Dissertations / Theses on the topic 'Acoustic equalization'
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1
Zhang, Wancheng. "Robust equalization of multichannel acoustic systems." Thesis, Imperial College London, 2010. http://hdl.handle.net/10044/1/5882.
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In most real-world acoustical scenarios, speech signals captured by distant microphones from a source are reverberated due to multipath propagation, and the reverberation may impair speech intelligibility. Speech dereverberation can be achieved by equalizing the channels from the source to microphones. Equalization systems can be computed using estimates of multichannel acoustic impulse responses. However, the estimates obtained from system identification always include errors; the fact that an equalization system is able to equalize the estimated multichannel acoustic system does not mean that it is able to equalize the true system. The objective of this thesis is to propose and investigate robust equalization methods for multichannel acoustic systems in the presence of system identification errors. Equalization systems can be computed using the multiple-input/output inverse theorem or multichannel least-squares method. However, equalization systems obtained from these methods are very sensitive to system identification errors. A study of the multichannel least-squares method with respect to two classes of characteristic channel zeros is conducted. Accordingly, a relaxed multichannel least- squares method is proposed. Channel shortening in connection with the multiple- input/output inverse theorem and the relaxed multichannel least-squares method is discussed. Two algorithms taking into account the system identification errors are developed. Firstly, an optimally-stopped weighted conjugate gradient algorithm is proposed. A conjugate gradient iterative method is employed to compute the equalization system. The iteration process is stopped optimally with respect to system identification errors. Secondly, a system-identification-error-robust equalization method exploring the use of error models is presented, which incorporates system identification error models in the weighted multichannel least-squares formulation.
2
Yellepeddi, Atulya. "Direct-form adaptive equalization for underwater acoustic communication." Thesis, Massachusetts Institute of Technology, 2012. http://hdl.handle.net/1912/5281.
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Thesis (S.M.)--Joint Program in Applied Ocean Science and Engineering (Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science; and the Woods Hole Oceanographic Institution), 2012.Cataloged from PDF version of thesis.
Includes bibliographical references (p. 139-143).
Adaptive equalization is an important aspect of communication systems in various environments. It is particularly important in underwater acoustic communication systems, as the channel has a long delay spread and is subject to the effects of time- varying multipath fading and Doppler spreading. The design of the adaptation algorithm has a profound influence on the performance of the system. In this thesis, we explore this aspect of the system. The emphasis of the work presented is on applying concepts from inference and decision theory and information theory to provide an approach to deriving and analyzing adaptation algorithms. Limited work has been done so far on rigorously devising adaptation algorithms to suit a particular situation, and the aim of this thesis is to concretize such efforts and possibly to provide a mathematical basis for expanding it to other applications. We derive an algorithm for the adaptation of the coefficients of an equalizer when the receiver has limited or no information about the transmitted symbols, which we term the Soft-Decision Directed Recursive Least Squares algorithm. We will demonstrate connections between the Expectation-Maximization (EM) algorithm and the Recursive Least Squares algorithm, and show how to derive a computationally efficient, purely recursive algorithm from the optimal EM algorithm. Then, we use our understanding of Markov processes to analyze the performance of the RLS algorithm in hard-decision directed mode, as well as of the Soft-Decision Directed RLS algorithm. We demonstrate scenarios in which the adaptation procedures fail catastrophically, and discuss why this happens. The lessons from the analysis guide us on the choice of models for the adaptation procedure. We then demonstrate how to use the algorithm derived in a practical system for underwater communication using turbo equalization. As the algorithm naturally incorporates soft information into the adaptation process, it becomes easy to fit it into a turbo equalization framework. We thus provide an instance of how to use the information of a turbo equalizer in an adaptation procedure, which has not been very well explored in the past. Experimental data is used to prove the value of the algorithm in a practical context.
by Atulya Yellepeddi.
S.M.
3
Talantzis, Fotios. "Equalization and source separation techniques in acoustic reverberant environments." Thesis, Imperial College London, 2006. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.428487.
