• Title/Summary/Keyword: multi-level signal

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A Novel Multi-Channel Hearing Aid Algorithm with SMR(signal-to-masking ratio) Improvement (신호 대 마스킹 비 개선을 통한 다채널 보청 알고리즘)

  • 김헌중;홍민철;차형태
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.8
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    • pp.12-21
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    • 2000
  • In this paper, we propose a novel hearing aid algorithm for sensorinural hearing loss restoration with multi-channel(band) dynamic range compression and psychoacoustics. In this way, we can present a normal perception condition to the impaired listener. The proposed algorithm make loudness scaling function achieve proper loudness level, and analysis masking property for the signal will be perceived to impaired listener, and then, restore normal spectral contrast using SMR(signal-to-masking ratio) defined by distance between the level of each frequency and masking threshold.

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Effects of the Types of Noise and Signal-to-Noise Ratios on Speech Intelligibility in Dysarthria (소음 유형과 신호대잡음비가 마비말장애인의 말명료도에 미치는 영향)

  • Lee, Young-Mee;Sim, Hyun-Sub;Sung, Jee-Eun
    • Phonetics and Speech Sciences
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    • v.3 no.4
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    • pp.117-124
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    • 2011
  • This study investigated the effects of the types of noise and signal to noise ratios (SNRs) on speech intelligibility of an adult with dysartrhia. Speech intelligibility was judged by 48 naive listeners using a word transcription task. Repeated measures design was used with the types of noise (multi-talker babble/environmental noise) and SNRs (0, +10 dB, +20 dB) as within-subject factors. The dependent measure was the percentage of correctly transcribed words. Results revealed that two main effects were statistically significant. Listeners performed significantly worse in the multi-talker babble condition than the environmental noise condition, and they performed significantly better at higher levels of SNRs. The current results suggested that the multi-talker babble and lower level of SNRs decreased the speech intelligibility of adults with dysarthria, and speech-language pathologists should consider environmental factors such as the types of noise and SNRs in evaluating speech intelligibility of adults with dysarthria.

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ECG Signal Compression based on Adaptive Multi-level Code (적응적 멀티 레벨 코드 기반의 심전도 신호 압축)

  • Kim, Jungjoon
    • Journal of the Korean Institute of Intelligent Systems
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    • v.23 no.6
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    • pp.519-526
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    • 2013
  • ECG signal has the feature that is repeated in a cycle of P, Q, R, S, and T waves and is sampled at a high sampling frequency in general. By using the feature of periodic ECG signals, maximizing compression efficiency while minimizing the loss of important information for diagnosis is required. However, the periodic characteristics of such amplitude and period is not constant by measuring time and patients. Even though measured at the same time, the patient's characteristics display different periodic intervals. In this paper, an adaptive multi-level coding is provided by coding adaptively the dominant and non-dominant signal interval of the ECG signal. The proposed method can maximize the compression efficiency by using a multi-level code that applies different compression ratios considering information loss associated with the dominant signal intervals and non-dominant signal intervals. For the case of long time measurement, this method has a merit of maximizing compression ratio compared with existing compression methods that do not use the periodicity of the ECG signal and for the lossless compression coding of non-dominant signal intervals, the method has an advantage that can be stored without loss of information. The effectiveness of the ECG signal compression is proved throughout the experiment on ECG signal of MIT-BIH arrhythmia database.

Implementation of an FPGA-based Multi-Carrier PWM Techniques for Multilevel Inverter (FPGA기반 멀티레벨 인버터의 다중 반송신호 PWM 기법 구현)

  • Chun, Tae-Won;Lee, Hong-Hee;Kim, Heung-Geun;Nho, Eui-Cheol
    • The Transactions of the Korean Institute of Power Electronics
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    • v.15 no.4
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    • pp.288-295
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    • 2010
  • Multi-level inverters have drawn much of attention in recent years because it can meet the demand of high power applications and good power quality associated with reduced harmonic distortion. As the number of voltage level increases, field programmable gate arrays (FPGAs) are suitable for the implementation of multi-level modulation algorithm. This paper proposes the implementation method for generating PWM pulses at the three phase diode clamped five-level inverter using FPGA. The strategy for communicating stably the data of three-phase reference voltages between the DSP and FPGA is suggested. The techniques for generating PWM signals based on a multi-carrier modulation method are carried out through the experiments with 32-bit DSP and Cyclone-III FPGA.

THD Reduction by Using HBML-PWM Hybrid Type Inverter (HBML-PWM 혼용 인버터를 이용한 THD 저감)

  • Yi, Tao;Moon, C.J.;Song, S.G.;Park, S.J.;Cho, S.E.;Kim, J.D.
    • Proceedings of the KIPE Conference
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    • 2005.07a
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    • pp.655-658
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    • 2005
  • In this paper, we proposed the electric circuit using one common arm of H-Bridge Inverters to reduce the number of switching component in multi-level inverter combined with H-Bridge Inverters and Transformers. and furthermore we suggested the new multi-level PWU inverter using PWM level to reduce THD(Total Harmonic Distortion). and we used the switching method that can be same rate of usage at each transformer. Also, we tested the proposed prototype 9-level inverter to clarify the proposed electric circuit and reasonableness of control signal for the proposed multi-level PWM inverter.

