voice users. Copyright © 2017 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

  12. Sustainable Consumer Voices

    DEFF Research Database (Denmark)

    Klitmøller, Anders; Rask, Morten; Jensen, Nevena

    2011-01-01

    Aiming to explore how user driven innovation can inform high level design strategies, an in-depth empirical study was carried out, based on data from 50 observations of private vehicle users. This paper reports the resulting 5 consumer voices: Technology Enthusiast, Environmentalist, Design Lover...

  13. Marshall’s Voice

    Directory of Open Access Journals (Sweden)

    Halper Thomas

    2017-12-01

    Full Text Available Most judicial opinions, for a variety of reasons, do not speak with the voice of identifiable judges, but an analysis of several of John Marshall’s best known opinions reveals a distinctive voice, with its characteristic language and style of argumentation. The power of this voice helps to account for the influence of his views.

  14. Identification and analysis of OsttaDSP, a phosphoglucan phosphatase from Ostreococcus tauri.

    Directory of Open Access Journals (Sweden)

    Julieta B Carrillo

    Full Text Available Ostreococcus tauri, the smallest free-living (non-symbiotic eukaryote yet described, is a unicellular green alga of the Prasinophyceae family. It has a very simple cellular organization and presents a unique starch granule and chloroplast. However, its starch metabolism exhibits a complexity comparable to higher plants, with multiple enzyme forms for each metabolic reaction. Glucan phosphatases, a family of enzymes functionally conserved in animals and plants, are essential for normal starch or glycogen degradation in plants and mammals, respectively. Despite the importance of O. tauri microalgae in evolution, there is no information available concerning the enzymes involved in reversible phosphorylation of starch. Here, we report the molecular cloning and heterologous expression of the gene coding for a dual specific phosphatase from O. tauri (OsttaDSP, homologous to Arabidopsis thaliana LSF2. The recombinant enzyme was purified to electrophoretic homogeneity to characterize its oligomeric and kinetic properties accurately. OsttaDSP is a homodimer of 54.5 kDa that binds and dephosphorylates amylopectin. Also, we also determined that residue C162 is involved in catalysis and possibly also in structural stability of the enzyme. Our results could contribute to better understand the role of glucan phosphatases in the metabolism of starch in green algae.

  15. Digital Signal Processing for a Sliceable Transceiver for Optical Access Networks

    DEFF Research Database (Denmark)

    Saldaña Cercos, Silvia; Wagner, Christoph; Vegas Olmos, Juan José

    2015-01-01

    Methods to upgrade the network infrastructure to cope with current traffic demands has attracted increasing research efforts. A promising alternative is signal slicing. Signal slicing aims at re-using low bandwidth equipment to satisfy high bandwidth traffic demands. This technique has been used...... also for implementing full signal path symmetry in real-time oscilloscopes to provide performance and signal fidelity (i.e. lower noise and jitter). In this paper the key digital signal processing (DSP) subsystems required to achieve signal slicing are surveyed. It also presents, for the first time...... penalty is reported for 10 Gbps. Power savings of the order of hundreds of Watts can be obtained when using signal slicing as an alternative to 10 Gbps implemented access networks....

  16. Neonatal co-lesion by DSP-4 and 5,7-DHT produces adulthood behavioral sensitization to dopamine D(2) receptor agonists.

    Science.gov (United States)

    Nowak, Przemysław; Nitka, Dariusz; Kwieciński, Adam; Jośko, Jadwiga; Drab, Jacek; Pojda-Wilczek, Dorota; Kasperski, Jacek; Kostrzewa, Richard M; Brus, Ryszard

    2009-01-01

    To assess the possible modulatory effects of noradrenergic and serotoninergic neurons on dopaminergic neuronal activity, the noradrenergic and serotoninergic neurotoxins DSP-4 N-(2-chlorethyl)-N-ethyl-2-bromobenzylamine (50.0 mg/kg, sc) and 5,7-dihydroxytryptamine (5,7-DHT) (37.5 microg icv, half in each lateral ventricle), respectively, were administered toWistar rats on the first and third days of postnatal ontogeny, and dopamine (DA) agonist-induced behaviors were assessed in adulthood. At eight weeks, using an HPLC/ED technique, DSP-4 treatment was associated with a reduction in NE content of the corpus striatum (> 60%), hippocampus (95%), and frontal cortex (> 85%), while 5,7-DHT was associated with an 80-90% serotonin reduction in the same brain regions. DA content was unaltered in the striatum and the cortex. In the group lesioned with both DSP-4 and 5,7-DHT, quinpirole-induced (DA D(2) agonist) yawning, 7-hydroxy-DPAT-induced (DA D(3) agonist) yawning, and apomorphine-induced (non-selective DA agonist) stereotypies were enhanced. However, SKF 38393-induced (DA D(1) agonist) oral activity was reduced in the DSP-4 + 5,7-DHT group. These findings demonstrate that DA D(2)- and D(3)-agonist-induced behaviors are enhanced while DA D(1)-agonist-induced behaviors are suppressed in adult rats in which brain noradrenergic and serotoninergic innervation of the brain has largely been destroyed. This study indicates that noradrenergic and serotoninergic neurons have a great impact on the development of DA receptor reactivity (sensitivity).

  17. Investigating the Effects of Glottal Stop Productions on Voice in Children With Cleft Palate Using Multidimensional Voice Assessment Methods.

    Science.gov (United States)

    Aydınlı, Fatma Esen; Özcebe, Esra; Kulak Kayıkçı, Maviş E; Yılmaz, Taner; Özgür, Fatma F

    2016-11-01

    The aim was to investigate the effects of glottal stop productions (GS) on voice in children with cleft palate using multidimensional voice assessment methods. This is a prospective case-control study. Children with repaired cleft palate (n = 34) who did not have any vocal fold lesions were separated into two groups based on the results of the articulation test. The glottal stop group (GSG) consisted of 17 children who had GS. The control group (CG) consisted of an equal number of age- and gender-matched children who did not have GS. The voice evaluation protocol included acoustic analysis, Pediatric Voice Handicap Index (pVHI), and perceptual analysis (Grade, Roughness, Breathiness, Asthenia, Strain method). The velopharyngeal statuses of the groups were compared using the nasopharyngoscopy and the nasometer. The total pVHI score and the subscales of the pVHI were found to be significantly higher in the GSG. The F0, jitter, and shimmer were found to be numerically higher in the GSG with the difference being statistically significant in jitter (P speech and language pathology intervention including voice therapy techniques. Copyright © 2016 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

  18. A pneumatic Bionic Voice prosthesis-Pre-clinical trials of controlling the voice onset and offset.

    Science.gov (United States)

    Ahmadi, Farzaneh; Noorian, Farzad; Novakovic, Daniel; van Schaik, André

    2018-01-01

    Despite emergent progress in many fields of bionics, a functional Bionic Voice prosthesis for laryngectomy patients (larynx amputees) has not yet been achieved, leading to a lifetime of vocal disability for these patients. This study introduces a novel framework of Pneumatic Bionic Voice Prostheses as an electronic adaptation of the Pneumatic Artificial Larynx (PAL) device. The PAL is a non-invasive mechanical voice source, driven exclusively by respiration with an exceptionally high voice quality, comparable to the existing gold standard of Tracheoesophageal (TE) voice prosthesis. Following PAL design closely as the reference, Pneumatic Bionic Voice Prostheses seem to have a strong potential to substitute the existing gold standard by generating a similar voice quality while remaining non-invasive and non-surgical. This paper designs the first Pneumatic Bionic Voice prosthesis and evaluates its onset and offset control against the PAL device through pre-clinical trials on one laryngectomy patient. The evaluation on a database of more than five hours of continuous/isolated speech recordings shows a close match between the onset/offset control of the Pneumatic Bionic Voice and the PAL with an accuracy of 98.45 ±0.54%. When implemented in real-time, the Pneumatic Bionic Voice prosthesis controller has an average onset/offset delay of 10 milliseconds compared to the PAL. Hence it addresses a major disadvantage of previous electronic voice prostheses, including myoelectric Bionic Voice, in meeting the short time-frames of controlling the onset/offset of the voice in continuous speech.

  19. The singer's voice range profile: female professional opera soloists.

    Science.gov (United States)

    Lamarche, Anick; Ternström, Sten; Pabon, Peter

    2010-07-01

    This work concerns the collection of 30 voice range profiles (VRPs) of female operatic voice. We address the questions: Is there a need for a singer's protocol in VRP acquisition? Are physiological measurements sufficient or should the measurement of performance capabilities also be included? Can we address the female singing voice in general or is there a case for categorizing voices when studying phonetographic data? Subjects performed a series of structured tasks involving both standard speech voice protocols and additional singing tasks. Singers also completed an extensive questionnaire. Physiological VRPs differ from performance VRPs. Two new VRP metrics, the voice area above a defined level threshold and the dynamic range independent from the fundamental frequency (F(0)), were found to be useful in the analysis of singer VRPs. Task design had no effect on performance VRP outcomes. Voice category differences were mainly attributable to phonation frequency-based information. Results support the clinical importance of addressing the vocal instrument as it is used in performance. Equally important is the elaboration of a protocol suitable for the singing voice. The given context and instructions can be more important than task design for performance VRPs. Yet, for physiological VRP recordings, task design remains critical. Both types of VRPs are suggested for a singer's voice evaluation. Copyright (c) 2010 The Voice Foundation. Published by Mosby, Inc. All rights reserved.

  20. Voice similarity in identical twins.

    Science.gov (United States)

    Van Gysel, W D; Vercammen, J; Debruyne, F

    2001-01-01

    If people are asked to discriminate visually the two individuals of a monozygotic twin (MT), they mostly get into trouble. Does this problem also exist when listening to twin voices? Twenty female and 10 male MT voices were randomly assembled with one "strange" voice to get voice trios. The listeners (10 female students in Speech and Language Pathology) were asked to label the twins (voices 1-2, 1-3 or 2-3) in two conditions: two standard sentences read aloud and a 2.5-second midsection of a sustained /a/. The proportion correctly labelled twins was for female voices 82% and 63% and for male voices 74% and 52% for the sentences and the sustained /a/ respectively, both being significantly greater than chance (33%). The acoustic analysis revealed a high intra-twin correlation for the speaking fundamental frequency (SFF) of the sentences and the fundamental frequency (F0) of the sustained /a/. So the voice pitch could have been a useful characteristic in the perceptual identification of the twins. We conclude that there is a greater perceptual resemblance between the voices of identical twins than between voices without genetic relationship. The identification however is not perfect. The voice pitch possibly contributes to the correct twin identifications.

  1. Tips for Healthy Voices

    Science.gov (United States)

    ... prevent voice problems and maintain a healthy voice: Drink water (stay well hydrated): Keeping your body well hydrated by drinking plenty of water each day (6-8 glasses) is essential to maintaining a healthy voice. The ...

  2. Your Cheatin' Voice Will Tell on You: Detection of Past Infidelity from Voice.

    Science.gov (United States)

    Hughes, Susan M; Harrison, Marissa A

    2017-01-01

    Evidence suggests that many physical, behavioral, and trait qualities can be detected solely from the sound of a person's voice, irrespective of the semantic information conveyed through speech. This study examined whether raters could accurately assess the likelihood that a person has cheated on committed, romantic partners simply by hearing the speaker's voice. Independent raters heard voice samples of individuals who self-reported that they either cheated or had never cheated on their romantic partners. To control for aspects that may clue a listener to the speaker's mate value, we used voice samples that did not differ between these groups for voice attractiveness, age, voice pitch, and other acoustic measures. We found that participants indeed rated the voices of those who had a history of cheating as more likely to cheat. Male speakers were given higher ratings for cheating, while female raters were more likely to ascribe the likelihood to cheat to speakers. Additionally, we manipulated the pitch of the voice samples, and for both sexes, the lower pitched versions were consistently rated to be from those who were more likely to have cheated. Regardless of the pitch manipulation, speakers were able to assess actual history of infidelity; the one exception was that men's accuracy decreased when judging women whose voices were lowered. These findings expand upon the idea that the human voice may be of value as a cheater detection tool and very thin slices of vocal information are all that is needed to make certain assessments about others.

  3. Voice-band Modems: A Device to Transmit Data Over Telephone ...

    Indian Academy of Sciences (India)

    Voice-band Modems: A Device to Transmit Data. Over Telephone Networks. 1. Basic Principles of Data Trans.mission v U Reddy is with the. Electrical Communica- tion Engineering. Department, Indian. Institute of Science. His research areas are adaptive signal process- ing, multirate filtering and wavelets, and multi-.

  4. Perceiving a stranger's voice as being one's own: a 'rubber voice' illusion?

    Directory of Open Access Journals (Sweden)

    Zane Z Zheng

    2011-04-01

    Full Text Available We describe an illusion in which a stranger's voice, when presented as the auditory concomitant of a participant's own speech, is perceived as a modified version of their own voice. When the congruence between utterance and feedback breaks down, the illusion is also broken. Compared to a baseline condition in which participants heard their own voice as feedback, hearing a stranger's voice induced robust changes in the fundamental frequency (F0 of their production. Moreover, the shift in F0 appears to be feedback dependent, since shift patterns depended reliably on the relationship between the participant's own F0 and the stranger-voice F0. The shift in F0 was evident both when the illusion was present and after it was broken, suggesting that auditory feedback from production may be used separately for self-recognition and for vocal motor control. Our findings indicate that self-recognition of voices, like other body attributes, is malleable and context dependent.

  5. 基于DSP的虚拟仪器设计与实现%Design and Implementation of Virtual Instrument Based on DSP

    Institute of Scientific and Technical Information of China (English)

    智伟敏; 朱德森; 贺新华

    2001-01-01

    主要介绍虚拟示波器的开发.讨论了以数字信号处理器(DSP)为核心的虚拟示波器硬件数据采集板卡的结构及各器件之间的逻辑关系,阐述了软件的组成与功能,提出了在印染机械控制系统中进行故障检测的新方案.

  6. Unfamiliar voice identification: Effect of post-event information on accuracy and voice ratings

    Directory of Open Access Journals (Sweden)

    Harriet Mary Jessica Smith

    2014-04-01

    Full Text Available This study addressed the effect of misleading post-event information (PEI on voice ratings, identification accuracy, and confidence, as well as the link between verbal recall and accuracy. Participants listened to a dialogue between male and female targets, then read misleading information about voice pitch. Participants engaged in verbal recall, rated voices on a feature checklist, and made a lineup decision. Accuracy rates were low, especially on target-absent lineups. Confidence and accuracy were unrelated, but the number of facts recalled about the voice predicted later lineup accuracy. There was a main effect of misinformation on ratings of target voice pitch, but there was no effect on identification accuracy or confidence ratings. As voice lineup evidence from earwitnesses is used in courts, the findings have potential applied relevance.

  7. A pneumatic Bionic Voice prosthesis-Pre-clinical trials of controlling the voice onset and offset.

    Directory of Open Access Journals (Sweden)

    Farzaneh Ahmadi

    Full Text Available Despite emergent progress in many fields of bionics, a functional Bionic Voice prosthesis for laryngectomy patients (larynx amputees has not yet been achieved, leading to a lifetime of vocal disability for these patients. This study introduces a novel framework of Pneumatic Bionic Voice Prostheses as an electronic adaptation of the Pneumatic Artificial Larynx (PAL device. The PAL is a non-invasive mechanical voice source, driven exclusively by respiration with an exceptionally high voice quality, comparable to the existing gold standard of Tracheoesophageal (TE voice prosthesis. Following PAL design closely as the reference, Pneumatic Bionic Voice Prostheses seem to have a strong potential to substitute the existing gold standard by generating a similar voice quality while remaining non-invasive and non-surgical. This paper designs the first Pneumatic Bionic Voice prosthesis and evaluates its onset and offset control against the PAL device through pre-clinical trials on one laryngectomy patient. The evaluation on a database of more than five hours of continuous/isolated speech recordings shows a close match between the onset/offset control of the Pneumatic Bionic Voice and the PAL with an accuracy of 98.45 ±0.54%. When implemented in real-time, the Pneumatic Bionic Voice prosthesis controller has an average onset/offset delay of 10 milliseconds compared to the PAL. Hence it addresses a major disadvantage of previous electronic voice prostheses, including myoelectric Bionic Voice, in meeting the short time-frames of controlling the onset/offset of the voice in continuous speech.

  8. A pneumatic Bionic Voice prosthesis—Pre-clinical trials of controlling the voice onset and offset

    Science.gov (United States)

    Noorian, Farzad; Novakovic, Daniel; van Schaik, André

    2018-01-01

    Despite emergent progress in many fields of bionics, a functional Bionic Voice prosthesis for laryngectomy patients (larynx amputees) has not yet been achieved, leading to a lifetime of vocal disability for these patients. This study introduces a novel framework of Pneumatic Bionic Voice Prostheses as an electronic adaptation of the Pneumatic Artificial Larynx (PAL) device. The PAL is a non-invasive mechanical voice source, driven exclusively by respiration with an exceptionally high voice quality, comparable to the existing gold standard of Tracheoesophageal (TE) voice prosthesis. Following PAL design closely as the reference, Pneumatic Bionic Voice Prostheses seem to have a strong potential to substitute the existing gold standard by generating a similar voice quality while remaining non-invasive and non-surgical. This paper designs the first Pneumatic Bionic Voice prosthesis and evaluates its onset and offset control against the PAL device through pre-clinical trials on one laryngectomy patient. The evaluation on a database of more than five hours of continuous/isolated speech recordings shows a close match between the onset/offset control of the Pneumatic Bionic Voice and the PAL with an accuracy of 98.45 ±0.54%. When implemented in real-time, the Pneumatic Bionic Voice prosthesis controller has an average onset/offset delay of 10 milliseconds compared to the PAL. Hence it addresses a major disadvantage of previous electronic voice prostheses, including myoelectric Bionic Voice, in meeting the short time-frames of controlling the onset/offset of the voice in continuous speech. PMID:29466455

  9. Auditory and Visual Modulation of Temporal Lobe Neurons in Voice-Sensitive and Association Cortices

    Science.gov (United States)

    Perrodin, Catherine; Kayser, Christoph; Logothetis, Nikos K.