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Sifferlen, James F. "Iterative equalization and decoding applied to underwater acoustic communication." Diss., Connect to a 24 p. preview or request complete full text in PDF format. Access restricted to UC campuses, 2008. http://wwwlib.umi.com/cr/ucsd/fullcit?p3331419.
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Thesis (Ph. D.)--University of California, San Diego, 2008.Title from first page of PDF file (viewed Dec. 16, 2008). Available via ProQuest Digital Dissertations. Vita. Includes bibliographical references (p. 131-134).
5
Allander, Martin. "Channel Equalization Using Machine Learning for Underwater Acoustic Communications." Thesis, Linköpings universitet, Kommunikationssystem, 2020. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-166643.
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Wireless underwater acoustic (UWA) communications is a developing field with various applications. The underwater acoustic communication channel is very special and its behavior is environment-dependent. The UWA channel is characterized by low available bandwidth, and severe motion-introduced Doppler effectcompared to wireless radio communication. Recent literature suggests that machine learning (ML)-based channel estimation and equalization offer benefits overtraditional techniques (a decision feedback equalizer), in UWA communications. ML can be advantageous due to the difficultly in designing algorithms for UWA communication, as finding general channel models have proven to be difficult. This study aims to explore if ML-based channel estimation and equalization as a part of a sophisticated physical layer structure can offer improved performance. In the study, supervised ML using a deep neural network and a recurrent neural network will be utilized to improve the bit error rate. A channel simulator with environment-specific input is used to study a wide range of channels. The simulations are utilized to study in which environments ML should be tested. It is shown that in highly time-varying channels, ML outperforms traditional techniques if trained with prior information of the channel. However, utilizing ML without prior information of the channel yielded no improvement of the performance.
6
Kuchler, Ryan J. "Comparison of channel equalization filtering techniquies in underwater acoustic communications." Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2002. http://library.nps.navy.mil/uhtbin/hyperion-image/02Jun%5FKuchler.pdf.
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Puikkonen, Panu Tapani. "Development of an Adaptive Equalization Algorithm Using Acoustic Energy Density." BYU ScholarsArchive, 2009. https://scholarsarchive.byu.edu/etd/1686.
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Sound pressure equalization of audio signals using digital signal processors has been a subject of ongoing study for many years. The traditional approach is to equalize sound at a point in a listening environment, but because of its specific dependence on the room frequency response between a source and receiver position, this equalization generally causes the spectral response to worsen significantly at other locations in the room. This work presents both a time-invariant and a time-varying implementation of an adaptive acoustic energy density equalization filter for a one-dimensional sound field. Energy density equalization addresses the aforementioned challenge and others that relate to sound equalization. The theory and real-time implementation of time-invariant sound pressure and energy density equalizers designed using the least-squares method are presented, and their performances are compared. An implementation of a time-varying energy density equalizer is also presented. Time-invariant equalization results based on real-time measurements in a plane-wave tube are presented. A sound pressure equalizer results in a nearly flat spectral magnitude at the point of equalization. However, it causes the frequencies corresponding to spatial nulls at that point to be undesirably boosted elsewhere in the sound field, where those nulls do not exist at the same frequencies. An energy density equalization filter identifies and compensates for all resonances and other global spectral effects of the tube and loudspeaker. It does not attempt to equalize the spatially varying frequency nulls caused by local pressure nodes at the point of equalization. An implementation of a time-varying energy density equalizer is also presented. This method uses the filtered-x filter update to adjust the filter coefficients in real-time to adapt to changes in the sound field. Convergence of the filter over time is demonstrated as the closed end of the tube is opened, then closed once again. Thus, the research results demonstrate that an acoustic energy density filter can be used to time-adaptively equalize global spectral anomalies of a loudspeaker and a one-dimensional sound field.
