Visible to the public Biblio

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2015
S. R. Islam, S. P. Maity, A. K. Ray.  2015.  "On compressed sensing image reconstruction using linear prediction in adaptive filtering". 2015 International Conference on Advances in Computing, Communications and Informatics (ICACCI). :2317-2323.

Compressed sensing (CS) or compressive sampling deals with reconstruction of signals from limited observations/ measurements far below the Nyquist rate requirement. This is essential in many practical imaging system as sampling at Nyquist rate may not always be possible due to limited storage facility, slow sampling rate or the measurements are extremely expensive e.g. magnetic resonance imaging (MRI). Mathematically, CS addresses the problem for finding out the root of an unknown distribution comprises of unknown as well as known observations. Robbins-Monro (RM) stochastic approximation, a non-parametric approach, is explored here as a solution to CS reconstruction problem. A distance based linear prediction using the observed measurements is done to obtain the unobserved samples followed by random noise addition to act as residual (prediction error). A spatial domain adaptive Wiener filter is then used to diminish the noise and to reveal the new features from the degraded observations. Extensive simulation results highlight the relative performance gain over the existing work.

2018
Sun, Xuguang, Zhou, Yi, Shu, Xiaofeng.  2018.  Multi-Channel Linear Prediction Speech Dereverberation Algorithm Based on QR-RLS Adaptive Filter. Proceedings of the 3rd International Conference on Multimedia Systems and Signal Processing. :109–113.

This paper proposes a multi-channel linear prediction (MCLP) speech dereverberation algorithm based on QR-decomposition recursive least squares (QR-RLS) adaptive filter, which can avoid the possible instability caused by the RLS algorithm, and achieve same speech dereverberation performance as the prototype MCLP dereverberation algorithm based on RLS. This can be confirmed by the theoretical derivation and experiments. Thus, the proposed algorithm can be a good alternative for practical speech applications.

Shen, Sheng, Roy, Nirupam, Guan, Junfeng, Hassanieh, Haitham, Choudhury, Romit Roy.  2018.  MUTE: Bringing IoT to Noise Cancellation. Proceedings of the 2018 Conference of the ACM Special Interest Group on Data Communication. :282–296.

Active Noise Cancellation (ANC) is a classical area where noise in the environment is canceled by producing anti-noise signals near the human ears (e.g., in Bose's noise cancellation headphones). This paper brings IoT to active noise cancellation by combining wireless communication with acoustics. The core idea is to place an IoT device in the environment that listens to ambient sounds and forwards the sound over its wireless radio. Since wireless signals travel much faster than sound, our ear-device receives the sound in advance of its actual arrival. This serves as a glimpse into the future, that we call lookahead, and proves crucial for real-time noise cancellation, especially for unpredictable, wide-band sounds like music and speech. Using custom IoT hardware, as well as lookahead-aware cancellation algorithms, we demonstrate MUTE, a fully functional noise cancellation prototype that outperforms Bose's latest ANC headphone. Importantly, our design does not need to block the ear - the ear canal remains open, making it comfortable (and healthier) for continuous use.

Martinek, Radek, Kahankova, Radana, Bilik, Petr, Nedoma, Jan, Fajkus, Marcel, Blaha, Petr.  2018.  Speech Quality Assessment Based on Virtual Instrumentation. Proceedings of the 10th International Conference on Computer Modeling and Simulation. :49–53.

This paper introduces a program for objective and subjective evaluation of speech quality. Using this environment, a lot of speech recordings and various indoor and outdoor noises were processed. As a subjective speech evaluation method, the Dynamic time warping (DTW) method was selected, with PARCOR coefficients being chosen as symptom vectors. For the filtration of the noise in the recording, adaptive filtering based on LMS and RLS algorithms was used and the performance of the adaptive filtering was assessed. Similarity ranged from 70% to 95% for both algorithms. In terms of signal to noise ratio, the RLS algorithm ranged from 36 dB to 42 dB, while the LMS algorithm only varied from 20 dB to 29 dB.

2020
Kulikov, G. V., Tien, D. T., Kulagin, V. P..  2020.  Adaptive filtering of non-fluctuation interference when receiving signals with multi-position phase shift keying. 2020 Moscow Workshop on Electronic and Networking Technologies (MWENT). :1—4.

{The paper considers the efficiency of an adaptive non-recursive filter using the adjustment algorithm for weighting coefficients taking into account the constant envelope of the desired signal when receiving signals with multi-position phase shift keying against the background of noise and non-fluctuation interference. Two types of such interference are considered - harmonic and retranslated. The optimal filter parameters (adaptation coefficient and length) are determined by using simulation; the effect of the filter on the noise immunity of a quadrature coherent signal receiver with multi-position phase shift keying for different combinations of interference and their intensity is estimated. It is shown that such an adaptive filter can successfully deal with the most dangerous sighting harmonic interference}.