• KSII Transactions on Internet and Information Systems (TIIS)
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• 제12권10호
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• pp.4873-4888
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• 2018

A joint blind parameter estimation algorithm based on minimum channel stability function aimed at the non-cooperative high-order modulated paired carrier multiple access (PCMA) signals is proposed. The method, which uses hierarchical search to estimate time delay, amplitude and frequency offset and the estimation of phase offset, including finite ambiguity, is presented simultaneously based on the derivation of the channel stability function. In this work, the structure of hierarchical iterative processing is used to enhance the performance of the algorithm, and the improved algorithm is used to reduce complexity. Compared with existing data-aided algorithms, this algorithm does not require a priori information. Therefore, it has significant advantage in solving the problem of blind parameter estimation of non-cooperative high-order modulated PCMA signals. Simulation results show the performance of the proposed algorithm is similar to the modified Cramer-Rao bound (MCRB) when the signal-to-noise ratio is larger than 16 dB. The simulation results also verify the practicality of the proposed algorithm.

• Journal of Communications and Networks
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• 제6권1호
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• pp.19-25
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• 2004

In this paper, we present a trellis-based blind channel estimation and equalization technique coupling two kinds of adaptive Viterbi algorithms. First, the initial blind channel estimation is accomplished by incorporating the list parallel Viterbi algorithm with the least mean square (LMS) updating approach. In this operation, multiple trellis mappings are preserved simultaneously and ranked in terms of path metrics. Equivalently, multiple channel estimates are maintained and updated once a single symbol is received. Second, the best channel estimate from the above operation will be adopted to set up the whole trellis. The conventional adaptive Viterbi algorithm is then applied to detect the signal and further update the channel estimate alternately. A small delay is introduced for the symbol detection and the decision feedback to smooth the noise impact. An automatic switch between the above two operations is also proposed by exploiting the evolution of path metrics and the linear constraint inherent in the trellis mapping. Simulation has shown an overall excellent performance of the proposed scheme in terms of mean square error (MSE) for channel estimation, robustness to the initial channel guess, computational complexity, and channel equalization.

• ETRI Journal
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• 제41권5호
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• pp.658-669
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• 2019

For dual-channel time-frequency (TF) overlapped signals with low sparsity in underdetermined blind source separation (UBSS), this paper proposes an effective method based on interval probability to estimate and expand the types of mixing matrices. First, the detection of TF single-source points (TF-SSP) is used to improve the TF sparsity of each source. For more distinguishability, as the ratios of the coefficients from different columns of the mixing matrix are close, a local peak-detection mechanism based on interval probability (LPIP) is proposed. LPIP utilizes uniform subintervals to optimize and classify the TF coefficient ratios of the detected TF-SSP effectively in the case of a high level of TF overlap among sources and reduces the TF interference points and redundant signal features greatly to enhance the estimation accuracy. The simulation results show that under both noiseless and noisy cases, the proposed method performs better than the selected mainstream traditional methods, has good robustness, and has low algorithm complexity.

• 한국멀티미디어학회논문지
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• 제6권5호
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• pp.816-823
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• 2003

이 논문에서는 토널 특성을 이용한 브라인드 오디오 워터마킹을 제안한다. 먼저 기존의 심리음향연구를 통해 토널의 인지영향에 대해 살펴보고, 토널 성분이 여러 신호처리 후 변동측면에서 매우 안정적인 특성을 가짐을 다른 워터마크에 이용되는 특성들과 비교하여 보였다. 이를 기반으로 토널 마스커를 구성하는 주파수 신호들의 관계를 이용한 브라인드 오디 오 워터마킹(blind audio watermarking) 기법을 제안하였다. 이 기법이 적용된 오디오에 대한 SDG(Subjective Diff-Grades) 음질평가에서 평균 SDG 0.27의 결과를 얻었고 이는 비지각성 면에서 토널의 인지 영향을 이용한 워터마킹이 유용하다고 볼 수 있다. 또한 time shift를 제외한 여러 신호처리 후의 워터마크 추출 결과는 98%이상으로 제안한 워터마킹의 강인성을 보였다. Time shift처리에 대해서는 시간 축 상에서 최적의 위치를 찾아 추출하는 새로운 방법을 적용하여 추출율 90%의 결과를 얻었다.

• Journal of Communications and Networks
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• 제14권5호
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• pp.481-486
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• 2012

Accurate and efficient estimation of the number of sources is critical for providing the parameter of targets in problems of array signal processing and blind source separation among other such problems. When conventional estimators work in unfavorable scenarios, e.g., at low signal-to-noise ratio (SNR), with a small number of snapshots, or for sources with a different strength, it is challenging to maintain good performance. In this paper, the detection limit of the minimum description length (MDL) estimator and the signal strength required for reliable detection are first discussed. Though a comparison, we analyze the reason that performances of classical estimators deteriorate completely in unfavorable scenarios. After discussing the limiting distribution of eigenvalues of the sample covariance matrix, we propose a new approach for estimating the number of sources which is based on a sequential hypothesis test. The new estimator performs better in unfavorable scenarios and is consistent in the traditional asymptotic sense. Finally, numerical evaluations indicate that the proposed estimator performs well when compared with other traditional estimators at low SNR and in the finite sample size case, especially when weak signals are superimposed on the strong signals.

