| 1. |
- Cornelius, Per, et al.
(författare)
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A Spatially Constrained Subband Beamforming Algorithm for speech enhancement
- 2004
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Konferensbidrag (refereegranskat)abstract
- This paper discusses speech enhancement in an enclosed environment such as communication in a motorcycle helmet. A new constrained subband adaptive beamformer is proposed, which uses the concept of an earlier proposed calibrated beamformer mainly developed for a hands-free in-car environment. The highly non-stationary nature of the disturbing sound field encountered in an motorcycle helmet and the fact that the source is situated in the extreme nearfield of the array, causes the beamformer to produce an unwanted fluctuation in the output level. The spatially constrained beamformer proposed in this paper makes sure that the output maintains a constant gain, as long as the corresponding source originates from the desired location.
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| 2. |
- Fai, Cedric Yiu Ka, et al.
(författare)
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A New Design Method for Broadband Microphone Arrays for Speech Input in Automobiles
- 2002
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Ingår i: IEEE Signal Processing Letters. - : IEEE. - 1070-9908 .- 1558-2361. ; 9:7, s. 222-224
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Tidskriftsartikel (refereegranskat)abstract
- A new design method for broadband microphone arrays is presented. Using sequences of calibration signals, the method is able to design finite-impulse response (FIR) filters with specific performance. The method can control and adjust the speech distortion, noise suppression, and echo cancellation directly. It turns out that a significantly shorter filter length can be applied to achieve better overall performance than the least-squares method or the signal-to-noise plus interference method.
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| 3. |
- Fundin, Per, et al.
(författare)
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Minimal Aliasing Subband System Identification
- 2001
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Konferensbidrag (refereegranskat)abstract
- Subband adaptive filters have been proposed to avoid the drawbacks of slow convergence and high computational complexity associated with time domain adaptive filters. While the computational complexity is reduced, other undesired properties, such as signal delays and signal aliasing, are introduced. Aliasing effects may result in loss of perception in speech applications. A method for the design of oversampled filter banks is proposed to reduce these effects. The design method aims at reducing the inband aliasing as well as the reconstruction aliasing for the reason of achieving robustness when weighting in the subbands alters the subband signal phase and magnitude.
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| 4. |
- Grbic, Nedelko, et al.
(författare)
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A New Pilot-Signal based Space-Time Adaptive Algorithm
- 2001
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Konferensbidrag (refereegranskat)abstract
- In the application of adaptive antenna arrays to wireless communications, a known pilot signal sequence may be used for estimating the array response at the beginning of each data frame. This pilot sequence is usually very short and conventional training methods which estimate the array response, based solely on this training sequence, may incur large estimation errors. In this paper, we propose an online modified weighted recursive least squares type of training algorithm for estimating the optimal array response by exploiting information from the whole frame of the received signal. The benefits of the proposed algorithm is that tracking of coherent noise and interference signals is substantially improved, and the overall performance is increased. Simulation results show that the proposed method offers substantial improvement when compared to the conventional least squares method.
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| 5. |
- Grbic, Nedelko, et al.
(författare)
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Blind signal separation using overcomplete subband representation
- 2001
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Ingår i: IEEE transactions on speech and audio processing. - : IEEE. - 1063-6676 .- 1558-2353. ; 9:5, s. 524-533
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Tidskriftsartikel (refereegranskat)abstract
- 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.
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| 6. |
- Grbic, Nedelko, et al.
(författare)
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Design of Oversampled Uniform DFT Filter Banks with Reduced Inband Aliasing and Delay Constraints
- 2001
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Konferensbidrag (refereegranskat)abstract
- Subband adaptive filters have been proposed to avoid the drawbacks of slow convergence and high computational complexity associated with time domain adaptive filters. Subband processing introduces transmission delays caused by the filter bank and signal degradations due to aliasing effects. One efficient way to reduce the aliasing effects is to allow a higher sample rate than critically needed in the subbands and thus reduce subband signal degradation. We suggest a design method, for a uniform DFT filter bank with any over sampling factor, where the total filter bank group delay may be specified, and where the aliasing and magnitude/phase distortions are minimized.
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| 7. |
- Grbic, Nedelko, et al.
(författare)
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Limits in FIR subband beamforming for spatially spread near-field speech sources
- 2003
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Konferensbidrag (refereegranskat)abstract
- This paper analyses optimal subband beamforming performance mainly aimed at speech enhancement and acoustic echo suppression for personal communication devices, personal computers and wireless cellular telephones. The focus is on theoretical limits of finite impulse response (FIR) beamformers for spatially spread sources in the array near-field. Performance of the Wiener solution is compared to the direct maximization of the array gain for different lengths of the FIR filters and different source interference spreads. The evaluation is performed individually in subbands with constant increasing logarithmic bandwidth. Results show that the difference between the Wiener solution and the direct array gain maximization is less than 2 dB in the measure of Signal-to-Noise plus Interference Ratio (SNIR), for small interference spread. With increasing interference spread the difference in SNIR performance increases, in favor of the array gain maximization.
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| 8. |
- Grbic, Nedelko
(författare)
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Optimal and Adaptive Subband Beamforming
- 2001
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Doktorsavhandling (övrigt vetenskapligt/konstnärligt)abstract
- The increased use of personal communication devices, personal computers and wireless cellular telephones enables the development of new inter-personal communication systems. The merge between computers and telephony technologies brings up the demand for convenient hands-free communications. In such systems the users wish to lead a conversation in much the same way as in a normal person-to-person conversation. The advantages of hands-free telephones are safety, convenience and greater flexibility. In many countries and regions, hand held telephony in cars is prohibited by legislation. By placing the microphone far away from the user a number of disadvantages are introduced, which results in substantial speech distortion and poor sound quality. These disturbances are mainly caused by room reverberation and background noise. Furthermore, acoustic feedback generated at the near-end side is a problem for the far-end side talker, who will hear his/her own voice echoed with 100-200 ms delay, making speech conversation substantially more difficult. Digital filtering may be used to obtain a similar sound quality as for hand held telephony. Three major tasks must be addressed in order to improve the quality of hands-free communication systems; noise suppression, room reverberation suppression, and acoustic feedback cancellation of the hands-free loudspeaker. The filtering operation must perform the above mentioned tasks without causing severe near-end speech distortion. A properly designed broad-band microphone array is able to perform all the given tasks, i.e. speech enhancement, echo cancellation and reverberation suppression, in a concise and effective manner. This is due to the fact that the spatial domain may be utilized as well as the temporal domain. This thesis deals with the problem of specification and design of beamformers used to extract the source signal information. A new subband adaptive beamforming algorithm is proposed, where many of the drawbacks embedded in conventional adaptive beamforming are eliminated. Evaluation in a car hands-free situation show the benefits of the proposed method. Blind signal separation is discussed and a new structure based on frequency domain inverse channel identification and time domain separation, is proposed. Further, filter-bank properties and design are discussed together with performance limitations in subband beamforming structures.
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| 9. |
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| 10. |
- Grbic, Nedelko, et al.
(författare)
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Optimal FIR Subband Beamforming for Speech Enhancement in Multipath Environments
- 2003
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Ingår i: IEEE Signal Processing Letters. - : IEEE. - 1070-9908 .- 1558-2361. ; 10:11, s. 335-338
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Tidskriftsartikel (refereegranskat)abstract
- This paper provides an analysis of optimal finite impulse response subband beamforming for speech enhancement in multipath environments. A modification of the direct path standard Wiener formulation is shown to give near-optimal performance when it comes to signal-to-noise plus interference ratio, by including coherent multipath propagation into the criterion.
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