• 제목/요약/키워드: training signal

검색결과 496건 처리시간 0.03초

훈련음성 데이터에 적응시킨 필터뱅크 기반의 MFCC 특징파라미터를 이용한 전화음성 연속숫자음의 인식성능 향상에 관한 연구 (A study on the recognition performance of connected digit telephone speech for MFCC feature parameters obtained from the filter bank adapted to training speech database)

  • 정성윤;김민성;손종목;배건성;강점자
    • 대한음성학회:학술대회논문집
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    • 대한음성학회 2003년도 5월 학술대회지
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    • pp.119-122
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    • 2003
  • In general, triangular shape filters are used in the filter bank when we get the MFCCs from the spectrum of speech signal. In [1], a new feature extraction approach is proposed, which uses specific filter shapes in the filter bank that are obtained from the spectrum of training speech data. In this approach, principal component analysis technique is applied to the spectrum of the training data to get the filter coefficients. In this paper, we carry out speech recognition experiments, using the new approach given in [1], for a large amount of telephone speech data, that is, the telephone speech database of Korean connected digit released by SITEC. Experimental results are discussed with our findings.

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호흡-바이오피드백 앱 개발을 위한 PPG기반의 호흡 추정 알고리즘 (Breathing Information Extraction Algorithm from PPG Signal for the Development of Respiratory Biofeedback App)

  • 최병훈
    • 전기학회논문지
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    • 제67권6호
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    • pp.794-798
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    • 2018
  • There is a growing need for a care system that can continuously monitor, manage and effectively relieve stress for modern people. In recent years, mobile healthcare devices capable of measuring heart rate have become popular, and many stress monitoring techniques using heart rate variability analysis have been actively proposed and commercialized. In addition, respiratory biofeedback methods are used to provide stress relieving services in environments using mobile healthcare devices. In this case, breathing information should be measured well to assess whether the user is doing well in biofeedback training. In this study, we extracted the heart beat interval signal from the PPG and used the oscillator based notch filter based on the IIR band pass filter to track the strongest frequency in the heart beat interval signal. The respiration signal was then estimated by filtering the heart beat interval signal with this frequency as the center frequency. Experimental results showed that the number of breathing could be measured accurately when the subject was guided to take a deep breath. Also, in the timeing measurement of inspiration and expiration, a time delay of about 1 second occurred. It is expected that this will provide a respiratory biofeedback service that can assess whether or not breathing exercise are performed well.

인체 수관절 근육의 진동 응답 (Vibration Response of a Human Carpal Muscle)

  • 전한용;김진오;박광훈
    • 한국소음진동공학회논문집
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    • 제21권1호
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    • pp.31-40
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    • 2011
  • This paper examines the dynamic characteristics of a human carpal muscle through theoretical analysis and experiment. The carpal muscle was modeled as a 1-DOF vibration system and vibration response due to a ramp function force was calculated. The electromyogram signal corresponding to the muscle excitation force was measured, and the excitation force function of an envelope curve from the electromyogram signal was extracted. The ramp input function of electrical stimulation to the carpal muscle was applied by using a device for functional electrical stimulation, and the angular displacements corresponding to steady state response were measured. Theoretical calculations of the vibration response displacements were compared with the experimental results of the angular displacements, and have shown a good agreement with the result that is linearly proportional to the excitation force magnitude. As a result, the relationship between the input current of the electrical stimulation and the excitation force magnitude was inferred. The result was shown that it can be applied to develop rehabilitation training devices.

