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[Keyword] signal separation(12hit)

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  • Deep Multiplicative Update Algorithm for Nonnegative Matrix Factorization and Its Application to Audio Signals

    Hiroki TANJI  Takahiro MURAKAMI  

     
    PAPER-Digital Signal Processing

      Pubricized:
    2023/01/19
      Vol:
    E106-A No:7
      Page(s):
    962-975

    The design and adjustment of the divergence in audio applications using nonnegative matrix factorization (NMF) is still open problem. In this study, to deal with this problem, we explore a representation of the divergence using neural networks (NNs). Instead of the divergence, our approach extends the multiplicative update algorithm (MUA), which estimates the NMF parameters, using NNs. The design of the extended MUA incorporates NNs, and the new algorithm is referred to as the deep MUA (DeMUA) for NMF. While the DeMUA represents the algorithm for the NMF, interestingly, the divergence is obtained from the incorporated NN. In addition, we propose theoretical guides to design the incorporated NN such that it can be interpreted as a divergence. By appropriately designing the NN, MUAs based on existing divergences with a single hyper-parameter can be represented by the DeMUA. To train the DeMUA, we applied it to audio denoising and supervised signal separation. Our experimental results show that the proposed architecture can learn the MUA and the divergences in sparse denoising and speech separation tasks and that the MUA based on generalized divergences with multiple parameters shows favorable performances on these tasks.

  • Blind Signal Separation for Array Radar Measurement Using Mathematical Model of Pulse Wave Propagation Open Access

    Takuya SAKAMOTO  

     
    PAPER-Sensing

      Pubricized:
    2022/02/18
      Vol:
    E105-B No:8
      Page(s):
    981-989

    This paper presents a novel blind signal separation method for the measurement of pulse waves at multiple body positions using an array radar system. The proposed method is based on a mathematical model of pulse wave propagation. The model relies on three factors: (1) a small displacement approximation, (2) beam pattern orthogonality, and (3) an impulse response model of pulse waves. The separation of radar echoes is formulated as an optimization problem, and the associated objective function is established using the mathematical model. We evaluate the performance of the proposed method using measured radar data from participants lying in a prone position. The accuracy of the proposed method, in terms of estimating the body displacements, is measured using reference data taken from laser displacement sensors. The average estimation errors are found to be 10-21% smaller than those of conventional methods. These results indicate the effectiveness of the proposed method for achieving noncontact measurements of the displacements of multiple body positions.

  • Polarization Filtering Based Transmission Scheme for Wireless Communications

    Zhangkai LUO  Zhongmin PEI  Bo ZOU  

     
    LETTER-Digital Signal Processing

      Vol:
    E102-A No:10
      Page(s):
    1387-1392

    In this letter, a polarization filtering based transmission (PFBT) scheme is proposed to enhance the spectrum efficiency in wireless communications. In such scheme, the information is divided into several parts and each is conveyed by a polarized signal with a unique polarization state (PS). Then, the polarized signals are added up and transmitted by the dual-polarized antenna. At the receiver side, the oblique projection polarization filters (OPPFs) are adopted to separate each polarized signal. Thus, they can be demodulated separately. We mainly focus on the construction methods of the OPPF matrix when the number of the separate parts is 2 and 3 and evaluate the performance in terms of the capacity and the bit error rate. In addition, we also discuss the probability of the signal separation when the number of the separate parts is equal or greater than 4. Theoretical results and simulation results demonstrate the performance of the proposed scheme.

  • Derivation of Update Rules for Convolutive NMF Based on Squared Euclidean Distance, KL Divergence, and IS Divergence

    Hiroki TANJI  Ryo TANAKA  Kyohei TABATA  Yoshito ISEKI  Takahiro MURAKAMI  Yoshihisa ISHIDA  

     
    PAPER

      Vol:
    E97-A No:11
      Page(s):
    2121-2129

    In this paper, we present update rules for convolutive nonnegative matrix factorization (NMF) in which cost functions are based on the squared Euclidean distance, the Kullback-Leibler (KL) divergence and the Itakura-Saito (IS) divergence. We define an auxiliary function for each cost function and derive the update rule. We also apply this method to the single-channel signal separation in speech signals. Experimental results showed that the convergence of our KL divergence-based method was better than that in the conventional method, and our method achieved single-channel signal separation successfully.

