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  • 1. Grbic, Nedelko
    et al.
    Tao, Xiao-Jiao
    Nordholm, Sven
    Claesson, Ingvar
    Blind signal separation using overcomplete subband representation2001In: IEEE transactions on speech and audio processing, ISSN 1063-6676, E-ISSN 1558-2353, Vol. 9, no 5, p. 524-533Article in journal (Refereed)
    Abstract [en]

    This paper discusses a multirate filterbank-based extended infomax algorithm for real-world signal separation, i.e., convolved mixtures separation. Since convolution in the time domain corresponds to instantaneous mixing in the frequency domain, polyphase subband projection naturally becomes an efficient alternative to the Fourier transform based frequency domain approach. The online implementation proposed is featured by a simultaneous inverse channel identification in the frequency domain and signal filtering in the time domain. It is shown that an over-representation structure reduces aliasing between different bands and results in more accurate inverse channel estimates. Therefore, it provides better performance than the Fourier transform based structure in the measures of both separation and distortion. The performance limitation of the method is also evaluated in terms of the Wiener solution.

  • 2. Gustafsson, Harald
    et al.
    Nordholm, Sven
    Claesson, Ingvar
    Spectral Subtraction Using Reduced Delay Convolution and Adaptive Averaging2001In: IEEE transactions on speech and audio processing, ISSN 1063-6676, E-ISSN 1558-2353, Vol. 9, no 8, p. 799-807Article in journal (Refereed)
    Abstract [en]

    In handsfree speech communication the signal to noise ratio is often poor, which makes it difficult to have a relaxed conversation. By using noise suppression, the conversation quality can be improved. This paper describes a noise suppression algorithm based on spectral subtraction. The method employs a noise and speech dependent gain function for each frequency component. Proper measures have been taken to obtain a corresponding causal filter and also to ensure that the circular convolution originating from FFT filtering yields a truly linear filtering. A novel method that uses spectrum-dependent adaptive averaging to decrease the variance of the gain function is also presented. The results show a 10-dB background noise reduction for all input SNR situations tested in the range -6 to 16 dB, as well as improvement in speech quality and reduction of noise artifacts as compared with conventional spectral subtraction methods.

  • 3. Haan, Jan Mark de
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    Nordholm, Sven
    Filter Bank Design for Subband Adaptive Microphone Arrays2003In: IEEE transactions on speech and audio processing, ISSN 1063-6676, E-ISSN 1558-2353, Vol. 11, no 1, p. 14-23Article in journal (Refereed)
    Abstract [en]

    This paper presents a new method for the design of oversampled uniform DFT-filter banks for the special application of subband adaptive beamforming with microphone arrays. Since array applications rely on the fact that different source positions give rise to different signal delays, a beamformer alters the phase information of the signals. This in turn leads to signal degradations when perfect reconstruction filter banks are used for the subband decomposition and reconstruction. The objective of the filter bank design is to minimize the magnitude of all aliasing components individually, such that aliasing distortion is minimized although phase alterations occur in the subbands. The proposed method is evaluated in a car hands-free mobile telephony environment and the results show that the proposed method offers better performance regarding suppression levels of disturbing signals and much less distortion to the source speech.

  • 4. Johansson, Sven
    et al.
    Nordebo, Sven
    Claesson, Ingvar
    Convergence Analysis of a Twin-Reference Complex Least-Mean-Squares Algorithm2002In: IEEE transactions on speech and audio processing, ISSN 1063-6676, E-ISSN 1558-2353, Vol. 10, no 4, p. 213-221Article in journal (Refereed)
    Abstract [en]

    In many noise control applications, the noise is dominated by low frequencies and generated by several independent periodic sources. In such situations the tonal noise may be suppressed by using a narrowband multiple-reference feedforward controller. The performance characteristics of the control system, e.g., the convergence behavior and noise reduction are directly related to the controller adaptation rate as well as the frequency separation of the tonal components in the noise, i.e., the beat frequency. This paper treats the convergence performance of a complex least-mean-squares (LMS) algorithm using two reference signals. An analysis of its convergence behavior is presented as well as the results from computer simulations validating the convergence behavior. The convergence of the filter weights and the decrease rate of the squared error (the learning curve) for noise control applications are also discussed.

  • 5. Nordholm, Sven
    et al.
    Claesson, Ingvar
    Dahl, Mattias
    Adaptive Microphone Array Employing Calibration Signals. An Analytical Evaluation1999In: IEEE transactions on speech and audio processing, ISSN 1063-6676, E-ISSN 1558-2353, Vol. 7, no 3, p. 241-52Article in journal (Refereed)
    Abstract [en]

    This paper gives an analytical description of an adaptive microphone array that facilitates a simple built-in calibration to the environment and instrumentation. This method, suggested for use in hands-free mobile telephones and speech recognition systems for cars, provides speech enhancement and acoustic echo-cancellation. The scheme offers several advantages, such as a simple calibration procedure, suppression of directional sources, versatile robust beamforming, and reduced target signal distortion. The analysis employs noncausal Wiener filters yielding compact and effective theoretical suppression limits

  • 6. Nordholm, Sven
    et al.
    Claesson, Ingvar
    Grbic, Nedelko
    Performance Limits in Subband Beamforming2003In: IEEE transactions on speech and audio processing, ISSN 1063-6676, E-ISSN 1558-2353, Vol. 11, no 3, p. 193-203Article in journal (Refereed)
    Abstract [en]

    This paper analyzes subband beamforming schemes mainly aimed at speech enhancement and acoustic echo suppression applications such as hands-free telephony for both mobile and office environments, internet telephony and video conferencing. Analytical descriptions of both causal finite-length and noncausal infinite-length subband microphone array structures are given. More specifically, this paper compares finite Wiener filter performance with the noncausal Wiener solution, giving a comprehensive theoretical suppression limit. It is shown that even short filters will yield a good approximation of the infinite solution, provided that the element spacing and temporal sampling is matched to the frequency band of interest. Typically, 10-20 FIR taps are sufficient in each subband.

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