    2014-01-01

    Effective interactions between conspecific individuals can depend upon the receiver forming a coherent multisensory representation of communication signals, such as merging voice and face content. Neuroimaging studies have identified face- or voice-sensitive areas (Belin et al., 2000; Petkov et al., 2008; Tsao et al., 2008), some of which have been proposed as candidate regions for face and voice integration (von Kriegstein et al., 2005). However, it was unclear how multisensory influences occur at the neuronal level within voice- or face-sensitive regions, especially compared with classically defined multisensory regions in temporal association cortex (Stein and Stanford, 2008). Here, we characterize auditory (voice) and visual (face) influences on neuronal responses in a right-hemisphere voice-sensitive region in the anterior supratemporal plane (STP) of Rhesus macaques. These results were compared with those in the neighboring superior temporal sulcus (STS). Within the STP, our results show auditory sensitivity to several vocal features, which was not evident in STS units. We also newly identify a functionally distinct neuronal subpopulation in the STP that appears to carry the area's sensitivity to voice identity related features. Audiovisual interactions were prominent in both the STP and STS. However, visual influences modulated the responses of STS neurons with greater specificity and were more often associated with congruent voice-face stimulus pairings than STP neurons. Together, the results reveal the neuronal processes subserving voice-sensitive fMRI activity patterns in primates, generate hypotheses for testing in the visual modality, and clarify the position of voice-sensitive areas within the unisensory and multisensory processing hierarchies. PMID:24523543

  10. Auditory and visual modulation of temporal lobe neurons in voice-sensitive and association cortices.

    Science.gov (United States)

    Perrodin, Catherine; Kayser, Christoph; Logothetis, Nikos K; Petkov, Christopher I

    2014-02-12

    Effective interactions between conspecific individuals can depend upon the receiver forming a coherent multisensory representation of communication signals, such as merging voice and face content. Neuroimaging studies have identified face- or voice-sensitive areas (Belin et al., 2000; Petkov et al., 2008; Tsao et al., 2008), some of which have been proposed as candidate regions for face and voice integration (von Kriegstein et al., 2005). However, it was unclear how multisensory influences occur at the neuronal level within voice- or face-sensitive regions, especially compared with classically defined multisensory regions in temporal association cortex (Stein and Stanford, 2008). Here, we characterize auditory (voice) and visual (face) influences on neuronal responses in a right-hemisphere voice-sensitive region in the anterior supratemporal plane (STP) of Rhesus macaques. These results were compared with those in the neighboring superior temporal sulcus (STS). Within the STP, our results show auditory sensitivity to several vocal features, which was not evident in STS units. We also newly identify a functionally distinct neuronal subpopulation in the STP that appears to carry the area's sensitivity to voice identity related features. Audiovisual interactions were prominent in both the STP and STS. However, visual influences modulated the responses of STS neurons with greater specificity and were more often associated with congruent voice-face stimulus pairings than STP neurons. Together, the results reveal the neuronal processes subserving voice-sensitive fMRI activity patterns in primates, generate hypotheses for testing in the visual modality, and clarify the position of voice-sensitive areas within the unisensory and multisensory processing hierarchies.

  11. FPGA based, DSP integrated, 8-channel SIMCON, ver. 3.0. Initial results for 8-channel algorithm

    Energy Technology Data Exchange (ETDEWEB)

    Giergusiewicz, W.; Koprek, W.; Jalmuzna, W.; Pozniak, K.T.; Romaniuk, R.S. [Warsaw Univ. of Technology (Poland). Inst. of Electronic Systems

    2005-07-01

    The paper describes design, construction and initial measurements of an eight channel electronic LLRF device predicted for building of the control system for the VUV-FEL accelerator at DESY (Hamburg). The device, referred in the paper to as the SIMCON 3.0 (from the SC cavity simulator and controller) consists of a 16 layer, VME size, PCB, a large FPGA chip (VirtexII-4000 by Xilinx), eight fast ADCs and four DACs (by Analog Devices). To our knowledge, the proposed device is the first of this kind for the accelerator technology in which there was achieved (the FPGA based) DSP latency below 200 ns. With the optimized data transmission system, the overall LLRF system latency can be as low as 500 ns. The SIMCON 3.0 sub-system was applied for initial tests with the ACC1 module of the VUV FEL accelerator (eight channels) and with the CHECHIA test stand (single channel), both at the DESY. The promising results with the SIMCON 3.0. encouraged us to enter the design of SIMCON 3.1. possessing 10 measurement and control channels and some additional features to be reported in the next technical note. SIMCON 3.0. is a modular solution, while SIMCON 3.1. will be an integrated board of the all-in-one type. Two design approaches - modular and all-in-one, after branching off in this version of the Simcon, will be continued. (orig.)

  12. Bilateral Manipulandum to Synthesize Ground Referenced and Interlimb Viscoelastic Loads

    National Research Council Canada - National Science Library

    Gallasch, E

    2001-01-01

    .... The mechatronics consists of two angular voice coil actuators (+/- 40 Nm) with embedded rotary (+/- 20 degrees) and torque sensors driven by voltage controlled current sources, DSP software routines to synthesize isotonic...

  13. Multi-dimensional Analysis for SLB Transient in ATLAS Facility as Activity of DSP (Domestic Standard Problem)

    International Nuclear Information System (INIS)

    Bae, B. U.; Park, Y. S.; Kim, J. R.; Kang, K. H.; Choi, K. Y.; Sung, H. J.; Hwang, M. J.; Kang, D. H.; Lim, S. G.; Jun, S. S.

    2015-01-01

    Participants of DSP-03 were divided in three groups and each group has focused on the specific subject related to the enhancement of the code analysis. The group A tried to investigate scaling capability of ATLAS test data by comparing to the code analysis for a prototype, and the group C studied to investigate effect of various models in the one-dimensional codes. This paper briefly summarizes the code analysis result from the group B participants in the DSP-03 of the ATLAS test facility. The code analysis by Group B focuses highly on investigating the multi-dimensional thermal hydraulic phenomena in the ATLAS facility during the SLB transient. Even though the one-dimensional system analysis code cannot simulate the whole system of the ATLAS facility with a nodalization of the CFD (Computational Fluid Dynamics) scale, a reactor pressure vessel can be considered with multi-dimensional components to reflect the thermal mixing phenomena inside a downcomer and a core. Also, the CFD could give useful information for understanding complex phenomena in specific components such as the reactor pressure vessel. From the analysis activity of Group B in ATLAS DSP-03, participants adopted a multi-dimensional approach to the code analysis for the SLB transient in the ATLAS test facility. The main purpose of the analysis was to investigate prediction capability of multi-dimensional analysis tools for the SLB experiment result. In particular, the asymmetric cooling and thermal mixing phenomena in the reactor pressure vessel could be significantly focused for modeling the multi-dimensional components

  14. SEU mitigation technique by Dynamic Reconfiguration method in FPGA based DSP application

    International Nuclear Information System (INIS)

    Dey, Madhusudan; Singh, Abhishek; Roy, Amitava

    2012-01-01

    Field Programmable Gate Array (FPGA), an SRAM based configurable devices meant for implementation of any digital circuits is susceptible to malfunction in the harsh radiation environment. It causes the corruption of the configuration memory of FPGA and the digital circuits starts malfunctioning. There is a need to restore the system as early as possible. This paper discusses about one such technique named dynamic partial reconfiguration (DPR) method. This paper also touches upon the signal processing by DPR method. The framework consisting of ADC, DAC and ICAP controllers designed using dedicated state machines to study the best possible downtime also for verifying the performance of digital filters for signal processing

  15. Performance evaluation of the QIAGEN EZ1 DSP Virus Kit with Abbott RealTime HIV-1, HBV and HCV assays.

    Science.gov (United States)

    Schneider, George J; Kuper, Kevin G; Abravaya, Klara; Mullen, Carolyn R; Schmidt, Marion; Bunse-Grassmann, Astrid; Sprenger-Haussels, Markus

    2009-04-01

    Automated sample preparation systems must meet the demands of routine diagnostics laboratories with regard to performance characteristics and compatibility with downstream assays. In this study, the performance of QIAGEN EZ1 DSP Virus Kit on the BioRobot EZ1 DSP was evaluated in combination with the Abbott RealTime HIV-1, HCV, and HBV assays, followed by thermalcycling and detection on the Abbott m2000rt platform. The following performance characteristics were evaluated: linear range and precision, sensitivity, cross-contamination, effects of interfering substances and correlation. Linearity was observed within the tested ranges (for HIV-1: 2.0-6.0 log copies/ml, HCV: 1.3-6.9 log IU/ml, HBV: 1.6-7.6 log copies/ml). Excellent precision was obtained (inter-assay standard deviation for HIV-1: 0.06-0.17 log copies/ml (>2.17 log copies/ml), HCV: 0.05-0.11 log IU/ml (>2.09 log IU/ml), HBV: 0.03-0.07 log copies/ml (>2.55 log copies/ml)), with good sensitivity (95% hit rates for HIV-1: 50 copies/ml, HCV: 12.5 IU/ml, HBV: 10 IU/ml). No cross-contamination was observed, as well as no negative impact of elevated levels of various interfering substances. In addition, HCV and HBV viral load measurements after BioRobot EZ1 DSP extraction correlated well with those obtained after Abbott m2000sp extraction. This evaluation demonstrates that the QIAGEN EZ1 DSP Virus Kit provides an attractive solution for fully automated, low throughput sample preparation for use with the Abbott RealTime HIV-1, HCV, and HBV assays.

  16. Voice amplification for primary school teachers with voice disorders: A randomized clinical trial

    Directory of Open Access Journals (Sweden)

    Roberto Bovo

    2013-06-01

    Full Text Available Objectives: Several studies have demonstrated a high prevalence of voice disorders in teachers, together with the personal, professional and economical consequences of the problem. Good primary prevention should be based on 3 aspects: 1 amelioration of classroom acoustics, 2 voice care programs for future professional voice users, including teachers and 3 classroom or portable amplification systems. The aim of the study was to assess the benefit obtained from the use of portable amplification systems by female primary school teachers in their occupational setting. Materials and Methods: Forty female primary school teachers attended a course about professional voice care, which comprised two theoretical lectures, each 60 min long. Thereafter, they were randomized into 2 groups: the teachers of the first group were asked to use a portable vocal amplifier for 3 months, till the end of school-year. The other 20 teachers were part of the control group, matched for age and years of employment. All subjects had a grade 1 of dysphonia with no significant organic lesion of the vocal folds. Results: Most teachers of the experimental group used the amplifier consistently for the whole duration of the experiment and found it very useful in reducing the symptoms of vocal fatigue. In fact, after 3 months, Voice Handicap Index (VHI scores in "course + amplifier" group demonstrated a significant amelioration (p = 0.003. The perceptual grade of dysphonia also improved significantly (p = 0.0005. The same parameters changed favourably also in the "course only" group, but the results were not statistically significant (p = 0.4 for VHI and p = 0.03 for perceptual grade. Conclusions: In teachers, and particularly in those with a constitutional weak voice and/or those who are prone to vocal fold pathology, vocal amplifiers may be an effective and low-cost intervention to decrease potentially damaging vocal loads and may represent a necessary form of prevention.

  17. [The prevalence, causes and specific features of voice disturbances in teachers].

    Science.gov (United States)

    Orlova, O S; Vasilenko, Iu S; Zakharova, A F; Samokhvalova, L O; Kozlova, P A

    2000-01-01

    The paper analyzes voice disturbances, their causes and specific features in teachers based on the questionnaires filled by 934 general educational school teachers. The teachers have been found to associate voice disturbances not only with changes in the voice timbre, but with different subjective feelings that make their professional activity difficult. The major factors that cause voice disturbances are the voice overloads that differ in teachers of different specialities, their inability to use the voice, psychoemotional stresses, and frequent colds, as well as a combination of several factors. The incidence of vocal apparatus diseases does not tend to decrease, which makes it necessary to implement combined medical and pedagogical prophylactic measures to prevent dysphonia.

  18. Towards an information ecosystem for animal disease surveillance using voice services

    CSIR Research Space (South Africa)

    Sharma Grover, A

    2013-01-01

    Full Text Available In this paper we introduce a solution for disease surveillance and monitoring in the primary animal health care (PAHC) domain that uses inbound voice-based services and voice- and text-based outbound services for connecting rural veterinarians...

  19. 8-channel, FPGA based, DSP integrated cavity simulator and controller for VUV-FEL. SIMCON 3.0 Ver. 3.0. rev. 1, 06.2005 - Hardware manual

    International Nuclear Information System (INIS)

    Pozniak, K.T.; Czarski, T.; Koprek, W.; Giergusiewicz, W.; Romaniuk, R.S.

    2005-01-01

    The note describes integrated, eight channel system of hardware controller and simulator of the resonant superconducting, narrowband niobium cavity, originally considered for the TTF and TESLA in DESY, Hamburg (now tested for the VUV FEL and developed for X-Ray FEL). The controller bases on a programmable circuit Xilinx VirtexII V4000. The solution uses DSP EMBEDDED BOARD module positioned on a Modular LLRF Control Platform. The algorithm and FPGA circuit configuration was done in the VHDL language. The internal hardware multiplication components, present in Virtex II chips, were used, to improve the floating point calculation efficiency. The implementation was achieved of a device working in the real time, according to the demands of the LLRF control system for the TESLA Test Facility (now associated with the VUV FEL machine). The device under consideration will be referred to as superconducting cavity (SCCav) SIMCON throughout this work. The manual describes hardware features of SIMCON, ver. 3.0 in modular solution. The following components are described here in detail: functional layer, parameter programming, foundations of control of particular blocks and monitoring of the real time processes. This note is accompanied by the one describing the multichannel DOOCS interface for the described hardware system. The interface was prepared in DOOCS for Solaris and in Windows. The hardware and software of 8-channel SIMCON was tested in CHECIA and ACC1 module of VUV FEL linac. The measurements results are presented. While giving all necessary technical details required to understand the work of the integrated hardware controller and simulator and to enable its practical copying, this document is a unity with other TESLA technical notes published by the same team on the subject. Thus, some modeling and other subjects were omitted, as they were addressed in detail in the quoted references. Keywords: Super conducting cavity, cavity simulator, CAVITIES CONTROLLER, SIMCON

  20. Amino acid sequences mediating vascular cell adhesion molecule 1 binding to integrin alpha 4: homologous DSP sequence found for JC polyoma VP1 coat protein

    Directory of Open Access Journals (Sweden)

    Michael Andrew Meyer

    2013-07-01

    Full Text Available The JC polyoma viral coat protein VP1 was analyzed for amino acid sequences homologies to the IDSP sequence which mediates binding of VLA-4 (integrin alpha 4 to vascular cell adhesion molecule 1. Although the full sequence was not found, a DSP sequence was located near the critical arginine residue linked to infectivity of the virus and binding to sialic acid containing molecules such as integrins (3. For the JC polyoma virus, a DSP sequence was found at residues 70, 71 and 72 with homology also noted for the mouse polyoma virus and SV40 virus. Three dimensional modeling of the VP1 molecule suggests that the DSP loop has an accessible site for interaction from the external side of the assembled viral capsid pentamer.

  1. The Role of Occupational Voice Demand and Patient-Rated Impairment in Predicting Voice Therapy Adherence.

    Science.gov (United States)

    Ebersole, Barbara; Soni, Resha S; Moran, Kathleen; Lango, Miriam; Devarajan, Karthik; Jamal, Nausheen

    2018-05-01

    Examine the relationship among the severity of patient-perceived voice impairment, perceptual dysphonia severity, occupational voice demand, and voice therapy adherence. Identify clinical predictors of increased risk for therapy nonadherence. A retrospective cohort study of patients presenting with a chief complaint of persistent dysphonia at an interdisciplinary voice center was done. The Voice Handicap Index-10 (VHI-10) and the Voice-Related Quality of Life (V-RQOL) survey scores, clinician rating of dysphonia severity using the Grade score from the Grade, Roughness Breathiness, Asthenia, and Strain scale, occupational voice demand, and patient demographics were tested for associations with therapy adherence, defined as completion of the treatment plan. Classification and Regression Tree (CART) analysis was performed to establish thresholds for nonadherence risk. Of 166 patients evaluated, 111 were recommended for voice therapy. The therapy nonadherence rate was 56%. Occupational voice demand category, VHI-10, and V-RQOL scores were the only factors significantly correlated with therapy adherence (P demand are significantly more likely to be nonadherent with therapy than those with high occupational voice demand (P 40 is a significant cutoff point for predicting therapy nonadherence (P demand and patient perception of impairment are significantly and independently correlated with therapy adherence. A VHI-10 score of ≤9 or a V-RQOL score of >40 is a significant cutoff point for predicting nonadherence risk. Copyright © 2018 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

  2. Diagnostic value of voice acoustic analysis in assessment of occupational voice pathologies in teachers.

    Science.gov (United States)

    Niebudek-Bogusz, Ewa; Fiszer, Marta; Kotylo, Piotr; Sliwinska-Kowalska, Mariola

    2006-01-01

    It has been shown that teachers are at risk of developing occupational dysphonia, which accounts for over 25% of all occupational diseases diagnosed in Poland. The most frequently used method of diagnosing voice diseases is videostroboscopy. However, to facilitate objective evaluation of voice efficiency as well as medical certification of occupational voice disorders, it is crucial to implement quantitative methods of voice assessment, particularly voice acoustic analysis. The aim of the study was to assess the results of acoustic analysis in 66 female teachers (aged 40-64 years), including 35 subjects with occupational voice pathologies (e.g., vocal nodules) and 31 subjects with functional dysphonia. The acoustic analysis was performed using the IRIS software, before and after a 30-minute vocal loading test. All participants were subjected also to laryngological and videostroboscopic examinations. After the vocal effort, the acoustic parameters displayed statistically significant abnormalities, mostly lowered fundamental frequency (Fo) and incorrect values of shimmer and noise to harmonic ratio. To conclude, quantitative voice acoustic analysis using the IRIS software seems to be an effective complement to voice examinations, which is particularly helpful in diagnosing occupational dysphonia.

  3. The Development of a Portable ECG Monitor Based on DSP

    Science.gov (United States)

    Nan, CHI Jian; Tao, YAN Yan; Meng Chen, LIU; Li, YANG

    With the advent of global information, researches of Smart Home system are in the ascendant, the ECG real-time detection, and wireless transmission of ECG become more useful. In order to achieve the purpose we developed a portable ECG monitor which achieves the purpose of cardiac disease remote monitoring, and will be used in the physical and psychological disease surveillance in smart home system, we developed this portable ECG Monitor, based on the analysis of existing ECG Monitor, using TMS320F2812 as the core controller, which complete the signal collection, storage, processing, waveform display and transmission.

  4. Objective voice parameters in Colombian school workers with healthy voices

    NARCIS (Netherlands)

    L.C. Cantor Cutiva (Lady Catherine); A. Burdorf (Alex)

    2015-01-01

    textabstractObjectives: To characterize the objective voice parameters among school workers, and to identify associated factors of three objective voice parameters, namely fundamental frequency, sound pressure level and maximum phonation time. Materials and methods: We conducted a cross-sectional

  5. High quality voice synthesis middle ware; Kohinshitsu onsei gosei middle war

    Energy Technology Data Exchange (ETDEWEB)

    NONE

    2000-03-01

    Toshiba Corp. newly developed a natural voice synthesis system, TOS Drive TTS (TOtally speaker Driven Text-To-Speech) system, in which natural high-quality read-aloud is greatly improved, and also developed as its application a voice synthesis middle ware. In the newly developed system, using as a model a narrator's voice recorded preliminarily, a metrical control dictionary is automatically learned that reproduces the characteristics of metrical patters such as intonation or rhythm of a human voice, as is a voice bases dictionary that reproduces the characteristics of a voice quality, enabling natural voice synthesis to be realized that picks up human voice characteristics. The system is high quality and also very compact, while the voice synthesis middle ware utilizing this technology is adaptable to various platforms such as MPU or OS. The system is very suitable for audio response in the ITS field having car navigation systems as the core; besides, expanded application is expected to an audio response system that used to employ a sound recording and reproducing system. (translated by NEDO)

  6. Sounds like a winner: voice pitch influences perception of leadership capacity in both men and women.

    Science.gov (United States)

    Klofstad, Casey A; Anderson, Rindy C; Peters, Susan

    2012-07-07

    It is well known that non-human animals respond to information encoded in vocal signals, and the same can be said of humans. Specifically, human voice pitch affects how speakers are perceived. As such, does voice pitch affect how we perceive and select our leaders? To answer this question, we recorded men and women saying 'I urge you to vote for me this November'. Each recording was manipulated digitally to yield a higher- and lower-pitched version of the original. We then asked men and women to vote for either the lower- or higher-pitched version of each voice. Our results show that both men and women select male and female leaders with lower voices. These findings suggest that men and women with lower-pitched voices may be more successful in obtaining positions of leadership. This might also suggest that because women, on average, have higher-pitched voices than men, voice pitch could be a factor that contributes to fewer women holding leadership roles than men. Additionally, while people are free to choose their leaders, these results clearly demonstrate that these choices cannot be understood in isolation from biological influences.

