포먼트 이동과 스펙트럼 기울기의 변환을 이용한 음색 변환

손성용,한민수,Son Song-Young,Hahn Min-Soo 대한음성학회 2003 말소리 Vol.45 No.-

The purpose of voice color conversion is to change the speaker identity perceived from the speech signal. In this paper, we propose a new voice color conversion algorithm through the formant shifting and the spectrum-tilt modification in the frequency domain. The basic idea of this technique is to convert the positions of source formants into those of target speaker's formants through interpolation and decimation and to modify the spectrum-tilt by utilizing the information of both speakers' spectrum envelops. The LPC spectrum is adopted to evaluate the position of formant and the information of spectrum-tilt. Our algorithm enables us to convert the speaker identity rather successfully while maintaining good speech quality, since it modifies speech waveforms directly in the frequency domain.

허밍 운율정보를 이용한 곡목 검색 기술

이지연,한민수,Lee Ji-Yeoun,Hahn Min-Soo 대한음성학회 2002 말소리 Vol.44 No.-

Music query by humming is a challenging problem since the humming signal inevitably contains much variation and inaccuracy. In this paper, we suggest an algorithm for querying a wanted song from music database by humming its melody. In order to suit or adapt the inaccurate peoples humming, a new melody representation technique is proposed. Our algorithm is basically a pitch and duration information-based one and performs fairly well. 85% of correct query rate of the song is achieved for the top 3 matches when tested with 20 songs.

지능형 서비스 로봇을 위한 원거리 음원 추적 기술

이지연,한민수,Lee Ji-Yeoun,Hahn Min-Soo 대한음성학회 2006 말소리 Vol.57 No.-

This paper suggests an algorithm that can estimate the direction of the sound source in real time. The algorithm uses the time difference and sound intensity information among the recorded sound source by four microphones. Also, to deal with noise of robot itself, the Kalman filter is implemented. The proposed method can take shorter execution time than that of an existing algorithm to fit the real-time service robot. Also, using the Kalman filter, signal ratio relative to background noise, SNR, is approximately improved to 8 dB. And the estimation result of azimuth shows relatively small error within the range of ${\pm}7$ degree.

손실이후 프레임 정보에 의한 패킷손실은닉 알고리즘 개선

김재현,한민수,Kim Jae-Hyun,Hahn Min-Soo 대한음성학회 2002 말소리 Vol.43 No.-

In real time packetized voice application, missing packets are major source of voice quality degradation. Thus packet loss concealment (PLC) algorithms are needed to guarantee QoS of VoIP. In this paper, we describe packet loss concealment scheme utilizing the next good frame which follows loss packets. When this scheme is combined with other PLC algorithms, such as G.711 pitch waveform replication recommended by ITU-T LP based PLC algorithm, additional voice quality improvement is obtained for consecutive packet loss larger than 60 msec.

TTS DB 압축을 위한 광대역 파형보간 부호기 구현

양희식,한민수,Yang, Hee-Sik,Hahn, Min-Soo 대한음성학회 2005 말소리 Vol.55 No.-

The adequate compression algorithm is essential to achieve high quality embedded TTS system. in this paper, we Propose waveform interpolation coder for TTS corpus compression after many speech coder investigation. Unlike speech coders in communication system, compression rate and anality are more important factors in TTS DB compression than other performance criteria. Thus we select waveform interpolation algorithm because it provides good speech quality under high compression rate at the cost of complexity. The implemented coder has bit rate 6kbps with quality degradation 0.47. The performance indicates that the waveform interpolation is adequate for TTS DB compression with some further study.

VoIP 환경에서의 잡음제거를 위한 최적화된 위너 필터

정상배,이성독,한민수,Jeong, Sang-Bae,Lee, Sung-Doke,Hahn, Min-Soo 대한음성학회 2007 말소리 Vol.64 No.-

Noise reduction technologies are indispensable to achieve acceptable speech quality in VoIP systems. This paper proposes a Wiener filter optimized to the estimated SNR of noisy speech for the noise reduction in VoIP environments. The proposed noise canceller is applied as a pre-processor before speech encoding. The performance of the proposed method is evaluated by the PESQ in various noisy conditions. In this paper, the proposed algorithm is applied to G.711, G.723.1, and G.729A which are all VoIP speech codecs. The PESQ results show that the performance of our proposed noise reduction scheme outperforms those of the noise suppression in the IS-127 EVRC and the ETSI standard for the advanced distributed speech recognition front-end.

파형보간 코더에서 파라미터간 거리차를 이용한 가변비트율 기법

양희식,정상배,한민수,Yang, Hee-Sik,Jeong, Sang-Bae,Hahn, Min-Soo 대한음성학회 2008 말소리 Vol.65 No.-

In this paper, we propose a new variable bit-rate speech coder based on the waveform interpolation concept. After the coder extracted all parameters, the amounts of the distortions between the current and the predicted parameters which are estimated by extrapolation using past two parameters are measured for all parameters. A parameter would not be transmitted unless the distortion exceeds the preset threshold. At the decoder side, the non-transmitted parameter is reconstructed by extrapolation with past two parameters used to synthesize signals. In this way, we can reduce 26% of the total bit rate while retaining the speech quality degradation below 0.1 PESQ score.