FES 보행을 위한 보행 이벤트 검출 (Gait-Event Detection for FES Locomotion)

  • 허지운;김철승;엄광문
    • 한국정밀공학회지
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    • 제22권3호
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    • pp.170-178
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    • 2005
  • The purpose of this study is to develop a gait-event detection system, which is necessary for the cycle-to-cycle FES control of locomotion. Proposed gait event detection system consists of a signal measurement part and gait event detection part. The signal measurement was composed of the sensors and the LabVIEW program for the data acquisition and synchronization of the sensor signals. We also used a video camera and a motion capture system to get the reference gait events. Machine learning technique with ANN (artificial neural network) was adopted for automatic detection of gait events. 2 cycles of reference gait events were used as the teacher signals for ANN training and the remnants ($2\sim5$ cycles) were used fur the evaluation of the performance in gait-event detection. 14 combinations of sensor signals were used in the training and evaluation of ANN to examine the relationship between the number of sensors and the gait-event detection performance. The best combinations with minimum errors of event-detection time were 1) goniometer, foot-switch and 2) goniometer, foot-switch, accelerometer x(anterior-posterior) component. It is expected that the result of this study will be useful in the design of cycle-to-cycle FES controller.

음성 및 잡음 인식 알고리즘을 이용한 환경 배경잡음의 제거 (Reduction of Environmental Background Noise using Speech and Noise Recognition)

  • 최재승
    • 한국정보통신학회논문지
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    • 제15권4호
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    • pp.817-822
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    • 2011
  • 본 논문에서는 먼저 신경회로망의 학습에 오차역전파 학습 알고리즘을 사용하여 각 프레임에서의 음성 및 잡음 구간의 검출에 의한 음성인식 알고리즘을 제안한다. 그리고 신경회로망에 의하여 음성 및 잡음 구간의 검출에 따라서 각 프레임에서 잡음을 제거하는 스펙트럼 차감법을 제안한다. 본 실험에서는 제안한 음성인식알고리즘의 성능을 원음성에 백색잡음 및 자동차 잡음을 부가하여 인식율을 평가한다. 또한 인식시스템에 의하여 검출된 음성 및 잡음 구간을 이용하여 각 프레임에서의 스펙트럼 차감법에 의한 잡음제거의 실험결과를 나타낸다. 잡음에 의하여 오염된 음성에 대하여 신호대잡음비를 사용하여 본 알고리즘이 유효하다는 것을 확인한다.

Modeling and assessment of VWNN for signal processing of structural systems

  • Lin, Jeng-Wen;Wu, Tzung-Han
    • Structural Engineering and Mechanics
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    • 제45권1호
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    • pp.53-67
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    • 2013
  • This study aimed to develop a model to accurately predict the acceleration of structural systems during an earthquake. The acceleration and applied force of a structure were measured at current time step and the velocity and displacement were estimated through linear integration. These data were used as input to predict the structural acceleration at next time step. The computation tool used was the Volterra/Wiener neural network (VWNN) which contained the mathematical model to predict the acceleration. For alleviating problems of relatively large-dimensional and nonlinear systems, the VWNN model was utilized as the signal processing tool, including the Taylor series components in the input nodes of the neural network. The number of the intermediate layer nodes in the neural network model, containing the training and simulation stage, was evaluated and optimized. Discussions on the influences of the gradient descent with adaptive learning rate algorithm and the Levenberg-Marquardt algorithm, both for determining the network weights, on prediction errors were provided. During the simulation stage, different earthquake excitations were tested with the optimized settings acquired from the training stage to find out which of the algorithms would result in the smallest error, to determine a proper simulation model.

Automatic melody extraction algorithm using a convolutional neural network

  • Lee, Jongseol;Jang, Dalwon;Yoon, Kyoungro
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • 제11권12호
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    • pp.6038-6053
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    • 2017
  • In this study, we propose an automatic melody extraction algorithm using deep learning. In this algorithm, feature images, generated using the energy of frequency band, are extracted from polyphonic audio files and a deep learning technique, a convolutional neural network (CNN), is applied on the feature images. In the training data, a short frame of polyphonic music is labeled as a musical note and a classifier based on CNN is learned in order to determine a pitch value of a short frame of audio signal. We want to build a novel structure of melody extraction, thus the proposed algorithm has a simple structure and instead of using various signal processing techniques for melody extraction, we use only a CNN to find a melody from a polyphonic audio. Despite of simple structure, the promising results are obtained in the experiments. Compared with state-of-the-art algorithms, the proposed algorithm did not give the best result, but comparable results were obtained and we believe they could be improved with the appropriate training data. In this paper, melody extraction and the proposed algorithm are introduced first, and the proposed algorithm is then further explained in detail. Finally, we present our experiment and the comparison of results follows.