  7. Voices of Women Teachers about Gender Inequalities and Gender-Based Violence in Rural South Africa

    Science.gov (United States)

    de Lange, Naydene; Mitchell, Claudia; Bhana, Deevia

    2012-01-01

    Gender-based violence is a reality in many societies and is linked to the spread of HIV and AIDS. There have been numerous studies that have attempted to acquire an understanding of the breadth and depth of the issues around gender-based violence. However, one area that has received scant attention is the voices of women teachers. Thus, in this…

  8. Pedagogic Voice: Student Voice in Teaching and Engagement Pedagogies

    Science.gov (United States)

    Baroutsis, Aspa; McGregor, Glenda; Mills, Martin

    2016-01-01

    In this paper, we are concerned with the notion of "pedagogic voice" as it relates to the presence of student "voice" in teaching, learning and curriculum matters at an alternative, or second chance, school in Australia. This school draws upon many of the principles of democratic schooling via its utilisation of student voice…

  9. Efficient voice activity detection in reverberant enclosures using far field microphones

    DEFF Research Database (Denmark)

    Petsatodis, Theodore; Boukis, Christos

    2009-01-01

    An algorithm suitable for voice activity detection under reverberant conditions is proposed in this paper. Due to the use of far-filed microphones the proposed solution processes speech signals of highly-varying intensity and signal to noise ratio, that are contaminated with several echoes....... The core of the system is a pair of Hidden Markov Models, that effectively model the speech presence and speech absence situations. To minimise mis-detections an adaptive threshold is used, while a hang-over scheme caters for the intra-frame correlation of speech signals. Experimental results conducted...

  10. Voice Savers for Music Teachers

    Science.gov (United States)

    Cookman, Starr

    2012-01-01

    Music teachers are in a class all their own when it comes to voice use. These elite vocal athletes require stamina, strength, and flexibility from their voices day in, day out for hours at a time. Voice rehabilitation clinics and research show that music education ranks high among the professionals most commonly affected by voice problems.…

  11. A Robust Multimodal Bio metric Authentication Scheme with Voice and Face Recognition

    International Nuclear Information System (INIS)

    Kasban, H.

    2017-01-01

    This paper proposes a multimodal biometric scheme for human authentication based on fusion of voice and face recognition. For voice recognition, three categories of features (statistical coefficients, cepstral coefficients and voice timbre) are used and compared. The voice identification modality is carried out using Gaussian Mixture Model (GMM). For face recognition, three recognition methods (Eigenface, Linear Discriminate Analysis (LDA), and Gabor filter) are used and compared. The combination of voice and face biometrics systems into a single multimodal biometrics system is performed using features fusion and scores fusion. This study shows that the best results are obtained using all the features (cepstral coefficients, statistical coefficients and voice timbre features) for voice recognition, LDA face recognition method and scores fusion for the multimodal biometrics system

  12. Mechanics of human voice production and control.

    Science.gov (United States)

    Zhang, Zhaoyan

    2016-10-01

    As the primary means of communication, voice plays an important role in daily life. Voice also conveys personal information such as social status, personal traits, and the emotional state of the speaker. Mechanically, voice production involves complex fluid-structure interaction within the glottis and its control by laryngeal muscle activation. An important goal of voice research is to establish a causal theory linking voice physiology and biomechanics to how speakers use and control voice to communicate meaning and personal information. Establishing such a causal theory has important implications for clinical voice management, voice training, and many speech technology applications. This paper provides a review of voice physiology and biomechanics, the physics of vocal fold vibration and sound production, and laryngeal muscular control of the fundamental frequency of voice, vocal intensity, and voice quality. Current efforts to develop mechanical and computational models of voice production are also critically reviewed. Finally, issues and future challenges in developing a causal theory of voice production and perception are discussed.

  13. You're a What? Voice Actor

    Science.gov (United States)

    Liming, Drew

    2009-01-01

    This article talks about voice actors and features Tony Oliver, a professional voice actor. Voice actors help to bring one's favorite cartoon and video game characters to life. They also do voice-overs for radio and television commercials and movie trailers. These actors use the sound of their voice to sell a character's emotions--or an advertised…

  14. Voice - How humans communicate?

    Science.gov (United States)

    Tiwari, Manjul; Tiwari, Maneesha

    2012-01-01

    Voices are important things for humans. They are the medium through which we do a lot of communicating with the outside world: our ideas, of course, and also our emotions and our personality. The voice is the very emblem of the speaker, indelibly woven into the fabric of speech. In this sense, each of our utterances of spoken language carries not only its own message but also, through accent, tone of voice and habitual voice quality it is at the same time an audible declaration of our membership of particular social regional groups, of our individual physical and psychological identity, and of our momentary mood. Voices are also one of the media through which we (successfully, most of the time) recognize other humans who are important to us-members of our family, media personalities, our friends, and enemies. Although evidence from DNA analysis is potentially vastly more eloquent in its power than evidence from voices, DNA cannot talk. It cannot be recorded planning, carrying out or confessing to a crime. It cannot be so apparently directly incriminating. As will quickly become evident, voices are extremely complex things, and some of the inherent limitations of the forensic-phonetic method are in part a consequence of the interaction between their complexity and the real world in which they are used. It is one of the aims of this article to explain how this comes about. This subject have unsolved questions, but there is no direct way to present the information that is necessary to understand how voices can be related, or not, to their owners.

  15. Speaker's voice as a memory cue.

    Science.gov (United States)

    Campeanu, Sandra; Craik, Fergus I M; Alain, Claude

    2015-02-01

    Speaker's voice occupies a central role as the cornerstone of auditory social interaction. Here, we review the evidence suggesting that speaker's voice constitutes an integral context cue in auditory memory. Investigation into the nature of voice representation as a memory cue is essential to understanding auditory memory and the neural correlates which underlie it. Evidence from behavioral and electrophysiological studies suggest that while specific voice reinstatement (i.e., same speaker) often appears to facilitate word memory even without attention to voice at study, the presence of a partial benefit of similar voices between study and test is less clear. In terms of explicit memory experiments utilizing unfamiliar voices, encoding methods appear to play a pivotal role. Voice congruency effects have been found when voice is specifically attended at study (i.e., when relatively shallow, perceptual encoding takes place). These behavioral findings coincide with neural indices of memory performance such as the parietal old/new recollection effect and the late right frontal effect. The former distinguishes between correctly identified old words and correctly identified new words, and reflects voice congruency only when voice is attended at study. Characterization of the latter likely depends upon voice memory, rather than word memory. There is also evidence to suggest that voice effects can be found in implicit memory paradigms. However, the presence of voice effects appears to depend greatly on the task employed. Using a word identification task, perceptual similarity between study and test conditions is, like for explicit memory tests, crucial. In addition, the type of noise employed appears to have a differential effect. While voice effects have been observed when white noise is used at both study and test, using multi-talker babble does not confer the same results. In terms of neuroimaging research modulations, characterization of an implicit memory effect

  16. Smartphone-based ecological momentary assessment and intervention in a coping-focused intervention for hearing voices (SAVVy): study protocol for a pilot randomised controlled trial.

    Science.gov (United States)

    Bell, Imogen H; Fielding-Smith, Sarah F; Hayward, Mark; Rossell, Susan L; Lim, Michelle H; Farhall, John; Thomas, Neil

    2018-05-02

    Smartphone-based ecological momentary assessment and intervention (EMA/I) show promise for enhancing psychological treatments for psychosis. EMA has the potential to improve assessment and formulation of experiences which fluctuate day-to-day, and EMI may be used to prompt use of therapeutic strategies in daily life. The current study is an examination of these capabilities in the context of a brief, coping-focused intervention for distressing voice hearing experiences. This is a rater-blinded, pilot randomised controlled trial comparing a four-session intervention in conjunction with use of smartphone EMA/I between sessions, versus treatment-as-usual. The recruitment target is 34 participants with persisting and distressing voice hearing experiences, recruited through a Voices Clinic based in Melbourne, Australia, and via wider advertising. Allocation will be made using minimisation procedure, balancing of the frequency of voices between groups. Assessments are completed at baseline and 8 weeks post-baseline. The primary outcomes of this trial will focus on feasibility and acceptability of the intervention and trial methodology, with secondary outcomes examining preliminary clinical effects related to overall voice severity, the emotional and functional impact of the voices, and emotional distress. This study offers a highly novel examination of specific smartphone capabilities and their integration with traditional psychological treatment for distressing voices. Such technology has potential to enhance psychological interventions and promote adaptation to distressing experiences. Australian New Zealand Clinical Trial Registry, ACTRN12617000348358 . Registered on 7 March 2017.

  17. Accelerometer-based automatic voice onset detection in speech mapping with navigated repetitive transcranial magnetic stimulation.

    Science.gov (United States)

    Vitikainen, Anne-Mari; Mäkelä, Elina; Lioumis, Pantelis; Jousmäki, Veikko; Mäkelä, Jyrki P

    2015-09-30

    The use of navigated repetitive transcranial magnetic stimulation (rTMS) in mapping of speech-related brain areas has recently shown to be useful in preoperative workflow of epilepsy and tumor patients. However, substantial inter- and intraobserver variability and non-optimal replicability of the rTMS results have been reported, and a need for additional development of the methodology is recognized. In TMS motor cortex mappings the evoked responses can be quantitatively monitored by electromyographic recordings; however, no such easily available setup exists for speech mappings. We present an accelerometer-based setup for detection of vocalization-related larynx vibrations combined with an automatic routine for voice onset detection for rTMS speech mapping applying naming. The results produced by the automatic routine were compared with the manually reviewed video-recordings. The new method was applied in the routine navigated rTMS speech mapping for 12 consecutive patients during preoperative workup for epilepsy or tumor surgery. The automatic routine correctly detected 96% of the voice onsets, resulting in 96% sensitivity and 71% specificity. Majority (63%) of the misdetections were related to visible throat movements, extra voices before the response, or delayed naming of the previous stimuli. The no-response errors were correctly detected in 88% of events. The proposed setup for automatic detection of voice onsets provides quantitative additional data for analysis of the rTMS-induced speech response modifications. The objectively defined speech response latencies increase the repeatability, reliability and stratification of the rTMS results. Copyright © 2015 Elsevier B.V. All rights reserved.

  18. Integrating cues of social interest and voice pitch in men's preferences for women's voices

    OpenAIRE

    Jones, Benedict C; Feinberg, David R; DeBruine, Lisa M; Little, Anthony C; Vukovic, Jovana

    2008-01-01

    Most previous studies of vocal attractiveness have focused on preferences for physical characteristics of voices such as pitch. Here we examine the content of vocalizations in interaction with such physical traits, finding that vocal cues of social interest modulate the strength of men's preferences for raised pitch in women's voices. Men showed stronger preferences for raised pitch when judging the voices of women who appeared interested in the listener than when judging the voices of women ...

  19. Controlling An Electric Car Starter System Through Voice

    Directory of Open Access Journals (Sweden)

    A.B. Muhammad Firdaus

    2015-04-01

    Full Text Available Abstract These days automotive has turned into a stand out amongst the most well-known modes of transportation on the grounds that a large number of Malaysians could bear to have an auto. There are numerous decisions of innovations in auto that have in the market. One of the engineering is voice controlled framework. Voice Recognition is the procedure of consequently perceiving a certain statement talked by a specific speaker focused around individual data included in discourse waves. This paper is to make an car controlled by voice of human. An essential pre-processing venture in Voice Recognition systems is to recognize the vicinity of noise. Sensitivity to speech variability lacking recognition precision and helplessness to mimic are among the principle specialized obstacles that keep the far reaching selection of speech-based recognition systems. Voice recognition systems work sensibly well with a quiet conditions however inadequately under loud conditions or in twisted channels. The key focus of the project is to control an electric car starter system.

  20. Research on control law accelerator of digital signal process chip TMS320F28035 for real-time data acquisition and processing

    Science.gov (United States)

    Zhao, Shuangle; Zhang, Xueyi; Sun, Shengli; Wang, Xudong

    2017-08-01

    TI C2000 series digital signal process (DSP) chip has been widely used in electrical engineering, measurement and control, communications and other professional fields, DSP TMS320F28035 is one of the most representative of a kind. When using the DSP program, need data acquisition and data processing, and if the use of common mode C or assembly language programming, the program sequence, analogue-to-digital (AD) converter cannot be real-time acquisition, often missing a lot of data. The control low accelerator (CLA) processor can run in parallel with the main central processing unit (CPU), and the frequency is consistent with the main CPU, and has the function of floating point operations. Therefore, the CLA coprocessor is used in the program, and the CLA kernel is responsible for data processing. The main CPU is responsible for the AD conversion. The advantage of this method is to reduce the time of data processing and realize the real-time performance of data acquisition.

  1. Digital Signal Processing for Medical Imaging Using Matlab

    CERN Document Server

    Gopi, E S

    2013-01-01

    This book describes medical imaging systems, such as X-ray, Computed tomography, MRI, etc. from the point of view of digital signal processing. Readers will see techniques applied to medical imaging such as Radon transformation, image reconstruction, image rendering, image enhancement and restoration, and more. This book also outlines the physics behind medical imaging required to understand the techniques being described. The presentation is designed to be accessible to beginners who are doing research in DSP for medical imaging. Matlab programs and illustrations are used wherever possible to reinforce the concepts being discussed.  ·         Acts as a “starter kit” for beginners doing research in DSP for medical imaging; ·         Uses Matlab programs and illustrations throughout to make content accessible, particularly with techniques such as Radon transformation and image rendering; ·         Includes discussion of the basic principles behind the various medical imaging tec...

  2. FonaDyn - A system for real-time analysis of the electroglottogram, over the voice range

    Science.gov (United States)

    Ternström, Sten; Johansson, Dennis; Selamtzis, Andreas

    2018-01-01

    From soft to loud and low to high, the mechanisms of human voice have many degrees of freedom, making it difficult to assess phonation from the acoustic signal alone. FonaDyn is a research tool that combines acoustics with electroglottography (EGG). It characterizes and visualizes in real time the dynamics of EGG waveforms, using statistical clustering of the cycle-synchronous EGG Fourier components, and their sample entropy. The prevalence and stability of different EGG waveshapes are mapped as colored regions into a so-called voice range profile, without needing pre-defined thresholds or categories. With appropriately 'trained' clusters, FonaDyn can classify and map voice regimes. This is of potential scientific, clinical and pedagogical interest.

  3. Foetal response to music and voice.

    Science.gov (United States)

    Al-Qahtani, Noura H

    2005-10-01

    To examine whether prenatal exposure to music and voice alters foetal behaviour and whether foetal response to music differs from human voice. A prospective observational study was conducted in 20 normal term pregnant mothers. Ten foetuses were exposed to music and voice for 15 s at different sound pressure levels to find out the optimal setting for the auditory stimulation. Music, voice and sham were played to another 10 foetuses via a headphone on the maternal abdomen. The sound pressure level was 105 db and 94 db for music and voice, respectively. Computerised assessment of foetal heart rate and activity were recorded. 90 actocardiograms were obtained for the whole group. One way anova followed by posthoc (Student-Newman-Keuls method) analysis was used to find if there is significant difference in foetal response to music and voice versus sham. Foetuses responded with heart rate acceleration and motor response to both music and voice. This was statistically significant compared to sham. There was no significant difference between the foetal heart rate acceleration to music and voice. Prenatal exposure to music and voice alters the foetal behaviour. No difference was detected in foetal response to music and voice.

  4. Influence of classroom acoustics on the voice levels of teachers with and without voice problems: a field study

    DEFF Research Database (Denmark)

    Pelegrin Garcia, David; Lyberg-Åhlander, Viveka; Rydell, Roland

    2010-01-01

    of the classroom. The results thus suggest that teachers with voice problems are more aware of classroom acoustic conditions than their healthy colleagues and make use of the more supportive rooms to lower their voice levels. This behavior may result from an adaptation process of the teachers with voice problems...... of the voice problems was made with a questionnaire and a laryngological examination. During teaching, the sound pressure level at the teacher’s position was monitored. The teacher’s voice level and the activity noise level were separated using mixed Gaussians. In addition, objective acoustic parameters...... of Reverberation Time and Voice Support were measured in the 30 empty classrooms of the study. An empirical model shows that the measured voice levels depended on the activity noise levels and the voice support. Teachers with and without voice problems were differently affected by the voice support...

  5. DSP-Based Focusing over Optical Fiber Using Time Reversal

    DEFF Research Database (Denmark)

    Piels, Molly; Porto da Silva, Edson; Estaran Tolosa, Jose Manuel

    2014-01-01

    A time-reversal array in multimode fiber is proposed for lossless switching using passive optical splitters. Numerical investigations are performed, and a two-transmitter array that routes a 3GBd QPSK signal through the physical layer is demonstrated experimentally.......A time-reversal array in multimode fiber is proposed for lossless switching using passive optical splitters. Numerical investigations are performed, and a two-transmitter array that routes a 3GBd QPSK signal through the physical layer is demonstrated experimentally....

  6. Automatic Assessment of Acoustic Parameters of the Singing Voice: Application to Professional Western Operatic and Jazz Singers.

    Science.gov (United States)

    Manfredi, Claudia; Barbagallo, Davide; Baracca, Giovanna; Orlandi, Silvia; Bandini, Andrea; Dejonckere, Philippe H

    2015-07-01

    The obvious perceptual differences between various singing styles like Western operatic and jazz rely on specific dissimilarities in vocal technique. The present study focuses on differences in vibrato acoustics and in singer's formant as analyzed by a novel software tool, named BioVoice, based on robust high-resolution and adaptive techniques that have proven its validity on synthetic voice signals. A total of 48 professional singers were investigated (29 females; 19 males; 29 Western operatic; and 19 jazz). They were asked to sing "a cappella," but with artistic expression, a well-known musical phrase from Gershwin's Porgy and Bess, in their own style: either operatic or jazz. A specific sustained note was extracted for detailed vibrato analysis. Beside rate (s(-1)) and extent (cents), duration (seconds) and regularity were computed. Two new concepts are introduced: vibrato jitter and vibrato shimmer, by analogy with the traditional jitter and shimmer of voice signals. For the singer's formant, on the same sustained tone, the ratio of the acoustic energy in formants 1-2 to the energy in formants 3, 4, and 5 was automatically computed, providing a quality ratio (QR). Vibrato rates did not differ among groups. Extent was significantly larger in operatic singers, particularly females. Vibrato jitter and vibrato shimmer were significantly smaller in operatic singers. Duration of vibrato was also significantly longer in operatic singers. QR was significantly lower in male operatic singers. Some vibrato characteristics (extent, regularity, and duration) very clearly differentiate the Western operatic singing style from the jazz singing style. The singer's formant is typical of male operatic singers. The new software tool is well suited to provide useful feedback in a pedagogical context. Copyright © 2015 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

  7. Emotional state and its impact on voice authentication accuracy

    Science.gov (United States)

    Voznak, Miroslav; Partila, Pavol; Penhaker, Marek; Peterek, Tomas; Tomala, Karel; Rezac, Filip; Safarik, Jakub

    2013-05-01

    The paper deals with the increasing accuracy of voice authentication methods. The developed algorithm first extracts segmental parameters, such as Zero Crossing Rate, the Fundamental Frequency and Mel-frequency cepstral coefficients from voice. Based on these parameters, the neural network classifier detects the speaker's emotional state. These parameters shape the distribution of neurons in Kohonen maps, forming clusters of neurons on the map characterizing a particular emotional state. Using regression analysis, we can calculate the function of the parameters of individual emotional states. This relationship increases voice authentication accuracy and prevents unjust rejection.

