• Proceedings of the IEEK Conference
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• 2003.11a
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• pp.57-60
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• 2003

In this paper, we propose a new excitation enhancement technique to improve the speech quality of low bit rate speech coders. The proposed technique is based on a harmonic model and it is employed only in the decoding process of speech coders without any additional bits. We develop the procedure of harmonic model parameters estimation and harmonic generation. and apply the technique to a current state of the art low bit rate speech coder, ITU-T G.729 Annex D. Also its performance is measured by using the ITU-T P.862 PESQ score and compared to those of the phase dispersion filter and the long-term postfilter applied to the decoded excitation. It is shown that the proposed excitation enhancement technique can improve the quality of decoded speech and provide better quality for male speech than other techniques.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.591-596
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• 1999

In this paper, we present a joint quality control system to be able to accurately control the relative picture quality among the video programs in terms of PSNR. The joint quality control system allows variable bit rate (VBR) for each video program to maintain the pre-determined relative picture quality among the aggregated video programs while keeping a constant sum of the bit rates for all programs to be transmitted over a single constant bit rate (CBR) channel. This is achieved by simultaneous controlling the video encoders to generate VBR video streams at the central controller. Furthermore we also suggest buffer regulation method based on the analysis of the constraints imposed by sender/receiver buffer sizes and total transmission rate. Through various simulation results, it is found that our quality control systems guarantee that the video buffers do not overflow and underflow and the quality control errors do not exceed 0.1 ㏈.

• Journal of the Optical Society of Korea
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• v.20 no.4
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• pp.464-469
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• 2016

All-optical non-return-to-zero (NRZ) -to- return-to-zero (RZ) data-format conversion has been successfully demonstrated using a semiconductor optical amplifier in a fiber-loop mirror (so-called SOA-loop mirror) with a continuous-wave (CW) holding beam. The converted RZ signal after pulse compression has been used to create a 40 Gb/s OTDM (Optical Time Division Multiplexing) signal. Here is proposed an NRZ-to-RZ conversion method without any additional optical clocks, unlike conventional methods based on optical AND logic. In addition, it has the merit of operating at various bit-rate speeds without any controlling device. Moreover, it has a simple structure, and it can be used for all-optical bit-rate-flexible clock recovery.

• Speech Sciences
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• v.11 no.2
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• pp.259-269
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• 2004

In this paper, we propose a new excitation enhancement technique to improve the speech quality of low bit-rate code-excited linear prediction (CELP) coders. The proposed technique is based on a harmonic model and it is employed only in the decoding process of speech coders without any additional bits. We develop the procedure of harmonic model parameter estimation and harmonic generation, and apply this technique to a current state-of-the-art low bit rate speech coder, ITU-T G.729 Annex D. Also, its performance is measured by using the ITU-T P.862 PESQ score and compared to those of the phase dispersion filter and the long-term postfilter applied to the decoded excitation. It is shown that the proposed excitation enhancement technique can improve the quality of decoded speech and provide better quality for male speech than other techniques.

• Journal of Communications and Networks
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• v.11 no.5
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• pp.480-488
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• 2009

Cooperative transmission is an effective solution to improve the performance of wireless communications over fading channels without the need for physical co-located antenna arrays. In this paper, selection combining is used at the destination instead of maximal ratio combing to optimize the structure of destination and to reduce power consumption in selective decode-and-forward relaying networks. For an arbitrary number of relays, an exact and closed-form expression of the bit error rate (BER) is derived for M-PAM, M-QAM, and M-PSK, respectively, in both independent identically distributed and independent but not identically distributed Rayleigh fading channels. A variety of simulations are performed and show that they match exactly with analytic ones. In addition, our results show that the optimum number of relays depend not only on channel conditions (operating SNRs) but also on modulation schemes which to be used.

• Journal of the Korean Institute of Telematics and Electronics
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• v.27 no.4
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• pp.484-490
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• 1990

In speech coding scheme, because information bits have different error sensitivities over channel errors, the channel coder for combining with speech coding should be realized by the variable coding rate considering the bit importance of speech information bits. In realizing the 4 kbps channel coder for 12kbps speech, this paper have chosen the channel coding method by analyzing the hard-decision post-decoding error rate of RCPC(Rate Compatible Punctured Convolutional) codes and bit error sensitivity of 12 kbps speech. Under the coherent QPSK and Rayleigh fading channel, the performance analysis has showed that 10dB gain was obtained in speech SEGSNR by 4-level uneuqal error protection, which was compared with the caseof no channel coding at 7dB channel SNR.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.349-350
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• 2006

This paper presents very low bit rate speech coder, ZFE-CELP(ZINC Function Excitation-Code Excited Linear Prediction). The ZFE-CELP speech codec is based on a ZINC function and CELP modeling of the excitation signal respectively according to the frame characteristic such as a voiced speech and an unvoiced speech. And this paper suggest strategies to improve the speech quality of the very low bit rate speech coder.

• ETRI Journal
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• v.26 no.1
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• pp.48-52
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• 2004

In this letter, we consider a new packet scheduling algorithm for an orthogonal frequency division multiplexing access/frequency division duplex (OFDMA/FDD)-based system, e.g., mobile broadband wireless access or high-speed portable internet systems, in which the radio resources of both time and frequency slots are dynamically shared by all users under a proper scheduling policy. Our design objective is to increase the number of non-realtime service (e.g., WWW) users that can be supported in the system, especially when the minimum bit rate requirement is imposed on them. The simulation results show that our proposed algorithm can provide a significant improvement in the average outage probability performance for the NRT service, i.e., significantly increasing the number of NRT users without much compromising of the cell throughput.

• Proceedings of the IEEK Conference
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• 2007.07a
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• pp.423-424
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• 2007

In this paper, we propose a scalable high-quality speech reconstruction method for distributed speech recognition (DSR). It is difficult to reconstruct speech of high quality with MFCCs at the DSR server. Depending on the bit-rate available by the DSR system, we can send additional information associated with speech coding to the DSR sorrel, where the bit-rate is variable from 4.8 kbit/s to 11.4 kbit/s. The experimental results show that the speech quality reproduced by the proposed method when the bit-rate is 11.4 kbit/s is comparable with that of ITU-T G.729 under both ideal channel and frame error channel conditions while the performance of DSR is maintained to that of wireline speech recognition.

• The Journal of the Acoustical Society of Korea
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• v.30 no.2
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• pp.86-91
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• 2011

In the underwater communications, the acoustic channel is in poor communication conditions, such as long propagation delay, narrow bandwidth, and high bit-error rate. For these bad acoustic channels, we propose an efficient automatic repeat request (ARQ) for multi-hop underwater network by using the concepts of concurrent bi-directional transmission, multiple sub-packets, and overhearing data packet instead of the acknowledgement signal. Our results show that the proposed ARQ significantly reduces the transmission latency especially in high BER compared with the existing Stop and Wait ARQ.