8
Lewis, Matthew Robert S. M. Massachusetts Institute of Technology. "Evaluation of vector sensors for adaptive equalization in underwater acoustic communication." Thesis, Massachusetts Institute of Technology, 2014. http://hdl.handle.net/1721.1/93793.
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Thesis: S.M., Joint Program in Applied Ocean Science and Engineering (Massachusetts Institute of Technology, Department of Mechanical Engineering; and the Woods Hole Oceanographic Institution), 2014.Cataloged from PDF version of thesis.
Includes bibliographical references (pages 123-125).
Underwater acoustic communication is an extremely complex field that faces many challenges due to the time-varying nature of the ocean environment. Vector sensors are a proven technology that when utilizing their directional sensing capabilities allows us to minimize the effect of interfering noise sources. A traditional pressure sensor array has been the standard for years but suffers at degraded signal to noise ratios (SNR) and requires maneuvers or a lengthly array aperture to direction find. This thesis explores the effect of utilizing a vector sensor array to steer to the direction of signal arrival and the effect it has on equalization of the signal at degraded SNRs. It was demonstrated that utilizing a single vector sensor we were able steer to the direction of arrival and improve the ability of an equalizer to determine the transmitted signal. This improvement was most prominent when the SNR was degraded to levels of 0 and 10 dB where the performance of the vector sensor outperformed that of the pressure sensor in nearly 100% of cases. Finally, this performance improvement occurred with a savings in computational expense.
by Matthew Robert Lewis.
S.M.
9
Kuchler, Ryan J. "Comparison of channel equalization filtering techniques in underwater acoustic communications." Thesis, Monterey California. Naval Postgraduate School, 2002. http://hdl.handle.net/10945/5887.
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In this thesis, underwater acoustic communications signal processing techniques, which are used to equalize the distortional effects associated with the ocean as a communications channel, are investigated for a shallow water ocean environment. The majority of current signal processing techniques employ a Finite Impulse Response (FIR) filter. Three equalization filters were investigated and presented as alternatives; they were the passive time-reversed filter, the inverse filter, and the Infinite Impulse Response (IIR) filter. The main advantage of the passive time-reversed filter and the inverse filter is simplicity of design. Bit error rates for the time-reversed filter were consistently around 10-1 and those for the inverse filter were greater than 10-1. However, inability of the passive time-reversed filter to completely eliminate multipath components and the ill-conditioned nature of the inverse filter made it difficult to achieve Probability of Error results below 10-1. Research into the development of an array receiver using a time-reversed filter should improve calculated bit error rates. Simulations of the IIR filter were conducted with limited success. The main advantage of an IIR filter is that fewer parameters are required in the design of the filter. However, the potential for instability in the filter is a significant limitation. Probability of Error results were found to be on the order of those for current FIR filters at close ranges. Unfortunately, instability issues arose for receivers as range from the source increased. This research on the IIR filter is still in the embryonic stage, whereas research using FIR filters is relatively highly developed. Further research is needed to address the issue of instability in IIR filters in order to make them an effective signal processing technique employable in underwater acoustic communications.
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Puikkonen, Panu. "Development of an adaptive equalization algorithm using acoustic energy density /." Diss., CLICK HERE for online access, 2009. http://contentdm.lib.byu.edu/ETD/image/etd2899.pdf.
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Blair, Ballard J. S. (Ballard Justin Smith). "Analysis of and techniques for adaptive equalization for underwater acoustic communication." Thesis, Massachusetts Institute of Technology, 2011. http://hdl.handle.net/1721.1/68436.
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Thesis (Ph. D.)--Joint Program in Oceanography/Applied Ocean Science and Engineering (Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science; and the Woods Hole Oceanographic Institution), 2011.Cataloged from PDF version of thesis.
Includes bibliographical references (p. 203-215).