  8. Deviant vocal fold vibration as observed during videokymography : the effect on voice quality

    NARCIS (Netherlands)

    Verdonck-de Leeuw, I M; Festen, J.M.; Mahieu, H.F.

    Videokymographic images of deviant or irregular vocal fold vibration, including diplophonia, the transition from falsetto to modal voice, irregular vibration onset and offset, and phonation following partial laryngectomy were compared with the synchronously recorded acoustic speech signals. A clear

  9. Understanding the 'Anorexic Voice' in Anorexia Nervosa.

    Science.gov (United States)

    Pugh, Matthew; Waller, Glenn

    2017-05-01

    In common with individuals experiencing a number of disorders, people with anorexia nervosa report experiencing an internal 'voice'. The anorexic voice comments on the individual's eating, weight and shape and instructs the individual to restrict or compensate. However, the core characteristics of the anorexic voice are not known. This study aimed to develop a parsimonious model of the voice characteristics that are related to key features of eating disorder pathology and to determine whether patients with anorexia nervosa fall into groups with different voice experiences. The participants were 49 women with full diagnoses of anorexia nervosa. Each completed validated measures of the power and nature of their voice experience and of their responses to the voice. Different voice characteristics were associated with current body mass index, duration of disorder and eating cognitions. Two subgroups emerged, with 'weaker' and 'stronger' voice experiences. Those with stronger voices were characterized by having more negative eating attitudes, more severe compensatory behaviours, a longer duration of illness and a greater likelihood of having the binge-purge subtype of anorexia nervosa. The findings indicate that the anorexic voice is an important element of the psychopathology of anorexia nervosa. Addressing the anorexic voice might be helpful in enhancing outcomes of treatments for anorexia nervosa, but that conclusion might apply only to patients with more severe eating psychopathology. Copyright © 2016 John Wiley & Sons, Ltd. Experiences of an internal 'anorexic voice' are common in anorexia nervosa. Clinicians should consider the role of the voice when formulating eating pathology in anorexia nervosa, including how individuals perceive and relate to that voice. Addressing the voice may be beneficial, particularly in more severe and enduring forms of anorexia nervosa. When working with the voice, clinicians should aim to address both the content of the voice and how

  10. DolphinAtack: Inaudible Voice Commands

    OpenAIRE

    Zhang, Guoming; Yan, Chen; Ji, Xiaoyu; Zhang, Taimin; Zhang, Tianchen; Xu, Wenyuan

    2017-01-01

    Speech recognition (SR) systems such as Siri or Google Now have become an increasingly popular human-computer interaction method, and have turned various systems into voice controllable systems(VCS). Prior work on attacking VCS shows that the hidden voice commands that are incomprehensible to people can control the systems. Hidden voice commands, though hidden, are nonetheless audible. In this work, we design a completely inaudible attack, DolphinAttack, that modulates voice commands on ultra...

  11. All-optical VPN utilizing DSP-based digital orthogonal filters access for PONs

    Science.gov (United States)

    Zhang, Xiaoling; Zhang, Chongfu; Chen, Chen; Jin, Wei; Qiu, Kun

    2018-04-01

    Utilizing digital filtering-enabled signal multiplexing and de-multiplexing, a cost-effective all-optical virtual private network (VPN) system is proposed, for the first time to our best knowledge, in digital filter multiple access passive optical networks (DFMA-PONs). Based on the DFMA technology, the proposed system can be easily designed to meet the requirements of next generation network's flexibility, elasticity, adaptability and compatibility. Through dynamic digital filter allocation and recycling, the proposed all-optical VPN system can provide dynamic establishments and cancellations of multiple VPN communications with arbitrary traffic volumes. More importantly, due to the employment of DFMA technology, the system is not limited to a fixed signal format and different signal formats such as pulse amplitude modulation (PAM), quadrature amplitude modulation (QAM) and orthogonal frequency division multiplexing (OFDM) can be used. Moreover, one transceiver is sufficient to simultaneously transmit upstream (US)/VPN data to optical line terminal (OLT) or other VPN optical network units (ONUs), thus leading to great reduction in network constructions and operation expenditures. The proposed all-optical VPN system is demonstrated with the transceiver incorporating the formats of QAM and OFDM, which can be made transparent to downstream (DS), US and VPN communications. The bit error rates (BERs) of DS, US and VPN for OFDM signals are below the forward-error-correction (FEC) limit of 3 . 8 × 10-3 when the received optical powers are about -16.8 dBm, -14.5 dBm and -15.7 dBm, respectively.

  12. A single-board NMR spectrometer based on a software defined radio architecture

    International Nuclear Information System (INIS)

    Tang, Weinan; Wang, Weimin

    2011-01-01

    A single-board software defined radio (SDR) spectrometer for nuclear magnetic resonance (NMR) is presented. The SDR-based architecture, realized by combining a single field programmable gate array (FPGA) and a digital signal processor (DSP) with peripheral radio frequency (RF) front-end circuits, makes the spectrometer compact and reconfigurable. The DSP, working as a pulse programmer, communicates with a personal computer via a USB interface and controls the FPGA through a parallel port. The FPGA accomplishes digital processing tasks such as a numerically controlled oscillator (NCO), digital down converter (DDC) and gradient waveform generator. The NCO, with agile control of phase, frequency and amplitude, is part of a direct digital synthesizer that is used to generate an RF pulse. The DDC performs quadrature demodulation, multistage low-pass filtering and gain adjustment to produce a bandpass signal (receiver bandwidth from 3.9 kHz to 10 MHz). The gradient waveform generator is capable of outputting shaped gradient pulse waveforms and supports eddy-current compensation. The spectrometer directly acquires an NMR signal up to 30 MHz in the case of baseband sampling and is suitable for low-field (<0.7 T) application. Due to the featured SDR architecture, this prototype has flexible add-on ability and is expected to be suitable for portable NMR systems

  13. Performer's attitudes toward seeking health care for voice issues: understanding the barriers.

    Science.gov (United States)

    Gilman, Marina; Merati, Albert L; Klein, Adam M; Hapner, Edie R; Johns, Michael M

    2009-03-01

    Contemporary commercial music (CCM) performers rely heavily on their voice, yet may not be aware of the importance of proactive voice care. This investigation intends to identify perceptions and barriers to seeking voice care among CCM artists. This cross-sectional observational study used a 10-item Likert-based response questionnaire to assess current perceptions regarding voice care in a population of randomly selected participants of professional CCM conference. Subjects (n=78) were queried regarding their likelihood to seek medical care for minor medical problems and specifically problems with their voice. Additional questions investigated anxiety about seeking voice care from a physician specialist, speech language pathologist, or voice coach; apprehension regarding findings of laryngeal examination, laryngeal imaging procedures; and the effect of medical insurance on the likelihood of seeking medical care. Eighty-two percent of subjects reported that their voice was a critical part of their profession; 41% stated that they were not likely to seek medical care for problems with their voice; and only 19% were reluctant to seek care for general medical problems (Peducation about the importance of voice care is needed in this population of vocal performers.

  14. Design of digital voice storage and playback system

    Science.gov (United States)

    Tang, Chao

    2018-03-01

    Based on STC89C52 chip, this paper presents a single chip microcomputer minimum system, which is used to realize the logic control of digital speech storage and playback system. Compared with the traditional tape voice recording system, the system has advantages of small size, low power consumption, The effective solution of traditional voice recording system is limited in the use of electronic and information processing.

  15. Development of a compact and cost effective multi-input digital signal processing system

    Science.gov (United States)

    Darvish-Molla, Sahar; Chin, Kenrick; Prestwich, William V.; Byun, Soo Hyun

    2018-01-01

    A prototype digital signal processing system (DSP) was developed using a microcontroller interfaced with a 12-bit sampling ADC, which offers a considerably inexpensive solution for processing multiple detectors with high throughput. After digitization of the incoming pulses, in order to maximize the output counting rate, a simple algorithm was employed for pulse height analysis. Moreover, an algorithm aiming at the real-time pulse pile-up deconvolution was implemented. The system was tested using a NaI(Tl) detector in comparison with a traditional analogue and commercial digital systems for a variety of count rates. The performance of the prototype system was consistently superior to the analogue and the commercial digital systems up to the input count rate of 61 kcps while was slightly inferior to the commercial digital system but still superior to the analogue system in the higher input rates. Considering overall cost, size and flexibility, this custom made multi-input digital signal processing system (MMI-DSP) was the best reliable choice for the purpose of the 2D microdosimetric data collection, or for any measurement in which simultaneous multi-data collection is required.

  16. Voice-to-Phoneme Conversion Algorithms for Voice-Tag Applications in Embedded Platforms

    Directory of Open Access Journals (Sweden)

    Yan Ming Cheng

    2008-08-01

    Full Text Available We describe two voice-to-phoneme conversion algorithms for speaker-independent voice-tag creation specifically targeted at applications on embedded platforms. These algorithms (batch mode and sequential are compared in speech recognition experiments where they are first applied in a same-language context in which both acoustic model training and voice-tag creation and application are performed on the same language. Then, their performance is tested in a cross-language setting where the acoustic models are trained on a particular source language while the voice-tags are created and applied on a different target language. In the same-language environment, both algorithms either perform comparably to or significantly better than the baseline where utterances are manually transcribed by a phonetician. In the cross-language context, the voice-tag performances vary depending on the source-target language pair, with the variation reflecting predicted phonological similarity between the source and target languages. Among the most similar languages, performance nears that of the native-trained models and surpasses the native reference baseline.

  17. Collaborative learning using VoiceThread in an online graduate course

    Directory of Open Access Journals (Sweden)

    Yu-Hui Ching

    2013-09-01

    Full Text Available Collaborative learning enables participants in a learning community to externalize and share knowledge, experiences, and practice. However, collaborative learning in an online environment can be challenging due to the lack of face-to face interaction. This current study examined twenty graduate students’ experiences of using VoiceThread for a collaborative activity in an entirely online course to explore students’ perceptions of using multi-modal communication for collaboration and knowledge sharing. The results of this study revealed that graduate students had very positive experiences toward using VoiceThread for collaborative learning. The participants found VoiceThread easy to learn and use, and reported that audio and video interaction on VoiceThread helped connect them with their peers. More than half of the participants interacted with peers using audio, followed by text and then by video. Half of the students felt they were more connected to peers; however, feeling more connected did not result in more participation as most of the students only participated at the level that met the course requirement. Participants identified benefits and drawbacks of using VoiceThread for collaboration as compared to using text-based discussion forums. The most frequently mentioned benefit of using VoiceThread for collaboration exemplifies its multi-modal affordance that enables learners to communicate emotion, personality, and other non-verbal cues conducive to better understanding and interpretation of meanings. About half of the participants indicated that they preferred VoiceThread to text-based discussion forums for collaborative learning activity. Challenges and implications for future research are also discussed.

  18. Risk factors for voice problems in teachers.

    NARCIS (Netherlands)

    Kooijman, P.G.C.; Jong, F.I.C.R.S. de; Thomas, G.; Huinck, W.J.; Donders, A.R.T.; Graamans, K.; Schutte, H.K.

    2006-01-01

    In order to identify factors that are associated with voice problems and voice-related absenteeism in teachers, 1,878 questionnaires were analysed. The questionnaires inquired about personal data, voice complaints, voice-related absenteeism from work and conditions that may lead to voice complaints

  19. Risk factors for voice problems in teachers

    NARCIS (Netherlands)

    Kooijman, P. G. C.; de Jong, F. I. C. R. S.; Thomas, G.; Huinck, W.; Donders, R.; Graamans, K.; Schutte, H. K.

    2006-01-01

    In order to identify factors that are associated with voice problems and voice-related absenteeism in teachers, 1,878 questionnaires were analysed. The questionnaires inquired about personal data, voice complaints, voice-related absenteeism from work and conditions that may lead to voice complaints

  20. A framework for the design of a voice-activated, intelligent, and hypermedia-based aircraft maintenance manual

    Science.gov (United States)

    Patankar, Manoj Shashikant

    Federal Aviation Regulations require Aviation Maintenance Technicians (AMTs) to refer to approved maintenance manuals when performing maintenance on airworthy aircraft. Because these manuals are paper-based, larger the size of the aircraft, more cumbersome are the manuals. Federal Aviation Administration (FAA) recognized the difficulties associated with the use of large manuals and conducted studies on the use of electronic media as an alternative to the traditional paper format. However, these techniques do not employ any artificial intelligence technologies and the user interface is limited to either a keyboard or a stylus pen. The primary emphasis of this research was to design a generic framework that would allow future development of voice-activated, intelligent, and hypermedia-based aircraft maintenance manuals. A prototype (VIHAMS-Voice-activated, Intelligent, and Hypermedia-based Aircraft Maintenance System) was developed, as a secondary emphasis, using the design and development techniques that evolved from this research. An evolutionary software design approach was used to design the proposed framework and the structured rapid prototyping technique was used to produce the VIHAMS prototype. VoiceAssist by Creative Labs was used to provide the voice interface so that the users (AMTs) could keep their hands free to work on the aircraft while maintaining complete control over the computer through discrete voice commands. KnowledgePro for Windows sp{TM}, an expert system shell, provided "intelligence" to the prototype. As a result of this intelligence, the system provided expert guidance to the user. The core information contained in conventional manuals was available in a hypermedia format. The prototype's operating hardware included a notebook computer with a fully functional audio system. An external microphone and the built-in speaker served as the input and output devices (along with the color monitor), respectively. Federal Aviation Administration

  1. Instantaneous and Frequency-Warped Signal Processing Techniques for Auditory Source Separation.

    Science.gov (United States)

    Wang, Avery Li-Chun

    which require a small fraction of the computational power of conventional FIR implementations. This design strategy is based on truncated and stabilized IIR filters. These signal-processing methods have been applied to the problem of auditory source separation, resulting in voice separation from complex music that is significantly better than previous results at far lower computational cost.

  2. Web-Based Learning Enhancements: Video Lectures through Voice-Over PowerPoint in a Majors-Level Biology Course

    Science.gov (United States)

    Lents, Nathan H.; Cifuentes, Oscar E.

    2009-01-01

    This study is an experimental introduction of web-based lecture delivery into a majors-level introductory biology course. Web-based delivery, achieved through the use of prerecorded Voice-Over PowerPoint video lectures, was introduced on a limited basis to an experimental section while a control group, with the same instructor, received standard…

  3. Resource Planning for Voice over Wireless Both-Way Transmission Media

    Directory of Open Access Journals (Sweden)

    B. Tsankov

    2008-04-01

    Full Text Available The medium in IEEE 802.11- and IEEE 802.16-based networks for voice communications can be considered "both-way" - for transmission and reception. Therefore, the packet arrivals for voice dialogue services in such networks are not strictly independent. In this paper, we discuss the traffic capacity in the call (network layer and suggest accounting for the impact of the correlated nature of two-way voice conversations on performance estimation. We present analytical results and numerical examples.

  4. Perceptual and acoustic outcomes of voice therapy for male-to-female transgender individuals immediately after therapy and 15 months later.

    Science.gov (United States)

    Gelfer, Marylou Pausewang; Tice, Ruthanne M

    2013-05-01

    The present study examined how effectively listeners' perceptions of gender could be changed from male to female for male-to-female (MTF) transgender (TG) clients based on the voice signal alone, immediately after voice therapy and at long-term follow-up. Short- and long-term changes in masculinity and femininity ratings and acoustic measures of speaking fundamental frequency (SFF) and vowel formant frequencies were also investigated. Prospective treatment study. Five MTF TG clients, five control female speakers, and five control male speakers provided a variety of speech samples for later analysis. The TG clients then underwent 8 weeks of voice therapy. Voice samples were collected immediately at the termination of therapy and again 15 months later. Two groups of listeners were recruited to evaluate gender and provide masculinity and femininity ratings. Perceptual results revealed that TG subjects were perceived as female 1.9% of the time in the pretest, 50.8% of the time in the immediate posttest, and 33.1% of the time in the long-term posttest. The TG speakers were also perceived as significantly less masculine and more feminine in the immediate posttest and the long-term posttest compared with the pre-test. Some acoustic measures showed significant differences between the pretest and the immediate posttest and long-term posttest. It appeared that 8 weeks of voice therapy could result in vocal changes in MTF TG individuals that persist at least partially for up to 15 months. However, some TG subjects were more successful with voice feminization than others. Copyright © 2013 The Voice Foundation. Published by Mosby, Inc. All rights reserved.

  5. Voice quality in relation to voice complaints and vocal fold condition during the screening of female student teachers.

    Science.gov (United States)

    Meulenbroek, Leo F P; de Jong, Felix I C R S

    2011-07-01

    The purpose of this study was to compare the perceptual examination of voice quality with the condition of the vocal folds and voice complaints during voice screening in female student teachers. This research was a cross-sectional study in 214 starting student teachers using the four-point grade scale of the GRBAS and laryngostroboscopic assessment of the vocal folds. The voice quality was assessed by speech pathologists using the ordinal 4-point G-scale (overall dysphonia) of the GRBAS method in a running speech sample. Glottal closure and vocal fold lesions were recorded. A questionnaire was used for assessing voice complaints. More students with an insufficient glottal closure (89%) were rated dysphonic compared with students with sufficient glottal closure (80%). Students with sufficient glottal closure had a significantly lower mean G-score (1.21) compared with the group with insufficient glottal closure (1.52) (P = 0.038). This study showed a larger percentage of students with vocal fold lesions (96%) labeled a dysphonic voice compared to students with no vocal fold problems (81%). Students with no vocal fold lesions had a significantly lower mean G-score (1.20) compared with the group with vocal fold lesions (2.05) (P=0.002). A dysphonic voice (G≥1) was rated in 76% of the students without voice complaints compared with 86% of the students with voice complaints. Students with no voice complaints had a lower mean G-score (1.07) compared with the group with voice complaints (1.41) (P=0.090). The present study showed that perceptual assessment of the voice and voice complaints is not sufficient to check if the future professional is at risk. Therefore, preventive measures are needed to detect students at risk early in their education and this depends on broader assessment: on the one hand, assessing voice quality and voice complaints and on the other hand, examination of the vocal folds of all starting students. Copyright © 2011 The Voice Foundation

  6. Voice Quality Estimation in Combined Radio-VoIP Networks for Dispatching Systems

    Directory of Open Access Journals (Sweden)

    Jiri Vodrazka

    2016-01-01

    Full Text Available The voice quality modelling assessment and planning field is deeply and widely theoretically and practically mastered for common voice communication systems, especially for the public fixed and mobile telephone networks including Next Generation Networks (NGN - internet protocol based networks. This article seeks to contribute voice quality modelling assessment and planning for dispatching communication systems based on Internet Protocol (IP and private radio networks. The network plan, correction in E-model calculation and default values for the model are presented and discussed.

