• Journal of the Korean Institute of Intelligent Systems
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• v.13 no.1
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• pp.45-50
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• 2003

Recently, robots technique has been developed remarkably. Emotion recognition is necessary to make an intimate robot. This paper shows the simulator and simulation result which recognize or classify emotions by learning pitch pattern. Also, because the pitch is not sufficient for recognizing emotion, we added acoustic elements. For that reason, we analyze the relation between emotion and acoustic elements. The simulator is composed of the DRNN(Dynamic Recurrent Neural Network), Feature extraction. DRNN is a learning algorithm for pitch pattern.

• Proceedings of the KIEE Conference
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• 2003.11c
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• pp.511-514
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• 2003

The human being uses speech signals to exchange information. When background noise is present, speech recognizers experience performance degradations. Speech recognition through speech enhancement in the noisy environment was studied. Histogram method as a reliable noise estimation approach for spectral subtraction was introduced using MFCC method. The experiment results show the effectiveness of the proposed algorithm.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.8 no.3
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• pp.64-70
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• 2009

This paper presents a speech enhancement technique based on generalized Gamma distribution in order to obtain robust speech recognition performance. For robust speech enhancement, the noise estimation based on a spectral noise floor controled recursive averaging spectral values is applied to speech estimation under the generalized Gamma distribution and spectral gain floor. The proposed speech enhancement technique is based on spectral component, spectral amplitude, and log spectral amplitude. The performance of three different methods is measured by recognition accuracy of automatic speech recognition (ASR).

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.8 no.4
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• pp.28-32
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• 2009

This paper compares two speech estimators under a generalized Gamma distribution for DFT-based single-microphone speech enhancement methods. For the speech enhancement, the noise estimation based on recursive averaging spectral values by spectral minimum noise is applied to two speech estimators based on the generalized Gamma distribution using $\kappa$=1 or $\kappa$=2. The performance of two speech enhancement algorithms is measured by recognition accuracy of automatic speech recognition(ASR) in car noisy environment.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2008.02a
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• pp.125-128
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• 2008

IPTV의 상용화에 맞추어 사용자와 TV간의 정보 교환에 의한 대화형 서비스들이 제공되고 있으며, 특히 음성인식 기술은 이러한 서비스를 실현하기 위한 중요한 기술 중의 하나로 대두되고 있다. TV에서의 음성인식 수행을 위해서는 가정환경과 같은 제한된 공간에서 효율적으로 사용자의 음성을 TV에 전송할 수 있는 근거리 무선통신 수단이 필요하게 된다. 특히, 리모트 컨트롤러와 같은 저전력 시스템 환경에서 구현이 가능해야 한다. 따라서 이러한 제한된 조건에서 최적의 성능을 갖는 음성 전송 시스템 개발이 요구되고 있다. 본 논문에서는 블루투스 환경 하에서 음성인식을 위해 필요한 음성전송 시스템을 실시간 구현한다. 효율적인 음성전송을 위해 G.711을 기본 코덱으로 사용하며, 음성전송 시 발생하는 패킷손실에 따른 음성 품질 저하를 줄이기 위해 G.711 패킷손실 은닉 알고리즘을 음성전송 시스템에 적용한다. 특히 G.711 패킷 손실 은닉 알고리즘 수행을 위해 블루투스 프로토콜 스택application layer에 RTP 프로토콜을 적용하여 패킷 손실 여부를 확인하고, 패킷 손실 발생 시 패킷손실 은닉 알고리즘을 통해 음성의 품질 저하를 줄인다. 구현된 시스템의 성능을 평가한 결과, G.711 패킷 손실 알고리즘을 적용하여 2~10%의 패킷손실 환경에서 14.7%의 음질개선을 얻을 수 있었다.

