태권도 품새 수련이 아동의 심박수, 심근산소소비량, 에너지 대사에 미치는 영향

정성태,전태원,박익렬,정덕조,우재홍,이광희,이동기,강현주 서울대학교 체육연구소 2000 서울大學校 體育硏究所論集 Vol.21 No.1

The Taegeuk Poomse 1 through 8 of Taekwondo are based upon eight major branches of philosophical theories. The propose of this study was to investigate the nfluence of Taekwondo poomse trainning on heart rate, MVO2, energy metabolism in primary school students. Subjects for this study were eight male the fifth∼sixth grades students who participated in exercise program consist of Taekwondo poomse trainning in dojang which is located in Shinlim-2dong, Kwanak-Gu, Seoul(12.7± 0.5years, 150.8 ±4.6cm, 45.1 ± 9.4kg, 24.9 ± 9trainning length). The results of thet this study are as follows: 1) The average of heart rate was Taegeuk Poomse 1 through 8 of Taekwondo 90.57± 4.gbeats/min in rest and 139.7 ± 17.5beats/min in exercise. The minimum heart rate was 89.0±2.8, peak heart rate was 167.1 ±6.4 in exercise. It was ranged from 42% to 80% 2) of the max heart rate and ranged 42 to 65%HRmax. Taegeuk 1 Jang was differenced with others Jangs significantly(p<0.05), 2) The time of the Taegeuk 1 through 8 Jang of Taekwondo was 5minutes 12seconds. Heart rate was ranged from 90.6 ± 7.85beats/min to 171.8 ±8.Ibeats/min and increased linearly. 3) The average of the MVO2 was 11.1 ±0.78mmHg · bpm-3 in rest and 19.1 ± 1.2mmHg ㆍbpm-3 in exercise. MVO2 was ranged from 10.5mmHg · bpm-3 to 20.3mmHg · bpm-3. There were not difference between Taegeuk Poomses(p<0.05). 4) The energy metabolism was as follows: Mean VO2 Respiratory exchange ratio, energy consumption were 0.51(V02L/min),0.71(RQ),2.32Ckca1/min) in rest respectively. but In exercise Mean VO2 Respiratory exchange ratio, energy consumption were 0.89 ± 0.18( V 02L/min), 0.79±0.08(RQ), 4.26±0.51(kcal/min) respectively . The difference was only between Taegeuk 1 Jang and Taegeuk 3 Jang(p<0.05). This study showed that the Taekwondo Poomse trainning was proper exercise for fitness, body composition and weight control in normal, abnormal(obese) children because the Taekwondo trainning are dynamic and exciting sport.

DSP를 이용한 가변어휘 음성인식기 구현에 관한 연구

정익주 한국음성과학회 2004 음성과학 Vol.11 No.3

In this paper, we implemented a vocabulary-independent speech recognizer using the TMS320VC33 DSP. For this implementation, we had developed very small-sized recognition engine based on diphone sub-word unit, which is especially suited for embedded applications where the system resources are severely limited. The recognition accuracy of the developed recognizer with 1 mixture per state and 4 states per diphone is 94.5% when tested on frequently-used 2000 words set. The design of the hardware was focused on minimal use of parts, which results in reduced material cost. The finally developed hardware only includes a DSP, 512 Kword flash ROM and a voice codec. In porting the recognition engine to the DSP, we introduced several methods of using data and program memory efficiently and developed the versatile software protocol for host interface. Finally, we also made an evaluation board for testing the developed hardware recognition module.

실시간 Time Scaler 개발

정익주,조돈하 강원대학교 정보통신연구소 1999 정보통신논문지 Vol.3 No.-

Time Scale Modification refers to the process of compressing or expanding the time-scale of an audio segment. A signal which is time-scale compressed has a shorter duration, while a time-scale expanded signal is longer in duration. Simply speeding up the playback rate of a digital signal resampling causes the local pitch periods to be shortened. A signal created in this way has a shorter time-scale, but is largely unintelligible. A properly time-scale modified signal maintains properties of the original signal shch as local pitch period, speaker identity, and intelligibility. It does so by preserving the prominent features of the signal associated with these properties, namely the local pitch period. When this is performed on a voice signal, the resulting signal sounds as though the same person is talking faster or slower in the same voice. Pitch detection is used by autocorrelation using signal only. This is synthesized using PSOLA method. It shows good quality from 8khz to 22.05khz sampling rate regardless of speaker's sex or age.

TMS320F28335 DSP를 이용한 화자독립 음성인식기 구현

정익주 江原大學校 産業技術硏究所 2009 産業技術硏究 Vol.29 No.A

In this paper, we implemented a speaker-independent speech recognizer using the TMS320F28335 DSP which is optimized for control applications. For this implementation, we used a small-sized commercial DSP module and developed a peripheral board including a codec, signal conditioning circuits and I/O interfaces. The speech signal digitized by the TLV320AIC23 codec is analyzed based on MFCC feature extraction methed and recognized using the continuous-density HMM. Thanks to the internal SRAM and flash memory on the TMS320F28335 DSP, we did not need any external memory devices. The internal flash memory contains ADPCM data for voice response as well as HMM data. Since the TMS320F28335 DSP is optimized for control applications, the recognizer may play a good role in the voice-activated control areas in aspect that it can integrate speech recognition capability and inherent control functions into the single DSP.

