Dissertations / Theses on the topic 'Noisy environments'
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Aschauer, Hans. "Quantum communication in noisy environments." Diss., lmu, 2005. http://nbn-resolving.de/urn:nbn:de:bvb:19-35882.
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Jaisimha, M. Y. "Compound document retrieval in noisy environments /." Thesis, Connect to this title online; UW restricted, 1996. http://hdl.handle.net/1773/6007.
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Parikh, Devangi Nikunj. "Improving the quality of speech in noisy environments." Diss., Georgia Institute of Technology, 2012. http://hdl.handle.net/1853/45889.
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In this thesis, we are interested in processing noisy speech signals that are meant to be heard by humans, and hence we approach the noise-suppression problem from a perceptual perspective. We develop a noise-suppression paradigm that is based on a model of the human auditory system, where we process signals in a way that is natural to the human ear. Under this paradigm, we transform an audio signal in to a perceptual domain, and processes the signal in this perceptual domain. This approach allows us to reduce the background noise and the audible artifacts that are seen in traditional noise-suppression algorithms, while preserving the quality of the processed speech. We develop a single- and dual-microphone algorithm based on this perceptual paradigm, and conduct subjecting tests to show that this approach outperforms traditional noise-suppression techniques. Moreover, we investigate the cause of audible artifacts that are generated as a result of suppressing the noise in noisy signals, and introduce constraints on the noise-suppression gain such that these artifacts are reduced.
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Nayfeh, Taysir H. "Multi-signal processing for voice recognition in noisy environments." Thesis, This resource online, 1991. http://scholar.lib.vt.edu/theses/available/etd-10222009-125021/.
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Chen, Aimin. "Speech recognition and enhancement in noisy cellular mobile environments." Thesis, Brunel University, 2002. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.251198.
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Laidler, Jonathan. "Modelling of glimpses for speech recognition in noisy environments." Thesis, University of Sheffield, 2012. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.575364.
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Noisy environments pose significant problems to automatic speech recognition (ASR) systems. A common scenario is the cocktail party problem, where there are competing speakers. Human listeners perform well in these situations despite the fact that the target and the noise share similar characteristics. However, traditional ASR systems struggle to deal with non-stationary noise. The glimpsing theory of speech perception states that human listeners are able to focus their attention on spectre-temporal glimpses where the target speech is not masked by noise. Glimpses of clean speech are highly available when the noise source is another speaker, due to the sparse nature of spectra-temporal representations of speech. ASR systems which aim to model the behaviour of human listeners should also take advantage of glimpses. Existing studies have detected glimpses based on features such as pitch, which is known to be valuable for separating competing talkers. This thesis takes the opposite approach, using no prior knowledge of speech features but rather learning the features of glimpses from samples of clean speech. This is considered to be a model-driven approach, in contrast to previous source-driven approaches. This thesis draws inspiration from computational vision, where the analogous problem is that of partial object recognition. The proposed glimpse detection system identifies spectre-temporal interest points which are small patches of speech, then forms glimpses from connected regions of interest points. In addition to a detailed description of the novel ASR framework, the thesis presents three new investigations. The first discovers what size of spectra-temporal speech patch can be recognised by human listeners. The second investigates what kind of encoding should be applied to patches in order to best capture the features of clean speech. The third takes grouping algorithms that are popular in vision research and compares their success in creating glimpses from speech interest points. Finally the full end-to-end ASR system is evaluated on a speech separation task.
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Jimenez, Blazquez Lara. "Mathematical Methods for Maritime Signal Curation in Noisy Environments." Thesis, Mälardalens högskola, Akademin för utbildning, kultur och kommunikation, 2019. http://urn.kb.se/resolve?urn=urn:nbn:se:mdh:diva-43653.
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QTAGG has designed a real-time autonomous system that continuously calculates an optimum propulsion plan controlling the engines and propellers of a vessel. In this way, the precision of the signals that are used is very important, as any little error in the signal can produce incorrect control effects and cause critical damages to the equipment or passengers. This thesis describes the mathematics and implementation of a system to detect and correct disturbances in the data signals of a vessel. The system applies a signal curation based on mathematical modelling and statistics leading to clean data to use in QTAGG’s control system.
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Mahfoudia, Osama. "DVB-T based bistatic passive radars in noisy environments." Doctoral thesis, Universite Libre de Bruxelles, 2017. https://dipot.ulb.ac.be/dspace/bitstream/2013/258499/5/contratOM.pdf.