• Journal of the Korean Institute of Intelligent Systems
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• v.20 no.2
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• pp.208-213
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• 2010

Generally, the mal factor for speech recognition is the background noise in speech recognition. The noise is the reason to reduce the speech recognition performance. Owing to the fact, the place to recognize is very important. To improve the recognition performance from the sound having noise, we implemented the noise filtered Wiener filter at the signal process step which adopted the FIR filter. In FIR filter, it deal with the filtered speech signal which is appropriate frequency range of human speech frequency range. Therefore, we make the recognition system distinguish between noise and speech sound from the incoming speech signal.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.7 no.5
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• pp.90-96
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• 2008

This paper presents an algorithm of a single chip acoustic speech enhancement for telematics device. The algorithm consists of two stages, i.e. noise reduction and echo cancellation. An adaptive filter based on cross spectral estimation is used to cancel echo. The external background noise is eliminated and the clear speech is estimated by using MMSE log-spectral magnitude estimation. To be suitable for use in consumer electronics, we also design a low cost, high speed and flexible hardware architecture. The performance of the proposed speech enhancement algorithms were measured both by the signal-to-noise ratio(SNR) and recognition accuracy of an automatic speech recognition(ASR) and yields better results compared with the conventional methods.

• Journal of the Korea Society of Computer and Information
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• v.15 no.11
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• pp.91-99
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• 2010

Recently, recognizing user resource for personalized service has been needed in digital content service fields. Especially, to analyze user taste, recommend music and service music related information need recognition of user music file in case of online music service. Music related information service is offered through recognizing user music based on tag information. Recognition error has grown by weak points like changing and removing of tag information. Techniques of content based user music recognition with music signal itself are researched for solving upper problems. In this paper, we propose user music recognition on the internet by extracted feature from music signal. Features are extracted after suitable preprocessing for structure of content based user music recognition. Recognizing on music server consist of feature form are progressed with extracted feature. Through this, user music can be recognized independently of tag data. 600 music was collected and converted to each 5 music qualities for proving of proposed recognition. Converted 3000 experiment music on this method is used for recognition experiment on music server including 300,000 music. Average of recognition ratio was 85%. Weak points of tag based music recognition were overcome through proposed content based music recognition. Recognition performance of proposed method show a possibility that can be adapt to online music service in practice.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2021.05a
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• pp.584-588
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• 2021

The impedance of the speaker is likely to be recognized as a fixed value. However, speaker impedance continues to vary with frequency variation, especially larger in resonant frequency region. The sound pressure level of loudspeakers is determined by the current flowing throughout the coil that consists loudspeakers. If loudspeakers are driven by voltage, sound pressure level of the loudspeaker is distorted by the variation of loudspeaker impedance. Current-drive of loudspeakers can solve this problem, but distortion of sound pressure level occurs in low frequencies due to resonance. The distortion can degrade the sound quality of the sound system. So to solve this problem, In this paper, we propose a resonance compensation circuit using DSP. we simulates audio systems using an equivalent model of loudspeakers to verify distortion of sound pressure level due to impedance variation and propose a circuit to compensate it. The proposed circuit is configured using a state variable filter and it can adjust the center frequency and output, so it will be used various sound systems.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.113-120
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• 1998

정보화 시대가 발전함에 따라 음성 통신 및 저장 시스템은 점점 더 우리 생활 깊숙이 자리잡아 가고 있다 따라서 급증하는 수요에 보다 더 효과적으로 대처하기 위한 연구가 진행되어 왔다. 그 한가지 예가 기존의 음성 부호화 시스템의 음질을 유지하면서 압축율을 크게 높일 수 있는 부호화 방법에 대한 연구 및 표준화 작업이다. 본 논문에서는 최근 확정된 음성 부호화기 표준안인 US DoD 2.4 kbps MELP, MPEG-4 HVXC, CDMA 용 IS-127 EVRC 음성 부호화기에 대해 비교적 자세히 설명하고, 현재 진행중인 ITU-T 4kbps 표준안으로 제안된 부호화 방법들이 경향을 살펴본다. 또한 새로운 연구 분야인 인터넷 전화기와 인식-합성 기법을 이용한 아주 낮은 전송율 음성 부호화기에 대한 연구 동향을 소개한다.