웨이브릿 기반의 움직임 추정 기법에 관한 연구

김남일,정익주 강원대학교 정보통신연구소 2002 정보통신논문지 Vol.6 No.-

The International standard such as H.261, MPEG(Moving Picture Expert Group) has been used block-based motion estimation technique. Wavelet transform is very popular in these video processing recently, because wavelet based compression has the advantages of efficient decorrelation each consecutive image frame and reduced-complexity in multiresolution motion estimation technique (MRME). We propose a technique to improve coding performance of baseline wavelet-based coder. The motion vectors are calculated using adaptive threshold method from subimage. In order to find wrong-searched block, threshold factor is defined for comparing with energy of reference block. The experiment results show that the proposed method improved bit-rate and PSNR of decoder.

Streaming SIMD Extension(SSE)에 적합한 영상 압축 방식에 대한 연구

배현성,정익주 강원대학교 정보통신연구소 2003 정보통신논문지 Vol.7 No.-

The objective of this paper is to present the new FourCC to use the power of general CPU, to the highest degree, supporting SIMD that is being used the most. This is the form that can be applied to all the picture processing compressed by block. The most ideal form is that the used data is in cache memories of CPU. But the video needs very big memories, so there is little probabilily that the data is in the cache memories. If FourCC suggested in this paper is used, there is always the data in the cache memories. Also, it means that this form can receive profits very much in terms of speed. This paper has simulation using DCT-based vision compression that is the most typical form coding based on block segmentation. We can recognize that the speed improves through this simulation.

서브랜드 적응필터를 이용한 잡음제거기

최지현,정익주 강원대학교 정보통신연구소 2008 정보통신논문지 Vol.12 No.-

In this paper, we proposed a subband adaptive noise canceller(SANL) where each band has different number of filter taps. Since most energy of voice signal is located in low frequency area, the proposed algorithm allocates more filter taps in lower frequency band than in higher frequency band with keeping total number of filter taps constant.' The performance of the proposed SANC is better than that of the full band ANC. In this experiment where we used the analysis filter with the filter order 80, we consider two cases, 2-band SANC and 4-band SANC, the 2-band SANC outperformed the 4-band SANC. This is because the 4-band SANC has more transition bands than 2-band SANC, and therefore transition bands caused more aliasing. That is, increased number of bands could not compensate for the degradation caused from aliasing due to relatively wide transition bands. This problem can be alleviated using analysis filter with higher filter order, but this requires more computational performance. In addition, we implement the proposed SANC using TMS320C6713 DSP for real-time operation.

반음절기반의 한국어 연속숫자음인식과 그 후처리에 대한 연구

정재부,정훈,정익주 한국음성과학회 2001 음성과학 Vol.8 No.4

This paper describes the effect of new recognition unit, a unit based on semi-syllable, and its post processing method. A recognition unit based on semi-syllable expresses Korean connected digit's coarticulation effect. An existing method using semi-syllable limits next models, derived from current recognized models, to make complete connected digit sequence. However, this paper uses a new method to make complete connected digit sequence. The new post-processing method recognizes isolated digit words which include digits sequence from the digit combinations being able to occur from current recognized semi-syllable sequence. This method gives an improved accuracy rate than that of existing method. This new post processing provides two advantages. 1) It corrects current rnis-recognized semi-syllable unit. 2) When people say each digit, they say it without regard to saying duration.

Mel-Scale 기반의 다중 대역 주파수 차감법을 이용한 음질향상

함영준,정익주 강원대학교 정보통신연구소 2004 정보통신논문지 Vol.8 No.-

The noise in digital signal processing always exists, posing an important subject to solve. Especially, in speech signal processing, it is very important to remove or reduce the noise using speech enhancement techniques. For this, various algorithms have been suggested. The basic spectral subtraction algorithm is very simple to implement. But for the better system performance, this paper presents some of modified algorithms and additional conception. The main contribution of our proposed algorithm is as follows. Firstly, the introduction of multi-band conception divides frequency domain into various bands. Secondly, when we separate bands, it does not use normal scale but Mel-scale. Finally, process parameters are adapted in the environment by VAD, and alpha value varies as SNR changes.

TMS320VC5510 DSK를 이용한 음성인식 로봇

최지현,정익주 강원대학교 산업기술연구소 2007 産業技術硏究 Vol.27 No.A

As demands for interaction of humans and robots are increasing, robots are expected to be equipped with intelligibility which humans have. Especially, for natural communication, hearing capabilities are so essential that speech recognition technology for robot is getting more important. In this paper, we implement a speech recognizer suitable for robot applications. One of the major problem in robot speech recognition is poor speech quality captured when a speaker talks distant from the microphone a robot is mounted with. To cope with this problem, we used wireless transmission of commands recognized by the speech recognizer implemented using TMS320VC5510 DSK. In addition, as for implementation, since TMS320VC5510 DSP is a fixed-point device, we represent efficient realization of HMM algorithm using fixed-point arithmetic.