• Journal of Korea Multimedia Society
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• v.13 no.9
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• pp.1322-1328
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• 2010

Recently, keyword spotting system is greatly in the limelight as UI(User Interface) technology of ubiquitous home network system. Keyword spotting system is vulnerable to non-stationary noises such as TV, radio, dialogue. Especially, speech recognition rate goes down drastically under the embedded DSP(Digital Signal Processor) environments because it is relatively low in the computational capability to process input speech in real-time. In this paper, we propose a new keyword spotting system using the call-command method, which is consisted of small number of recognition networks. We select the call-command such as 'narae', 'home manager' and compose the small network as a token which is consisted of silence with the noise and call commands to carry the real-time recognition continuously for input speeches.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.2
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• pp.120-126
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• 2009

In order to reduce the noise quantity in a conversation under the noisy environment it is necessary for the signal processing system to process adaptively according to the noise quantity in order to enhance the performance. Therefore this paper presents a recognition method for noise quantity by linear predictive coefficient using a three layered neural network, which is trained using three kinds of speech that is degraded by various background noises. The performance of the proposed method for the noise quantity was evaluated based on the recognition rates for various noises. In the experiment, the average values of the recognition results were 98.4% or more for such noise using Aurora2 database.

• Proceedings of the IEEK Conference
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• summer
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• pp.237-243
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• 2004

A multichannel smart sound sensor capable to detect and identify sound events in noisy conditions is presented in this paper. Sound information extraction is a complex task and the main difficulty consists is the extraction of high­level information from an one-dimensional signal. The input of smart sound sensor is composed of data collected by 5 microphones and its output data is sent through a network. For a real time working purpose, the sound analysis is divided in three steps: sound event detection for each sound channel, fusion between simultaneously events and sound identification. The event detection module find impulsive signals in the noise and extracts them from the signal flow. Our smart sensor must be capable to identify impulsive signals but also speech presence too, in a noisy environment. The classification module is launched in a parallel task on the channel chosen by data fusion process. It looks to identify the event sound between seven predefined sound classes and uses a Gaussian Mixture Model (GMM) method. Mel Frequency Cepstral Coefficients are used in combination with new ones like zero crossing rate, centroid and roll-off point. This smart sound sensor is a part of a medical telemonitoring project with the aim of detecting serious accidents.

• Proceedings of the Korea Society of Information Technology Applications Conference
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• 2005.11a
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• pp.45-48
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• 2005

In this paper, we propose a new fast running FIR filter structure that improves the convergence speed of adaptive signal processing and reduces the computational complexity. The proposed filter is applied to wavelet based adaptive algorithm. Actually we compared the performance of the proposed algorithm with other algorithm using computer simulation of adaptive noise canceler based on synthesis speech. As the result, We know the proposed algorithm is prefer than the existent algorithm.

• Proceedings of the KIEE Conference
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• 2006.10c
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• pp.419-421
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• 2006

The importance of automatic discrimination image data has evolved as a research topic over recent years. We have used forward neural network as a classifier using sound data features within image data, our initial tests have shown encouraging results that indicate the viability of our approach.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.661-666
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• 1999

웨이브렛 이론은 응용수학에서 처음 소개된 후 다중해상도 표면 및 이산신호의 부대역 분해방법 등에 대한 단일화된 이론을 제공하고 있으며 최근 신호처리 전반에 걸쳐 널리 이용되고 있는 이론이다. 본 논문에서는 최근 들어 신호저리분야의 새로운 기법으로 제시된 웨이브렛 이론에 대한 소개와 더불어 이를 이용하여 음성개선, 유성음/무성음/묵음 판별, 끝점검출, 피치 및 성문 폐쇄시점 검출 등의 음성신호처리에 적용한 예들을 소개한다.

• Proceedings of the IEEK Conference
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• 2000.06e
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• pp.155-158
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• 2000

In this paper, For operator's conveniency of the mobile robot, achieved the system which control the robot by adopting the speaker independently isolated word recognition and by implementing the real time with TMS320C31. and This paper using the Tri-ultrasonics range finder to detect obstacles and implements the mobile robot. In this paper, DSP processor (TMS320C31) is used signal processing for speech recognition in the real time and Micro processor(80C196KC) is controling the ultrasonics range finders.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.17-23
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• 1995

최근 우리나라에서는 음성신호처리 기술을 바탕으로한 여러 가지 시스템이 상용화되고, 또 그에 따라 관련분야의 연구도 더욱 활발해지고 있다. 본 고에서는 최근 몇 년간 발표되었던 연구결과들을 바탕으로 현재 국내에서 dam성신호처리 관련분야에서의 연구현황을 소개하고 향후의 연구방향 및 미래의 연구 경향을 예측해보고자 g나다. 이를 위해서, 음성신호처리 분야를 음성분석, 음성 합성, 음성 인식, 음성 부호화의 네 세부 분야로 나누고 각 분야별로 국내 현황 그리고 앞으로의 전망을 제시한다.

• Proceedings of the Korean Information Science Society Conference
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• 2002.10d
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• pp.502-504
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• 2002

군중잡음(crowd noise)이 발생하는 환경에서 음성 통화 및 화자 인식을 할 때에는 음성에 파열음이나 마찰음과 같은 유색잡음(colored noise)이 부가되어 원래 음성이 왜곡된다. 이와 같이 왜곡된 음성 신호를 처리할 때에는 군중잡음을 제거하는 과정이 반드시 필요하다. 본 논문에서는 군중잡음의 특성을 분석하고, 그 결과를 이용하여 음성 신호처리 시에 효과적으로 군중잡음만을 제거할 수 있는 모델을 제안한다. 제안된 모델은 시간 영역에서는 침묵 구간을 검출하여 마찰음과 파열음을 제거하는 과정과 주파수 영역에서는 잡음 평균을 생성하고 이를 이용한 스펙트럼 차감법(spectral subtraction)으로 군중 잡음을 제거하는 과정으로 이루어진다.

• The Journal of Korea Robotics Society
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• v.3 no.3
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• pp.237-244
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• 2008

Nowadays, research on human-robot interaction has been getting increasing attention. In the research field of human-robot interaction, speech signal processing in particular is the source of much interest. In this paper, we report a speaker localization system with six microphones for a humanoid robot called MAHRU from KIST and propose a time delay of arrival (TDOA)-based feature matrix with its algorithm based on the minimum sum of absolute errors (MSAE) for sound source localization. The TDOA-based feature matrix is defined as a simple database matrix calculated from pairs of microphones installed on a humanoid robot. The proposed method, using the TDOA-based feature matrix and its algorithm based on MSAE, effortlessly localizes a sound source without any requirement for calculating approximate nonlinear equations. To verify the solid performance of our speaker localization system for a humanoid robot, we present various experimental results for the speech sources at all directions within 5 m distance and the height divided into three parts.