• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.8 no.5
• /
• pp.977-982
• /
• 2004

In this paper, I propose a new method of Multi-Pulse Coding(IP-MPC) use individual pitch pulses in order to accommodate the changes in each pitch interval and reduce pitch errors. The extraction rate of individual pitch pulses was 85% for female voice and 96% for male voice respectively. 1 evaluate the MPC by using pitch information of autocorrelation method and the IP-MPC by using individual pitch pulses. As a result, I knew that synthesis speech of the IP-MPC was better in speech quality than synthesis speech of the MPC.

• The Journal of the Korea Contents Association
• /
• v.6 no.2
• /
• pp.59-64
• /
• 2006

In this paper, 1 propose a new method of Multi-Pulse Coding(IP-MPC) use individual pitch pulses in order to accommodate the changes in each pitch interval and reduce pitch errors. The extraction rate of individual pitch pulses was $85\%$ for female voice and $96\%$ for male voice respectively, 1 evaluate the MPC by using pitch information of autocorrelation method and the IP-MPC by using individual pitch pulses. As a result, 1 knew that synthesis speech of the IP-MPC was better in speech quality than synthesis speech of the MPC.

• Journal of Digital Convergence
• /
• v.11 no.5
• /
• pp.285-290
• /
• 2013

In a MPC coding using excitation source of voiced and unvoiced, it would be a distortion of speech waveform. This is caused by normalization of synthesis speech waveform of voiced in the process of restoration the multi-pulses of representation section. To solve this problem, this paper present a method of position compensation(PC-MPC) in a multi-pulses each pitch interval in order to reduce distortion of speech waveform. I was confirmed that the method can be synthesized close to the original speech waveform. And I evaluate the MPC and PC-MPC using multi-pulses position compensation method. As a result, $SNR_{seg}$ of PC-MPC was improved 0.4dB for female voice and 0.5dB for male voice respectively. Compared to the MPC, $SNR_{seg}$ of PC-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

• Journal of the Korea Academia-Industrial cooperation Society
• /
• v.12 no.9
• /
• pp.4119-4124
• /
• 2011

In a MPC coding using excitation source of voiced and unvoiced, it would be a distortion of speech waveform in case of increasing or decreasing of speech signal amplitude in a frame. This is caused by normalization of synthesis speech signal in the process of restoration the multi-pulses of representation section. To solve this problem, this paper present a method of amplitude compensation(AC-MPC) in a multi-pulses each pitch interval in order to reduce distortion of speech waveform. I was confirmed that the method can be synthesized close to the original speech waveform. And I evaluate the MPC and AC-MPC using amplitude compensation method. As a result, SNRseg of AC-MPC was improved 0.7dB for female voice and 0.7dB for male voice respectively. Compared to the MPC, SNRseg of AC-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

• Journal of Internet Computing and Services
• /
• v.18 no.5
• /
• pp.17-22
• /
• 2017

In a MPC coding using excitation source of voiced and unvoiced, it would be a distortion of speech waveform. This is caused by normalization of synthesis speech waveform of voiced in the process of restoration the multi-pulses of representation section. This paper present PCFBD-MPC( Position Compensation Frequency Band Division-Multi Pulse Coding ) used V/UV/S( Voiced / Unvoiced / Silence ) switching, position compensation in a multi-pulses each pitch interval and Unvoiced approximate-synthesis by using specific frequency in order to reduce distortion of synthesis waveform. Also, I was implemented that the PCFBD-MPC( Position Compensation Frequency Band Division-Multi Pulse Coding ) system and evaluate the SNRseg of PCFBD-MPC in coding condition of 8kbps. As a result, SNRseg of PCFBD-MPC was 13.4dB for female voice and 13.8dB for male voice respectively. In the future, I will study the evaluation of the sound quality of 8kbps speech coding method that simultaneously compensation the amplitude and position of multi-pulse source. These methods are expected to be applied to a method of speech coding using sound source in a low bit rate such as a cellular phone or a smart phone.

• Journal of Korea Multimedia Society
• /
• v.7 no.9
• /
• pp.1233-1239
• /
• 2004

In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech qualify in case coexist with a voiced and an unvoiced consonants in a frame. This paper present a new multi-pulse coding method by using V/S/TSIUVC switching, individual pitch pulses and TSIUVC approximation-synthesis method in order to restrict a distortion of speech quality. The TSIUVC is extracted by using the zero crossing rate and individual pitch pulse. And the TSIUVC extraction rate was 91% for female voice and 96.2% for male voice respectively. The important thing is that the frequency information of 0.347kHz below and 2.813kHz above can be made with high quality synthesis waveform within TSIUVC. I evaluate the MPC use V/UV and the FBD-MPC use V/S/TSIUVC. As a result, I knew that synthesis speech of the FBD-MPC was better in speech quality than synthesis speech of the MPC.