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Passive coherent location (PCL) radars employ illuminators of opportunity to detect and track targets. This silent operating mode provides many advantages such as low cost and interception immunity. Many radiation sources have been exploited as illumination sources such as broadcasting and telecommunication transmitters. The classical architecture of the bistatic PCL radars involves two receiving channels: a reference channel and a surveillance channel. The reference channel captures the direct-path signal from the transmitter, and the surveillancesignal collects the possible target echoes. The two major challenges for the PCL radars are the reference signal noise and the surveillance signal static clutter. A noisy reference signal degrades the detection probability by increasing the noise-floor level of the detection filter output. And the static clutter presence in the surveillance signal reduces the detector dynamic range and buries low magnitude echoes.In this thesis, we consider a PCL radar based on the digital video broadcasting-terrestrial (DVB-T) signals, and we propose a set of improved methods to deal with the reference signal noise and the static clutter in the surveillance signal. The DVB-T signals constitute an excellentcandidate as an illumination source for PCL radars; they are characterized by a wide bandwidth and a high radiated power. In addition, they provide the possibility of reconstructing the reference signal to enhance its quality, and they allow a straightforward static clutter suppressionin the frequency domain. This thesis proposes an optimum method for the reference signal reconstruction and an improved method for the static clutter suppression.The optimum reference signal reconstruction minimizes the mean square error between the reconstructed signal and the exact one. And the improved static clutter suppression method exploits the possibility of estimating the propagation channel. These two methods extend thefeasibility of a single receiver PCL radar, where the reference signal is extracted from the direct-path signal present in the surveillance signal.Doctorat en Sciences de l'ingénieur et technologie
info:eu-repo/semantics/nonPublished
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Gajic, Bojana. "Feature Extraction for Automatic Speech Recognition in Noisy Acoustic Environments." Doctoral thesis, Norwegian University of Science and Technology, Faculty of Information Technology, Mathematics and Electrical Engineering, 2002. http://urn.kb.se/resolve?urn=urn:nbn:no:ntnu:diva-441.
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This thesis presents a study of alternative speech feature extraction methods aimed at increasing robustness of automatic speech recognition (ASR) against additive background noise.
Spectral peak positions of speech signals remain practically unchanged in presence of additive background noise. Thus, it was expected that emphasizing spectral peak positions in speech feature extraction would result in improved noise robustness of ASR systems. If frequency subbands are properly chosen, dominant subband frequencies can serve as reasonable estimates of spectral peak positions. Thus, different methods for incorporating dominant subband frequencies into speech feature vectors were investigated in this study.
To begin with, two earlier proposed feature extraction methods that utilize dominant subband frequency information were examined. The first one uses zero-crossing statistics of the subband signals to estimate dominant subband frequencies, while the second one uses subband spectral centroids. The methods were compared with the standard MFCC feature extraction method on two different recognition tasks in various background conditions. The first method was shown to improve ASR performance on both recognition tasks at sufficiently high noise levels. The improvement was, however, smaller on the more complex recognition task. The second method, on the other hand, led to some reduction in ASR performance in all testing conditions.
Next, a new method for incorporating subband spectral centroids into speech feature vectors was proposed, and was shown to be considerably more robust than the standard MFCC method on both ASR tasks. The main difference between the proposed method and the zero-crossing based method is in the way they utilize dominant subband frequency information. It was shown that the performance improvement due to the use of dominant subband frequency information was considerably larger for the proposed method than for the ZCPA method, especially on the more complex recognition task. Finally, the computational complexity of the proposed method is two orders of magnitude lower than that of the zero-crossing based method, and of the same order of magnitude as the standard MFCC method.
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Wu, Mingyang. "Pitch tracking and speech enhancement in noisy and reverberant environments." Columbus, Ohio : Ohio State University, 2003. http://rave.ohiolink.edu/etdc/view?acc%5Fnum=osu1064341479.
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Thesis (Ph. D.)--Ohio State University, 2003.Title from first page of PDF file. Document formatted into pages; contains xvi, 149 p.; also includes graphics. Includes abstract and vita. Advisor: DeLiang Wang, Dept. of Computer and Information Science. Includes bibliographical references (p. 136-149).
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Bettadapura, Raghuprasad Shivatejas. "Echo Delay Estimation to Aid Source Localization in Noisy Environments." Thesis, Virginia Tech, 2014. http://hdl.handle.net/10919/50517.
