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  • 1. Borgh, Markus
    et al.
    Berggren, Magnus
    Schüldt, Christian
    Lindström, Fredric
    Claesson, Ingvar
    An improved adaptive gain equalizer for noise reduction with low speech distortion2011In: EURASIP Journal on Audio, Speech, and Music Processing, ISSN 1687-4714, E-ISSN 1687-4722, Vol. 7Article in journal (Refereed)
    Abstract [en]

    In high-quality conferencing systems, it is desired to perform noise reduction with as limited speech distortion as possible. Previous work, based on time varying amplification controlled by signal-to-noise ratio estimation in different frequency subbands, has shown promising results in this regard but can suffer from problems in situations with intense continuous speech. Further, the amount of noise reduction cannot exceed a certain level in order to avoid artifacts. This paper establishes the problems and proposes several improvements. The improved algorithm is evaluated with several different noise characteristics, and the results show that the algorithm provides even less speech distortion, better performance in a multi-speaker environment and improved noise suppression when speech is absent compared with previous work.

  • 2. Lindström, Fredric
    et al.
    Schüldt, Christian
    Claesson, Ingvar
    Efficient Multichannel NLMS Implementation for Acoustic Echo Cancellation2007In: EURASIP Journal on Audio, Speech, and Music Processing, ISSN 1687-4714, E-ISSN 1687-4722, Vol. 2007Article in journal (Refereed)
    Abstract [en]

    An acoustic echo cancellation structure with a single loudspeaker and multiple microphones is, from a system identification perspective, generally modelled as a single-input multiple-output system. Such a system thus implies specific echo-path models (adaptive filter) for every loudspeaker to microphone path. Due to the often large dimensionality of the filters, which is required to model rooms with standard reverberation time, the adaptation process can be computationally demanding. This paper presents a selective updating normalized least mean square (NLMS)-based method which reduces complexity to nearly half in practical situations, while showing superior convergence speed performance as compared to conventional complexity reduction schemes. Moreover, the method concentrates the filter adaptation to the filter which is most misadjusted, which is a typically desired feature.

  • 3. Sällberg, Benny
    et al.
    Sattar, Farook
    Claesson, Ingvar
    On a Method for Improving Impulsive Sounds Localization in Hearing Defenders2008In: EURASIP Journal on Audio, Speech, and Music Processing, ISSN 1687-4714, E-ISSN 1687-4722, Vol. 2008, p. 1-7Article in journal (Refereed)
    Abstract [en]

    This paper proposes a new algorithm for a directional aid with hearing defenders. Users of existing hearing defenders experience distorted information, or in worst cases, directional information may not be perceived at all. The users of these hearing defenders may therefore be exposed to serious safety risks. The proposed algorithm improves the directional information for the users of hearing defenders by enhancing impulsive sounds using interaural level difference (ILD). This ILD enhancement is achieved by incorporating a new gain function. Illustrative examples and performance measures are presented to highlight the promising results. By improving the directional information for active hearing defenders, the new method is found to serve as an advanced directional aid.

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