• 방송공학회논문지
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• 제6권3호
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• pp.270-282
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• 2001

이 논문은 현재 잘 알려진 3종류의 블라인드 워터마크 삽입 및 검출방법을 신호처리 관점에서 소개한다. 이들 3가지는 각각 상 관관계기반 방법, 에코기반 방법, 그리고 패치워크 방법이다. 이들 방법은 시간영역 (또는 공간영역) 또는 변환영역에서 적용할 수 있다. 이 논문에서는 이들 세 방법을 구현하는데 필요한 기초이론 및 구현방법을 제공한다. 아울러 실제 약간의 실험 결과들을 포함시켰다.

• 한국정보통신학회논문지
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• 제17권4호
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• pp.789-796
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• 2013

시간에 따라 중심주파수를 바꾸는 도약신호를 도약주기, 도약 주파수 등에 대한 정보 없이 검출하는 것은 대단히 어렵다고 알려져 있다. 본 논문에서는 도약 신호가 존재하는 광대역 샘플링 신호로부터 도약신호의 중심주파수, 도약 주기 등의 정보를 검출하는 알고리즘을 제안하였다. 도약 신호를 검출하기 위한 일반적인 방법으로는 다수의 협대역 필터가 필요하지만 이러한 구현은 비효율적이므로 본 논문에서는 다상 DFT 필터뱅크를 도입하였다. 또한 다상 DFT 필터뱅크의 출력으로부터 도약신호를 검출하는 알고리즘을 제안하였다. 제안하는 검출 알고리즘은 메모리 사이즈나 구현 복잡도를 줄이기 위해 이진 이미지 신호처리에 기반하여 개발되었다. 제안하는 알고리즘의 성능은 모의실험과 FPGA (field programmable gate array) 구현을 통하여 확인하였다.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2003년도 학술회의 논문집 정보 및 제어부문 A
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• pp.91-94
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• 2003

In this paper, we achieve implementation of a Time-Frequency Domain Reflectometry(TFDR) system through comparatively low performance(100MS/s) PCI extensions for Instrumentation(PXI). The TFDR is the general methodology of Time Domain Reflectometry(TDR) and Frequency Domain Reflectometry(FDR). This methodology is robust in Gaussian noises, because the fixed frequency bandwidth is used. Moreover, the methodology can get more information of the fault by using the normalized time-frequency cross correlation function. The Arbitrary Waveform Generator(AWG) module generates the input signal, and the digital oscilloscope module acquires the input and reflected signals, while PXI controller module performs the control of the total PXI modules and execution of the main algorithm. The maximum range of measurement and the blind spot are calculated according ta variations of time duration and frequency bandwidth. On the basis of above calculations, the algorithm and the design of input signals used in the TFDR system are verified by real experiments. The correlation function is added to the TDR methodology for reduction of the blind spot in the TFDR system.

• The Journal of the Acoustical Society of Korea
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• 제23권2E호
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• pp.56-67
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• 2004

Despite abundant research outcomes of blind source separation (BSS) in many types of simulated environments, their performances are still not satisfactory to be applied to the real environments. The major obstacle may seem the finite filter length of the assumed mixing model and the nonlinear sensor noises. This paper presents a two-step speech enhancement method with multiple microphone inputs. The first step performs a frequency-domain BSS algorithm to produce multiple outputs without any prior knowledge of the mixed source signals. The second step further removes the remaining cross-channel interference by a spectral cancellation approach using a probabilistic source absence/presence detection technique. The desired primary source is detected every frame of the signal, and the secondary source is estimated in the power spectral domain using the other BSS output as a reference interfering source. Then the estimated secondary source is subtracted to reduce the cross-channel interference. Our experimental results show good separation enhancement performances on the real recordings of speech and music signals compared to the conventional BSS methods.

• ETRI Journal
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• 제37권3호
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• pp.491-501
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• 2015

Spectrum sensing is an elementary function in cognitive radio designed to monitor the existence of a primary user (PU). To achieve a high rate of detection, most techniques rely on knowledge of prior spectrum patterns, with a trade-off between high computational complexity and long sensing time. On the other hand, blind techniques ignore pattern matching processes to reduce processing time, but their accuracy degrades greatly at low signal-to-noise ratios. To achieve both a high rate of detection and short sensing time, we propose fast spectrum sensing with coordinate system (FSC) - a novel technique that decomposes a spectrum with high complexity into a new coordinate system of salient features and that uses these features in its PU detection process. Not only is the space of a buffer that is used to store information about a PU reduced, but also the sensing process is fast. The performance of FSC is evaluated according to its accuracy and sensing time against six other well-known conventional techniques through a wireless microphone signal based on the IEEE 802.22 standard. FSC gives the best performance overall.