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AR 파라메타의 강인한 추정방법과 음성향상을 위한 적용
하판봉,이기용 國立 昌原大學校 産業技術硏究所 1992 産技硏論文集 Vol.6 No.-
기존 선형예측방법은 성도특성을 추정하는데 두가지의 주요한 문제점(음원의 특성에 의한 추정의 정확도, 가산배경잡음에 의한 추정정확도의 감소)을 가지고 있다. 이러한 문제를 다음과 같이 해결한다: 첫째, 구동잡음이 가우시안과 라프라시안의 혼합으로 본 강인한 AR모델을 사용하여 성도의 파라메타를 구한다. 이때, 잡음이 없는 환경에서 얻어진 음성신호를 이용한다. 두번째로는 위의 방법과 강인한 Kalman 필터를 사용하는 방법을 제안하고, 잡음에 오염된 음성을 향상시키기 위하여 사용하였다.
실시간 Multi-Tasking Processor Scheduler에 관한 연구
하판봉 國立 昌原大學校 産業技術硏究所 1990 産技硏論文集 Vol.4 No.-
Real-time multi-tasking processor scheduler(PRCSCH)consists of four function blocks : task dispatcher (XSCAN), high-priority queue handler(XSETHI), and d-bus input queue handler(XBIQ). The last function will be supplied by inter-processor communication module and will not be described here. XSCAN schedules bus input tasks, high-priority tasks, timeout tasks, timeout tasks, normal tasks, and free-time tasks according to their status bytes. Tasks are run in the following way, XSCAN takes out top message in the d-bus input queue, translates the communication type byte(comty) contained in the message to its corresponding task index value, and runs the task when its ready bit is set. If its ready bit is reset or no message is waiting in the d-us input queue, XSCAN checks the high-priority queue. If any task is waiting in it, the task is run before any normal tasks. After bus input tasks and high-priority tasks are run. XSCAN runs normal tasks sequentially according to their status bytes. At extreme case that there are no normal tasks to be run, XSCAN runs free-time tasks(for example memory test, interface test, etc) of which free bit is set. XSETHI allows each task to upgrade the priority of the ask to high priority, thus permitting it to be run before any normal task. Multiple high-priority requests can be queued. And XTIME supplies to each task a timer that can be used to suspend its execution for a specified duration of time. These are useful for i/o timeout. If XTIME receives timeout request from a task with tick count(multiples of real-time clock period), the task will be run by XSCAN when timeout expires.
Robust Thresholding을 이용한 Wavelet 기반의 영상잡음 제거연구
하판봉 國立 昌原大學校 産業技術硏究所 2002 産技硏論文集 Vol.16 No.-
Approaches to wavelet-based denoising (or signal enhancement) have so far relied on the assumption of normally distributed perturbations. To relax this assumption, which is often violated in practice, we derive a robust wavelet thresholding technique based on the minimax description length principle. We first determine the least favorable distribution in the ε-cintaminated normal family as the member that maximizes the entropy. We show that this distribution and the best estimate, constitute a saddle point. This results in a threshold that is more resistant to heavy-tailed noise, but for which the estimation error is still potentially unbounded. We address the practical case where the underlying signal is known to be bounded, and derive a two-sided thresholding technique that is resistant to outliers and has bounded errors. We provide illustrative examples.
김외철,하판봉 國立 昌原大學校 産業技術硏究所 1996 産技硏論文集 Vol.10 No.-
Speech recognition systems based on phonemes have mainly two problems. First, phoneme segmentations are done by hand labeling. Second, coarticulation is not considered. We tried to solve the first problem by segmenting phonemes automatically using cepstrum distance measure and change detection thory, and the second problem by recognizing speech by a diphone basis. In addition, end point detections are made using the O-th cepstral coefficient and its rate of change while automatic phoneme segmentations are performed. For starting consonants, the start points of speech were detected by the proposed method more accurately than by Rabiner and Sambur's. Digits of 21 Speakers(11 males and 10 females) were used for training and those of the other four for the test data. In this paper, detection rates for phonemes and diphones were shown 55% and 85%, respectively. Detected diphones were fed into the MLP for training. In case of the speakers not participating in the training, recognition rates were shown to be 94%.
저전압/고용량 DRAM 회로를 위한 Half-VDD 전압 발생기 설계
이재형,우엄찬,박무훈,최중경,하판봉,김영희 국립7개대학공동논문집간행위원회 2005 공업기술연구 Vol.5 No.-
A half-VDD voltage generator(VHDD) using PMOS pull-up and NMOS pull-down transistors, and each of the gate voltage is able to swing at CMOS voltage level was newly proposed. Designed with these, pull-up current was made to be proportional to (VDD-|V_(TP)|)^2 and pull-down current was made to be proportional to (VDD-|V_(TN)|)^2. Also, being turned on pull-up and pull-down transistors at the same time was prevented by using short-circuit current protection circuit. Test chips which were designed by using 0.35μm CMOS twin-well process technology were fabricated and the result of measurement was suitable for low-boltage DRAMs.
Ha, Pan-Bong 國立 昌原大學校 産業技術硏究所 2002 産技硏論文集 Vol.16 No.-
Automatic control of the step size and filter length is applied to the dual-mode normalized constant modulus algorithm(DMNCMA) with joint blind carrier recovery. The proposed method results in substantial shortening of the convergence time.
Ha, Pan-Bong 창원대학교 정보통신연구소 2000 精報通信論文集 Vol.4 No.-
Modified automatic control of the step size and filter length is applied to the dual-mode normalized stop and go algorithm (DMNSAG). The proposed method results in substantial shortening of the convergence time.
Realization of Korean Word Recognizer with RASTA-PLP Parameters
Ha, Pan-Bong 國立 昌原大學校 精報通信硏究所 1998 精報通信論文集 Vol.2 No.-
Most speech parameter estimation techniques are easily influenced by the frequency response of the communication channel. With this problem in mind, we adopted a technique of RASTA-PLP[1, 2] that is more robust to such steady-state spectral factors in speech. Extracted RASTA-PLP parameters (from Korean word) were used as the input of continuous-HMM recognizer. In this paper, we want to find the recognition efficiency of RASTA-PLP speech analysis technique when Korean word (part of ETRI 445 words which were based on Koran phonation) were applied. We assumed that Korean consists of 40 phonemes including silence[3]. Korean words were trained by frequently used 3-state left-to-right CHMM(Continuous Hidden Markov Models). Experiment were conducted by varying the number of mixtures from 1 to 5 but the order of autoregressive model was fixed at 13 at present. We experimented each of the speaker-independent and speaker-dependent systems with context-dependency or context-independency. According to the experiments, the recognition rate of the speaker-independent context-independent recognizer was shown to be 91.11% with 3 mixtures and made no difference for the speaker-independent context-dependent(or tri-phone) one.
하판봉,장기원 國立 昌原大學校 情報通信硏究所 1997 精報通信論文集 Vol.1 No.-
In this paper, phonemes (phones, strictly speaking) of Korean digit speech are classified by LVQ algorithms. The LVQs are trained by the corresponding phonemes selected randomly since the lengths of phoneme data are different from one another. We are going to use the obtained VQ codebooks to train HMMs to improve their recognition rates, later on. The recognition rates range from 61.83% to 89.53% for training data and they are 60% on the average for testing data. The better and more stable results will be obtained if much more accurately labeled testing data are used and delta and acceleration cepstrums are added.