• Proceedings of the Korean Information Science Society Conference
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• 2007.06b
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• pp.344-349
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• 2007

무선 통신 환경에서 실시간 화상 통신을 위해 제안된 3G-324M 프로토콜은 회선 교환 통신을 사용함으로 인해 제한된 대역폭을 이용해야만 하므로 전송 시 여러 가지 제약이 발생하게 된다. 특히 송수신의 스케줄링을 효율적으로 하지 못할 경우 H.223 프로토콜에서 매 타임 슬롯마다 전송 가능한 최대 크기의 데이터를 전송 버퍼에 채워주지 못해 전송 지연 시간이 발생하게 된다. 본 논문에서는 이러한 문제들을 해결하기 위한 통신 스케줄링 기법을 제시한다. 이 통신 스케줄링 기법은 화상 프레임의 종류에 따라 크기가 크게 변한다는 사실에 기초해 H.223 프로토콜의 실행 비율을 임시적으로 증가시켜 지연시간이 늘어나는 것을 최소화하고 이로 인해 내부 버퍼 사용량을 줄일 수 있도록 한다. 또 수신 버퍼에 처리해야 할 데이터가 많은 경우, 임시로 H.223 프로토콜의 실행 비율을 증가시켜 불필요한 수신 지연 시간이 발생하지 않도록 한다. 실험은 내부 버퍼의 사용량은 제안한 통신 스케줄링 기법이 다른 통신 스케줄링 기법들에 비해 효율적으로 관리 되며 패킷의 손실률, 수신 단말기에서의 지연이 줄어드는 것을 보여준다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.4B
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• pp.298-307
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• 2002

This paper concentrates on developing and analyzing an advanced space-time multiuser receiver for WCDMA forward link. Through exploring the interference caused by co-channel users and delayed multipaths in CDMA, a constrained linear adaptive filter is adopted to cancel the interference. Furthermore, by utilizing the space-time diversity, an optimum diversity receiver, called a constrained adaptive space-time interference canceller (C-ASTIC), is proposed. For comparison, the statistical analysis of multiuser performance for WCDMA system with space-time diversity is described. The result of the simulation showed that C-ASTIC performs better than both the space-time diversity combining maximal ratio combiner (MRC) and the single-antenna adaptive interference canceller in a multipath fading channel. Also, the efficiency of the C-ASTIC in the multimedia communication environment is investigated under the multiuser, multi-transmit antenna, and multirate WCDMA system in a multipath fading channel. From the results, the C-ASTIC was validated to be useful for multi-rate WCDMA system through improving the performance gain by more than 3 dB at BER of 10$\^$-3/ in a half or more loaded system.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.5
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• pp.963-969
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• 2000

In this paper, we define two input-output gains of linear periodic time-varying systems. One is the ratio of output with worst-case l2-norm over all inputs with unit 12-norm. It denotes G($\iota_2,\iota_2$.The other is the ratio of output with worst-case RMS value over all inputs with unit RMS value. It denotes G(RMS, RMS) .It is fact that these two gains are equivalent for linear time-invariant system. In this paper, we prove these two gains are also equivalent for linear periodic time-varying system. In addition, the relationship between two method of obtaining the generalized frequency responses for linear periodic time-varying system is derived. Finally, we apply the defined input-output gains to M-channel filter-bank which is multi-rate signal Processing system, used to speech coding. In the filter-bank, generally, aliasing distortion, magnitude distortion, and phase distortion are present. It is shown that these are kept small if the filter-bank is designed by a method that optimizes the gain G($\iota_2,\iota_2$ of an error system.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.55 no.7
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• pp.335-343
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• 2006

We have designed the system that measure one channel ECG by two electrode and extract real-time EDR with more related resipiration and comportable to subject by using conductive textile. On the assumption that relation between RL electrode and potential measurement electrode is coupled with RC connected model, we designed RL drive output to feedback two electrode for reduction of common mode signal. The conductive textile which was used for two ECG electrode was offered more comfort during night sleep in bed than any other method using attachments. In the method of single-lead EDR, R wave point or QRS interval area could be used for EDR estimation in traditional method, it is, so to speak, the amplitude modulation(AM) method for EDR. Alternatively, R-R interval could be used for frequency modulation(FM) method based on Respiratory Sinus Arrhythmia(RSA). For evaluation of performance on AM EDR and FM EDR from 14 subject, ECG lead III was measured. Each EDR was compared with both temperature around nose(direct measurement of respiration) and respiration signal from thoracic belt(indirect measurement of respiration) on mean squared error(MSE), cross correlation(Xcorr), and Coherence. The upsampling interpolation technique of multirate signal processing is applied to interpolating data instead of cubic spline interpolation. As a result, we showed the real-time EDR extraction processing to be implemented at micro-controller.

