The acoustic theory for multichannel sound reproduction systems usually assumes free-field conditions for the listening environment. However, their performance in real-world listening environments may be impaired by reflections at the walls. This impairment can be reduced by suitable compensation measures. For systems with many channels, active compensation is an option, since the compensating waves can be created by the reproduction loudspeakers. Due to the time-varying nature of room acoustics, the compensation signals have to be determined by an adaptive system. The problems associated with the successful operation of multichannel adaptive systems are addressed in this contribution. First, a method for decoupling the adaptation problem is introduced. It is based on a generalized singular value decomposition and is called eigenspace adaptive filtering. Unfortunately, it cannot be implemented in its pure form, since the continuous adaptation of the generalized singular value decomposition matrices to the variable room acoustics is numerically very demanding. However, a combination of this mathematical technique with the physical description of wave propagation yields a realizable multichannel adaptation method with good decoupling properties. It is called wave domain adaptive filtering and is discussed here in the context of wave field synthesis.
Abstract. For hands-free acoustic humadmachine interfaces, as required, e.8.. for automatic speech recognition, teleconferencing, and other multimedia services. microphone arrays using generalized sidelobe cancellers (GSCs) in conjunction with acoustic echo cancellation (AEC) can be efficiently applied for optimum communication. This contribution first devises a new structure for combining AEC and GSC in the frequency domain. We show that computational complexity is reduced by more than a factor of ten compared to a time-domain mngement. Second, robustness issues of the GSC adaptation mechanism are addressed. We illustrate how robust adaptation is assured in the new structure. Third, we propose alternatives for the fixed beamformer part of the GSC, which allow to specify (a) the width of the target tracking region and (b) improve the suppression of low frequencies. GSC robustness constraints are included using linear optimization with configurable penalty functions.
A concept of robust adaptive beamforming integrating stereophonic acoustic echo cancellation is presented which reconciles the need for low-computational complexity and efficient adaptive filtering with versatility and robustness in real-world scenarios. The synergetic combination of a robust generalized sidelobe canceller and a stereo acoustic echo canceller is designed in the frequency domain based on a general framework for multichannel adaptive filtering in the frequency domain. Theoretical analysis and real-time experiments show the superiority of this concept over comparable time-domain approaches in terms of computational complexity and adaptation behaviour. The real-time implementation confirms that the concept is robust and meets well the practical requirements of real-world scenarios, which makes it a promising candidate for commercial products
In adaptive filtering, undetected noise bursts often disturb the adaptation and may lead to h t abilities and divergence of the adaptive filter, The sensitivity against noise bursts increases with the convergence speed of the adaptive filter and limits the performance of signal processing methods where fast convergence is required. Typical applications which are sensitive against noise bursts are adaptive beamforming for audio signal acquisition or acoustic echo cancellation, where noise bursts are frequent due to undetected double-talk, In this paper, we apply double-talk resistant adaptive filtering (Gaensler, 1998) using a non-linear optimization criterion to adaptive beamforming in the discrete Fourier transform domain for bin-wise adaptation controls. We show the efficiency of double-talk resilient adaptive filtering for a generalized sidelobe canceller for speech and audio signal acquisition. The improved robustness leads to faster convergence, to higher noise-reduction, and to a better output signal quality in turn.Proposed is a method for removing reflected waves from a mixed wave consisting of a direct signal and reflected waves. The method is a kind of waveform subtraction referring to auto-correlation functions(ACFs) of multi channel speech signals. A reflected wave is assumed to have two parameters; path amplitude and delay time. The method estimates these parameters based on ACFs of signals received by microphones. The delay time of a particular path is estimated as the time lag that gives the maximum difference between the ACF of the channel in concem and the average ACF of the other rest channels. The delayed wave is subtracted from the received wave using an estimated delay only for vocalic segments, fricative-like and nasal-like segments are left as they are, and conventional spectral subtraction is applied to the rest of the input speech. The rate of waveform subtraction, or the path amplitude of a reflected wave, is estimated by minimizing the difference between the ACF of the signal in concem and the average ACF of the rests at the time delay attributed to the reflection path in concern. The proposed method can be realized without a priori knowledge about room characteristics or the target speech. Speech recognition rate for the signals picked up with 3 microphones in a reverberant environment is improved about 8% employing the proposed method. 16
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