  7. A Semi-Continuous State-Transition Probability HMM-Based Voice Activity Detector

    Directory of Open Access Journals (Sweden)

    H. Othman

    2007-02-01

    Full Text Available We introduce an efficient hidden Markov model-based voice activity detection (VAD algorithm with time-variant state-transition probabilities in the underlying Markov chain. The transition probabilities vary in an exponential charge/discharge scheme and are softly merged with state conditional likelihood into a final VAD decision. Working in the domain of ITU-T G.729 parameters, with no additional cost for feature extraction, the proposed algorithm significantly outperforms G.729 Annex B VAD while providing a balanced tradeoff between clipping and false detection errors. The performance compares very favorably with the adaptive multirate VAD, option 2 (AMR2.

  8. A Semi-Continuous State-Transition Probability HMM-Based Voice Activity Detector

    Directory of Open Access Journals (Sweden)

    Othman H

    2007-01-01

    Full Text Available We introduce an efficient hidden Markov model-based voice activity detection (VAD algorithm with time-variant state-transition probabilities in the underlying Markov chain. The transition probabilities vary in an exponential charge/discharge scheme and are softly merged with state conditional likelihood into a final VAD decision. Working in the domain of ITU-T G.729 parameters, with no additional cost for feature extraction, the proposed algorithm significantly outperforms G.729 Annex B VAD while providing a balanced tradeoff between clipping and false detection errors. The performance compares very favorably with the adaptive multirate VAD, option 2 (AMR2.

  9. Oral cavity anaerobic pathogens in biofilm formation on voice prostheses

    NARCIS (Netherlands)

    Bertl, Kristina; Zijnge, Vincent; Zatorska, Beata; Leonhard, Matthias; Schneider-Stickler, Berit; Harmsen, Hermie J. M.

    BACKGROUND: A polymerase chain reaction (PCR)-based method has been used to identify oral anaerobic pathogens in biofilms on voice prostheses. The purpose of the present study was to determine the location of those pathogens inside the biofilms. METHODS: Biofilms of 15 voice prostheses were sampled

  10. Using Hierarchical Time Series Clustering Algorithm and Wavelet Classifier for Biometric Voice Classification

    Directory of Open Access Journals (Sweden)

    Simon Fong

    2012-01-01

    Full Text Available Voice biometrics has a long history in biosecurity applications such as verification and identification based on characteristics of the human voice. The other application called voice classification which has its important role in grouping unlabelled voice samples, however, has not been widely studied in research. Lately voice classification is found useful in phone monitoring, classifying speakers’ gender, ethnicity and emotion states, and so forth. In this paper, a collection of computational algorithms are proposed to support voice classification; the algorithms are a combination of hierarchical clustering, dynamic time wrap transform, discrete wavelet transform, and decision tree. The proposed algorithms are relatively more transparent and interpretable than the existing ones, though many techniques such as Artificial Neural Networks, Support Vector Machine, and Hidden Markov Model (which inherently function like a black box have been applied for voice verification and voice identification. Two datasets, one that is generated synthetically and the other one empirically collected from past voice recognition experiment, are used to verify and demonstrate the effectiveness of our proposed voice classification algorithm.

  11. The VOICES/VOCES Success Story: Effective Strategies for Training, Technical Assistance and Community-Based Organization Implementation

    Science.gov (United States)

    Hamdallah, Myriam; Vargo, Sue; Herrera, Jennifer

    2006-01-01

    The Centers for Disease Control and Prevention's Diffusion of Effective Behavioral Interventions (DEBI) project successfully disseminated VOICES/VOCES, a brief video-based HIV risk reduction intervention targeting African American and Latino heterosexual men and women at risk for HIV infection. Elements of the dissemination strategy included a…

  12. Digital signal processor for silicon audio playback devices; Silicon audio saisei kikiyo digital signal processor

    Energy Technology Data Exchange (ETDEWEB)

    NONE

    2000-03-01

    The digital audio signal processor (DSP) TC9446F series has been developed silicon audio playback devices with a memory medium of, e.g., flash memory, DVD players, and AV devices, e.g., TV sets. It corresponds to AAC (advanced audio coding) (2ch) and MP3 (MPEG1 Layer3), as the audio compressing techniques being used for transmitting music through an internet. It also corresponds to compressed types, e.g., Dolby Digital, DTS (digital theater system) and MPEG2 audio, being adopted for, e.g., DVDs. It can carry a built-in audio signal processing program, e.g., Dolby ProLogic, equalizer, sound field controlling, and 3D sound. TC9446XB has been lined up anew. It adopts an FBGA (fine pitch ball grid array) package for portable audio devices. (translated by NEDO)

  13. VOICE QUALITY BEFORE AND AFTER THYROIDECTOMY

    Directory of Open Access Journals (Sweden)

    Dora CVELBAR

    2016-04-01

    Full Text Available Introduction: Voice disorders are a well-known complication which is often associated with thyroid gland diseases and because voice is still the basic mean of communication it is very important to maintain its quality healthy. Objectives: The aim of this study referred to questions whether there is a statistically significant difference between results of voice self-assessment, perceptual voice assessment and acoustic voice analysis before and after thyroidectomy and whether there are statistically significant correlations between variables of voice self-assessment, perceptual assessment and acoustic analysis before and after thyroidectomy. Methods: This scientific research included 12 participants aged between 41 and 76. Voice self-assessment was conducted with the help of Croatian version of Voice Handicap Index (VHI. Recorded reading samples were used for perceptual assessment and later evaluated by two clinical speech and language therapists. Recorded samples of phonation were used for acoustic analysis which was conducted with the help of acoustic program Praat. All of the data was processed through descriptive statistics and nonparametric statistical methods. Results: Results showed that there are statistically significant differences between results of voice self-assessments and results of acoustic analysis before and after thyroidectomy. Statistically significant correlations were found between variables of perceptual assessment and acoustic analysis. Conclusion: Obtained results indicate the importance of multidimensional, preoperative and postoperative assessment. This kind of assessment allows the clinician to describe all of the voice features and provides appropriate recommendation for further rehabilitation to the patient in order to optimize voice outcomes.

  14. Experiment and practice on signal processing

    International Nuclear Information System (INIS)

    2002-11-01

    The contents of this book contains basic practice of CEM Tool, discrete time signal and experiment and practice of system, experiment and practice of discrete time signal sampling, practice of frequency analysis, experiment of digital filter design, application of digital signal processing, project related voice, basic principle of signal processing, the technique of basic image signal processing, biology astronomy and Robot soccer with apply of image signal processing technique, control video signal and project related image. It also has an introduction of CEM Linker I. O in the end.

  15. Experiment and practice on signal processing

    Energy Technology Data Exchange (ETDEWEB)

    NONE

    2002-11-15

    The contents of this book contains basic practice of CEM Tool, discrete time signal and experiment and practice of system, experiment and practice of discrete time signal sampling, practice of frequency analysis, experiment of digital filter design, application of digital signal processing, project related voice, basic principle of signal processing, the technique of basic image signal processing, biology astronomy and Robot soccer with apply of image signal processing technique, control video signal and project related image. It also has an introduction of CEM Linker I. O in the end.

  16. Aerodynamic and sound intensity measurements in tracheoesophageal voice

    NARCIS (Netherlands)

    Grolman, Wilko; Eerenstein, Simone E. J.; Tan, Frédérique M. L.; Tange, Rinze A.; Schouwenburg, Paul F.

    2007-01-01

    BACKGROUND: In laryngectomized patients, tracheoesophageal voice generally provides a better voice quality than esophageal voice. Understanding the aerodynamics of voice production in patients with a voice prosthesis is important for optimizing prosthetic designs and successful voice rehabilitation.

  17. Voice over IP in Wireless Heterogeneous Networks

    DEFF Research Database (Denmark)

    Fathi, Hanane; Chakraborty, Shyam; Prasad, Ramjee

    with the deployment of wireless heterogeneous systems, both speech and data traffic are carrried over wireless links by the same IP-based packet-switched infrastructure. However, this combination faces some challenges due to the inherent properties of the wireless network. The requirements for good quality VoIP...... communications are difficult to achieve in a time-varying environment due to channel errors and traffic congestion and across different systems. The provision of VoIP in wireless heterogeneous networks requires a set of time-efficient control mechanisms to support a VoIP session with acceptable quality....... The focus of Voice over IP in Wierless Heterogeneous Networks is on mechanisms that affect the VoIP user satisfaction  while not explicitly involved in the media session. This relates to the extra delays introduced by the security and the signaling protocols used to set up an authorized VoIP session...

  18. Fast response Antiwindup PI speed controller of Brushless DC motor drive: Modeling, simulation and implementation on DSP

    Directory of Open Access Journals (Sweden)

    Mohd Tariq

    2016-05-01

    Full Text Available Most of the Brushless DC (BLDC motors drive adopts proportional, integral and derivative (PID controller and pulse width modulation (PWM scheme for speed control. Hence, BLDC motor drive has strong saturation characteristics. The saturation results in a typical windup phenomenon. The paper presents an Antiwindup drive for BLDC motor. An Antiwindup controller (AWC has been used in the paper. AWC has been modeled in MATLAB/Simulink and comparison has been done between conventional PI controller and AWC at different starting loads. Dynamic characteristics of the BLDC motor drive have been examined and results are presented and discussed in detail in this paper. Details of DSP based experimental validation of the simulated results are also presented here.

  19. Crossing Cultures with Multi-Voiced Journals

    Science.gov (United States)

    Styslinger, Mary E.; Whisenant, Alison

    2004-01-01

    In this article, the authors discuss the benefits of using multi-voiced journals as a teaching strategy in reading instruction. Multi-voiced journals, an adaptation of dual-voiced journals, encourage responses to reading in varied, cultured voices of characters. It is similar to reading journals in that they prod students to connect to the lives…

  20. [Applicability of Voice Handicap Index to the evaluation of voice therapy effectiveness in teachers].

    Science.gov (United States)

    Niebudek-Bogusz, Ewa; Kuzańska, Anna; Błoch, Piotr; Domańska, Maja; Woźnicka, Ewelina; Politański, Piotr; Sliwińska-Kowalska, Mariola

    2007-01-01

    The aim of this study was to assess the applicability of Voice Handicap Index (VHI) to the evaluation of effectiveness of functional voice disorders treatment in teachers. The subjects were 45 female teachers with functional dysphonia who evaluated their voice problems according to the subjective VHI scale before and after phoniatric management. Group I (29 patients) were subjected to vocal training, whereas group II (16 patients) received only voice hygiene instructions. The results demonstrated that differences in the mean VHI score before and after phoniatric treatment were significantly higher in group 1 than in group II (p teacher's dysphonia.

  1. Voice parameters and videonasolaryngoscopy in children with vocal nodules: a longitudinal study, before and after voice therapy.

    Science.gov (United States)

    Valadez, Victor; Ysunza, Antonio; Ocharan-Hernandez, Esther; Garrido-Bustamante, Norma; Sanchez-Valerio, Araceli; Pamplona, Ma C

    2012-09-01

    Vocal Nodules (VN) are a functional voice disorder associated with voice misuse and abuse in children. There are few reports addressing vocal parameters in children with VN, especially after a period of vocal rehabilitation. The purpose of this study is to describe measurements of vocal parameters including Fundamental Frequency (FF), Shimmer (S), and Jitter (J), videonasolaryngoscopy examination and clinical perceptual assessment, before and after voice therapy in children with VN. Voice therapy was provided using visual support through Speech-Viewer software. Twenty patients with VN were studied. An acoustical analysis of voice was performed and compared with data from subjects from a control group matched by age and gender. Also, clinical perceptual assessment of voice and videonasolaryngoscopy were performed to all patients with VN. After a period of voice therapy, provided with visual support using Speech Viewer-III (SV-III-IBM) software, new acoustical analyses, perceptual assessments and videonasolaryngoscopies were performed. Before the onset of voice therapy, there was a significant difference (ptherapy period, a significant improvement (pvocal nodules were no longer discernible on the vocal folds in any of the cases. SV-III software seems to be a safe and reliable method for providing voice therapy in children with VN. Acoustic voice parameters, perceptual data and videonasolaryngoscopy were significantly improved after the speech therapy period was completed. Copyright © 2012 Elsevier Ireland Ltd. All rights reserved.

  2. Interactive Augmentation of Voice Quality and Reduction of Breath Airflow in the Soprano Voice.

    Science.gov (United States)

    Rothenberg, Martin; Schutte, Harm K

    2016-11-01

    In 1985, at a conference sponsored by the National Institutes of Health, Martin Rothenberg first described a form of nonlinear source-tract acoustic interaction mechanism by which some sopranos, singing in their high range, can use to reduce the total airflow, to allow holding the note longer, and simultaneously enrich the quality of the voice, without straining the voice. (M. Rothenberg, "Source-Tract Acoustic Interaction in the Soprano Voice and Implications for Vocal Efficiency," Fourth International Conference on Vocal Fold Physiology, New Haven, Connecticut, June 3-6, 1985.) In this paper, we describe additional evidence for this type of nonlinear source-tract interaction in some soprano singing and describe an analogous interaction phenomenon in communication engineering. We also present some implications for voice research and pedagogy. Copyright © 2016 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

  3. Performance of wavelet analysis and neural networks for pathological voices identification

    Science.gov (United States)

    Salhi, Lotfi; Talbi, Mourad; Abid, Sabeur; Cherif, Adnane

    2011-09-01

    Within the medical environment, diverse techniques exist to assess the state of the voice of the patient. The inspection technique is inconvenient for a number of reasons, such as its high cost, the duration of the inspection, and above all, the fact that it is an invasive technique. This study focuses on a robust, rapid and accurate system for automatic identification of pathological voices. This system employs non-invasive, non-expensive and fully automated method based on hybrid approach: wavelet transform analysis and neural network classifier. First, we present the results obtained in our previous study while using classic feature parameters. These results allow visual identification of pathological voices. Second, quantified parameters drifting from the wavelet analysis are proposed to characterise the speech sample. On the other hand, a system of multilayer neural networks (MNNs) has been developed which carries out the automatic detection of pathological voices. The developed method was evaluated using voice database composed of recorded voice samples (continuous speech) from normophonic or dysphonic speakers. The dysphonic speakers were patients of a National Hospital 'RABTA' of Tunis Tunisia and a University Hospital in Brussels, Belgium. Experimental results indicate a success rate ranging between 75% and 98.61% for discrimination of normal and pathological voices using the proposed parameters and neural network classifier. We also compared the average classification rate based on the MNN, Gaussian mixture model and support vector machines.

  4. Embedded DSP-based telehealth radar system for remote in-door fall detection.

    Science.gov (United States)

    Garripoli, Carmine; Mercuri, Marco; Karsmakers, Peter; Jack Soh, Ping; Crupi, Giovanni; Vandenbosch, Guy A E; Pace, Calogero; Leroux, Paul; Schreurs, Dominique

    2015-01-01

    Telehealth systems and applications are extensively investigated nowadays to enhance the quality-of-care and, in particular, to detect emergency situations and to monitor the well-being of elderly people, allowing them to stay at home independently as long as possible. In this paper, an embedded telehealth system for continuous, automatic, and remote monitoring of real-time fall emergencies is presented and discussed. The system, consisting of a radar sensor and base station, represents a cost-effective and efficient healthcare solution. The implementation of the fall detection data processing technique, based on the least-square support vector machines, through a digital signal processor and the management of the communication between radar sensor and base station are detailed. Experimental tests, for a total of 65 mimicked fall incidents, recorded with 16 human subjects (14 men and two women) that have been monitored for 320 min, have been used to validate the proposed system under real circumstances. The subjects' weight is between 55 and 90 kg with heights between 1.65 and 1.82 m, while their age is between 25 and 39 years. The experimental results have shown a sensitivity to detect the fall events in real time of 100% without reporting false positives. The tests have been performed in an area where the radar's operation was not limited by practical situations, namely, signal power, coverage of the antennas, and presence of obstacles between the subject and the antennas.

  5. Voice loops as coordination aids in space shuttle mission control.

    Science.gov (United States)

    Patterson, E S; Watts-Perotti, J; Woods, D D

    1999-01-01

    Voice loops, an auditory groupware technology, are essential coordination support tools for experienced practitioners in domains such as air traffic management, aircraft carrier operations and space shuttle mission control. They support synchronous communication on multiple channels among groups of people who are spatially distributed. In this paper, we suggest reasons for why the voice loop system is a successful medium for supporting coordination in space shuttle mission control based on over 130 hours of direct observation. Voice loops allow practitioners to listen in on relevant communications without disrupting their own activities or the activities of others. In addition, the voice loop system is structured around the mission control organization, and therefore directly supports the demands of the domain. By understanding how voice loops meet the particular demands of the mission control environment, insight can be gained for the design of groupware tools to support cooperative activity in other event-driven domains.

  6. PPM-based System for Guided Waves Communication Through Corrosion Resistant Multi-wire Cables

    Science.gov (United States)

    Trane, G.; Mijarez, R.; Guevara, R.; Pascacio, D.

    Novel wireless communication channels are a necessity in applications surrounded by harsh environments, for instance down-hole oil reservoirs. Traditional radio frequency (RF) communication schemes are not capable of transmitting signals through metal enclosures surrounded by corrosive gases and liquids. As an alternative to RF, a pulse position modulation (PPM) guided waves communication system has been developed and evaluated using a corrosion resistant 4H18 multi-wire cable, commonly used to descend electronic gauges in down-hole oil applications, as the communication medium. The system consists of a transmitter and a receiver that utilizes a PZT crystal, for electrical/mechanical coupling, attached to each extreme of the multi-wire cable. The modulator is based on a microcontroller, which transmits60 kHz guided wave pulses, and the demodulator is based on a commercial digital signal processor (DSP) module that performs real time DSP algorithms. Experimental results are presented, which were obtained using a 1m corrosion resistant 4H18multi-wire cable, commonly used with downhole electronic gauges in the oil sector. Although there was significant dispersion and multiple mode excitations of the transmitted guided wave energy pulses, the results show that data rates on the order of 500 bits per second are readily available employing PPM and simple communications techniques.

  7. Hardware Implementation of LMS-Based Adaptive Noise Cancellation Core with Low Resource Utilization

    Directory of Open Access Journals (Sweden)

    Omid Sharifi Tehrani

    2011-10-01

    Full Text Available A hardware implementation of adaptive noise cancellation (ANC core is proposed. Adaptive filters are widely used in different applications such as adaptive noise cancellation, prediction, equalization, inverse modeling and system identification. FIR adaptive filters are mostly used because of their low computation costs and their linear phase. Least mean squared algorithm (LMS is used to train FIR adaptive filter weights. Advances in semiconductor technology especially in digital signal processors (DSP and field programmable gate arrays (FPGA with hundreds of mega hertz in speed, will allow digital designers to embed essential digital signal processing units in small chips. But designing a synthesizable core on an FPGA is not always as simple as DSP chips due to complexity and limitations of FPGAs. In this paper we design anLMS-based FIR adaptive filter for adaptive noise cancellation based on VHDL97 hardware description language (HDL and Xilinx SPARTAN3E (XC3S500E which utilizes low resources and is high performance and FPGA-brand independent so can be implemented on different FPGA brands (Xilinx, ALTERA, ACTEL. Simulations are done in MODELSIM and MATLAB and implementation is done with Xilinx ISE. Finally, result are compared with other papers for better judgment.