음성 특성을 이용한 G.711 패킷 손실 은닉 알고리즘의 성능개선

한승호,김진술,이현우,류원,한민수,Han Seung-Ho,Kim Jin-Sul,Lee Hyun-Woo,Ryu Won,Hahn Min-Soo 대한음성학회 2006 말소리 Vol.57 No.-

Because a packet loss brings about degradation of speech quality, VoIP speech coders have PLC (Packet Loss Concealment) mechanism. G.711, which is a mandatory VoIP speech coder, also has the PLC algorithm based on pitch period replication. However, it is not robust to burst losses. Thus, we propose two methods to improve the performance of the original PLC algorithm in G.711. One adaptively utilizes voiced/unvoiced information of adjacent good frames regarding to the current lost frame. The other is based on adaptive gain control according to energy variation across the frames. We evaluate the performance of the proposed PLC algorithm by measuring a PESQ value under different random and burst packet loss simulating conditions. It is shown from the experiments that the performance of the proposed PLC algorithm outperforms that of PLC employed in ITU-T Recommendation G.711.

광용적맥파의 정량적 맥파형 분류에 관한 연구

장대근,우말 파르크,박승훈,한민수,Jang, Dae-Jeun,Farooq, Umar,Park, Seung-Hun,Hahn, Min-Soo 대한의용생체공학회 2010 의공학회지 Vol.31 No.4

Over the past few years, a considerable number of methods have been proposed and applied for the classification of photoplethysmography (PPG). Most of the previous studies, however, focused on the qualitative description of the pulse type according to specific disease and thus provided ambiguous criteria to interpreters. In order to screen out this problem, we present a quantitative method for the pulse type classification including the second derivative of photoplethysmography (SDPTG). In the PPG signal, we have classified the signal as 4 types using the position and the presence of the dicrotic wave. In addition, we have categorized the SDPTG signal as 7 types using the position and the presence of "c" and "d" wave and the sign of "c" wave. In order to check the efficacy of the proposed pulse type classification rule, we collected pulse signals from 155 subjects with different ages and sex. From the correlation analysis, Class 1(p<0.01) and Class 2(p<0.01) in the PPG signal are significantly correlated with ages. In a similar manner Class A(p<0.01), Class C(p<0.05), Class D(p<0.01), and Class F(p<0.01) in the SDPTG signal are considerably correlated with the ages. From these observations, and some earlier ones [4], [5], we can conclude that since the newly proposed method has objectivity and clarity in pulse type classification, this method can be used as an alternative of previous classification rules including similar age-related characteristics.

LED를 사용한 무대 표현 기술 확장을 위한 디지털 의상 플랫폼

오승원(Seung-Won Oh),한민수(Min-Soo Hahn) 한국콘텐츠학회 2012 한국콘텐츠학회논문지 Vol.12 No.2

최근 LED는 조명, 광고판, 디스플레이 등 다양한 분야에 널리 활용되고 있다. 이번 연구에서 우리는 공연에서 사용하기 위한 무대 의상에 LED 기술을 융합시키는 방법을 다룬다. 우리는 공연 환경의 요구 조건들을 충족시키기 위해서 무선 네트워크, 임베디드 시스템, 컨트롤 재킷, 디지털 스킨, 패턴저작도구로 구성된 디지털 의상 플랫폼을 제안하며, 이것은 LED 조명 효과를 가진 디지털 의상을 만드는데 활용된다. 임베디드 시스템은 무선 네트워크를 통해 공연 제어 시스템으로부터 제어 신호를 받고 그에 따라 실시간으로 LED 빛의 색 변화를 제어한다. 임베디드 시스템을 장착할 수 있도록 설계된 조끼 형태의 컨트롤 재킷은 임베디드 시스템과 LED를 연결하는 커넥터 역할을 하고, LED가 디자인된 외피 개념의 디지털 스킨은 연출가의 의도에 따라 의상에 디자인된 LED에 의해 그 자체로 발광체가 된다. 패턴저작도구는 타임라인 기반으로 LED 빛의 색이 변화하는 다양한 패턴들을 만드는 것을 도와준다. 디지털 의상은 새로운 형식의 움직이는 조명으로 활용되며 무대 표현 기술의 확장을 가능하게 한다. 우리는 디지털 의상 플랫폼을 적용한 디지털 의상을 제작하여 실제 공연에서 사용함으로써 디지털 의상 플랫폼의 안정적인 사용성과 활용가능성을 검증하였다. Recently, LEDs are used in various fields such as lightings, billboards, displays, and so on. In this paper, we tried to combine a costume with LEDs in a performance environment. To meet the environmental requirements, we propose a digital costume platform which consists of wireless network, an embedded system, a control jacket, a digital skin, and a pattern authoring tool. It is utilized for making a digital costume with LED lighting effects. The embedded system controls LED lighting according to the signals from a show control system through wireless networks. The control jacket can carry the embedded system and connect it with LEDs. The digital skin means a costume with LEDs. The LEDs were designed to fit in performance concept in point of aesthetic. The pattern authoring tool enables users to make timeline-based changing patterns of LED lighting. The proposed platform is a new lighting element in performances. It can expand stage representation techniques and be used as a special effect. It will play a role as a moving light in performances. We evaluated stable operations of our platform by employing it in live performances.