| 1. |
- Cornelius, Per, et al.
(author)
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A Spatially Constrained Subband Beamforming Algorithm for speech enhancement
- 2004
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Conference paper (peer-reviewed)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. |
- Cornelius, Per, et al.
(author)
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Microphone array system for speech enhancement in a motorcycle helmet
- 2005
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Reports (other academic/artistic)abstract
- In this report a real case study of the sound environment within a helmet while driving motorcycle is investigated. A solution to perform speech enhancement is proposed for the purpose of mobile speech communication. A microphone array, mounted onto the face shield in front of the user's mouth, is used to capture the spatio-temporal properties of the acoustic wave ¯eld inside the helmet. The power of the spatially spread noise within the helmet is small when standing still while it may heavily exceed the power of the speech when driving at high speeds. This will result in dramatically reduced speech intelligibility in the communication channel. The highly dynamic noise level imposes a challenge for existing speech enhancement solutions. We propose a subband adaptive system for speech enhancement which consists of a soft constrained beamformer in cascade with a signal-to-noise ratio dependent single microphone solution. The beamformer make use of a calibration signal gathered in the actual environment from the speaker's position. This calibration procedure e±ciently captures the acoustical properties in the environment. Evaluation of the beamformer and the single microphone algorithm, both as either parts by them selves and as a cascaded structure, together with the optimal subband Wiener solution is presented. It is shown that a cascaded combination of the calibrated subband beamforming technique together with the single channel solution outperforms either one by it self, and provides near optimal results at all noise levels.
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| 3. |
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| 4. |
- Fundin, Per, et al.
(author)
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Minimal Aliasing Subband System Identification
- 2001
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Conference paper (peer-reviewed)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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| 5. |
- Grbic, Nedelko, et al.
(author)
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A New Pilot-Signal based Space-Time Adaptive Algorithm
- 2001
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Conference paper (peer-reviewed)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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| 6. |
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| 7. |
- Grbic, Nedelko, et al.
(author)
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Acoustic Echo Cancelling and Noise Suppression with Microphone Arrays
- 1999
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Reports (other academic/artistic)abstract
- This report presents a method to achieve acoustic echo canceling and noise suppression using microphone arrays. The method employs a digital self-calibrating microphone system. The on-site calibration process is a simple indirect calibration which adapts in each specific case to the environment and the electronic equipment used. The method also continuously reduces environmental disturbances such as car engine noise and fan noise. The method is primarily aimed at hands free mobile telephones by suppressing the hands free loudspeaker and car cabin noise simultaneously. The report also contains an evaluation of the impact of echo and noise suppression on a real conversation, accomplished in a car using a microphone array.
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| 8. |
- Grbic, Nedelko, et al.
(author)
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Blind signal separation using overcomplete subband representation
- 2001
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In: IEEE transactions on speech and audio processing. - : IEEE. - 1063-6676 .- 1558-2353. ; 9:5, s. 524-533
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Journal article (peer-reviewed)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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| 9. |
- Grbic, Nedelko, et al.
(author)
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Design of Oversampled Uniform DFT Filter Banks with Reduced Inband Aliasing and Delay Constraints
- 2001
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Conference paper (peer-reviewed)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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| 10. |
- Grbic, Nedelko, et al.
(author)
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Neural Network Based Adaptive Microphone Array System for Speech Enhancement
- 1998
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Conference paper (peer-reviewed)abstract
- Presents a microphone array system for use in handsfree mobile telephone equipment. The array is based on a fast and efficient “on-site” and “self- calibration” scheme. The performance in suppressing the interior car cabin noise and the far-end speech is approximately 17 dB, respectively, while maintaining the near-end speaker level. The near-end signal is almost undistorted. The performance of two different algorithms, normalized least-mean-square (NLMS) and fully connected backpropagation supervised neural network (MLP-NN) are evaluated. The proposed microphone array calibration scheme can also be used in other situations such as speech recognition devices.
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