• Journal of the Institute of Electronics Engineers of Korea SD
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• v.38 no.12
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• pp.66-74
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• 2001

In this paper, we proposed the new structure of the Rake Finger using Walsh Switch, the shared accumulator, and the pipeline FWHT(Fast Walsh Hadamard Transform) algorithm for reducing the signal processing complexity resulting from the increase of the number of data correlators. The function simulation of the proposed architecture is performed by Synopsys tool and the timing simulation is performed by Compass tool. The number of computational operation in the proposed data correlators is 160 additions and the conventional ones is 512 additions when the number of walsh code channels is 4. As a result, it is reduced about 3.2 times other than the number of computational operation of the conventional ones. Also, the result shows that the data processing time of the proposed Rake Finger architecture is 90,496[ns] and the conventional ones is 110,696[ns]. It is 18.3% faster than the data processing time of the conventional Rake Finger architecture.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.13-20
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• 2001

Adaptive digital filters with long impulse response such as acoustic echo canceller and active noise controller suffer from slow convergence and computational burden. Subband techniques and multirate signal processing have been recently developed to improve the problem of computational complexity and slow convergence in conventional adaptive filter. Any FIR transfer function can be realized as a serial connection of interpolators followed by subfilters with a sparse impulse response. In this case, each interpolator which is related to the column vector of Hadamard matrix has band-pass magnitude response characteristics shifted uniformly. Subband technique using Hadamard transform and decimation of subband signal to reduce sampling rate are adapted to system modeling and acoustic noise cancellation In this paper, delayless subband structure with nonuniform bandwidth has been proposed to improve the performance of the convergence speed without aliasing due to decimation, where input signal is split into subband one using tree-structured filter bank, and the subband signal is decimated by a decimator to reduce the sampling rate in each channel, then subfilter with sparse impulse response is transformed to full band adaptive filter coefficient using Hadamard transform. It is shown by computer simulations that the proposed method can be adapted to general adaptive filtering.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.1435-1438
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• 2005

In hands-free telephone systems, the received speech signal is fed back to the microphone and constitutes the so-called echo. To cancel the effect of this time-varying echo path, it is necessary to device an adaptive filter between the receiving and the transmitting ends. For a typical FIR realization, the length of the fullband adaptive filter results in high computational complexity and low convergence rate. Consequently, subband adaptive filtering schemes have been proposed to improve the performance. In this work, we use deterministic approach to analyze the relationship between fullband and subband adaptive filtering structures. With block adaptive filtering structure as an intermediate stage, the analysis is divided into two parts. First, to avoid aliasing, it is found that the matrix of block adaptive filters is in the form of pseudocirculant, and the elements of this matrix are the polyphase components of the fullband adaptive filter. Second, to transmit the near-end voice signal faithfully, the analysis and the synthesis filter banks in the subband adaptive filtering structure must form a perfect reconstruction pair. Using polyphase representation, the relationship between the block and the subband adaptive filters is derived.

• 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 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.

• 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 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.