• Journal of KIISE:Computing Practices and Letters
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• v.14 no.2
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• pp.191-195
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• 2008

This paper proposes a technique for improving IEEE 802.11e EDCA's performance by dynamically adjusting each class's contention window size based on each class's traffic amount. For providing differentiated service differently from 802.11, 802.11e EDCA maintains four classes each of which specifies different static minimum and maximum contention window sizes. Since the initial window sites significantly affect the 802.11e performance, several window adjustment schemes have been proposed. One of the schemes known as CWminAS (CWmin Adaptation Scheme) dynamically and synchronously determines the four windows' site based on the periodically measured collision rate. This method, however, can lower the send probability of high priority classes since it can't differentiate their collisions from those of low priority classes, leading to the channel underutilization. For solving this problem, we propose ACATICT(Adaptive Contention-window Adjustment Technique based on Individual Class Traffic) algorithm which adapts each class window size based on each individual collision rate rather than one average collision rate. Our simulation experiments show that ACATICT achieves better utilization by around 10% at maximum.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.8B
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• pp.1424-1435
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• 2000

A bit-rate controller that can achieve a constant bit rate when coding object-based video sequences is an important part to achieve an adaptation to bit-rate constraints, desired video quality, distribution of bits among objects, relationship between texture and shape coding, and determination of frame skip or not. Therefore we design content-based bit rate controller which will be used for relevant bit-rate control. The implementation is an extension of MPEG-4 rate control algorithm which employs a quadratic rate-quantizer model. The importance of different objects in a video is analyzed and segmented into a number of VOPs which are adaptively bit-allocated using the object-based modelling. Some test sequences are observed by a number of non-experts and interests in each object are analysed. The initial total target bit-rate for all objects is obtained by using the proposed technique. Then the total target bits are jointly analyzed for preventing from overflow or underflow of the buffer fullness. The target bits are distributed to each object in view of its importance, not only of statistical analysis such as motion vector magnitude, size of object shape, and coding distortion of previous frame. The scheme is compared with the rate controller adopted by the MPEG-4 VM8 video coder by representing their statistics and performance.

• Speech Sciences
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• v.9 no.3
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• pp.17-24
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• 2002

The goal of this research is the real-time implementation of an AEC (Acoustic Echo Canceller) using the floating-point digital signal processor of TMS320C31. We employ an FIR-type adaptive filter with the conventional NLMS (Normalized Least Mean Square) algorithm for the adaptation of filter coefficients. We program and optimize the system in the assembler level to make it run in real-time. With 8 kHz sampling rate, the implemented AEC requires $46\;\mu$sec and $77\;\mu$sec computational time per sample for 128-and 256-tap filter, respectively. It corresponds to 37% and 62% of maximum computational ability of TMS320C31 DSP.

• Journal of Sensor Science and Technology
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• v.17 no.4
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• pp.317-324
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• 2008

This paper presents an optical doppler tomography(ODT) system using a speaker as a method to achieve depth measurement in a flowing sample. The use of the speaker provides easy implementation with a low cost. The nonlinear characteristics of the speaker has hindered its adaptation because it produces inconsistent fringe frequencies at different depths. This paper reports an adaptive algorithm to compensate the nonlinear characteristics, and could, resultantly, acquire the Doppler frequency shift caused by the sample. The experiment utilizes a flowing scattering particle solution in a capillary tube at a certain flow rate. The Doppler frequency profile over the lumen was calculated by using spectrogram method. and we obtained the velocity image of the sample.

• The Journal of the Acoustical Society of Korea
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• v.15 no.3E
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• pp.60-65
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• 1996

This paper describes on the Korean continuous speech recognition system for flight reservation. It adopts a frame-synchronous One-Pass DP search algorithm driven by syntactic constraints of context free grammar(CFG). For recognition, 48 phoneme-like units(PLU) were defined and used as basic units for acoustic modeling of Korean. This modeling was conducted using a HMM technique, where each model has 4-states 3-continuous output probability distributions and 3-discrete-duration distributions. Language modeling by CFG was also applied to the task domain of flight reservation, which consisted of 346 words and 422 rewriting rules. In the tests, the sentence recognition rate of 62.6% was obtained after speaker adaptation.

