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| 2. |
- Grbic, Nedelko, et al.
(författare)
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Acoustic Echo Cancelling and Noise Suppression with Microphone Arrays
- 1999
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Rapport (övrigt vetenskapligt/konstnärligt)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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| 3. |
- Grbic, Nedelko, et al.
(författare)
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Neural Network Based Adaptive Microphone Array System for Speech Enhancement
- 1998
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Konferensbidrag (refereegranskat)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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| 4. |
- Grbic, Nedelko, et al.
(författare)
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Performance Improvement of Multiple Array 3D Sonic Digitizer System via Calibration
- 1999
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Konferensbidrag (refereegranskat)abstract
- Multiple array 3D sonic digitizer system is a product made for measuring the geometric structure of an arbitrary object in three dimension. This paper gives a theoretical performance evaluation of the system. The Cramer-Rao Lower Bound of the parameter estimates of a linear segment is derived. In addition, the distortion effect caused by the time-delay bias, including rescaling, rotation and bending of the original object, is clarified. A simple method using a line segment is proposed to calibrate the bias term arised in the different parts of the system. Finally a simulation gives a numerical demonstration of the previous analysis and performance improvement of the suggested method.
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| 5. |
- Grbic, Nedelko
(författare)
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Speech Signal Extraction : A Multichannel Approach
- 1999
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Licentiatavhandling (övrigt vetenskapligt/konstnärligt)abstract
- In speech signal extraction the aim is to extract human speech in a physical environment by using microphones. In any real world environment there are many disturbance sources that cause unwanted sound pressure, which in turn may degrade the comprehension of the speech at the microphones. This licentiate thesis deals with the solution to this problem. Different approaches are proposed to solve the problem for many accounted scenarios in real life.
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| 6. |
- Grbic, Nedelko, et al.
(författare)
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Subband Acoustic Echo Cancelling using LMS and RLS
- 1999
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Rapport (övrigt vetenskapligt/konstnärligt)abstract
- The increasing use of modern hands free communication systems such as video conferencing, computer communications, and vehicle mounted cellular telephones brings the demand for high-quality acoustic echo cancellation up to focus. In these applications the echo path which has to be identified typically has long time duration, the order of 100 ms. For this identification the length of the filter will be long. This report evaluates the Normalized Least Mean Square (NLMS) and the Weighted Recursive Least Square (WRLS) algorithms for acoustic echo cancelling using a delayless subband scheme. Subband signal processing has shown to be efficient both when it comes to convergence rate and level of echo suppression. The evaluation is performed for real speech signals sampled from a conversation using a hands free set mounted in an automobile, and a conversation using conference telephony equipment in a conference room. A comparison of subband and fullband algorithms is made both with respect to the computational cost and level of echo suppression. Results show that when the impulse response is very long, i.e. in such environments as conference rooms, the subband approach is beneficial. In a car environment the size of enclosure and damping means that the response is quite short and a conventional echo canceller could perform as well as a subband echo canceller. In the study, finite word length effects have not been considered.
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| 7. |
- Tao, Xiao-Jiao, et al.
(författare)
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Behind Acoustic Doppler Current Profiler : Narrowband Volume Backscattering Signal Model, Analysis and Estimation
- 1999
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Rapport (övrigt vetenskapligt/konstnärligt)abstract
- This report examines the statistical properties of the narrowband Doppler volume backscattering process and discusses its spectral centroid estimation problem. After clarifying the mechanism of both the finite duration Doppler effect and the continuously space-shifted integration process, the first two order time-varying statistics under a more general assumption, i.e. von Mises distribution, of random phase are derived. The generalization permits nonuniform phase tendency, which occurs in layered medium scattering. Based on the locally stationary process model, the evolutionary spectrum of the signal is derived where the variation of the backscattering strength enters as an amplitude modulation. On the contrary, the variation of the random phase distribution acts as both the amplitude modulation and the frequency modulation. The observability and the estimation of the average flow speed are discussed. It is shown that even under the homogeneous condition, the spectral centroid does not coincide with the aver age flow speed. Finally, a semiparametric spectral centroid estimation method, which simply contains an AR(2) model with its time-varying coefficients adapted by a wavelet shrinkage is proposed.
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