연속 잡음 음성 인식을 위한 다 모델 기반 인식기의 성능 향상에 대한 연구 (Performance Improvement in the Multi-Model Based Speech Recognizer for Continuous Noisy Speech Recognition)

  • 정용주
    • 음성과학
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    • 제15권2호
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    • pp.55-65
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    • 2008
  • Recently, the multi-model based speech recognizer has been used quite successfully for noisy speech recognition. For the selection of the reference HMM (hidden Markov model) which best matches the noise type and SNR (signal to noise ratio) of the input testing speech, the estimation of the SNR value using the VAD (voice activity detection) algorithm and the classification of the noise type based on the GMM (Gaussian mixture model) have been done separately in the multi-model framework. As the SNR estimation process is vulnerable to errors, we propose an efficient method which can classify simultaneously the SNR values and noise types. The KL (Kullback-Leibler) distance between the single Gaussian distributions for the noise signal during the training and testing is utilized for the classification. The recognition experiments have been done on the Aurora 2 database showing the usefulness of the model compensation method in the multi-model based speech recognizer. We could also see that further performance improvement was achievable by combining the probability density function of the MCT (multi-condition training) with that of the reference HMM compensated by the D-JA (data-driven Jacobian adaptation) in the multi-model based speech recognizer.

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재활운동에 참가한 뇌졸중환자의 운동과학적 연구 (The Scientific Research of Rehabilitation Training Program Participants in Stroke Patients)

  • 진영완
    • 생명과학회지
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    • 제20권11호
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    • pp.1704-1710
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    • 2010
  • 본 연구는 뇌졸중 발병 후 재활운동에 참가하는 환자들을 대상으로 재활운동 3개월이 지난 시점에서 1차 실험을 하였고, 6개월이 지난 시점에서 2차 실험을 하여 운동역학적 비교분석을 하였다. 실험에 사용된 장비는 영상분석기, 족저압분석기, 근전도분석기를 사용하였다. 대상자는 7명으로 하였으며 통계방법은 t-test분석 이용하였다. 결과는 엉덩관절의 최대신전 피크 값과 최대신전 피크 모멘트 값에서 통계적으로 유의한 차이를 나타내었다(p<0.05). 족저압의 비교에서는 환측 다리의 족저압에서 일반적인 걷기 동작시에 족저압의 중심이동거리에서 통계적으로 유의한 차이를 나타내었다(p<0.05). 근전도 분석에서는 대퇴사두근 중의 대퇴직근과 외측광근의 근력에서 통계적으로 유의한 차이를 나타내었다(p<0.05).

새로운 갱신조건을 적용한 부호책 생성 알고리즘 (A Codebook Generation Algorithm Using a New Updating Condition)

  • 김형철;조제황
    • 융합신호처리학회논문지
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    • 제5권3호
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    • pp.205-209
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    • 2004
  • 벡터양자화에서 사용되는 부호책 생성 알고리즘들 중에서 가장 널리 사용되는 방법은 K-means 알고리즘이다. 본 논문에서는 부호책의 성능 개선을 위해 새로운 갱신조건을 적용한 부호책 생성 알고리즘을 제안한다. 기존의 K-means 알고리즘은 모든 학습반복 과정 동안 부호벡터 갱신 시 거리의 가중치를 고정하지만, 제안된 방법은 학습반복 과정에서 새로운 부호벡터의 갱신 조건에 따라서 다른 가중치를 적용하여 부호책을 구한다. 따라서, 갱신 조건에 의해 부호벡터에 다른 가중치를 적용할 수 있고, 학습반복 과정마다 가변되는 가중치를 적용하는 효과를 얻을 수 있다. 실험 결과 K-means 알고리즘보다 부호책의 성능이 향상됨을 확인하였다.

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