  • Music Signal Separation Based on Supervised Nonnegative Matrix Factorization with Orthogonality and Maximum-Divergence Penalties

    Daichi KITAMURA  Hiroshi SARUWATARI  Kosuke YAGI  Kiyohiro SHIKANO  Yu TAKAHASHI  Kazunobu KONDO  

     
    LETTER-Engineering Acoustics

      Vol:
    E97-A No:5
      Page(s):
    1113-1118

    In this letter, we address monaural source separation based on supervised nonnegative matrix factorization (SNMF) and propose a new penalized SNMF. Conventional SNMF often degrades the separation performance owing to the basis-sharing problem. Our penalized SNMF forces nontarget bases to become different from the target bases, which increases the separated sound quality.

  • Bayesian Nonparametric Approach to Blind Separation of Infinitely Many Sparse Sources

    Hirokazu KAMEOKA  Misa SATO  Takuma ONO  Nobutaka ONO  Shigeki SAGAYAMA  

     
    PAPER

      Vol:
    E96-A No:10
      Page(s):
    1928-1937

    This paper deals with the problem of underdetermined blind source separation (BSS) where the number of sources is unknown. We propose a BSS approach that simultaneously estimates the number of sources, separates the sources based on the sparseness of speech, estimates the direction of arrival of each source, and performs permutation alignment. We confirmed experimentally that reasonably good separation was obtained with the present method without specifying the number of sources.

  • Super Resolution TOA Estimation Algorithm with Maximum Likelihood ICA Based Pre-Processing

    Tetsuhiro OKANO  Shouhei KIDERA  Tetsuo KIRIMOTO  

     
    PAPER-Sensing

      Vol:
    E96-B No:5
      Page(s):
    1194-1201

    High-resolution time of arrival (TOA) estimation techniques have great promise for the high range resolution required in recently developed radar systems. A widely known super-resolution TOA estimation algorithm for such applications, the multiple-signal classification (MUSIC) in the frequency domain, has been proposed, which exploits an orthogonal relationship between signal and noise eigenvectors obtained by the correlation matrix of the observed transfer function. However, this method suffers severely from a degraded resolution when a number of highly correlated interference signals are mixed in the same range gate. As a solution for this problem, this paper proposes a novel TOA estimation algorithm by introducing a maximum likelihood independent component analysis (MLICA) approach, in which multiple complex sinusoidal signals are efficiently separated by the likelihood criteria determined by the probability density function (PDF) of a complex sinusoid. This MLICA schemes can decompose highly correlated interference signals, and the proposed method then incorporates the MLICA into the MUSIC method, to enhance the range resolution in richly interfered situations. The results from numerical simulations and experimental investigation demonstrate that our proposed pre-processing method can enhance TOA estimation resolution compared with that obtained by the original MUSIC, particularly for lower signal-to-noise ratios.

  • Supervised Single-Channel Speech Separation via Sparse Decomposition Using Periodic Signal Models

    Makoto NAKASHIZUKA  Hiroyuki OKUMURA  Youji IIGUNI  

     
    PAPER-Engineering Acoustics

      Vol:
    E95-A No:5
      Page(s):
    853-866

    In this paper, we propose a method for supervised single-channel speech separation through sparse decomposition using periodic signal models. The proposed separation method employs sparse decomposition, which decomposes a signal into a set of periodic signals under a sparsity penalty. In order to achieve separation through sparse decomposition, the decomposed periodic signals have to be assigned to the corresponding sources. For the assignment of the periodic signal, we introduce clustering using a K-means algorithm to group the decomposed periodic signals into as many clusters as the number of speakers. After the clustering, each cluster is assigned to its corresponding speaker using preliminarily learnt codebooks. Through separation experiments, we compare our method with MaxVQ, which performs separation on the frequency spectrum domain. The experimental results in terms of signal-to-distortion ratio show that the proposed sparse decomposition method is comparable to the frequency domain approach and has less computational costs for assignment of speech components.

