• Title/Summary/Keyword: 신호변환

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Smart Glove Gimbal Control that Improves the Convenience of Drone Control (드론 제어의 편의성을 향상한 스마트 글러브 짐벌 제어)

  • Lee, Seung Ho;Shin, Soo Young
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.26 no.6
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    • pp.890-896
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    • 2022
  • In this paper, gimbal camera control through smart gloves was implemented to increase convenience and accessibility to the control of drones used in various fields. Smart gloves identify human gestures and transmit signals through Bluetooth. The received signal is converted into a signal suitable for the drone through a GCS (Gound Control Station). Signals from smart gloves are expressed in a quaternion method to prevent gimbal locks, but for gimbal cameras, conversion is required to use Roll, Pitch, and Yaw methods. The data conversion mission is performed in the GCS. The GCS transmits an input signal to the control board of the drone through Wi-Fi. The control board generates and outputs the transmitted signal in a PWM manner. The output signal is input to the gimbal camera through the SBUS method and controlled. The input signal of the smart glove averaged 0.093 s and up to 0.099 s to output to the gimbal camera, showing that there was no problem in real-time use.

Development of Parallel Signal Processing Algorithm for FMCW LiDAR based on FPGA (FPGA 고속병렬처리 구조의 FMCW LiDAR 신호처리 알고리즘 개발)

  • Jong-Heon Lee;Ji-Eun Choi;Jong-Pil La
    • The Journal of the Korea institute of electronic communication sciences
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    • v.19 no.2
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    • pp.335-343
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    • 2024
  • Real-time target signal processing techniques for FMCW LiDAR are described in this paper. FMCW LiDAR is gaining attention as the next-generation LiDAR for self-driving cars because of its detection robustness even in adverse environmental conditions such as rain, snow and fog etc. in addition to its long range measurement capability. The hardware architecture which is required for high-speed data acquisition, data transfer, and parallel signal processing for frequency-domain signal processing is described in this article. Fourier transformation of the acquired time-domain signal is implemented on FPGA in real time. The paper also details the C-FAR algorithm for ensuring robust target detection from the transformed target spectrum. This paper elaborates on enhancing frequency measurement resolution from the target spectrum and converting them into range and velocity data. The 3D image was generated and displayed using the 2D scanner position and target distance data. Real-time target signal processing and high-resolution image acquisition capability of FMCW LiDAR by using the proposed parallel signal processing algorithms based on FPGA architecture are verified in this paper.

디지틀 이동통신의 최근 부품 개발 동향

  • 한경호
    • 전기의세계
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    • v.43 no.1
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    • pp.20-22
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    • 1994
  • 이동통신은 그 방식이 아날로그든 디지틀이든 같은 기능의 음성신호를 송신하고 수신하는 것이다. 아나로그형은 크게 RF 수신부, RF 송신부, 신호처리부 제어부 그리고 전원부로 나누어 진다. 통화시 송,수신이 동시에 이루어지므로 송신신호와 수신신호가 서로 간섭하지 않도록 정교환 신호 여과기가 필요하다. Philips사의 경우 몇개의 칩으로 RF/IF 변환, 아나로그 신호처리 그리고 신호제어를 할 수 있도록 설계하였고 몇몇 선두회사들은 하나의 아나로그 처리기로 baseband 아나로그 신호를 처리 할 수 있도록 설계하였다. RF 부분은 아직 별도의 PCB에 제작되는데 이유는 IF 부분의 송.수신부가 공간을 많이 차지하며, RF 부분에는 가격을 내리기 위해 개별 소자들이 많이 쓰이기 때문이다.

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Study on Noise Reduction of ECG Signal using Wavelets Transform (심전도신호의 잡음제거를 위한 웨이브렛변환의 적용에 관한 연구)

  • Chang, Doo-Bong;Lee, Sang-Min;Shin, Tae-Min;Lee, Gun-Ki
    • Journal of the Korean Institute of Telematics and Electronics S
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    • v.35S no.8
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    • pp.39-46
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    • 1998
  • One of the main techniques for diagnosing heart disease is by examining the electrocardiogram(ECG). Many studies on detecting the QRS complex, P, and T waves have been performed because meaningful information is contained in these parameters. However, the earlier detection techniques can not effectively extract those parameters from the ECG that is severely contaminated by noise source. In this paper, we performed the extracting parameters from and recovering the ECG signal using wavelets transform that has recently been applying to various fields.

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Design and Implementation the USB Multi-signal Transmission System (USB 다중 신호 전송 시스템 설계 및 구현)

  • Chae, Jung-Sik;Kim, Jong-Moon;Kim, Chang-Su;Jung, Hoe-Kyung
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2013.05a
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    • pp.623-625
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    • 2013
  • This signal can be distributed using a single USB cable, computer, video, audio, USB, LAN, one USB multi-signal transmission system was designed and implemented. USB multi-signal transmission was implemented in order to convert a single DVI, audio, and multiple USB, LAN, USB signal converter modules. This USB DVI port supports up to 1920 * 1090 resolution. USB multi-signal transmission system by sending multiple signals into a single cable installation costs of the various cable and using the replication feature of the screen, will provide schools and institutes, etc., providing the convenience of the river, and the scalability of computer peripheral ports.