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Time-delay estimation (TDE) finds application in a variety of problems, be it locating
fractures or steering cameras towards the speaker in a multi-participant conference application. Underwater acoustic OFDM source localization is another important application of TDE. Existing underwater acoustic source localization techniques use a microphone array consisting of three or four sensors in order to effectively locate the source. Analog-to-digital (ADC) converters at these sensors call for a non-nominal investment in terms of circuitry and memory. A relatively inexpensive source localization algorithm is needed that works with the output of a single sensor. Since an inexpensive process for estimating the location of the source is desired, the ADC used at the sensor is capable only of a relatively low sampling rate. For a given delay, a low sampling rate leads to sub-sample interval delays, which the desired algorithm must be able to estimate. Prevailing TDE algorithms make some a priori assumptions about the nature of the received signal, such as Gaussianity, wide-sense stationarity, or periodicity. The desired algorithm must not be restrictive in so far as the nature of the transmitted signal is concerned.
A time-delay estimation algorithm based on the time-frequency ratio of mixtures (TFRM) method is proposed. The experimental set-up consists of two microphones/sensors placed at some distances from the source. The method accepts as input the received signal which consists of the sum of the signal received at the nearer sensor and the signal received at the farther sensor and noise. The TFRM algorithm works in the time-frequency domain and seeks to perform successive source cancellation in the received burst. The key to performing source cancellation is to estimate the ratio in which the sources combine and this ratio is estimated by means of taking a windowed mean of the ratio of the spectrograms of any two pulses in the received burst. The variance of the mean function helps identify single-source regions and regions in which the sources mix.
The performance of the TFRM algorithm is evaluated in the presence of noise and is compared against the Cramer-Rao lower bound. It is found that the variance of the estimates returned by the estimator diverge from the predictions of the Cramer-Rao inequality as the farther sensor is moved farther away. Conversely, the estimator becomes more reliable as the farther sensor is moved closer.
The time-delay estimates obtained from the TFRM algorithm are used for source localization. The problem of finding the source reduces to finding the locus of points such that the difference of its distances to the two sensors equals the time delay. By moving the pair of sensors to a different location, or having a second time delay sensor, an exact location for the source can be determined by finding the point of intersection of the two loci.
The TFRM method does not rely on a priori information about the signal. It is
applicable to OFDM sources as well as sinusoidal and chirp sources.Master of Science
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Wiltgen, Timothy Edward. "Adaptive Beamforming using ICA for Target Identification in Noisy Environments." Thesis, Virginia Tech, 2007. http://hdl.handle.net/10919/33118.
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The blind source separation problem has received a great deal of attention in previous years. The aim of this problem is to estimate a set of original source signals from a set of linearly mixed signals through any number of signal processing techniques. While many methods exist that attempt to solve the blind source separation problem, a new technique is being used that uniquely separates audio sources as they are received from a microphone array. In this thesis a new algorithm is proposed that that utilizes the ICA algorithm in conjunction with a filtering technique that separates source signals and then removes sources of interference so that a signal of interest can be accurately tracked. Experimental results will compare a common blind source separation technique to the new algorithm and show that the new algorithm can detect a signal of interest and accurately track it as it moves through an anechoic environment.Master of Science
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Rådsten-Ekman, Maria. "MAY NOISY SOUND ENVIRONMENTS BE IMPROVED BY ADDING PLEASANT WATER SOUNDS?" Thesis, Stockholms universitet, Psykologiska institutionen, 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:su:diva-43834.
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Munlin, Joyce C. "The effect of three variables on synthetic speech intelligibility in noisy environments." Thesis, Monterey, California. Naval Postgraduate School, 1990. http://hdl.handle.net/10945/30702.
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Approved for public release, distribution is unlimitedMilitary Command and Control (C2) requires easy access to information needed for the commander's situation assessment and direction of troops. Providing this information via synthetic speech is a viable alternative, but additional information is required before speech systems can be implemented for C2 functions. An experiment was conducted to study several factors which may affect the intelligibility of synthetic speech. The factors examined were: (1) speech rate; (2) synthetic speech messages presented at lower, the same, and higher frequencies than background noise frequency; (3) voice richness; and (4) interactions between speech rate, voice fundamental frequency, and voice richness. Response latency and recognition accuracy were measured. Results clearly indicate that increasing speech rate leads to an increase latency and a decrease in recognition accuracy, at least for the novice user. No effect of voice fundamental frequency or richness was demonstrated.
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Chuang, Chern. "Spectroscopy, relaxation, and transport of molecular excitons in noisy and disordered environments." Thesis, Massachusetts Institute of Technology, 2018. http://hdl.handle.net/1721.1/115803.
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Thesis: Ph. D., Massachusetts Institute of Technology, Department of Chemistry, 2018.Cataloged from PDF version of thesis.
Includes bibliographical references (pages 139-150).