• The Journal of the Acoustical Society of Korea
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• v.24 no.1
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• pp.52-57
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• 2005

The adaptive multirate wideband (AMR-WB) speech coder has an extended audio bandwidth from 50 Hz to 7 kBz and operates on nine speech coding bit-rates from 6.6 to 23.85 kbit/s. In this Paper, we present the real-time implementation of AMR-WB speech coder using 16bit fixed-point TMS320C5509 that has dual MAC units. Firstly, We implemented AMR-WB speech coder in C 1anguage level using intrinsics, and then performed optimization in assembly language. The computational complexity of the implemented AMR-WB coder at 23.85 kbit/s is 42.9 Mclocks. And this coder needs the program memory of 15.1 kwords, data ROM of 9.2 kwords and data RAM of 13.9 kwords.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.8C
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• pp.806-815
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• 2006

It is studied to design a low pass filter of the SRC(sample rate converter), which is used to change the sampling rate of digital signals such as in digital modulation and demodulation systems. The larger the conversion ratio of the sample rate becomes, the more signal processing is needed for the filter, which corresponds to the more complexity in circuit realization. Thus it is important to reduce the amount of signal processing for the case of high conversion ratio. In this paper it is presented a design method of a two-stage cascaded FIR filter, which proved to have reduced amount of signal processing in comparison with a conventional single-stage one. The reduction effect of signal processing turned out to be more noticeable for larger value of conversion ratio, for instance, giving down to 72% in complexity for the conversion ratio of 32. It has been shown that the reduction effect is dependent to specific combination of conversion ratios of the cascaded filters. So an exhaustive search has been performed in order to obtain the optimal combination for various values of the total conversion ratio. In this paper every filter is considered to be implemented in the form of a polyphase FIR filter, and its coefficients are determined by use of the Parks-McCllelan algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.7
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• pp.1707-1717
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• 1998

Polyphase filtering techniques is used to suppress the narrowband jammer signal such as USDC TDMA overlaying the band occupied by DS-CDMA system. In the proposed jammer suppression, the received signal is separated into 64 subchannels in two stages by polyphase filtering and the location of the narrowband jammer signal is determined by measuring each subchannel power and the contaminated subchannels are simply blocked. The $E_{b}/N_{0}$ 0/ improvement of the CDMA system from jammer suppession was outstanding. The $E_{b}/N_{0}$ degradation in comparison with a performance of no jammer is around 0.8dB in the worst case. The results are also compared with those of linear prediction jammer suppression. The implementation of the ployphase jammer suppression requires great amount of data processing and computation compared to linear predication filter. Thus it is more appropriate to implement with a ASIC rather than WITH several DSPs for user terminals of forward link.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.9C
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• pp.1337-1344
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• 2004

This paper deals with analysis and real-time Implementation of a wide band adaptive multirate speech codec (AMR-WB) using a fixed-point DSP of TI's TMS320C6201. In the AMR-WB codec, input speech is divided into two frequency bands, lower and upper bands, and processed independently. The lower band signal is encoded based on the ACELP algorithm and the upper band signal is processed using the random excitation with a linear prediction synthesis filter. The implemented AMR-WB system used 218 kbytes of program memory and 92 kbytes of data memory. And its proper operation was confirmed by comparing a decoded speech signal sample-by-sample with that of PC-based simulation. Maximum required time of 5 75 ms for processing a frame of 20 ms of speech validates real-time operation of the Implemented system.

• ETRI Journal
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• v.38 no.6
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• pp.1064-1073
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• 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.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.3
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• pp.43-50
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• 2010

DDC(Digital Down Converter) is a conversion technology to decimate to a lower sampling rate and DDC for the future development of communications technology has the necessary skills. So, it has been recognized in the wireless and the SDR(Software Defined Radio) system as essential components. In addition, research is underway on spectrum sensing for efficient communications environment due to the shortage of frequency resources. In this paper, the DDC systems were analyzed for CIC(Cascaded Integrator Comb) Filter, WDF(Wave Digital Filter), SRC(Sample Rate Conversion) each module. Moreover, we proposed a linkage effectively between DDC system and Spectrum Sensing for improve the efficiency of use of frequency by computer simulations. The simulation results of the DDC system was applied to the spectrum sensing capabilities. Also, performance and complexity of the results were derived and proposed system was the result of the check.