• 제목/요약/키워드: Signal distortion

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A Study On the Feedforward Linear Power Amplifier Using Imperfect Signal Cancellation And Feedback (Imperfect Signal Cancellation과 Feedback을 이용한 Feedforward 선형전력증폭기에 관한 연구)

  • Park, Jung-Min;Yang, Seung-In
    • Proceedings of the Korea Electromagnetic Engineering Society Conference
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    • 2003.11a
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    • pp.87-90
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    • 2003
  • In this paper, A feedforward linear power amplifier is analyzed for imperfect signal cancellation and negative feedback for basestaion of IMT2000 band. the distortion generatied by the error amplifier is reduced using an imperfect signal cancellation for a 1-carrier WCDMA source by 4.3dB at 2.5MHz offset and 6dB at 5MHz offset of IMSR(intermodulation signal power ratio) compared to a perfect signal cancellation system. additionally, An imperfect signal cancellation using negative feedback improved 1.3dB and 8.2dB at 2.5MHz and 5MHz offset of IMSR compared to an imperfect signal cancellation.

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Compensation Techniques for TWTA non-linear intermodulation of Satellite WiBro

  • Shrestha, Robin;Lee, Byung-Seub
    • Journal of Satellite, Information and Communications
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    • v.3 no.1
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    • pp.15-21
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    • 2008
  • The high peak to average power ratio (PAPR) of OFDM (Orthogonal Frequency Division Multiplexing) system introduces inevitable non-linear distortion in the transmission due to the amplifier non-linear property. This causes both in-band distortion and out of band spectrum re-growth. In this paper we tried to compensate the problem by using polynomial based pre-distortion. Estimation of both the non-linear and inverse non-linear polynomial is achieved using the Least Square Error (LSE) method. Using these parameters closed form pre-distorter can be easily created. We also used the 'partial peak cancellation and clipping' method to remove the high peak present in the non constant amplitude of the OFDM signal responsible to drive the amplifier in near saturation region for better performance of the system

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A Temporal Decomposition Method Based on a Rate-distortion Criterion (비트율-왜곡 기반 음성 신호 시간축 분할)

  • 이기승
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.3
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    • pp.315-322
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    • 2002
  • In this paper, a new temporal decomposition method is proposed. which takes into consideration not only spectral distortion but also bit rates. The interpolation functions, which are one of necessary parameters for temporal decomposition, are obtained from the training speech corpus. Since the interval between the two targets uniquely defines the interpolation function, the interpolation can be represented without additional information. The locations of the targets are determined by minimizing the bit rates while the maximum spectral distortion maintains below a given threshold. The proposed method has been applied to compressing the LSP coefficients which are widely used as a spectral parameter. The results of the simulation show that an average spectral distortion of about 1.4 dB can be achieved at an average bit rate of about 8 bits/Frame.

Study For Watermarking Technique In Medical Image (의료영상에서의 워터마킹 기법에 관한 연구)

  • 남기철;박무훈
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2002.11a
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    • pp.348-351
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    • 2002
  • Recently, the medical imaging field has been digitalized by the development of computer science and digitization of the medical devices. There are needs of the medical imaging database service and long term storage today because of the installation of PACS system following DICOM standards, telemedicine and etc. and ,also, the illegal distortion of the medical information, data authentication and copyright are being happened. In this paper, we propose watermarking technique as a method which can protect private informations and medical imaging from geometric distortion. Because many watermarks for images are sensitive to geometric distortion, we present a algorithm that is insensitive to RST distortion in medical image. we observed the robustness against several of the signal processing and attacks in medical imaging field by embedding watermark after making a region which is insensitive to RST distortion by using FFT and LPM transformation.

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Baseband Signal Compensation Scheme for Frequency Selective Fading Channel and RF Impairments in OFDM System (OFDM 시스템에서 주파수 선택적 페이딩 채널과 RF 불완전 변환 극복을 위한 기저대역 신호보상 기법)

  • Kim, Jae-Kil;Kim, Jeong-Been;Hwang, Jin-Yong;Shin, Dong-Chul;Ahn, Jae-Min
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.35 no.1C
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    • pp.55-64
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    • 2010
  • In this paper, we propose a new compensation scheme for combined channel distortions and RF impairments based on the analysis of the impacts of IQ(In-phase/Quadrature) imbalance and phase noise on the OFDM(Orthogonal Frequency Division Multiplexing) system in the direct conversion transceiver and frequency selective fading channel distortion. The proposed scheme estimates the combined distortion by the use of training symbols and the residual distortion by pilot symbols and compensates the combined distortion, including IQ imbalance, phase noise and multipath fading at the same time. The simulation results show that the proposed scheme compensates the combined distortion of IQ imbalance, phase noise and multipath fading simultaneously.