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Noise Reduction and Characteristic Points Detectoin of ECG Signal using Wavelet Transforms (웨이브렛 변환을 이용한 ECG신호의 잡음제거와 특징점 검출)

  • 장두봉;이상민;신태민;이건기
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.2 no.1
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    • pp.11-17
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    • 1998
  • One of the main techniques for diagnosing heart disease is by examining the electrocardiogram(ECG). Many studies on detecting the QRS complex, p, and T waves have been performed because meaningful information is contained in these parameters. However, the earlier detecting techniques can not effectively extract those parameters from the ECG that is severely contaminated by noise source. In this paper, we performed the extracting parameters from and recovering the ECG signal using wavelets transform that has recently been applying to various fields.

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Distribution of the Slopes of Autocovariances of Speech Signals in Frequency Bands (음성 신호의 주파수 대역별 자기 공분산 기울기 분포)

  • Kim, Seonil
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.17 no.5
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    • pp.1076-1082
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    • 2013
  • The frequency bands were discovered which maximize the slopes of autocovariances of speech signals in frequency domain to increase the possibility of segregation between speech signals and background noise signal. A speech signal is divided into blocks which include multiples of sampled data, then those blocks are transformed to frequency domain using Fast Fourier Transform(FFT). To find linear equation by Linear Regression, the coefficients of autocovariance within blocks of some frequency band are used. The slope of the linear equation which is called the slope of autocovariance is varied from band to band according to the characteristics of the speech signal. Using speech signals of a man which consist of 200 files, the coefficients of the slopes of autocovariances are analyzed and compared from band to band.

Design and implementation of power-controlled front-end module for direct conversion receiver (전력제어 직접변환수신 6단자 소자 설계 및 제작)

  • Kim, Young-Wan
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.14 no.11
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    • pp.2391-2396
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    • 2010
  • The power-controlled six-port element that can control the local oscillator signal power and receiving RF signal power was designed and implemented in this paper. The direct conversion six-port element configuration was proposed, which provides the constant six-port output power by controlling the six-port input power with various signal strength. The direct conversion six-port element protects the power detector element of six-port receiver from the saturation status and compensates the transmission performance degradation. For implementation of power-controlled six-port element, the power-controlled six-port element including the power controller was analyzed. The implemented power-controlled six-port element shows the power control capability of 36 dB and gain imbalance of about 1.6 dB, phase imbalance of about $4^{\circ}$ in the frequency range of 1.69 GHz. The measured results show the good performance as direct conversion front-end element.

Noise Reduction using Spectral Subtraction in the Discrete Wavelet Transform Domain (이산 웨이브렛 변환영역에서의 스펙트럼 차감법을 이용한 잡음제거)

  • 김현기;이상운;홍재근
    • Journal of Korea Multimedia Society
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    • v.4 no.4
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    • pp.306-315
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    • 2001
  • In noise reduction method from noisy speech for speech recognition in noisy environments, conventional spectral subtraction method has a disadvantage which distinction of noise and speech is difficult, and characteristic of noise can't be estimated accurately. Also, noise reduction method in the wavelet transform domain has a disadvantage which loss of signal is generated in the high frequency domain. In order to compensate theme disadvantage, this paper propose spectral subtraction method in continuous wavelet transform domain which speech and non- speech intervals is distinguished by standard deviation of wavelet coefficient, and signal is divided three scales at different scale. The proposed method extract accurately characteristic of noise in order to apply spectral subtraction method by end detection and band division. The proposed method shows better performance than noise reduction method using conventional spectral subtraction and wavelet transform from viewpoint signal to noise ratio and Itakura-Saito distance by experimental.

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Optimized Time Scale Modification (TSM) System Integrating G,729 Speech Decoder and Dual SOLA Algorithm (G.729 음성 복호화기와 듀얼 SOLA 알고리즘을 통합한 최적의 음성 속도 변환 시스템)

  • 박규식;오승록;김선영
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.3
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    • pp.293-303
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    • 2002
  • This paper implements optimized Time Scale Modification (TSM) system using ITU G.729 speech decoder and Dual SOLA algorithm. The proposed system assume 8 Kz sampling rate, 80 samples/frame input speech from the ITU G.729 speech Decoder and the TSM (Time Scale Modification) feature of Dual SOLA produces the high quality output speech that was slow-down or speed up as a user's choice. Especially, the proposed Optimized Dual SOLA base on various simulations and theoretical analysis, and the additional interpolation procedure of the speech makes it possible to setup high performance integrated TSM system at the maximum time scale modification rate. The system performance is analyzed and verified with various input speech and playback speed.