• 한국정보전자통신기술학회논문지
• /
• 제14권4호
• /
• pp.247-256
• /
• 2021

본 논문에서는 피치 하모닉 움직임 예측과 적응적 신호 크기 예측을 이용한 패킷 손실 알고리즘을 제안한다. 스펙트럼 움직임 예측 방법은 사용 가능한 이전 패킷의 스펙트럼 상의 움직임을 일정한 부대역으로 나누어 손실된 신호의 움직임을 예측하여 복원한다. 제안하는 알고리즘에서는 음성신호를 유성음과 무성음으로 구분하여 유성음의 경우 피치 주파수를 활용하여 피치 하모닉으로 나누어 손실된 신호의 피치 하모닉 움직임을 예측하여 복원하고 무성음의 경우 스펙트럼 움직임 예측 방법을 사용하여 신호를 복원한다. 음성 프레임의 연속 손실이 발생한 경우 LMS(Least Mean Square) 예측기를 사용하여 이전 프레임의 이득 정보를 활용하여 신호 크기를 예측하여 출력 신호의 이득을 조절하는 방법을 제안한다. 객관적 평가방법인 PESQ (Perceptual Evaluation of Speech Quality) 시험을 통해 제안된 알고리즘의 성능을 평가하였고 기존의 방법보다 MOS 0.1의 성능 개선을 보였다.

• 전자공학회논문지B
• /
• 제31B권5호
• /
• pp.26-33
• /
• 1994

Phase retrieval is concerned with the reconstruction of a signal from its Fourier transform magnitude (or intensity), which arises in many areas such as X-ray crystallography, optics, astronomy, or digital signal processing. In such areas, the Fourier transform phase of the desired signal is lost while measuring Fourier transform magnitude (F.T.M.). However, if a reference 'signal is added to the desired signal, then, in the Fourier trans form magnitude of the added signal, the Fourier transform phase of the desired signal is encoded. This paper addresses uniqueness and retrieval of the encoded Fourier phase of a multidimensional signal from the Fourier transform magnitude of the added signal along with the Fourier transform magnitude of the desired signal and the information of the additive reference signal. In Part I, several conditions under which the desired signal can be uniquely specified from the two Fourier transform magnitudes and the additive reference signal are presented. In Part II, the development of non-iterative algorithms and an iterative algorithm that may be used to reconstruct the desired signal(s) is considered.

• 전자공학회논문지B
• /
• 제31B권5호
• /
• pp.34-41
• /
• 1994

Phase retrieval is concerned with the reconstruction of a signal from its Fourier transform magnitude (or intensity), which arises in many areas such as X-ray crystallography, optics, astronomy, or digital signal processing In such areas, the Fourier transform phase of the desired signal is lost while measuring Fourier transform magnitude (F.T.M.). However, if a reference 'signal is added to the desired signal, then, in the Fourier trans form magnitude of the added signal, the Fourier transform phase of the desired signal is encoded This paper addresses uniqueness and retrieval of the encoded Fourier phase of a multidimensional signal from the Fourier transform magnitude of the added signal along with Fourier transform magnitude of the desired signal and the information of the additive reference signal In Part I, several conditions under which the desired signal can be uniquely specified from the two Fourier transform magnitudes and the additive reference signal are presented In Part II, the development of non-iterative algorithms and an iterative algorithm that may be used to reconstruct the desired signal (s) is considered

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• 제7권12호
• /
• pp.3118-3134
• /
• 2013

This paper deliberates for a multimedia transmission scheme combining multiple description coding (MDC) and network coding (NC). Our goal is to take advantage from the property of MDC to provide quantized and compressed independent and identically distributed (iid) descriptions and also from the benefit of network coding, which uses network resources efficiently to recover lost data in the network. Recently, p-cycle NC has been introduced to recover and protect any lost or distorted descriptions at the receiver part exactly without need of retransmission. So far, MDC have not been explored using this type of NC. Compressed and coded descriptions are transmitted through the network where p-cycle NC is applied. P-cycle based algorithm is proposed for single and multiple descriptions lost. Results show that in the fixed bit rate, the PSNR (Peak Signal to Noise Ratio) of our reconstructed image and also subjective evaluation is improved significantly compared to previous work which is averaging method joint with MDC in order to conceal lost descriptions.

• 천문학논총
• /
• 제28권2호
• /
• pp.17-23
• /
• 2013

Song I-Yeong (1619 ~ 1692), who was an astronomy professor of Gwansanggam (觀象監, Bureau of Astronomy), created the Honcheonsigye (渾天時計, Armillary Clock) in 1669 (10th year of King Hyeonjong Era). Honcheonsigye was a unique astronomical clock which combined an armillary sphere, the traditional astronomical instrument of the Far East, with the power mechanism of western alarm clock. The clock part of this armillary clock is composed of two major parts which are the going-train, power unit used the weight, and the time signal system in a wooden case. The time signal system is composed of four parts which are the time-annunciator, the striking train, the 12 different time-announcing medallions and the sound bell. This clock has been neglected for many years and its several components have been lost. This study is to understand the structure of time signal system and suggests the restoration process.