Soft-Decision Based Quantization of the Multimedia Signal Considering the Outliers in Rate-Allocation and Distortion (이상 비트율 할당과 신호왜곡 문제점을 고려한 멀티미디어 신호의 연판정 양자화 방법)

  • Lim, Jong-Wook;Noh, Myung-Hoon;Kim, Moo-Young
    • The Journal of the Acoustical Society of Korea
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    • v.29 no.4
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    • pp.286-293
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    • 2010
  • There are two major conventional quantization algorithms: resolution-constrained quantization (RCQ) and entropy-constrained quantization (ECQ). Although RCQ works well for fixed transmission-rate, it produces the distortion outliers since the cell sizes are different. Compared with RCQ, ECQ has the constraints on the cell size but it produces the rate outliers. We propose the cell-size constrained vector quantization (CCVQ) that improves the generalized Lloyd algorithm (GLA). The CCVQ algorithm is able to make a soft-decision between RCQ and ECQ by using the flexible penalty measure according to the cell size. Although the proposed method increases the small amount of overall mean-distortion, it can reduce the distortion outliers.

Performance Improvement in Optical CDMA System Under The Presence of Beat Noise Using a Cancellation Method

  • Benaree, Warut;Noppanakeepong, Suthichai;Leelaruji, Nipha
    • 제어로봇시스템학회:학술대회논문집
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    • 2005.06a
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    • pp.1206-1210
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    • 2005
  • This paper presents performance improvement in optical CDMA system under the presence of beat noise using a cancellation technique. Optical fibers and atmospheric optical communications have been proposed the connection between base stations and central station. The optical signal beat noise is due to interference between lightwave, many optical waves are simultaneously incident on each receiver photodiode. Since the photodiode acts as a square-law detector, beat noise can occur in the receiver. While A two-stage cancellation technique is analyzed and verified via simulation employed here because of its system simplicity. By using the random ingredients of all user signals are estimated, the beat noise is rebuilt and removed from the intended signal. In addition to cancellation technique cancel the inherent multiuser interference (MUI) in CDMA system and nonlinear distortion (NLD) in optical system. It is performed at the receiver of the central station where the random ingredients of all user signals are estimated and the MUI and the NLD are rebuilt and removed from the received signal. The validity of the cancellation technique is theoretically analyzed and shown by numerical results. The increasing of capacity in two stage cancellation are obtained.

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Speaker Identification Based on Vowel Classification and Vector Quantization (모음 인식과 벡터 양자화를 이용한 화자 인식)

  • Lim, Chang-Heon;Lee, Hwang-Soo;Un, Chong-Kwan
    • The Journal of the Acoustical Society of Korea
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    • v.8 no.4
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    • pp.65-73
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    • 1989
  • In this paper, we propose a text-independent speaker identification algorithm based on VQ(vector quantization) and vowel classification, and its performance is studied and compared with that of a conventional speaker identification algorithm using VQ. The proposed speaker identification algorithm is composed of three processes: vowel segmentation, vowel recognition and average distortion calculation. The vowel segmentation is performed automatlcally using RMS energy, BTR(Back-to-Total cavity volume Ratio)and SFBR(Signed Front-to-Back maximum area Ratio) extracted from input speech signal. If the Input speech signal Is noisy, particularity when the SNR is around 20dB, the proposed speaker identification algorithm performs better than the reference speaker identification algorithm when the correct vowel segmentation is done. The same result is obtained when we use the noisy telephone speech signal as an input, too.

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A Study on TSIUVC Approximate-Synthesis Method using Least Mean Square (최소 자승법을 이용한 TSIUVC 근사합성법에 관한 연구)

  • Lee, See-Woo
    • The KIPS Transactions:PartB
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    • v.9B no.2
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    • pp.223-230
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    • 2002
  • In a speech coding system using excitation source of voiced and unvoiced, it would be involves a distortion of speech waveform in case coexist with a voiced and an unvoiced consonants in a frame. This paper present a new method of TSIUVC (Transition Segment Including Unvoiced Consonant) approximate-synthesis by using Least Mean Square. The TSIUVC extraction is based on a zero crossing rate and IPP (Individual Pitch Pulses) extraction algorithm using residual signal of FIR-STREAK Digital Filter. As a result, This method obtain a high Quality approximation-synthesis waveform by using Least Mean Square. The important thing is that the frequency signals in a maximum error signal can be made with low distortion approximation-synthesis waveform. This method has the capability of being applied to a new speech coding of Voiced/Silence/TSIUVC, speech analysis and speech synthesis.

A Novel Predistorter design using a Balanced Type IM3 Generator (평형 구조 혼변조 발생기를 이용한 전치왜곡 선형화기 설계)

  • 정형태;김상원;김철동;장익수
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.41 no.2
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    • pp.65-70
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    • 2004
  • This paper presents a novel linearization scheme for a nonlinear RF amplifier It is based on the amplitude modulation with envelope signal. The 3rd order distortion generator is composed of two FETs and it adopts a balanced structure for the purpose of main carrier cancellation. The amplitude and phase of the IM3 component can be controlled at RF band. This predistorter is implemented and tested at the KOREA PCS Tx. band (1840∼1870MHz). Experimental results of two-tone test show that the IM3 cancellation is achieved about 30-40 ㏈ for the wide dynamic range. The adjacent channel power ratio is improved by over 10 ㏈ at the broad-band CDMA signal with a peak to average power ratio of l0㏈, and this improvement is maintained through a wide range of output power levels.