• Title/Summary/Keyword: chirp rate

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Prewhitening Method for LFM Reverberation by Linear Dechirping (선형 Dechirping 기법을 이용한 LFM 잔향의 백색화 기법)

  • Choi, Byung-Woong;Kim, Jeong-Soo;Lee, Kyun-Kyung
    • The Journal of the Acoustical Society of Korea
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    • v.26 no.3
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    • pp.129-135
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    • 2007
  • In this paper. we propose a prewhitening method for the km reverberation to enhance the target signal. The proposed algorithm uses the dechirping method which inversely compensates the frequency chirp rate of LFM and transforms the LFM reverberation to have stationary frequency property in each data block. Also, using the left and right adjacent beam signals as reference signals. we model frequency response of each data block by AR coefficients. From these coefficients, we implement inverse filter and prewhiten the LFM reverberation of the center beam efficiently.

Performance Analysis of Multitone FH/MFSK System with Stage Address Coding in Subband and Nonsegmented Frequency Band (서브밴드 및 넌세그먼트 주파수대에서 어드레스 코딩을 사용한 FH / MFSK 시스템의 성능 분석)

  • Moon-Seung Lee
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.7 no.5
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    • pp.418-429
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    • 1996
  • The number of bits per message and the number of tones in the frequency-hopping sequence are determined by the available bandwidth and the data rate of each user. These parameters in turn determine the tone duration which strongly influences the vulnerability of the system to transmission distortions. In this paper, an address code which is assigned to each individual user was employed in order to reduce the collisions or hit. Also the frequency band is divided into several subbands and each user transmits multitone frequency per subband per chip. And the new system which is to increase the duration of each tone by increasing the total number of system frequencies that has been proposed. It is found that an optimum value bit, tone, number of frequencies per chirp can improve the err performance. This flexibility slightly increases maximum efficiecy and makes the the system less vulnerable to multipath delay. So, It is found that as the nuber of user increased 50%, the efficiency as a tuncion of the bandwidth to user'rate ratio improve 20%.

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A study on the improvement of receiver antenna as elevation angle on optical satellite communication downlink for B-ISDN (B-ISDN용 광휘성통신 다운링크의 앙각에 따른 수신안테나 개선에 관한 연구)

  • 이상규;한종석;정진호;김영권
    • Journal of the Korean Institute of Telematics and Electronics A
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    • v.32A no.3
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    • pp.1-9
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    • 1995
  • In the B-ISDN using satellite between geo-satellites and earth stations, the laser having high security and broad band width has to be used as a carrier for transmitting massive information of visual, vocal, and high rate data. In this paper, by computer simulation we analyzed the number of optical detector array of optical satellite communication downlink in case of using channel coding and no channel coding for BISDN between geo-satelites and earth stations under clear weather condition. It was supposed that 1 watt semiconductor laser was used and as modulation method, the binary FSK was used. The data rate of 10Gbps was used for B-ISDN. Also, hardly affected by atmospheric absorption 1.55$\mu$m wave-length was used to reduce influence of dispersion and chirp generated at a high speed transmission. We analyzed the received power, SNR and BER. The number of optical detector array was determined to satisfy for the BER less than 10$^{-7}$. Also, we ananlyzed the possibility of reducting the number of optical detector array in case of using channel coding. the number of optical detector array is one in the region where the elevation nangle is between 38$^{\circ}$ and 90$^{\circ}$ and two where the elevation angle is between 33$^{\circ}$ and 37$^{\circ}$ and three where the elevation angle is between 30$^{\circ}$ and 32$^{\circ}$ and increases per one as the elevation angle decreases per 1.deg.. So in the region where the elevation angle is 25$^{\circ}$, the number of optical detector arrays is eight. In case of using channel coding, the number of optical detector arrays decreases to five in the region where the elevation angle is 25$^{\circ}$. Therefore, we remaark the advantage of the channelcoding to decrease the size of received antenna and the number ob optical detector arrays.