In this thesis contribution we theoretically investigate the spectroscopy, relaxation, and transport properties of Frenkel excitons in molecular aggregates, with extensive comparison to or prediction of experimental observables. Particular emphasis is devoted to the effects of thermal noise, static disorder, and system dimensionality. Our key contributions are summarized as the following. We study the spectroscopic signatures of excitonic molecular aggregates of dimensionality larger than unity as functions of temperature and disorder strength. These findings are applied to the determination of essential system characteristics and quantitatively explain the spectroscopic traits seen in experiments where either the temperature or disorder strength is altered. A classification scheme generalized from Kasha's seminal work on J- and H-aggregates is proposed that is compatible with experimental observations previously unexplained. We recognize the importance of long-wavelength approximations in understanding the density of states in two-dimensional excitonic aggregates. And for tubular aggregates this leads to a simple expression for the energy gap between the parallel- and the perpendicular-polarized peaks useful in inferring key system parameters. This long-wavelength approach is then extended to the analysis of 2D excitonic molecular aggregates in general. A universal scaling relation concerning the steady-state diffusive transport of excitons in molecular tubes is predicted and analyzed, where the key order parameter is identified as the ratio between the localization length of the exciton wavefunctions and the tube circumference. A unified theoretical framework is proposed to explain the relaxation of hot excitons generated in emissive conjugated polymers across three orders of magnitude in timescale, with quantitative agreements with experiments.
by Chern Chuang.
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健紘, 大田, and Kenko Ota. "Studies in signal processing for robust speech recognition in noisy and reverberant environments." Thesis, https://doors.doshisha.ac.jp/opac/opac_link/bibid/BB10268908/?lang=0, 2008. https://doors.doshisha.ac.jp/opac/opac_link/bibid/BB10268908/?lang=0.
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Novoa, Ilic José Eduardo. "Robust speech recognition in noisy and reverberant environments using deep neural network-based systems." Tesis, Universidad de Chile, 2018. http://repositorio.uchile.cl/handle/2250/168062.
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Doctor en Ingeniería EléctricaIn this thesis an uncertainty weighting scheme for deep neural network-hidden Markov model (DNN-HMM) based automatic speech recognition (ASR) is proposed to increase discriminability in the decoding process. To this end, the DNN pseudo-log-likelihoods are weighted according to the uncertainty variance assigned to the acoustic observation. The results presented here suggest that substantial reduction in word error rate (WER) is achieved with clean training. Moreover, modelling the uncertainty propagation through the DNN is not required and no approximations for non linear activation functions are made. The presented method can be applied to any network topology that delivers log likelihood-like scores. It can be combined with any noise removal technique and adds a minimal computational cost. This technique was exhaustively evaluated and combined with uncertainty-propagation-based schemes for computing the pseudo-log-likelihoods and uncertainty variance at the DNN output. Two proposed methods optimized the parameters of the weighting function by leveraging the grid search either on a development database representing the given task or on each utterance based on discrimination metrics. Experiments with Aurora-4 task showed that, with clean training, the proposed weighting scheme can reduce WER by a maximum of 21% compared with a baseline system with spectral subtraction and uncertainty propagation using the unscented transform. Additionally, it is proposed to replace the classical black box integration of automatic speech recognition technology in human-robot interaction (HRI) applications with the incorporation of the HRI environment representation and modeling, and the robot and user states and contexts. Accordingly, this thesis focuses on the environment representation and modeling by training a DNN-HMM based automatic speech recognition engine combining clean utterances with the acoustic channel responses and noise that were obtained from an HRI testbed built with a PR2 mobile manipulation robot. This method avoids recording a training database in all the possible acoustic environments given an HRI scenario. In the generated testbed, the resulting ASR engine provided a WER that is at least 26% and 38% lower than publicly available speech recognition application programming interfaces (APIs) with the loudspeaker and human speakers testing databases, respectively, with a limited amount of training data. This thesis demonstrates that even state-of-the-art DNN-HMM based speech recognizers can benefit by combining systems for which the acoustic models have been trained using different feature sets. In this context, the complementarity of DNN-HMM based ASR systems trained with the same data set but with different signal representations is discussed. DNN fusion methods based on flat-weight combination, the minimization of mutual information and the maximization of discrimination metrics were proposed and tested. Schemes that consider the combination of ASR systems with lattice combination and minimum Bayes risk decoding were also evaluated and combined with DNN fusion techniques. The experimental results were obtained using a publicly-available naturally-recorded highly reverberant speech data. Significant improvements in WER were observed by combining DNN-HMM based ASR systems with different feature sets, obtaining relative improvements of 10% with two classifiers and 18% with four classifiers, without any tuning or a priori information of the ASR accuracy.