• Journal of Communications and Networks
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• v.14 no.5
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• pp.471-480
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• 2012

Reliable detection of primary user activity increases the opportunity to access temporarily unused bands and prevents harmful interference to the primary system. By extracting a global decision from local sensing results, cooperative sensing achieves high reliability against multipath fading. For the effective combining of sensing results, which is generalized by a likelihood ratio test, the fusion center should learn some parameters, such as the probabilities of primary transmission, false alarm, and detection at the local sensors. During the training period in supervised learning, the on/off log of primary transmission serves as the output label of decision statistics from the local sensor. In this paper, we extend unsupervised learning techniques with an expectation maximization algorithm for cooperative spectrum sensing, which does not require an external primary transmission log. Local sensors report binary hard decisions to the fusion center and adjust their operating points to enhance learning performance. Increasing the number of sensors, the joint-expectation step makes a confident classification on the primary transmission as in the supervised learning. Thereby, the proposed scheme provides accurate parameter estimates and a fast convergence rate even in low signal-to-noise ratio regimes, where the primary signal is dominated by the noise at the local sensors.

• Journal of Broadcast Engineering
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• v.13 no.1
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• pp.25-33
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• 2008

Video telephony is a representative service in 3G mobile network. And there have been efforts to improve quality of video telephony service in different fields. In Korea, the leading mobile service provider SKTelecom and KTF service the WCDMA network as 3G mobile network. Now, more than a million people is using the network. In this paper, we study about video telephony over WCDMA network. and propose error minimizing algorithm using cross-layer adaptation between physical layer and video codec. We simulated 3G-324M protocol with MPEG-4 video codec, and simulation results show suggested algorithm improve packet transmission rate for improving quality of video telephony service.

• Journal of Advanced Navigation Technology
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• v.6 no.2
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• pp.158-167
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• 2002

In this paper, it is analyzed that WAP packet transmission time to improve performance of WAP using SAR-QT algorithm in Bluetooth channel. The order for SAR-QT algorithm to improve the transfer capability, it is fragmented in WTP total messages that are coming down from upper layer and then the packets are sent one at time in baseband. And it is studied that transmission time for WAP over Bluetooth according to DMI, DM3 or DM5 packet type using SAR-QT algorithm in Bluetooth piconet environment. This SAR-QT algorithm decreases WAP packet transmission time of L2CAP baseband packets by sending packet that are spanning multiple slots. From the results, in WAP over Bluetooth channels, it is found out that WTP packet size ought to be increased to decrease transmission time of WAP packet. In addition, considering BER in wireless channel, optimal WTP packet size is achieved for WAP over Bluetooth in a Rician fading environment.

• Journal of the Korean Institute of Intelligent Systems
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• v.11 no.3
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• pp.223-230
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• 2001

Since self-organizing map (SOM) preserves the topology of ordering in input spaces and trains itself by unsupervised algorithm, it is Llsed in many areas. However, SOM has a shortcoming: structure cannot be easily detcrmined without many trials-and-errors. Structure-adaptive self-orgnizing map (SASOM) which can adapt its structure as well as its weights overcome the shortcoming of self-organizing map: SASOM makes use of structure adaptation capability to place the nodes of prototype vectors into the pattern space accurately so as to make the decision boundmies as close to the class boundaries as possible. In this scheme, the initialization of weights of newly adapted nodes is important. This paper proposes a method which optimizes SASOM with genetic algorithm (GA) to determines the weight vector of newly split node. The leanling algorithm is a hybrid of unsupervised learning method and supervised learning method using LVQ algorithm. This proposed method not only shows higher performance than SASOM in terms of recognition rate and variation, but also preserves the topological order of input patterns well. Experiments with 2D pattern space data and handwritten digit database show that the proposed method is promising.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.12
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• pp.1806-1814
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• 1993

Efficient speech coders using tree coding combined with harmonic scaling are designed at the rate of 4.8 kilobitts/sec (kbps). A time domain harmonic scaling algorithm (TDHS) is used to compress input speech by a factor of two. This process allows the tree coder have 1.5 bits/sample for 4.8 kbps in the case of a 6.4 kHz sampling rate. In the backward adaptive tree coder, there are three components of the code generator, including a hybrid adaptive quantizer, a short-term predictor and a pitch predictor. The robustness of the tree coder is achieved by carefully choosing the input of the short term predictor adaptation. Also, inclusion of a smoother in the pitch predictor improves the error performance of tree coder in the noisy channel. Subjectively, tree coding combined with TDHS provides good quality speech at 4.8 kbps.