• The Journal of the Acoustical Society of Korea
• /
• v.10 no.1
• /
• pp.20-29
• /
• 1991

A fundamental assumption in conventional linear predictive coding (LPC) analysis procedure is that the input to an all-pole vocal tract filter is white process. In the case of periodic inputs, however, a pitch bias error is introduced into the conventional LP coefficient. Multi-pulse (MP) LP analysis can reduce this bias, provided that an estimate of the excitation is available. Since the prediction error of conventional LP analysis can be modeled as the sum of an MP excitation sequence and a random noise sequence, we can view extracting MP sequences from the prediction error as a classical detection and estimation problem. In this paper, we propose an algorithm in which the locations and amplitudes of the MP sequences are first obtained by applying a likelihood ratio test (LRT) to the prediction error, and LP coefficients free of pitch bias are then obtained from the MP sequences. To verify the performance enhancement, we iterate the above procedure with adaptive threshold at each step.

• Proceedings of the KIPE Conference
• /
• 2011.11a
• /
• pp.185-186
• /
• 2011

본 논문에서는 최근 직류송전용 컨버터로 많은 관심이 집중되고 있는 모듈형 멀티레벨 컨버터(Modular Multi-level Converter)에서 출력파형의 고조파를 저감하는 모듈레이션 방법과 각 모듈의 직류전압 불평형을 해소하는 알고리즘에 대해 기술하고 있다. 먼저 본 논문에서 임의의 개수로 반브리지 모듈이 주어졌을 때 고조파 레벨이 최소화되도록 다펄스 형태로 출력파형을 형성하는 방식을 제안하고 그 타당성을 PSCAD 소프트웨어를 이용한 시뮬레이션으로 검증하였다. 이 방식은 다펄스 출력파형의 각 계단을 형성하는 모듈의 턴온과 턴오프 시점을 보편화된 수식으로 정하는 방식으로 알고리즘 구현이 매우 용이하다. 또한 각 모듈의 직류전압 불평형을 바로잡는 알고리즘을 제안하고 그 타당성도 시뮬레이션으로 검증하였는데, 이 방식은 각 모듈이 교류 매 반주기마다 생성하는 펄스의 크기를 순차적으로 형성하는 것으로 알고리즘의 구현이 용이하다. 본 논문에서 제안하는 알고리즘은 향후 국내에서 모듈형 멀티레벨 컨버터를 개발할 때 유용하게 활용될 것으로 보인다.

• Proceedings of the KIPE Conference
• /
• 2012.07a
• /
• pp.61-62
• /
• 2012

본 논문에서는 다중 반송파 정현 펄스폭 변조방식으로 제어되는 Cascaded H-bridge 멀티레벨 PWM 인버터의 스위칭 손실 저감을 위한 스위칭 패턴을 제안한다. 부하 담당 전력이 상대적으로 큰 H-bridge 모듈의 스위치는 저주파의 기본 출력 전압 레벨을 형성하도록 동작시키며, 부하 담당 전력이 상대적으로 작은 H-bridge 모듈의 스위치는 고주파의 PWM 파형을 기본파에 가감하여 출력전압 파형이 사인파에 가까워지도록 스위칭 패턴을 형성한다. 본 논문에서는 제안된 스위칭 패턴을 PD, APOD 방식의 다중 반송파 정현 펄스폭 변조방식으로 구현하여 Cascaded H-bridge 멀티레벨 PWM 인버터에 적용시키고 실험을 통해 기존의 스위칭 패턴에 비해 스위칭 손실이 개선됨을 증명한다.

• Journal of the Korea Computer Industry Society
• /
• v.2 no.10
• /
• pp.1245-1252
• /
• 2001

In this paper, we have identified speakers who give digits in Korean. In order to identify speakers, we have utilized the specifie feature parameters which extracted from sound wave. We have noticed that multipulses are present in pitch periods of sound wave, which containes the personal information and depends on the speakers. In this experiment, we have extracted multipulses, and have attempted to identify the speaker by investigating the specific feature parameters of each speaker based on the extracted multipulses.