Underwater wireless communication is quickly becoming a necessity for applications in ocean science, defense, and homeland security. Acoustics remains the only practical means of accomplishing long-range communication in the ocean. The acoustic communication channel is fraught with difficulties including limited available bandwidth, long delay-spread, time-variability, and Doppler spreading. These difficulties reduce the reliability of the communication system and make high data-rate communication challenging. Adaptive decision feedback equalization is a common method to compensate for distortions introduced by the underwater acoustic channel. Limited work has been done thus far to introduce the physics of the underwater channel into improving and better understanding the operation of a decision feedback equalizer. This thesis examines how to use physical models to improve the reliability and reduce the computational complexity of the decision feedback equalizer. The specific topics covered by this work are: how to handle channel estimation errors for the time varying channel, how to use angular constraints imposed by the environment into an array receiver, what happens when there is a mismatch between the true channel order and the estimated channel order, and why there is a performance difference between the direct adaptation and channel estimation based methods for computing the equalizer coefficients. For each of these topics, algorithms are provided that help create a more robust equalizer with lower computational complexity for the underwater channel.
by Ballard J. S. Blair.
Ph.D.
12
Kodrasi, Ina [Verfasser]. "Dereverberation and Noise Reduction Techniques based on Acoustic Multi-Channel Equalization / Ina Kodrasi." München : Verlag Dr. Hut, 2016. http://d-nb.info/1113336366/34.
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Betlehem, Terence, and terenceb@rsise anu edu au. "Acoustic Signal Processing Algorithms for Reverberant Environments." The Australian National University. Research School of Information Sciences and Engineering, 2005. http://thesis.anu.edu.au./public/adt-ANU20051129.121453.
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This thesis investigates the design and the analysis of acoustic signal processing algorithms in reverberant rooms. Reverberation poses a major challenge to acoustic signal processing problems. It degrades speech intelligibility and causes many acoustic algorithms that process sound to perform poorly. Current solutions to the reverberation problem frequently only work in lightly reverberant environments. There is need to improve the reverberant performance of acoustic algorithms.¶
The approach of this thesis is to explore how the intrinsic properties of reverberation can be exploited to improve acoustic signal processing algorithms. A general approach to soundfield modelling using statistical room acoustics is applied to analyze the reverberant performance of several acoustic algorithms. A model of the underlying structure of reverberation is incorporated to create a new method
of soundfield reproduction.¶
Several outcomes resulting from this approach are: (i) a study of how more sound capture with directional microphones and beamformers can improve the robustness of acoustic equalization, (ii) an assessment of the extent to which source tracking can improve accuracy of source localization, (iii) a new method of soundfield reproduction for reverberant rooms, based upon a parametrization of the acoustic transfer function and (iv) a study of beamforming to directional sources, specifically exploiting the directionality of human speech.¶
The approach to soundfield modelling has permitted a study of algorithm performance on important parameters of the room acoustics and the algorithm design. The performance of acoustic equalization and source tracking have been found to depend not only on the levels of reverberation but also on the correlation of pressure between points in reverberant soundfields. This correlation can be increased by sound capture with directional capture devices. Work on soundfield reproduction has shown that, though reverberation significantly degrades the performance of conventional techniques, by accounting for the reverberation it is possible to design reproduction methods that function well in reverberant environments.
14
Chester, Ryan T. "Error Sensor Strategies for Active Noise Control and Active Acoustic Equalization in a Free Field." Diss., CLICK HERE for online access, 2008. http://contentdm.lib.byu.edu/ETD/image/etd2298.pdf.
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Pelekanos, Georgios N. "Performance of acoustic spread-spectrum signaling in simulated ocean channels." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2003. http://library.nps.navy.mil/uhtbin/hyperion-image/03Jun%5FPelekanos.pdf.
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Thesis (M.S. in Electrical Engineering and M.S. in Engineering Acoustics)--Naval Postgraduate School, June 2003.Thesis advisor(s): Roberto Cristi, Joseph Rice. Includes bibliographical references (p. 107-108). Also available online.
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Korst-Fagundes, Bruno Carleton University Dissertation Engineering Electronics. "Acoustical equalization at multiple listening positions." Ottawa, 1995.