  8. Interventions for preventing voice disorders in adults.

    Science.gov (United States)

    Ruotsalainen, J H; Sellman, J; Lehto, L; Jauhiainen, M; Verbeek, J H

    2007-10-17

    Poor voice quality due to a voice disorder can lead to a reduced quality of life. In occupations where voice use is substantial it can lead to periods of absence from work. To evaluate the effectiveness of interventions to prevent voice disorders in adults. We searched MEDLINE (PubMed, 1950 to 2006), EMBASE (1974 to 2006), CENTRAL (The Cochrane Library, Issue 2 2006), CINAHL (1983 to 2006), PsychINFO (1967 to 2006), Science Citation Index (1986 to 2006) and the Occupational Health databases OSH-ROM (to 2006). The date of the last search was 05/04/06. Randomised controlled clinical trials (RCTs) of interventions evaluating the effectiveness of treatments to prevent voice disorders in adults. For work-directed interventions interrupted time series and prospective cohort studies were also eligible. Two authors independently extracted data and assessed trial quality. Meta-analysis was performed where appropriate. We identified two randomised controlled trials including a total of 53 participants in intervention groups and 43 controls. One study was conducted with teachers and the other with student teachers. Both trials were poor quality. Interventions were grouped into 1) direct voice training, 2) indirect voice training and 3) direct and indirect voice training combined.1) Direct voice training: One study did not find a significant decrease of the Voice Handicap Index for direct voice training compared to no intervention.2) Indirect voice training: One study did not find a significant decrease of the Voice Handicap Index for indirect voice training when compared to no intervention.3) Direct and indirect voice training combined: One study did not find a decrease of the Voice Handicap Index for direct and indirect voice training combined when compared to no intervention. The same study did however find an improvement in maximum phonation time (Mean Difference -3.18 sec; 95 % CI -4.43 to -1.93) for direct and indirect voice training combined when compared to no

  9. Designing a Voice Controlled Interface For Radio : Guidelines for The First Generation of Voice Controlled Public Radio

    OpenAIRE

    Päärni, Anna

    2017-01-01

    From being a fictional element in sci-fi, voice control has become a reality, with inventions such as Apple's Siri, and interactive voice response (IVR) when calling your doctor's office. The combination of radio’s strength as a hands-free medium, public radio’s mission to reach across all platforms and the rise of voice makes up a relevant intersection; voice controlled public radio in Sweden. This thesis has aimed to investigate how radio listeners wish to interact using voice control to li...

  10. VOICES AGAINST EXTREMISM: A CASE STUDY OF A COMMUNITY-BASED CVE COUNTER-NARRATIVE CAMPAIGN

    Directory of Open Access Journals (Sweden)

    Logan Macnair

    2017-03-01

    Full Text Available This article presents a case study of the recently conceived and ongoing counter-extremism campaign, Voices Against Extremism, a campaign designed and implemented by university students from Vancouver, Canada. Through a multifaceted approach that includes extensive use of social media, academic research, and grassroots community activities and involvement, Voices Against Extremism operates under the mission statement of countering and preventing violent extremism and radicalization through the humanization of minority groups and through the education and engagement of the silent majority. This article examines the effectiveness of this campaign as a proactive counter-radicalization strategy by outlining its specific components and activities. Based on the results of this campaign, suggestions are then offered regarding specific counter-extremism and counter-radicalizations policies that may be adopted by law enforcement, policymakers – or any other organizations concerned with countering and preventing radicalization and violent extremism – with a specific focus on the potential benefits of proactive and long-term social and community engagement.

  11. Application of computer voice input/output

    International Nuclear Information System (INIS)

    Ford, W.; Shirk, D.G.

    1981-01-01

    The advent of microprocessors and other large-scale integration (LSI) circuits is making voice input and output for computers and instruments practical; specialized LSI chips for speech processing are appearing on the market. Voice can be used to input data or to issue instrument commands; this allows the operator to engage in other tasks, move about, and to use standard data entry systems. Voice synthesizers can generate audible, easily understood instructions. Using voice characteristics, a control system can verify speaker identity for security purposes. Two simple voice-controlled systems have been designed at Los Alamos for nuclear safeguards applicaations. Each can easily be expanded as time allows. The first system is for instrument control that accepts voice commands and issues audible operator prompts. The second system is for access control. The speaker's voice is used to verify his identity and to actuate external devices

  12. Presidential, But Not Prime Minister, Candidates With Lower Pitched Voices Stand a Better Chance of Winning the Election in Conservative Countries.

    Science.gov (United States)

    Banai, Benjamin; Laustsen, Lasse; Banai, Irena Pavela; Bovan, Kosta

    2018-01-01

    Previous studies have shown that voters rely on sexually dimorphic traits that signal masculinity and dominance when they choose political leaders. For example, voters exert strong preferences for candidates with lower pitched voices because these candidates are perceived as stronger and more competent. Moreover, experimental studies demonstrate that conservative voters, more than liberals, prefer political candidates with traits that signal dominance, probably because conservatives are more likely to perceive the world as a threatening place and to be more attentive to dangerous and threatening contexts. In light of these findings, this study investigates whether country-level ideology influences the relationship between candidate voice pitch and electoral outcomes of real elections. Specifically, we collected voice pitch data for presidential and prime minister candidates, aggregate national ideology for the countries in which the candidates were nominated, and measures of electoral outcomes for 69 elections held across the world. In line with previous studies, we found that candidates with lower pitched voices received more votes and had greater likelihood of winning the elections. Furthermore, regression analysis revealed an interaction between candidate voice pitch, national ideology, and election type (presidential or parliamentary). That is, having a lower pitched voice was a particularly valuable asset for presidential candidates in conservative and right-leaning countries (in comparison to presidential candidates in liberal and left-leaning countries and parliamentary elections). We discuss the practical implications of these findings, and how they relate to existing research on candidates' voices, voting preferences, and democratic elections in general.

  13. Voice and silence in organizations

    Directory of Open Access Journals (Sweden)

    Moaşa, H.

    2011-01-01

    Full Text Available Unlike previous research on voice and silence, this article breaksthe distance between the two and declines to treat them as opposites. Voice and silence are interrelated and intertwined strategic forms ofcommunication which presuppose each other in such a way that the absence of one would minimize completely the other’s presence. Social actors are not voice, or silence. Social actors can have voice or silence, they can do both because they operate at multiple levels and deal with multiple issues at different moments in time.

  14. FILTWAM and Voice Emotion Recognition

    NARCIS (Netherlands)

    Bahreini, Kiavash; Nadolski, Rob; Westera, Wim

    2014-01-01

    This paper introduces the voice emotion recognition part of our framework for improving learning through webcams and microphones (FILTWAM). This framework enables multimodal emotion recognition of learners during game-based learning. The main goal of this study is to validate the use of microphone

  15. Audio-visual identification of place of articulation and voicing in white and babble noise.

    Science.gov (United States)

    Alm, Magnus; Behne, Dawn M; Wang, Yue; Eg, Ragnhild

    2009-07-01

    Research shows that noise and phonetic attributes influence the degree to which auditory and visual modalities are used in audio-visual speech perception (AVSP). Research has, however, mainly focused on white noise and single phonetic attributes, thus neglecting the more common babble noise and possible interactions between phonetic attributes. This study explores whether white and babble noise differentially influence AVSP and whether these differences depend on phonetic attributes. White and babble noise of 0 and -12 dB signal-to-noise ratio were added to congruent and incongruent audio-visual stop consonant-vowel stimuli. The audio (A) and video (V) of incongruent stimuli differed either in place of articulation (POA) or voicing. Responses from 15 young adults show that, compared to white noise, babble resulted in more audio responses for POA stimuli, and fewer for voicing stimuli. Voiced syllables received more audio responses than voiceless syllables. Results can be attributed to discrepancies in the acoustic spectra of both the noise and speech target. Voiced consonants may be more auditorily salient than voiceless consonants which are more spectrally similar to white noise. Visual cues contribute to identification of voicing, but only if the POA is visually salient and auditorily susceptible to the noise type.

  16. SIGNAL RECONSTRUCTION PERFORMANCE OF THE ATLAS HADRONIC TILE CALORIMETER

    CERN Document Server

    Do Amaral Coutinho, Y; The ATLAS collaboration

    2013-01-01

    "The Tile Calorimeter for the ATLAS experiment at the CERN Large Hadron Collider (LHC) is a sampling calorimeter with steel as absorber and scintillators as active medium. The scintillators are readout by wavelength shifting fibers coupled to photomultiplier tubes (PMT). The analogue signals from the PMTs are amplified, shaped and digitized by sampling the signal every 25 ns. The TileCal front-end electronics allows to read out the signals produced by about 10000 channels measuring energies ranging from ~30 MeV to ~2 TeV. The read-out system is responsible for reconstructing the data in real-time fulfilling the tight time constraint imposed by the ATLAS first level trigger rate (100 kHz). The main component of the read-out system is the Digital Signal Processor (DSP) which, using an Optimal Filtering reconstruction algorithm, allows to compute for each channel the signal amplitude, time and quality factor at the required high rate. Currently the ATLAS detector and the LHC are undergoing an upgrade program tha...

  17. Part Summary of the Project ‘Speakers’ Comfort’: Teachers’ Voice use in Teaching Environments

    DEFF Research Database (Denmark)

    Lyberg-Åhlander, Viveka; Rydell, Roland; Löfqvist, Anders

    2015-01-01

    Classroom acoustics not always take the speaker’s comfort into consideration. The purpose of the presented papers was to investigate voice use, vocal behavior and prevalence of voice problems in Swedish teaching staff. Ratings of features in the work-environment on voice use were explored in n...... = 487 teachers. Based on their answers the respondents were split into two groups: teachers with self-assessed voice problems and voice-healthy teachers. Teachers with voice problems and were matched to a voice-healthy colleague from the same school and were investigated and compared for clinical...... findings and for vocal behavior. Acoustic properties of their teaching environments were measured. Teachers with voice-problems were more affected by any loading factor in the work-environment and were more aware of the room acoustics. Differences between the groups were found during field...

  18. A heterogeneous multi-core platform for low power signal processing in systems-on-chip

    DEFF Research Database (Denmark)

    Paker, Ozgun; Sparsø, Jens; Haandbæk, Niels

    2002-01-01

    is based on message passing. The mini-cores are designed as parameterized soft macros intended for a synthesis based design flow. A 520.000 transistor 0.25µm CMOS prototype chip containing 6 mini-cores has been fabricated and tested. Its power consumption is only 50% higher than a hardwired ASIC and more......This paper presents a low-power and programmable DSP architecture - a heterogeneous multiprocessor platform consisting of standard CPU/DSP cores, and a set of simple instruction set processors called mini-cores each optimized for a particular class of algorithm (FIR, IIR, LMS, etc.). Communication...

  19. Clinical voice analysis of Carnatic singers.

    Science.gov (United States)

    Arunachalam, Ravikumar; Boominathan, Prakash; Mahalingam, Shenbagavalli

    2014-01-01

    Carnatic singing is a classical South Indian style of music that involves rigorous training to produce an "open throated" loud, predominantly low-pitched singing, embedded with vocal nuances in higher pitches. Voice problems in singers are not uncommon. The objective was to report the nature of voice problems and apply a routine protocol to assess the voice. Forty-five trained performing singers (females: 36 and males: 9) who reported to a tertiary care hospital with voice problems underwent voice assessment. The study analyzed their problems and the clinical findings. Voice change, difficulty in singing higher pitches, and voice fatigue were major complaints. Most of the singers suffered laryngopharyngeal reflux that coexisted with muscle tension dysphonia and chronic laryngitis. Speaking voices were rated predominantly as "moderate deviation" on GRBAS (Grade, Rough, Breathy, Asthenia, and Strain). Maximum phonation time ranged from 4 to 29 seconds (females: 10.2, standard deviation [SD]: 5.28 and males: 15.7, SD: 5.79). Singing frequency range was reduced (females: 21.3 Semitones and males: 23.99 Semitones). Dysphonia severity index (DSI) scores ranged from -3.5 to 4.91 (females: 0.075 and males: 0.64). Singing frequency range and DSI did not show significant difference between sex and across clinical diagnosis. Self-perception using voice disorder outcome profile revealed overall severity score of 5.1 (SD: 2.7). Findings are discussed from a clinical intervention perspective. Study highlighted the nature of voice problems (hyperfunctional) and required modifications in assessment protocol for Carnatic singers. Need for regular assessments and vocal hygiene education to maintain good vocal health are emphasized as outcomes. Copyright © 2014 The Voice Foundation. Published by Mosby, Inc. All rights reserved.

  20. The Signal Validation method of Digital Process Instrumentation System on signal conditioner for SMART

    International Nuclear Information System (INIS)

    Moon, Hee Gun; Park, Sang Min; Kim, Jung Seon; Shon, Chang Ho; Park, Heui Youn; Koo, In Soo

    2005-01-01

    The function of PIS(Process Instrumentation System) for SMART is to acquire the process data from sensor or transmitter. The PIS consists of signal conditioner, A/D converter, DSP(Digital Signal Process) and NIC(Network Interface Card). So, It is fully digital system after A/D converter. The PI cabinet and PDAS(Plant Data Acquisition System) in commercial plant is responsible for data acquisition of the sensor or transmitter include RTD, TC, level, flow, pressure and so on. The PDAS has the software that processes each sensor data and PI cabinet has the signal conditioner, which is need for maintenance and test. The signal conditioner has the potentiometer to adjust the span and zero for test and maintenance. The PIS of SMART also has the signal conditioner which has the span and zero adjust same as the commercial plant because the signal conditioner perform the signal condition for AD converter such as 0∼10Vdc. But, To adjust span and zero is manual test and calibration. So, This paper presents the method of signal validation and calibration, which is used by digital feature in SMART. There are I/E(current to voltage), R/E(resistor to voltage), F/E(frequency to voltage), V/V(voltage to voltage). Etc. In this paper show only the signal validation and calibration about I/E converter that convert level, pressure, flow such as 4∼20mA into signal for AD conversion such as 0∼10Vdc

  1. Voice Biometrics for Information Assurance Applications

    National Research Council Canada - National Science Library

    Kang, George

    2002-01-01

    .... The ultimate goal of voice biometrics is to enable the use of voice as a password. Voice biometrics are "man-in-the-loop" systems in which system performance is significantly dependent on human performance...

  2. Analysis of failure of voice production by a sound-producing voice prosthesis

    NARCIS (Netherlands)

    van der Torn, M.; van Gogh, C.D.L.; Verdonck-de Leeuw, I M; Festen, J.M.; Mahieu, H.F.

    OBJECTIVE: To analyse the cause of failing voice production by a sound-producing voice prosthesis (SPVP). METHODS: The functioning of a prototype SPVP is described in a female laryngectomee before and after its sound-producing mechanism was impeded by tracheal phlegm. This assessment included:

  3. The relation of vocal fold lesions and voice quality to voice handicap and psychosomatic well-being

    NARCIS (Netherlands)

    Smits, R.; Marres, H.A.; de Jong, F.

    2012-01-01

    BACKGROUND: Voice disorders have a multifactorial genesis and may be present in various ways. They can cause a significant communication handicap and impaired quality of life. OBJECTIVE: To assess the effect of vocal fold lesions and voice quality on voice handicap and psychosomatic well-being.

  4. Estimation of Length and Order of Polynomial-based Filter Implemented in the Form of Farrow Structure

    Directory of Open Access Journals (Sweden)

    S. Vukotic

    2016-08-01

    Full Text Available Digital polynomial-based interpolation filters implemented using the Farrow structure are used in Digital Signal Processing (DSP to calculate the signal between its discrete samples. The two basic design parameters for these filters are number of polynomial-segments defining the finite length of impulse response, and order of polynomials in each polynomial segment. The complexity of the implementation structure and the frequency domain performance depend on these two parameters. This contribution presents estimation formulae for length and polynomial order of polynomial-based filters for various types of requirements including attenuation in stopband, width of transitions band, deviation in passband, weighting in passband/stopband.

  5. Voice Onset Time in Azerbaijani Consonants

    Directory of Open Access Journals (Sweden)

    Ali Jahan

    2009-10-01

    Full Text Available Objective: Voice onset time is known to be cue for the distinction between voiced and voiceless stops and it can be used to describe or categorize a range of developmental, neuromotor and linguistic disorders. The aim of this study is determination of standard values of voice onset time for Azerbaijani language (Tabriz dialect. Materials & Methods: In this description-analytical study, 30 Azeris persons whom were selected conveniently by simple selection, uttered 46 monosyllabic words initiating with 6 Azerbaijani stops twice. Using Praat software, the voice onset time values were analyzed by waveform and wideband spectrogram in milliseconds. Vowel effect, sex differences and the effect of place of articulation on VOT, were evaluated and data were analyzed by one-way ANOVA test. Results: There was no significant difference in voice onset time between male and female Azeris speakers (P<0.05. Vowel and place of articulation had significant correlation with voice onset time (P<0.001. Voice onset time values for /b/, /p/, /d/, /t/, /g/, /k/, and [c], [ɟ] allophones were 10.64, 86.88, 13.35, 87.09, 26.25, 100.62, 131.19, 63.18 mili second, respectively. Conclusion: Voice onset time values are the same for Azerbaijani men and women. However, like many other languages, back and high vowels and back place of articulation lengthen VOT. Also, voiceless stops are aspirated in this language and voiced stops have positive VOT values.

  6. Does CPAP treatment affect the voice?

    Science.gov (United States)

    Saylam, Güleser; Şahin, Mustafa; Demiral, Dilek; Bayır, Ömer; Yüceege, Melike Bağnu; Çadallı Tatar, Emel; Korkmaz, Mehmet Hakan

    2016-12-20

    The aim of this study was to investigate alterations in voice parameters among patients using continuous positive airway pressure (CPAP) for the treatment of obstructive sleep apnea syndrome. Patients with an indication for CPAP treatment without any voice problems and with normal laryngeal findings were included and voice parameters were evaluated before and 1 and 6 months after CPAP. Videolaryngostroboscopic findings, a self-rated scale (Voice Handicap Index-10, VHI-10), perceptual voice quality assessment (GRBAS: grade, roughness, breathiness, asthenia, strain), and acoustic parameters were compared. Data from 70 subjects (48 men and 22 women) with a mean age of 44.2 ± 6.0 years were evaluated. When compared with the pre-CPAP treatment period, there was a significant increase in the VHI-10 score after 1 month of treatment and in VHI- 10 and total GRBAS scores, jitter percent (P = 0.01), shimmer percent, noise-to-harmonic ratio, and voice turbulence index after 6 months of treatment. Vague negative effects on voice parameters after the first month of CPAP treatment became more evident after 6 months. We demonstrated nonsevere alterations in the voice quality of patients under CPAP treatment. Given that CPAP is a long-term treatment it is important to keep these alterations in mind.