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Adaptive-length pendulum smart tuned mass damper using shape-memory-alloy wire for tuning period in real time

  • Pasala, Dharma Theja Reddy;Nagarajaiah, Satish
    • Smart Structures and Systems
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    • v.13 no.2
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    • pp.203-217
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    • 2014
  • Due to the shift in paradigm from passive control to adaptive control, smart tuned mass dampers (STMDs) have received considerable attention for vibration control in tall buildings and bridges. STMDs are superior to tuned mass dampers (TMDs) in reducing the response of the primary structure. Unlike TMDs, STMDs are capable of accommodating the changes in primary structure properties, due to damage or deterioration, by tuning in real time based on a local feedback. In this paper, a novel adaptive-length pendulum (ALP) damper is developed and experimentally verified. Length of the pendulum is adjusted in real time using a shape memory alloy (SMA) wire actuator. This can be achieved in two ways i) by changing the amount of current in the SMA wire actuator or ii) by changing the effective length of current carrying SMA wire. Using an instantaneous frequency tracking algorithm, the dominant frequency of the structure can be tracked from a local feedback signal, then the length of pendulum is adjusted to match the dominant frequency. Effectiveness of the proposed ALP-STMD mechanism, combined with the STFT frequency tracking control algorithm, is verified experimentally on a prototype two-storey shear frame. It has been observed through experimental studies that the ALP-STMD absorbs most of the input energy associated in the vicinity of tuned frequency of the pendulum damper. The reduction of storey displacements up to 80 % when subjected to forced excitation (harmonic and chirp-signal) and a faster decay rate during free vibration is observed in the experiments.

A digital Audio Watermarking Algorithm using 2D Barcode (2차원 바코드를 이용한 오디오 워터마킹 알고리즘)

  • Bae, Kyoung-Yul
    • Journal of Intelligence and Information Systems
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    • v.17 no.2
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    • pp.97-107
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    • 2011
  • Nowadays there are a lot of issues about copyright infringement in the Internet world because the digital content on the network can be copied and delivered easily. Indeed the copied version has same quality with the original one. So, copyright owners and content provider want a powerful solution to protect their content. The popular one of the solutions was DRM (digital rights management) that is based on encryption technology and rights control. However, DRM-free service was launched after Steve Jobs who is CEO of Apple proposed a new music service paradigm without DRM, and the DRM is disappeared at the online music market. Even though the online music service decided to not equip the DRM solution, copyright owners and content providers are still searching a solution to protect their content. A solution to replace the DRM technology is digital audio watermarking technology which can embed copyright information into the music. In this paper, the author proposed a new audio watermarking algorithm with two approaches. First, the watermark information is generated by two dimensional barcode which has error correction code. So, the information can be recovered by itself if the errors fall into the range of the error tolerance. The other one is to use chirp sequence of CDMA (code division multiple access). These make the algorithm robust to the several malicious attacks. There are many 2D barcodes. Especially, QR code which is one of the matrix barcodes can express the information and the expression is freer than that of the other matrix barcodes. QR code has the square patterns with double at the three corners and these indicate the boundary of the symbol. This feature of the QR code is proper to express the watermark information. That is, because the QR code is 2D barcodes, nonlinear code and matrix code, it can be modulated to the spread spectrum and can be used for the watermarking algorithm. The proposed algorithm assigns the different spread spectrum sequences to the individual users respectively. In the case that the assigned code sequences are orthogonal, we can identify the watermark information of the individual user from an audio content. The algorithm used the Walsh code as an orthogonal code. The watermark information is rearranged to the 1D sequence from 2D barcode and modulated by the Walsh code. The modulated watermark information is embedded into the DCT (discrete cosine transform) domain of the original audio content. For the performance evaluation, I used 3 audio samples, "Amazing Grace", "Oh! Carol" and "Take me home country roads", The attacks for the robustness test were MP3 compression, echo attack, and sub woofer boost. The MP3 compression was performed by a tool of Cool Edit Pro 2.0. The specification of MP3 was CBR(Constant Bit Rate) 128kbps, 44,100Hz, and stereo. The echo attack had the echo with initial volume 70%, decay 75%, and delay 100msec. The sub woofer boost attack was a modification attack of low frequency part in the Fourier coefficients. The test results showed the proposed algorithm is robust to the attacks. In the MP3 attack, the strength of the watermark information is not affected, and then the watermark can be detected from all of the sample audios. In the sub woofer boost attack, the watermark was detected when the strength is 0.3. Also, in the case of echo attack, the watermark can be identified if the strength is greater and equal than 0.5.