• 대한전자공학회논문지TC
• /
• 제42권12호
• /
• pp.127-134
• /
• 2005

VoIP에서 패킷의 손실은 음성의 품질에 영향을 주는 가장 중요한 요인이다. 따라서 수신된 정보로부터 손실된 패킷을 복구하는 것은 중요하다. 따라서 본 논문은 VoIP에서 가장 많이 사용되는 CELP 부호화기를 위한 수신측 기반의 손실 패킷 복구방법을 제안한다. 제안하는 WSOLA(Waveform Shift OverLab Add)기반의 BS-PLC (Both Side Packet Loss Concealment) 방법은 패킷 손실이 발생하였을 경우 미래 패킷을 이용할 수 있는 경우와 그렇지 않을 경우로 나누고, LP(Linear prediction) 파라미터와 여기 신호를 복구한다. 미래 패킷을 이용할 수 없는 경우에는 과거에 전송된 정상 패킷만을 가지고 복원을 하며, 미래 패킷을 이용할 수 있을 경우에는 과거의 정상 패킷과 미래의 정상 패킷을 동시에 이용하여 손실된 패킷을 복구한다. 연속 패킷 손실 환경은 Gilbert 모델로 설정하였고, 제안한 알고리즘을 VoIP에서 가장 많이 사용되는 CELP 음성부호화기인 G.729에 적용하여 성능을 비교한다. 성능 비교를 위해 손실율을 변화시키면서 SNR(Signal to Noise)와 MOS(Mean Opinion Score)측정하였고, 제안한 방법을 G.729의 패킷 손실 은닉 방법과 비교하였다. 실험 결과, 평균 손실률이 $20\%$에서도 SNR은 2dB, MOS값은 0.3정도로의 음질 개선을 보였다.

• International Journal of Fuzzy Logic and Intelligent Systems
• /
• 제3권1호
• /
• pp.66-71
• /
• 2003

In the past, when there were few vehicles on the road, the T.O.D.(Time of Day) traffic signal worked very well. The T.O.D. signal operates on a preset signal cycling which cycles on the basis of the average number of average passenger cars in the memory device of an electric signal unit. Today, with increasing traffic and congested roads, the conventional traffic light creates startup-delay time and end lag time so that thirty to forty-five percent efficiency in traffic handling is lost, as well as adding to fuel costs. To solve this problem, this paper proposes a new concept of optimal green time algorithm, which reduces average vehicle waiting time while improving average vehicle speed using fuzzy rules and neural networks. Through computer simulation, this method has been proven to be much more efficient than fixed time interval signals. Fuzzy Neural Network will consistanly improve average waiting time, vehicle speed, and fuel consumption.

• 대한의용생체공학회:학술대회논문집
• /
• 대한의용생체공학회 1990년도 춘계학술대회
• /
• pp.25-27
• /
• 1990

By designing the gradient pulses to be velocity compensated during one pulse cycle in SSFP (steady state free precession) imaging, the flowing spins can be maintained in the steady state. In the new SSFP sequence the flow signal which might be lost in conventional SSFP imaging sequences can be restored owing to the signal contribution from the preceding pulse cycles. By using the proposed SSFP sequence substantial restoration of the flow signal has been observed for the CSF (cerebro spinal fluid) of the human head.

• 한국통신학회논문지
• /
• 제38B권5호
• /
• pp.328-336
• /
• 2013

디바이스 간 직접통신을 할 때 중요한 절차 중 하나는 근거리에 위치한 디바이스들을 발견하는 것이다. 정확한 디바이스 발견을 위해서는 모든 디바이스들이 한 번 이상 발견 신호를 전송하고 다른 디바이스들이 그 신호를 수신함으로써 모든 디바이스들 사이의 거리 정보를 파악해야 한다. 그러나 디바이스 발견을 위해서 주기적으로 모든 디바이스들이 한 번 이상 신호를 전송한다면 디바이스 신호 전송 주기가 너무 길어지거나 디바이스 발견을 위한 자원을 많이 사용해야 하는 문제가 발생할 수 있다. 또한 여러 실제적인 요인에 의해서 일부 정보들이 손실될 수 있으며 모든 필요한 정보를 얻는데 너무 많은 시간이 소요될 수 있다. 본 논문에서는 디바이스 발견을 위한 자원을 줄이고 발견 정보 손실이 있는 경우를 지원하기 위하여 일부 정보만을 활용하여 근거리에 있는 디바이스를 발견하는 방법을 제안한다. 또한 일부 발견 정보만이 있을 때의 발견 확률에 대해서 논한다.

• ETRI Journal
• /
• 제38권6호
• /
• pp.1064-1073
• /
• 2016

An adaptive speech streaming method to improve the perceived speech quality of a software-based multipoint control unit (SW-based MCU) over IP networks is proposed. First, the proposed method predicts whether the speech packet to be transmitted is lost. To this end, the proposed method learns the pattern of packet losses in the IP network, and then predicts the loss of the packet to be transmitted over that IP network. The proposed method classifies the speech signal into different classes of silence, unvoiced, speech onset, or voiced frame. Based on the results of packet loss prediction and speech classification, the proposed method determines the proper amount and bitrate of redundant speech data (RSD) that are sent with primary speech data (PSD) in order to assist the speech decoder to restore the speech signals of lost packets. Specifically, when a packet is predicted to be lost, the amount and bitrate of the RSD must be increased through a reduction in the bitrate of the PSD. The effectiveness of the proposed method for learning the packet loss pattern and assigning a different speech coding rate is then demonstrated using a support vector machine and adaptive multirate-narrowband, respectively. The results show that as compared with conventional methods that restore lost speech signals, the proposed method remarkably improves the perceived speech quality of an SW-based MCU under various packet loss conditions in an IP network.