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Nadakuditi, Rajesh Rao. "A channel subspace post-filtering approach to adaptive equalization." Thesis, Massachusetts Institute of Technology, 2002. http://hdl.handle.net/1721.1/87613.
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Thesis (S.M.)--Joint Program in Oceanography/Applied Ocean Science and Engineering (Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science; and the Woods Hole Oceanographic Institution), 2002.Includes bibliographical references (p. 151-154).
by Rajesh Rao Naduditi.
S.M.
18
Dickson, Crispin. "A few aspects of aircraft noise." Licentiate thesis, Stockholm : Teknisk akustik, Kungliga Tekniska högskolan, 2007. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-4510.
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Mi, Jing. "Acoustic source separation based on target equalization-cancellation." Thesis, 2018. https://hdl.handle.net/2144/27449.
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Normal-hearing listeners are good at focusing on the target talker while ignoring the interferers in a multi-talker environment. Therefore, efforts have been devoted to build psychoacoustic models to understand binaural processing in multi-talker environments and to develop bio-inspired source separation algorithms for hearing-assistive devices. This thesis presents a target-Equalization-Cancellation (target-EC) approach to the source separation problem. The idea of the target-EC approach is to use the energy change before and after cancelling the target to estimate a time-frequency (T-F) mask in which each entry estimates the strength of target signal in the original mixture. Once the mask is calculated, it is applied to the original mixture to preserve the target-dominant T-F units and to suppress the interferer-dominant T-F units. On the psychoacoustic modeling side, when the output of the target-EC approach is evaluated with the Coherence-based Speech Intelligibility Index (CSII), the predicted binaural advantage closely matches the pattern of the measured data. On the application side, the performance of the target-EC source separation algorithm was evaluated by psychoacoustic measurements using both a closed-set speech corpus and an open-set speech corpus, and it was shown that the target-EC cue is a better cue for source separation than the interaural difference cues.
20
Shih-WeiChuang and 莊士緯. "Novel low complexity and flexible psycho-acoustic model design for efficient audio equalization system." Thesis, 2015. http://ndltd.ncl.edu.tw/handle/vk87zj.
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碩士國立成功大學
電機工程學系
103
This paper presents a novel low complexity and flexible psychoacoustic model with proposed audiogram based efficient audio equalization system. Our proposed structure have the following characteristics: 1) With psychoacoustic model, we can reduce the processing points about 70% every frame according to the masking effect in our ears. Besides, based on different purpose, we keep only the necessary steps in psychoacoustic model which reducing 30%-40% operations without affecting the sound quality compared with original model in MPEG standard. Moreover, the flexible design with band-selectable strategy in our model lightening the burden on our system when we don’t need to calculate full-band model; 2) The equalization algorithm based on audiogram considering the hearing loss in normal people, we adjust the gain in frequency domain with the information given by audiogram making the equalization better and adaptive; 3) With the masking effect in psychoacoustic, the redundant signals were discarded, leading to the lower output power every frame. Compared with other audio equalizer with filter bank design, our proposed structure has the lowest computation complexity, lowest output power. Nevertheless, considering the property in human ear, our proposed is more suitable for normal using. Based on the above, the proposed structure would be a new solution for future application in audio processing.
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Chow, Y. F., and 周永富. "Acoustical Spatial Equalization." Thesis, 1994. http://ndltd.ncl.edu.tw/handle/77119023421739345465.
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碩士國立交通大學
電信研究所
82
This thesis describes a general theoretical basis for the design of multichannel acoustical spatial equalizers to provide a good acoustical sound field. We aim to equalize both the response of the loudspeakers and the listening room as well as to cancel the acoustic crosstalk. The work presented is applied first to the single-channel, two-channel and then extended to the multi-channel case. Several methods to obtain the solution of the equalization system are provided. Extension of acoustical equalization over a listening area is also discussed. Because the room impulse response varies with location, the issue of mismatched equalization based on matrix perturbation theory is addressed. Multi-channel equalization using adaptive filters offers an attractive solution to the reduction of computation and complexity, we will use the LMS approach to design the digital filters. Finally, computer simulations will justify the algorithms for acoustical spatial equalization.
