• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.127-134
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• 2005

Lost packet robustness is an most important quality measure for voice over IP networks(VoIP). Recovery of the lost packet from the received information is crucial to realize this robustness. So, this paper proposes the lost packet recovery method from the received information for real-time communication for CELP coder. The proposed BS-PLC (Both Side Packet Loss Concealment) based WSOLA(Waveform Shift OverLab Add) allow the lost packet to be recovered from both the 'previous' and 'next' good packet as the LP parameter and the excitation signal are respectively recovered. The burst of packet loss is modeled by Gilbert model. The proposed scheme is applied to G.729 most used in VoIP and is evaluated through the SNR(signal to noise) and the MOS(Mean Opinion Score) test. As a simulation result, The proposed scheme provide 0.3 higher in Mean Opinion Score and 2 dB higher in terms of SNR than an error concealment procedure in the decoder of G.729 at $20\%$ average packet loss rate.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.8 s.350
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• pp.69-76
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• 2006

In this paper, we propose blind adaptation strategies for decision feedback equalizer (DFE) optimizing the operation mode between acquisitionand tracking modes based on adjustable soft decision devices. The proposed schemes select an optimal soft decision device to generate feedback samples for the DFE at a given noise to signal ratio, and apply corresponding adaptation rules which combine a blind infinite impulse response (IIR) filtering adaptation and the decision-directed least mean squared (DD-LMS) DFE adaptation. These adaptation approaches attempt to achieve not only smooth transition between acquisition and tracking of DFE but also mitigation of error propagation.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.12
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• pp.46-54
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• 2015

SAR is a Radar to obtain the video information using a radio wave. Platform emit the radio wave, depending backscattered waves returned from the target object the signal to the distance, to create a topographical map is recorded in two-dimensional image. In this paper, through a simulation to apply a variety of window in the SAR image processing for SAR image recovery is to study the application effect of the window, as a result, at the side of the signal of the SNR, Flattop window to improve the best performance it was confirmed to show.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12A
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• pp.2027-2035
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• 2001

In this paper, we present an audio stream delivery using the AMR (Adaptive Multi-Rate) coder that was adopted by ETSI and 3GPP as a standard vocoder for next generation IMT-2000 service in which includes combined sender (FEC) and receiver reconstruction technique in the Internet. By use of the media-specific FEC scheme, the possibility to recover lost packets can be much increased due to the addition of repair data to a main data stream, by which the contents of lost packets can be recovered. The AMR codec is based on the code-excited linear predictive (CELP) coding model. So we use a frame erasure concealment for CELP-based coders. The proposed scheme is evaluated with ITU-T G.729 (CS-ACELP) coder and AMR - 12.2 kbit/s through the SNR (Signal to Noise Ratio) and the MOS (Mean Opinion Score) test. The proposed scheme provides 1.1 higher in Mean Opinion Score value and 5.61 dB higher than AMR - 12.2 kbit/s in terms of SNR in 10% packet loss, and maintains the communicab1e quality speech at frame erasure rates lop to 20%.

• The Journal of the Acoustical Society of Korea
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• v.28 no.4
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• pp.392-400
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• 2009

In this paper, we propose an effective speech reinforcement technique base on ITU-T G.729A CS-ACELP codec under the near-end background noise environments. In general, since the intelligibility of the far-end speech for the near-end listener is significantly reduced under near-end noise environments, we require a far-end speech reinforcement approach to avoid this phenomena. In contrast to the conventional speech reinforcement algorithm, we reinforce the excitation signal of the codec's parameters received from the far-end speech signal based on the G.729A speech codec under various background noise environments. Specifically, we first estimate the excitation signal of ambient noise at the near-end through the encoder of the G.729A speech codec, reinforcing the excitation signal of the far-end speech transmitted from the far-end. we specially propose a novel approach to directly reinforce the excitation signal of far-end speech signal based on the decoder of the G.729A. The performance of the proposed algorithm is evaluated by the CCR (Comparison Category Rating) test of the method for subjective determination of transmission quality in ITU-T P.800 under various noise environments and shows better performances compared with conventional SNR Recovery methods.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.1
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• pp.76-87
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• 2005