  • Signal Separation and Reconstruction Method for Simultaneously Received Multi-System Signals in Flexible Wireless System

    Takayuki YAMADA  Doohwan LEE  Hiroyuki SHIBA  Yo YAMAGUCHI  Kazunori AKABANE  Kazuhiro UEHARA  

     
    PAPER

      Vol:
    E95-B No:4
      Page(s):
    1085-1092

    We previously proposed a unified wireless system called “Flexible Wireless System”. Comprising of flexible access points and a flexible signal processing unit, it collectively receives a wideband spectrum that includes multiple signals from various wireless systems. In cases of simultaneous multiple signal reception, however, reception performance degrades due to the interference among multiple signals. To address this problem, we propose a new signal separation and reconstruction method for spectrally overlapped signals. The method analyzes spectral information obtained by the short-time Fourier transform to extract amplitude and phase values at each center frequency of overlapped signals at a flexible signal processing unit. Using these values enables signals from received radio wave data to be separated and reconstructed for simultaneous multi-system reception. In this paper, the BER performance of the proposed method is evaluated using computer simulations. Also, the performance of the interference suppression is evaluated by analyzing the probability density distribution of the amplitude of the overlapped interference on a symbol of the received signal. Simulation results confirmed the effectiveness of the proposed method.

  • Time-Domain Blind Signal Separation of Convolutive Mixtures via Multidimensional Independent Component Analysis

    Takahiro MURAKAMI  Toshihisa TANAKA  Yoshihisa ISHIDA  

     
    PAPER

      Vol:
    E92-A No:3
      Page(s):
    733-744

    An algorithm for blind signal separation (BSS) of convolutive mixtures is presented. In this algorithm, the BSS problem is treated as multidimensional independent component analysis (ICA) by introducing an extended signal vector which is composed of current and previous samples of signals. It is empirically known that a number of conventional ICA algorithms solve the multidimensional ICA problem up to permutation and scaling of signals. In this paper, we give theoretical justification for using any conventional ICA algorithm. Then, we discuss the remaining problems, i.e., permutation and scaling of signals. To solve the permutation problem, we propose a simple algorithm which classifies the signals obtained by a conventional ICA algorithm into mutually independent subsets by utilizing temporal structure of the signals. For the scaling problem, we prove that the method proposed by Koldovský and Tichavský is theoretically proper in respect of estimating filtered versions of source signals which are observed at sensors.

  • High Speed 3D IR Scanner for Home Service Robots

    Jehyuk RYU  Sungho YUN  Kyungjin SONG  Jundong CHO  Jongmoo CHOI  Sukhan LEE  

     
    PAPER-Image/Vision Processing

      Vol:
    E89-A No:3
      Page(s):
    678-685

    This paper introduces the hardware platform of the structured light processing based on depth imaging to perform a 3D modeling of cluttered workspace for home service robots. We have discovered that the degradation of precision and robustness comes mainly from the overlapping of multiple codes in the signal received at a camera pixel. Considering the criticality of separating the overlapped codes to precision and robustness, we proposed a novel signal separation code, referred to here as "Hierarchically Orthogonal Code (HOC)," for depth imaging. The proposed HOC algorithm was implemented by using hardware platform which applies the Xilinx XC2V6000 FPGA to perform a real time 3D modeling and the invisible IR (Infrared) pattern lights to eliminate any inconveniences for the home environment. The experimental results have shown that the proposed HOC algorithm significantly enhances the robustness and precision in depth imaging, compared to the best known conventional approaches. Furthermore, after we processed the HOC algorithm implemented on our hardware platform, the results showed that it required 34 ms of time to generate one 3D image. This processing time is about 24 times faster than the same implementation of HOC algorithm using software, and the real-time processing is realized.

  • Offset-Tolerant Design of Analog Chips for Independent Component Analysis

    Ki-Seok CHO  Soo-Young LEE  

     
    LETTER-Electronic Circuits

      Vol:
    E87-C No:8
      Page(s):
    1382-1387

    An analog neurochip for independent component analysis (ICA) is designed with on-line learning capability. Due to the limited dynamic range of analog device, the nonholonomic ICA algorithm is adopted. In order to accommodate the offsets due to device mismatches, a modified algorithm is developed with 2-quadrant multipliers and self-adjusting biases. Performance of the developed system was demonstrated by Monte-Carlo simulation.

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