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Matheson, Ryan. "An Investigation and Application of the Finite Difference Time Domain Method as a Tool for Solving Equalization Problems in Acoustics." Thesis, 2010. http://hdl.handle.net/10012/5001.
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This thesis investigates the issues in deriving the Finite Di erence Time Domain
Method, including the derivation of a unique method for exciting an FDTD system
that is physically realistic in terms of acoustics. It is also the goal of this thesis to use
the FDTD method as a tool for investigating various speaker placement con gurations
for use in bass equalization. A demerit function is then developed in order to assess
how well a particular equalization method performs relative to any others.
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Χατζηαντωνίου, Παναγιώτης. "Ανάπτυξη μεθόδων ψηφιακής ισοστάθμισης για ηλεκτρακουστικές εφαρμογές." 2005. http://nemertes.lis.upatras.gr/jspui/handle/10889/307.
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H Διδακτορική Διατριβή μελετά το πρόβλημα της ψηφιακής ισοστάθμισης,σκοπεύοντας στην ανάπτυξη αποτελεσματικών μεθόδων εξάλειψης των ηχητικών παραμορφώσεων, που εισάγονται κατά την ηχητική αναπαραγωγή εξαιτίας της απόκρισης, είτε των ηχείων (ανηχωική ισοστάθμιση), είτε των χώρων ακρόασης (εξάλειψη αντήχησης). Αναπτύσσονται πρωτότυπες μέθοδοι που αφενός εξασφαλίζουν ακριβείς μετρήσεις των ανηχωικών ηλεκτρακουστικών αποκρίσεων μέσα σε μη ανηχωικούς χώρους, αφετέρου πετυχαίνουν κατάλληλη εξομάλυνση των πολύπλοκων αποκρίσεων των ακουστικών συστημάτων για χρήση στην ψηφιακή ισοστάθμιση αλλά και για χρήση σε άλλες εφαρμογές της ακουστικής χώρων που απαιτούν ανάλυση συγκεκριμένων ιδιοτήτων αυτών των συστημάτων. Η συστηματική μελέτη της μεθόδου εξάλειψης αντήχησης που βασίζεται στην ιδανική αντιστροφή των αποκρίσεων χώρων οδηγεί στο πρωτότυπο συμπέρασμα ότι τα ακουστά οφέλη από την εφαρμογή της μεθόδου σε πραγματικό χρόνο είναι σημαντικά υποδεέστερα από τα αναμενόμενα που προκύπτουν από τα αντίστοιχα πειράματα εξομοίωσης αυτής της μεθόδου. Το πρόβλημα της εξάλειψης αντήχησης αντιμετωπίζεται για πρώτη φορά με έναν πρακτικά βιώσιμο τρόπο, με την εισαγωγή πρωτότυπης μεθόδου ισοστάθμισης που βασίζεται στην Μιγαδική Εξομάλυνση των αποκρίσεων χώρων.The dissertation studies the digital audio equalization problem, in order to develop methods that would effectively eliminate the audio distortions being introduced during the sound reproduction by either the loudspeakers(anechoic equalization) or the room response (dereverberation). Novel methods are introduced that ensure precise measurements of anechoic electracoustic responses inside reverberant enclosures and on the other hand, achieve appropriately smoothed acoustic responses, for use in digital equalization and also in other applications of room acoustics that require analysis of concrete properties of these systems. Novel conclusions have been drawn by the analytic study of the room acoustics dereverberation based on ideal inverse filtering, indicating that the application of such a method in real time yields a significantly degraded performance compared to that achieved by the corresponding simulated dereverberation experiments. The problem of dereverberation is faced with a practically viable solution, with the introduction of a novel method based on the room response Complex Smoothing.