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Sound Quality Enhancement in MPEG Surround by Using ILD Distortion (ILD DISTORTION을 이용한 MPEG SURROUND의 음질 개선)

  • Chon, Sang-Bae;Choi, In-Yong;Sung, Koeng-Mo
    • Proceedings of the IEEK Conference
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    • 2006.06a
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    • pp.241-242
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    • 2006
  • MPEG Surround is an audio coding technology that represents multi-channel audio signal with downmixed audio signal(s) and very low bitrate side information based on Binaural Cue Coding. The side information consists of Inter-Channel Level Difference, Inter-Channel Correlation, and payloads. These two parameters are correspondent to the well-known spatial parameters in psycho-acoustics, Inter-aural Level Difference (ILD) and Inter-Aural Cross Correlation (IACC). Though ICLD is to provide perceptually equivalent ILD to the listener, however, the ILD of the original multi-channel audio signal and that of the MPEG Surround encoded signal was different. The difference between two ILD values is defined as ILD Distortion (ILDD). This paper provides how ILDD can be applied to enhance sound quality in MPEG Surround and how much ILDD is decreased.

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A Multicoded-PPM Scheme for High Data Rate UWB Communication Systems

  • lung, Sung-Yoon;Park, Dong-Jo
    • Journal of Communications and Networks
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    • v.11 no.3
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    • pp.271-278
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    • 2009
  • A new modulation scheme called multicoded-pulse position modulation (MC-PPM) is proposed for an ultrawideband (UWB) impulse radio communication system. The multicoded signal is generated by using several orthogonal codes for transmitting data simultaneously. Then, each multi-level value of the multicoded signal is converted to pulse position which results in not only an improved data rate, but also a processing gain in reception, delivering the power-efficient benefit of PPM and guaranteeing the low pulse energy for UWB systems. We notice that the modulation of multi-level values of the multicoded signal to pulse position is more efficient in terms of achievable data rate than the modulation of transmitting data based on other PPM schemes within given bandwidth and pulse energy. Therefore, as a performance measure, we focus on the achievable data rate (link capacity) of the proposed scheme and analyze it theoretically. Through simulation, we compare the link capacity of the MC-PPM scheme and other PPM schemes, such as M -ary PPM and multiple PPM. With the fixed bandwidth and same pulse energy condition, the UWB system based on the proposed MC-PPM scheme shows good link capacity and an increased data rate as L increases, which is contrary to other PPM schemes.

Analysis of Signal-to-Noise Ratio in High Field Multi-dimensional Magnetic Resonance Imaging (고자장 다차원 자기공명영상에서 신호대잡음비 분석)

  • Ahn, C.B.;Kim, H.J.;Chang, K.S.
    • Proceedings of the KIEE Conference
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    • 2003.07d
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    • pp.2783-2785
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    • 2003
  • In multi-dimensional magnetic resonance imaging, data is obtained in the spatial frequency domain. Since the signal variation in the spatial frequency domain is much larger than that in the spatial domain, analog-to-digital converts with wide conversion bits are required. In this paper, the quantization noise in magnetic resonance imaging is analyzed. The signal-to-quantization noise ratio(SQNR) in the reconstructed image is derived from the level of quantization in the data acquisition. Since the quantization noise is proportional to the signal amplitude, it becomes more dominant in high field imaging. Using the derived formula the SQNR for several MRI systems are evaluated, and it is shown that the quantization noise can be a limiting factor in high field imaging, especially in three dimensional imaging in magnetic resonance imaging.

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A Modified Decision-Directed LMS Algorithm (수정된 DD LMS 알고리즘)

  • Oh, Kil Nam
    • Journal of the Institute of Electronics and Information Engineers
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    • v.53 no.7
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    • pp.3-8
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    • 2016
  • We propose a modified form of the decision-directed least mean square (DD LMS) algorithm that is widely used in the optimization of self-adaptive equalizers, and show the modified version greatly improves the initial convergence properties of the conventional algorithm. Existing DD LMS regards the difference between a equalizer output and a quantization value for it as an error, and achieves an optimization of the equalizer based on minimizing the mean squared error cost function for the equalizer coefficients. This error generating method is useful for binary signal or a single-level signals, however, in the case of multi-level signals, it is not effective in the initialization of the equalizer. The modified DD LMS solves this problem by modifying the error generation. We verified the usefulness and performance of the modified DD LMS through experiments with multi-level signals under distortions due to intersymbol interference and additive noise.

Design and Applications of a Generalized Software-Based GNSS IF Signal Generator

  • Lim, Deok-Won;Park, Chan-Sik;Lee, Sang-Jeong
    • Proceedings of the Korean Institute of Navigation and Port Research Conference
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    • v.2
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    • pp.211-215
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    • 2006
  • In this paper, design and applications of a generalized, versatile and customizable IF signal generator that can model the modernized GPS and Galileo signal is given. It generates IF sampled data that can be directly used by a software receiver. Entire constellation of satellites which is independent of satellite-user geometry is easily determined using a real or simulated ephemeris data. Since the IF center frequency, sampling frequency and quantization bit number are user location dependent parameters, their effects are also considered in IF signal generator. The generalized IF signal generator will be very well suited for the development phase of a software receiver due to its versatility. The full access to the sampling frequency, front-end filter definition and ADC parameters also offers a great opportunity for cost-effective analysis of tracking loops and error mitigation techniques at the receiver level. Interference sources can be easily added to the generator to simulate specific environments. This software IF signal generator can also be used to feed a multi-frequency multi-system software receiver for the prototyping of a combined GPS/Galileo receiver. The test result using the generated signals and a real software receiver shows the effectiveness of the implemented IF signal generator.

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