  7. Managing dysphonia in occupational voice users.

    Science.gov (United States)

    Behlau, Mara; Zambon, Fabiana; Madazio, Glaucya

    2014-06-01

    Recent advances with regard to occupational voice disorders are highlighted with emphasis on issues warranting consideration when assessing, training, and treating professional voice users. Findings include the many particularities between the various categories of professional voice users, the concept that the environment plays a major role in occupational voice disorders, and that biopsychosocial influences should be analyzed on an individual basis. Assessment via self-evaluation protocols to quantify the impact of these disorders is mandatory as a component of an evaluation and to document treatment outcomes. Discomfort or odynophonia has evolved as a critical symptom in this population. Clinical trials are limited and the complexity of the environment may be a limitation in experiment design. This review reinforced the need for large population studies of professional voice users; new data highlighted important factors specific to each group of voice users. Interventions directed at student teachers are necessities to not only improving the quality of future professionals, but also to avoid the frustration and limitations associated with chronic voice problems. The causative relationship between the work environment and voice disorders has not yet been established. Randomized controlled trials are lacking and must be a focus to enhance treatment paradigms for this population.

  8. Stated product formulation preferences for HIV pre-exposure prophylaxis among women in the VOICE-D (MTN-003D) study.

    Science.gov (United States)

    Luecke, Ellen H; Cheng, Helen; Woeber, Kubashni; Nakyanzi, Teopista; Mudekunye-Mahaka, Imelda C; van der Straten, Ariane

    2016-01-01

    The effectiveness of HIV pre-exposure prophylaxis (PrEP) requires consistent and correct product use, thus a deeper understanding of women's stated product formulation preferences, and the correlates of those preferences, can help guide future research. VOICE-D (MTN-003D), a qualitative ancillary study conducted after the VOICE trial, retrospectively explored participants' tablet and gel use, as well as their preferences for other potential PrEP product formulations. We conducted an analysis of quantitative and qualitative data from VOICE-D participants. During in-depth interviews, women were presented with pictures and descriptions of eight potential PrEP product formulations, including the oral tablet and vaginal gel tested in VOICE, and asked to discuss which product formulations they would prefer to use and why. Seven of the original product formulations displayed were combined into preferred product formulation categories based on exploratory factor and latent class analyses. We examined demographic and behavioural correlates of these preferred product formulation categories. In-depth interviews with participants were conducted, coded, and analysed for themes related to product preference. Of the 68 female participants who completed in-depth interviews (22 South Africa, 24 Zimbabwe, 22 Uganda), median age was 28 (range 21-41), 81% were HIV negative, and 49% were married or living with a partner. Four preferred product formulation categories were identified via exploratory factor analysis: 1) oral tablets; 2) vaginal gel; 3) injectable, implant, or vaginal ring; and 4) vaginal film or suppository. A majority of women (81%) expressed a preference for product formulations included in category 3. Characteristics significantly associated with each preferred product category differed. Attributes described by participants as being important in a preferred product formulation included duration of activity, ease of use, route of administration, clinic- versus self

  9. Implementation of medical monitor system based on networks

    Science.gov (United States)

    Yu, Hui; Cao, Yuzhen; Zhang, Lixin; Ding, Mingshi

    2006-11-01

    In this paper, the development trend of medical monitor system is analyzed and portable trend and network function become more and more popular among all kinds of medical monitor devices. The architecture of medical network monitor system solution is provided and design and implementation details of medical monitor terminal, monitor center software, distributed medical database and two kind of medical information terminal are especially discussed. Rabbit3000 system is used in medical monitor terminal to implement security administration of data transfer on network, human-machine interface, power management and DSP interface while DSP chip TMS5402 is used in signal analysis and data compression. Distributed medical database is designed for hospital center according to DICOM information model and HL7 standard. Pocket medical information terminal based on ARM9 embedded platform is also developed to interactive with center database on networks. Two kernels based on WINCE are customized and corresponding terminal software are developed for nurse's routine care and doctor's auxiliary diagnosis. Now invention patent of the monitor terminal is approved and manufacture and clinic test plans are scheduled. Applications for invention patent are also arranged for two medical information terminals.

  10. Voice Assessment of Student Work: Recent Studies and Emerging Technologies

    Science.gov (United States)

    Eckhouse, Barry; Carroll, Rebecca

    2013-01-01

    Although relatively little attention has been given to the voice assessment of student work, at least when compared with more traditional forms of text-based review, the attention it has received strongly points to a promising form of review that has been hampered by the limits of an emerging technology. A fresh review of voice assessment in light…

  11. Epidemiology of Voice Disorders in Latvian School Teachers.

    Science.gov (United States)

    Trinite, Baiba

    2017-07-01

    The prevalence of voice disorders in the teacher population in Latvia has not been studied so far and this is the first epidemiological study whose goal is to investigate the prevalence of voice disorders and their risk factors in this professional group. A wide cross-sectional study using stratified sampling methodology was implemented in the general education schools of Latvia. The self-administered voice risk factor questionnaire and the Voice Handicap Index were completed by 522 teachers. Two teachers groups were formed: the voice disorders group which included 235 teachers with actual voice problems or problems during the last 9 months; and the control group which included 174 teachers without voice disorders. Sixty-six percent of teachers gave a positive answer to the following question: Have you ever had problems with your voice? Voice problems are more often found in female than male teachers (68.2% vs 48.8%). Music teachers suffer from voice disorders more often than teachers of other subjects. Eighty-two percent of teachers first faced voice problems in their professional carrier. The odds of voice disorders increase if the following risk factors exist: extra vocal load, shouting, throat clearing, neglecting of personal health, background noise, chronic illnesses of the upper respiratory tract, allergy, job dissatisfaction, and regular stress in the working place. The study findings indicated a high risk of voice disorders among Latvian teachers. The study confirmed data concerning the multifactorial etiology of voice disorders. Copyright © 2017 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

  12. The stability of locus equation slopes across stop consonant voicing/aspiration

    Science.gov (United States)

    Sussman, Harvey M.; Modarresi, Golnaz

    2004-05-01

    The consistency of locus equation slopes as phonetic descriptors of stop place in CV sequences across voiced and voiceless aspirated stops was explored in the speech of five male speakers of American English and two male speakers of Persian. Using traditional locus equation measurement sites for F2 onsets, voiceless labial and coronal stops had significantly lower locus equation slopes relative to their voiced counterparts, whereas velars failed to show voicing differences. When locus equations were derived using F2 onsets for voiced stops that were measured closer to the stop release burst, comparable to the protocol for measuring voiceless aspirated stops, no significant effects of voicing/aspiration on locus equation slopes were observed. This methodological factor, rather than an underlying phonetic-based explanation, provides a reasonable account for the observed flatter locus equation slopes of voiceless labial and coronal stops relative to voiced cognates reported in previous studies [Molis et al., J. Acoust. Soc. Am. 95, 2925 (1994); O. Engstrand and B. Lindblom, PHONUM 4, 101-104]. [Work supported by NIH.

  13. Voice Over Internet Protocol (VoIP) in a Control Center Environment

    Science.gov (United States)

    Pirani, Joseph; Calvelage, Steven

    2010-01-01

    The technology of transmitting voice over data networks has been available for over 10 years. Mass market VoIP services for consumers to make and receive standard telephone calls over broadband Internet networks have grown in the last 5 years. While operational costs are less with VoIP implementations as opposed to time division multiplexing (TDM) based voice switches, is it still advantageous to convert a mission control center s voice system to this newer technology? Marshall Space Flight Center (MSFC) Huntsville Operations Support Center (HOSC) has converted its mission voice services to a commercial product that utilizes VoIP technology. Results from this testing, design, and installation have shown unique considerations that must be addressed before user operations. There are many factors to consider for a control center voice design. Technology advantages and disadvantages were investigated as they refer to cost. There were integration concerns which could lead to complex failure scenarios but simpler integration for the mission infrastructure. MSFC HOSC will benefit from this voice conversion with less product replacement cost, less operations cost and a more integrated mission services environment.

  14. Perceived Benefits and Drawbacks of Synchronous Voice-Based Computer-Mediated Communication in the Foreign Language Classroom

    Science.gov (United States)

    Bueno Alastuey, M. C.

    2011-01-01

    This study explored the benefits and drawbacks of synchronous voice-based computer-mediated communication (CMC) in a blended course of English for specific purposes. Quantitative and qualitative data from two groups following the same syllabus, except for the oral component, were compared. Oral tasks were carried out face-to-face with same L1…

  15. Identifying hidden voice and video streams

    Science.gov (United States)

    Fan, Jieyan; Wu, Dapeng; Nucci, Antonio; Keralapura, Ram; Gao, Lixin

    2009-04-01

    Given the rising popularity of voice and video services over the Internet, accurately identifying voice and video traffic that traverse their networks has become a critical task for Internet service providers (ISPs). As the number of proprietary applications that deliver voice and video services to end users increases over time, the search for the one methodology that can accurately detect such services while being application independent still remains open. This problem becomes even more complicated when voice and video service providers like Skype, Microsoft, and Google bundle their voice and video services with other services like file transfer and chat. For example, a bundled Skype session can contain both voice stream and file transfer stream in the same layer-3/layer-4 flow. In this context, traditional techniques to identify voice and video streams do not work. In this paper, we propose a novel self-learning classifier, called VVS-I , that detects the presence of voice and video streams in flows with minimum manual intervention. Our classifier works in two phases: training phase and detection phase. In the training phase, VVS-I first extracts the relevant features, and subsequently constructs a fingerprint of a flow using the power spectral density (PSD) analysis. In the detection phase, it compares the fingerprint of a flow to the existing fingerprints learned during the training phase, and subsequently classifies the flow. Our classifier is not only capable of detecting voice and video streams that are hidden in different flows, but is also capable of detecting different applications (like Skype, MSN, etc.) that generate these voice/video streams. We show that our classifier can achieve close to 100% detection rate while keeping the false positive rate to less that 1%.

  16. A FRAMEWORK FOR INTELLIGENT VOICE-ENABLED E-EDUCATION SYSTEMS

    Directory of Open Access Journals (Sweden)

    Azeta A. A.

    2009-07-01

    Full Text Available Although the Internet has received significant attention in recent years, voice is still the most convenient and natural way of communicating between human to human or human to computer. In voice applications, users may have different needs which will require the ability of the system to reason, make decisions, be flexible and adapt to requests during interaction. These needs have placed new requirements in voice application development such as use of advanced models, techniques and methodologies which take into account the needs of different users and environments. The ability of a system to behave close to human reasoning is often mentioned as one of the major requirements for the development of voice applications. In this paper, we present a framework for an intelligent voice-enabled e-Education application and an adaptation of the framework for the development of a prototype Course Registration and Examination (CourseRegExamOnline module. This study is a preliminary report of an ongoing e-Education project containing the following modules: enrollment, course registration and examination, enquiries/information, messaging/collaboration, e-Learning and library. The CourseRegExamOnline module was developed using VoiceXML for the voice user interface(VUI, PHP for the web user interface (WUI, Apache as the middle-ware and MySQL database as back-end. The system would offer dual access modes using the VUI and WUI. The framework would serve as a reference model for developing voice-based e-Education applications. The e-Education system when fully developed would meet the needs of students who are normal users and those with certain forms of disabilities such as visual impairment, repetitive strain injury (RSI, etc, that make reading and writing difficult.

  17. Prediction of body mass index status from voice signals based on machine learning for automated medical applications.

    Science.gov (United States)

    Lee, Bum Ju; Kim, Keun Ho; Ku, Boncho; Jang, Jun-Su; Kim, Jong Yeol

    2013-05-01

    The body mass index (BMI) provides essential medical information related to body weight for the treatment and prognosis prediction of diseases such as cardiovascular disease, diabetes, and stroke. We propose a method for the prediction of normal, overweight, and obese classes based only on the combination of voice features that are associated with BMI status, independently of weight and height measurements. A total of 1568 subjects were divided into 4 groups according to age and gender differences. We performed statistical analyses by analysis of variance (ANOVA) and Scheffe test to find significant features in each group. We predicted BMI status (normal, overweight, and obese) by a logistic regression algorithm and two ensemble classification algorithms (bagging and random forests) based on statistically significant features. In the Female-2030 group (females aged 20-40 years), classification experiments using an imbalanced (original) data set gave area under the receiver operating characteristic curve (AUC) values of 0.569-0.731 by logistic regression, whereas experiments using a balanced data set gave AUC values of 0.893-0.994 by random forests. AUC values in Female-4050 (females aged 41-60 years), Male-2030 (males aged 20-40 years), and Male-4050 (males aged 41-60 years) groups by logistic regression in imbalanced data were 0.585-0.654, 0.581-0.614, and 0.557-0.653, respectively. AUC values in Female-4050, Male-2030, and Male-4050 groups in balanced data were 0.629-0.893 by bagging, 0.707-0.916 by random forests, and 0.695-0.854 by bagging, respectively. In each group, we found discriminatory features showing statistical differences among normal, overweight, and obese classes. The results showed that the classification models built by logistic regression in imbalanced data were better than those built by the other two algorithms, and significant features differed according to age and gender groups. Our results could support the development of BMI diagnosis

  18. DSP30 enhances the immunosuppressive properties of mesenchymal stromal cells and protects their suppressive potential from lipopolysaccharide effects: A potential role of adenosine.

    Science.gov (United States)

    Sangiorgi, Bruno; De Freitas, Helder Teixeira; Schiavinato, Josiane Lilian Dos Santos; Leão, Vitor; Haddad, Rodrigo; Orellana, Maristela Delgado; Faça, Vitor Marcel; Ferreira, Germano Aguiar; Covas, Dimas Tadeu; Zago, Marco Antônio; Panepucci, Rodrigo Alexandre

    2016-07-01

    Multipotent mesenchymal stromal cells (MSC) are imbued with an immunosuppressive phenotype that extends to several immune system cells. In this study, we evaluated how distinct Toll-like receptor (TLR) agonists impact immunosuppressive properties of bone marrow (BM)-MSC and explored the potential mechanisms involved. We show that TLR4 stimulation by lipopolysaccharide (LPS) restricted the ability of MSC to suppress the proliferation of T lymphocytes, increasing the gene expression of interleukin (IL)-1β and IL-6. In contrast, stimulation of TLR9 by DSP30 induced proliferation and the suppressive potential of BM-MSC, coinciding with reducing tumor necrosis factor (TNF)-α expression, increased expression of transforming growth factor (TGF)-β1, increased percentages of BM-MSC double positive for the ectonucleotidases CD39+CD73+ and adenosine levels. Importantly, following simultaneous stimulation with LPS and DSP30, BM-MSC's ability to suppress T lymphocyte proliferation was comparable with that of non-stimulated BM-MSC levels. Moreover, stimulation of BM-MSC with LPS reduced significantly the gene expression levels, on co-cultured T lymphocyte, of IL-10 and interferon (IFN)γ, a cytokine with potential to enhance the immunosuppression mediated by MSC and ameliorate the clinical outcome of patients with graft-versus-host disease (GVHD). Altogether, our findings reiterate the harmful effects of LPS on MSC immunosuppression, besides indicating that DSP30 could provide a protective effect against LPS circulating in the blood of GVHD patients who receive BM-MSC infusions, ensuring a more predictable immunosuppressive effect. The novel effects and potential mechanisms following the stimulation of BM-MSC by DSP30 might impact their clinical use, by allowing the derivation of optimal "licensing" protocols for obtaining therapeutically efficient MSC. Copyright © 2016 International Society for Cellular Therapy. Published by Elsevier Inc. All rights reserved.

  19. Implementation of orthogonal frequency division multiplexing (OFDM) and advanced signal processing for elastic optical networking in accordance with networking and transmission constraints

    Science.gov (United States)

    Johnson, Stanley

    An increasing adoption of digital signal processing (DSP) in optical fiber telecommunication has brought to the fore several interesting DSP enabled modulation formats. One such format is orthogonal frequency division multiplexing (OFDM), which has seen great success in wireless and wired RF applications, and is being actively investigated by several research groups for use in optical fiber telecom. In this dissertation, I present three implementations of OFDM for elastic optical networking and distributed network control. The first is a field programmable gate array (FPGA) based real-time implementation of a version of OFDM conventionally known as intensity modulation and direct detection (IMDD) OFDM. I experimentally demonstrate the ability of this transmission system to dynamically adjust bandwidth and modulation format to meet networking constraints in an automated manner. To the best of my knowledge, this is the first real-time software defined networking (SDN) based control of an OFDM system. In the second OFDM implementation, I experimentally demonstrate a novel OFDM transmission scheme that supports both direct detection and coherent detection receivers simultaneously using the same OFDM transmitter. This interchangeable receiver solution enables a trade-off between bit rate and equipment cost in network deployment and upgrades. I show that the proposed transmission scheme can provide a receiver sensitivity improvement of up to 1.73 dB as compared to IMDD OFDM. I also present two novel polarization analyzer based detection schemes, and study their performance using experiment and simulation. In the third implementation, I present an OFDM pilot-tone based scheme for distributed network control. The first instance of an SDN-based OFDM elastic optical network with pilot-tone assisted distributed control is demonstrated. An improvement in spectral efficiency and a fast reconfiguration time of 30 ms have been achieved in this experiment. Finally, I

  20. DSP-Enabled Radio Astronomy: Towards IIIZW35 Reconquest

    Directory of Open Access Journals (Sweden)

    Alain Lecacheux

    2005-09-01

    Full Text Available In radio astronomy, the radio spectrum is used to detect weak emission from celestial sources. By spectral averaging, observation noise is reduced and weak sources can be detected. However, more and more observations are polluted by man-made radio frequency interferences (RFI. The impact of these RFIs on power spectral measurement ranges from total saturation to subtle distortions of the data. To some extent, elimination of artefacts can be achieved by blanking polluted channels in real time. With this aim in view, a complete real-time digital system has been implemented on a set of FPGA and DSP. The current functionalities of the digital system have high dynamic range of 70 dB, bandwidth selection facilities ranging from 875 kHz to 14 MHz, high spectral resolution through a polyphase filter bank with up to 8192 channels with 49 152 coefficients and real-time time-frequency blanking with a robust threshold detector. This receiver has been used to reobserve the IIIWZ35 astronomical source which has been scrambled by a strong satellite RFI for several years.

  1. Occupational risk factors and voice disorders.

    Science.gov (United States)

    Vilkman, E

    1996-01-01

    From the point of view of occupational health, the field of voice disorders is very poorly developed as compared, for instance, to the prevention and diagnostics of occupational hearing disorders. In fact, voice disorders have not even been recognized in the field of occupational medicine. Hence, it is obviously very rare in most countries that the voice disorder of a professional voice user, e.g. a teacher, a singer or an actor, is accepted as an occupational disease by insurance companies. However, occupational voice problems do not lack significance from the point of view of the patient. We also know from questionnaires and clinical studies that voice complaints are very common. Another example of job-related health problems, which has proved more successful in terms of its occupational health status, is the repetition strain injury of the elbow, i.e. the "tennis elbow". Its textbook definition could be used as such to describe an occupational voice disorder ("dysphonia professional is"). In the present paper the effects of such risk factors as vocal loading itself, background noise and room acoustics and low relative humidity of the air are discussed. Due to individual factors underlying the development of professional voice disorders, recommendations rather than regulations are called for. There are many simple and even relatively low-cost methods available for the prevention of vocal problems as well as for supporting rehabilitation.