In this paper, we propose a new profile, ROHC(RObust Header Compression) profile 7, for SCTP with ROHC for applying robust header compression SCTP, which is a transport layer protocol. The proposed new profile 7 adds a new field of 1 or 2 byte size on the existing SCTP packet header, which can make the SCTP stream to be diveded into acknowledgement stream and data stream. In addition, the classification of the stream can be used for recovering fault context. Consequently, in the case of using proposed ROHC-SCTP, it is possible to reduce the SCTP header overhead rate and also can be saved bandwidth.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2016.11a
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• pp.63-65
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• 2016

DVB (Digital Video Broadcasting)-S2 (Satellite - Second Generation) 표준은 현재 위성방송 시스템으로 가장 많이 사용되고 있는 표준이나, 추가적인 성능향상과 보다 다양한 응용분야에 적용하기 위해 DVB-S2 기술을 확장한 DVB-S2x (Satellite - Second Generation Extension) 시스템이 제정되었다. 그 중에서도 612,540 심볼 길이의 수퍼프레임 (Super-frame) 구조를 선택적으로 도입하여 긴 데이터 길이에 대해 스크렘블링 (Scrambling)을 적용하는 방법과 PLS (Physical Layer Signaling) 코드와 같은 참조 데이터 필드를 반복적으로 사용하는 방법 등을 사용할 수 있도록 하였다. 이를 통해 동일채널 간섭 (co-channel interference)에 대한 강인성을 증가시키고, 매우 낮은 SNR (Signal to Noise Ratio) 환경에서의 수신기 성능 향상 효과를 제공하게 된다 본 논문에서는 버스트 슈퍼프레임 전송 기반의 DVB-S2x 수신기를 위한 동기부를 설계하고 구조를 제안한다. 슈퍼프레임의 포맷은 DVB-S2x Annex E 의 규격 중 2 번 포맷을 이용하였으며, 2 번 슈퍼프레임 포맷은 버스트 (burst) 기반의 전송 방식에 용이한 측면이 있다. 동기부는 크게 버스트 검출부, 주파수 복구부, 신호 이득 조절부 그리고 심벌 타이밍 복구부로 구성된다.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.11
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• pp.63-71
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• 2003

This paper suggests the packet loss recovery method to communicate in real time in the Internet. To reduce the effects of packet loss, Forward Error Correction (FEC) that adds redundant information to voice packets can be used. Adaptive Multi Rate Wideband(AMR-WB) codec which is recently selected by the Third Generation Partnership Project(3GPP) for GSM and the third generation mobile communication WCDMA system and has also been standardized in ITU-T for providing wideband speech services is used. The major cause for speech qualitly degradation in IP-networks is packet loss. So, We recovered single lossy packet by using FEC method and concealed continued errors. The proposed scheme if evaluated in the Gilbert Internet channel model. The high quality of audio maintained up to 30% packet loss.

• The Journal of the Acoustical Society of Korea
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• v.27 no.7
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• pp.379-385
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• 2008

In this paper, we propose an effective speech reinforcement technique under the near-end and the far-end noise environments. In general, since the intelligibility of the far-end speech for the near-end listener is significantly reduced under near-end noise environments, we require a far-end speech reinforcement approach to avoid this phenomena. Specifically, based on the estimated background noise spectrum of the near-end, we reinforce the far-end speech spectrum by incorporating the more general cases under the near-end with background noise. Also, we propose the novel approach to reinforce the actual speech signal except for the noise signal in the far-end noisy speech signal. The performance of the proposed algorithm is evaluated by the CCR (Comparison Category Rating) test of the method for subjective determination of transmission quality in ITU-T P.800 under various noise environments and shows better performances compared with the conventional method.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.51 no.2
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• pp.78-85
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• 2002

We introduce a rotational decision-directed joint algorithm of blind equalization coupled with carrier recovery for 32-QAM demodulation with high symbol rate. The proposed carrier recovery, which we call a rotational decision-directed carrier recovery(RDDCR), removes the residual phase difference by rotating the decision boundary for the kth received symbol by the frequency detector output of the (k-1)th received symbol. Since the RDDCR includes the function of PLL loop filter by rotating the decision boundary, it gives a simpler demodulator structure. The rotational decision-directed blind equalization(RDDBE) with the rotated decision boundary based on the Stop-and-Go Algorithm(SGA) operated during tracking the frequency offset by the RDDCR and removes intersymbol interference due to multipaths and channel noise. Test results show that symbol error rate of $10^{-3}$ is obtained before the forward error correction when SNR equals 15dB with 150KHz of carrier frequency offset and two multipaths, which is the channel condition for 32-QAM receiver.