  2. A user configurable data acquisition and signal processing system for high-rate, high channel count applications

    International Nuclear Information System (INIS)

    Salim, Arwa; Crockett, Louise; McLean, John; Milne, Peter

    2012-01-01

    Highlights: ► The development of a new digital signal processing platform is described. ► The system will allow users to configure the real-time signal processing through software routines. ► The architecture of the DRUID system and signal processing elements is described. ► A prototype of the DRUID system has been developed for the digital chopper-integrator. ► The results of acquisition on 96 channels at 500 kSamples/s per channel are presented. - Abstract: Real-time signal processing in plasma fusion experiments is required for control and for data reduction as plasma pulse times grow longer. The development time and cost for these high-rate, multichannel signal processing systems can be significant. This paper proposes a new digital signal processing (DSP) platform for the data acquisition system that will allow users to easily customize real-time signal processing systems to meet their individual requirements. The D-TACQ reconfigurable user in-line DSP (DRUID) system carries out the signal processing tasks in hardware co-processors (CPs) implemented in an FPGA, with an embedded microprocessor (μP) for control. In the fully developed platform, users will be able to choose co-processors from a library and configure programmable parameters through the μP to meet their requirements. The DRUID system is implemented on a Spartan 6 FPGA, on the new rear transition module (RTM-T), a field upgrade to existing D-TACQ digitizers. As proof of concept, a multiply-accumulate (MAC) co-processor has been developed, which can be configured as a digital chopper-integrator for long pulse magnetic fusion devices. The DRUID platform allows users to set options for the integrator, such as the number of masking samples. Results from the digital integrator are presented for a data acquisition system with 96 channels simultaneously acquiring data at 500 kSamples/s per channel.

  3. Voice Response Systems Technology.

    Science.gov (United States)

    Gerald, Jeanette

    1984-01-01

    Examines two methods of generating synthetic speech in voice response systems, which allow computers to communicate in human terms (speech), using human interface devices (ears): phoneme and reconstructed voice systems. Considerations prior to implementation, current and potential applications, glossary, directory, and introduction to Input Output…

  4. Clinical Voices - an update

    DEFF Research Database (Denmark)

    Fusaroli, Riccardo; Weed, Ethan

    Anomalous aspects of speech and voice, including pitch, fluency, and voice quality, are reported to characterise many mental disorders. However, it has proven difficult to quantify and explain this oddness of speech by employing traditional statistical methods. In this talk we will show how...

  5. Application of 4G-based Emergency Communication System in Anhui Electric Power Communication%基于4G的应急通信在安徽电力的应用

    Institute of Scientific and Technical Information of China (English)

    谢小军; 梁本仁

    2013-01-01

    In order to improve the ability in responding to emergent hazards, Anhui Electric Power Company establishes an emergency communication system based on 4G communication technology. Multiple communication modes are proposed, including 3G public network and microwave self-organized network, thus to realize no-blind-area coverage in emergency zone. Meanwhile, a variety of voice network interconnection modules based on IP soft switch technology and DSP signal processing technology are designed, and the network communication problems in complex environment solved. Thus different types of speech communication terminal are integrated to achieve inter connectivity; while in video transmission, microwave communication technology is utilized to achieve real-time speech and video transmission when the public network breaks off. The system could provide a new communication means for emergency rescue personnel.%  为提高应对突发事件的能力,安徽省电力公司建设了基于4G通信技术的应急通信系统。提出了3G 公网,微波自组网多种通信模式互补通信的方案,实现应急区域无盲区覆盖;设计开发了基于 IP软交换技术及 DSP 信号处理技术的多种语音网络互连模块,可以将不同类型的语音通信终端融合起来,实现互连互通,在视频传输方面,利用微波通信技术,解决了公网中断的应急情况下语音视频的实时传输。为应急救援人员提供了新的通信手段。

  6. Increased chemokine signaling in a model of HIV1-associated peripheral neuropathy

    Directory of Open Access Journals (Sweden)

    Buchanan David J

    2009-08-01

    Full Text Available Abstract Painful distal sensory polyneuropathy (DSP is the most common neurological complication of HIV1 infection. Although infection with the virus itself is associated with an incidence of DSP, patients are more likely to become symptomatic following initiation of nucleoside reverse transcriptase inhibitor (NRTI treatment. The chemokines monocyte chemoattractant protein-1 (MCP1/CCL2 and stromal derived factor-1 (SDF1/CXCL12 and their respective receptors, CCR2 and CXCR4, have been implicated in HIV1 related neuropathic pain mechanisms including NRTI treatment in rodents. Utilizing a rodent model that incorporates the viral coat protein, gp120, and the NRTI, 2'3'-dideoxycytidine (ddC, we examined the degree to which chemokine receptor signaling via CCR2 and CXCR4 potentially influences the resultant chronic hypernociceptive behavior. We observed that following unilateral gp120 sciatic nerve administration, rats developed profound tactile hypernociception in the hindpaw ipsilateral to gp120 treatment. Behavioral changes were also present in the hindpaw contralateral to the injury, albeit delayed and less robust. Using immunohistochemical studies, we demonstrated that MCP1 and CCR2 were upregulated by primary sensory neurons in lumbar ganglia by post-operative day (POD 14. The functional nature of these observations was confirmed using calcium imaging in acutely dissociated lumbar dorsal root ganglion (DRG derived from gp120 injured rats at POD 14. Tactile hypernociception in gp120 treated animals was reversed following treatment with a CCR2 receptor antagonist at POD 14. Some groups of animals were subjected to gp120 sciatic nerve injury in combination with an injection of ddC at POD 14. This injury paradigm produced pronounced bilateral tactile hypernociception from POD 14–48. More importantly, functional MCP1/CCR2 and SDF1/CXCR4 signaling was present in sensory neurons. In contrast to gp120 treatment alone, the hypernociceptive behavior

  7. Changes after voice therapy in objective and subjective voice measurements of pediatric patients with vocal nodules.

    Science.gov (United States)

    Tezcaner, Ciler Zahide; Karatayli Ozgursoy, Selmin; Ozgursoy, Selmin Karatayli; Sati, Isil; Dursun, Gursel

    2009-12-01

    The aim of this study was to analyze the efficiency of the voice therapy in children with vocal nodules by using the acoustic analysis and subjective assessment. Thirty-nine patients with vocal fold nodules, aged between 7 and 14, were included in the study. Each subject had voice therapy led by an experienced voice therapist once a week. All diagnostic and follow-up workouts were performed before the voice therapy and after the third or the sixth month. Transoral and/or transnasal videostroboscopic examination and acoustic analysis were achieved using multi-dimensional voice program (MDVP) and subjective analysis with GRBAS scale. As for the perceptual assessment, the difference was significant for four parameters out of five. A significant improvement was found in the acoustic analysis parameters of jitter, shimmer, and noise-to-harmonic ratio. The voice therapy which was planned according to patients' needs, age, compliance and response to therapy had positive effects on pediatric patients with vocal nodules. Acoustic analysis and GRBAS may be used successfully in the follow-up of pediatric vocal nodule treatment.

  8. VoiceThread as a Peer Review and Dissemination Tool for Undergraduate Research

    Science.gov (United States)

    Guertin, L. A.

    2012-12-01

    VoiceThread has been utilized in an undergraduate research methods course for peer review and final research project dissemination. VoiceThread (http://www.voicethread.com) can be considered a social media tool, as it is a web-based technology with the capacity to enable interactive dialogue. VoiceThread is an application that allows a user to place a media collection online containing images, audio, videos, documents, and/or presentations in an interface that facilitates asynchronous communication. Participants in a VoiceThread can be passive viewers of the online content or engaged commenters via text, audio, video, with slide annotations via a doodle tool. The VoiceThread, which runs across browsers and operating systems, can be public or private for viewing and commenting and can be embedded into any website. Although few university students are aware of the VoiceThread platform (only 10% of the students surveyed by Ng (2012)), the 2009 K-12 edition of The Horizon Report (Johnson et al., 2009) lists VoiceThread as a tool to watch because of the opportunities it provides as a collaborative learning environment. In Fall 2011, eleven students enrolled in an undergraduate research methods course at Penn State Brandywine each conducted their own small-scale research project. Upon conclusion of the projects, students were required to create a poster summarizing their work for peer review. To facilitate the peer review process outside of class, each student-created PowerPoint file was placed in a VoiceThread with private access to only the class members and instructor. Each student was assigned to peer review five different student posters (i.e., VoiceThread images) with the audio and doodle tools to comment on formatting, clarity of content, etc. After the peer reviews were complete, the students were allowed to edit their PowerPoint poster files for a new VoiceThread. In the new VoiceThread, students were required to video record themselves describing their research

  9. Playful Interaction with Voice Sensing Modular Robots

    DEFF Research Database (Denmark)

    Heesche, Bjarke; MacDonald, Ewen; Fogh, Rune

    2013-01-01

    This paper describes a voice sensor, suitable for modular robotic systems, which estimates the energy and fundamental frequency, F0, of the user’s voice. Through a number of example applications and tests with children, we observe how the voice sensor facilitates playful interaction between child...... children and two different robot configurations. In future work, we will investigate if such a system can motivate children to improve voice control and explore how to extend the sensor to detect emotions in the user’s voice....

  10. Objective Voice Parameters in Colombian School Workers with Healthy Voices

    Directory of Open Access Journals (Sweden)

    Lady Catherine Cantor Cutiva

    2015-09-01

    Full Text Available Objectives: To characterize the objective voice parameters among school workers, and to identi­fy associated factors of three objective voice parameters, namely fundamental frequency, sound pressure level and maximum phonation time. Materials and methods: We conducted a cross-sectional study among 116 Colombian teachers and 20 Colombian non-teachers. After signing the informed consent form, participants filled out a questionnaire. Then, a voice sample was recorded and evaluated perceptually by a speech therapist and by objective voice analysis with praat software. Short-term environmental measurements of sound level, temperature, humi­dity, and reverberation time were conducted during visits at the workplaces, such as classrooms and offices. Linear regression analysis was used to determine associations between individual and work-related factors and objective voice parameters. Results: Compared with men, women had higher fundamental frequency (201 Hz for teachers and 209 for non-teachers vs. 120 Hz for teachers and 127 for non-teachers and sound pressure level (82 dB vs. 80 dB, and shorter maximum phonation time (around 14 seconds vs. around 16 seconds. Female teachers younger than 50 years of age evidenced a significant tendency to speak with lower fundamental frequen­cy and shorter mpt compared with female teachers older than 50 years of age. Female teachers had significantly higher fundamental frequency (66 Hz, higher sound pressure level (2 dB and short phonation time (2 seconds than male teachers. Conclusion: Female teachers younger than 50 years of age had significantly lower F0 and shorter mpt compared with those older than 50 years of age. The multivariate analysis showed that gender was a much more important determinant of variations in F0, spl and mpt than age and teaching occupation. Objectively measured temperature also contributed to the changes on spl among school workers.

  11. Using the Voice to Design Ceramics

    DEFF Research Database (Denmark)

    Hansen, Flemming Tvede; Jensen, Kristoffer

    2011-01-01

    Digital technology makes new possibilities in ceramic craft. This project is about how experiential knowledge that the craftsmen gains in a direct physical and tactile interaction with a responding material can be transformed and utilized in the use of digital technologies. The project presents...... to make ceramic results. The system demonstrates the close connection between digital technology and craft practice....... SoundShaping, a system to create ceramics from the human voice. Based on a generic audio feature extraction system, and the principal component analysis to ensure that the pertinent information in the voice is used, a 3D shape is created using simple geometric rules. This shape is output to a 3D printer...

  12. The Influence of Sleep Disorders on Voice Quality.

    Science.gov (United States)

    Rocha, Bruna Rainho; Behlau, Mara

    2017-09-19

    To verify the influence of sleep quality on the voice. Descriptive and analytical cross-sectional study. Data were collected by an online or printed survey divided in three parts: (1) demographic data and vocal health aspects; (2) self-assessment of sleep and vocal quality, and the influence that sleep has on voice; and (3) sleep and voice self-assessment inventories-the Epworth Sleepiness Scale (ESS), the Pittsburgh Sleep Quality Index (PSQI), and the Voice Handicap Index reduced version (VHI-10). A total of 862 people were included (493 women, 369 men), with a mean age of 32 years old (maximum age of 79 and minimum age of 18 years old). The perception of the influence that sleep has on voice showed a difference (P influence a voice handicap are vocal self-assessment, ESS total score, and self-assessment of the influence that sleep has on voice. The absence of daytime sleepiness is a protective factor (odds ratio [OR] > 1) against perceived voice handicap; the presence of daytime sleepiness is a damaging factor (OR influences voice. Perceived poor sleep quality is related to perceived poor vocal quality. Individuals with a voice handicap observe a greater influence of sleep on voice than those without. Copyright © 2017 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

  13. RNS Applications in Digital Signal Processing

    DEFF Research Database (Denmark)

    Cardarilli, Gian Carlo; Nannarelli, Alberto; Re, Marco

    2017-01-01

    In the past decades, the Residue Number System (RNS) has been adopted in DSP as an alternative to the traditional two’s complement number system (TCS) because of the high speed of the obtained architectures and the savings in area and power dissipation. However, with the shrinking of device featu......-offs, and we identify some trends for implementing DSP on ASIC and FPGA platforms....

  14. Research on moving object detection based on frog's eyes

    Science.gov (United States)

    Fu, Hongwei; Li, Dongguang; Zhang, Xinyuan

    2008-12-01

    On the basis of object's information processing mechanism with frog's eyes, this paper discussed a bionic detection technology which suitable for object's information processing based on frog's vision. First, the bionics detection theory by imitating frog vision is established, it is an parallel processing mechanism which including pick-up and pretreatment of object's information, parallel separating of digital image, parallel processing, and information synthesis. The computer vision detection system is described to detect moving objects which has special color, special shape, the experiment indicates that it can scheme out the detecting result in the certain interfered background can be detected. A moving objects detection electro-model by imitating biologic vision based on frog's eyes is established, the video simulative signal is digital firstly in this system, then the digital signal is parallel separated by FPGA. IN the parallel processing, the video information can be caught, processed and displayed in the same time, the information fusion is taken by DSP HPI ports, in order to transmit the data which processed by DSP. This system can watch the bigger visual field and get higher image resolution than ordinary monitor systems. In summary, simulative experiments for edge detection of moving object with canny algorithm based on this system indicate that this system can detect the edge of moving objects in real time, the feasibility of bionic model was fully demonstrated in the engineering system, and it laid a solid foundation for the future study of detection technology by imitating biologic vision.

  15. Twenty Years of Research on RNS for DSP: Lessons Learned and Future Perspectives

    DEFF Research Database (Denmark)

    Albicocco, Pietro; Cardarilli, Gian Carlo; Nannarelli, Alberto

    2014-01-01

    In this paper, we discuss a number of issues emerged from our twenty-year long experience in applying the Residue Number System (RNS) to DSP systems. In early days, RNS was mainly used to reach the maximum performance in speed. Today, RNS is also used to obtain powerefficient (tradeoffs speed......-power) and reliable systems (redundant RNS). Advances in microelectronics and CAD tools play an important role in favoring one technology over another, and a winning choice of the past may become at disadvantage today. In this paper, we address a number of factors influencing the choice of RNS as the winning solution...

  16. Central voice production and pathophysiology of spasmodic dysphonia.

    Science.gov (United States)

    Mor, Niv; Simonyan, Kristina; Blitzer, Andrew

    2018-01-01

    Our ability to speak is complex, and the role of the central nervous system in controlling speech production is often overlooked in the field of otolaryngology. In this brief review, we present an integrated overview of speech production with a focus on the role of central nervous system. The role of central control of voice production is then further discussed in relation to the potential pathophysiology of spasmodic dysphonia (SD). Peer-review articles on central laryngeal control and SD were identified from PUBMED search. Selected articles were augmented with designated relevant publications. Publications that discussed central and peripheral nervous system control of voice production and the central pathophysiology of laryngeal dystonia were chosen. Our ability to speak is regulated by specialized complex mechanisms coordinated by high-level cortical signaling, brainstem reflexes, peripheral nerves, muscles, and mucosal actions. Recent studies suggest that SD results from a primary central disturbance associated with dysfunction at our highest levels of central voice control. The efficacy of botulinum toxin in treating SD may not be limited solely to its local effect on laryngeal muscles and also may modulate the disorder at the level of the central nervous system. Future therapeutic options that target the central nervous system may help modulate the underlying disorder in SD and allow clinicians to better understand the principal pathophysiology. NA.Laryngoscope, 128:177-183, 2018. © 2017 The American Laryngological, Rhinological and Otological Society, Inc.

  17. Domestic dogs and puppies can use human voice direction referentially.

    Science.gov (United States)

    Rossano, Federico; Nitzschner, Marie; Tomasello, Michael

    2014-06-22

    Domestic dogs are particularly skilled at using human visual signals to locate hidden food. This is, to our knowledge, the first series of studies that investigates the ability of dogs to use only auditory communicative acts to locate hidden food. In a first study, from behind a barrier, a human expressed excitement towards a baited box on either the right or left side, while sitting closer to the unbaited box. Dogs were successful in following the human's voice direction and locating the food. In the two following control studies, we excluded the possibility that dogs could locate the box containing food just by relying on smell, and we showed that they would interpret a human's voice direction in a referential manner only when they could locate a possible referent (i.e. one of the boxes) in the environment. Finally, in a fourth study, we tested 8-14-week-old puppies in the main experimental test and found that those with a reasonable amount of human experience performed overall even better than the adult dogs. These results suggest that domestic dogs' skills in comprehending human communication are not based on visual cues alone, but are instead multi-modal and highly flexible. Moreover, the similarity between young and adult dogs' performances has important implications for the domestication hypothesis.

  18. Your Voice Counts: Listening to the Voice of High School Students with Autism Spectrum Disorder

    Science.gov (United States)

    Saggers, Beth; Hwang, Yoon-Suk; Mercer, K. Louise

    2011-01-01

    Supporting students with autism spectrum disorders (ASDs) in inclusive settings presents both opportunities and significant challenges to school communities. This study, which explored the lived experience of nine students with ASD in an inclusive high school in Australia, is based on the belief that by listening to the voices of students, school…

  19. Facing Sound - Voicing Art

    DEFF Research Database (Denmark)

    Lønstrup, Ansa

    2013-01-01

    This article is based on examples of contemporary audiovisual art, with a special focus on the Tony Oursler exhibition Face to Face at Aarhus Art Museum ARoS in Denmark in March-July 2012. My investigation involves a combination of qualitative interviews with visitors, observations of the audience´s...... interactions with the exhibition and the artwork in the museum space and short analyses of individual works of art based on reception aesthetics and phenomenology and inspired by newer writings on sound, voice and listening....

  20. [Design of standard voice sample text for subjective auditory perceptual evaluation of voice disorders].

    Science.gov (United States)

    Li, Jin-rang; Sun, Yan-yan; Xu, Wen

    2010-09-01

    To design a speech voice sample text with all phonemes in Mandarin for subjective auditory perceptual evaluation of voice disorders. The principles for design of a speech voice sample text are: The short text should include the 21 initials and 39 finals, this may cover all the phonemes in Mandarin. Also, the short text should have some meanings. A short text was made out. It had 155 Chinese words, and included 21 initials and 38 finals (the final, ê, was not included because it was rarely used in Mandarin). Also, the text covered 17 light tones and one "Erhua". The constituent ratios of the initials and finals presented in this short text were statistically similar as those in Mandarin according to the method of similarity of the sample and population (r = 0.742, P text were statistically not similar as those in Mandarin (r = 0.731, P > 0.05). A speech voice sample text with all phonemes in Mandarin was made out. The constituent ratios of the initials and finals presented in this short text are similar as those in Mandarin. Its value for subjective auditory perceptual evaluation of voice disorders need further study.