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Goeckel, Tom [Verfasser], Gerhard [Akademischer Betreuer] Lakemeyer, and Hermann [Akademischer Betreuer] Wagner. "Efficient Binaural Sound Localization in Noisy and Reverberant Environments / Tom Goeckel ; Gerhard Lakemeyer, Hermann Wagner." Aachen : Universitätsbibliothek der RWTH Aachen, 2015. http://d-nb.info/1130402738/34.
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Kuhr, Mark Gregory Hamilton John A. "An adaptive jam-resistant cross-layer protocol for mobile ad-hoc networks In noisy environments." Auburn, Ala., 2009. http://hdl.handle.net/10415/1611.
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Roberts, David James. "Applications of Artificial Neural Networks to Synthetic Aperture Radar for Feature Extraction in Noisy Environments." DigitalCommons@CalPoly, 2013. https://digitalcommons.calpoly.edu/theses/996.
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It is often that images generated from Synthetic Aperture Radar (SAR) are noisy, distorted, or incomplete pictures of a target or target region. As the goal for most SAR research pertains to automatic target recognition (ATR), extensive filtering and image processing is required in order to extract the features necessary to carry out ATR. This thesis investigates the use of Artificial Neural Networks (ANNs) in order to improve upon the feature extraction process by laying the foundation for ANN SAR ATR algorithms and programs. The first technique investigated is that of an ANN edge detector designed to be invariant to multiplicative speckle noise. The algorithm designed uses the Back Propagation (BP) algorithm to teach a multi-layer perceptron network to detect edges. In order to do so, several parameters within a Sliding Window (SW), are calculated as the inputs to the ANN. The ANN then outputs an edge map that includes the outer edge features of the target as well as some internal edge features. The next technique that is examined is a pattern recognition and target reconstruction algorithm based off of the associative memory ANN known as the Hopfield Network (HN). For this version of the HN, the network is trained with a collection of varying geometric shapes. The output of the network is a nearest-fit representation of the incomplete image data input. Because of the versatility of this program, it is also able to reconstruct incomplete 3D models determined from SAR data. The final technique investigated is an automatic rotation procedure to detect the change in perspective relative to the platform. This type of detection can prove useful if used for target tracking or 3D modeling where the direction vector or relative angle of the target is a desired piece of information.
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Gstach, Dieter. "Small sample performance of two approaches to technical efficiency estimation in noisy multiple output environments." Inst. für Volkswirtschaftstheorie und -politik, WU Vienna University of Economics and Business, 1998. http://epub.wu.ac.at/1190/1/document.pdf.
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This paper provides simulation evidence concerning some statistical properties of two different approaches to technical efficiency estimation for multiple-output production under noisy conditions: The Ray Frontier Approach (RFA) from Löthgren (1997) DEA+ proposed in Gstach (1996). RFA, unlike earlier approaches in the realm of stochastic frontier analysis, is capable of efficiency estimation in the case of multiple outputs as well and lends itself for comparison with DEA+. Several settings with varying sample sizes, noise to signal ratios and mean inefficiencies are investigated. (author's abstract)Series: Department of Economics Working Paper Series
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NAKAMURA, Satoshi, Kazuya TAKEDA, and Masakiyo FUJIMOTO. "CENSREC-3: An Evaluation Framework for Japanese Speech Recognition in Real Car-Driving Environments." Institute of Electronics, Information and Communication Engineers, 2006. http://hdl.handle.net/2237/15050.
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Braun, Sebastian [Verfasser], Emanuel [Gutachter] Habets, and Patrick [Gutachter] Naylor. "Speech dereverberation in noisy environments using time-frequency domain signal models / Sebastian Braun ; Gutachter: Emanuel Habets, Patrick Naylor." Erlangen : Friedrich-Alexander-Universität Erlangen-Nürnberg (FAU), 2018. http://d-nb.info/1155590597/34.
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Heidrich-Meisner, Verena [Verfasser], Gregor [Gutachter] Schöner, and Christian [Gutachter] Igel. "Evolutionary direct policy search in noisy environments / Verena Heidrich-Meisner ; Gutachter: Gregor Schöner, Christian Igel ; Fakultät für Physik und Astronomie." Bochum : Ruhr-Universität Bochum, 2011. http://d-nb.info/1217845216/34.
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Boczkowski, Lucas. "Search and broadcast in stochastic environments, a biological perspective." Thesis, Sorbonne Paris Cité, 2018. http://www.theses.fr/2018USPCC044.
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Cette thèse s’articule autour de deux séries de travaux motivés par des expériences sur des fourmis. Bien qu’inspirés par labiologie, les modèles que nous développons utilisent une terminologie et une approche typique de l’informatique théorique.Le premier modèle s’inspire du transport collaboratif de nourriture au sein de l’espèce P. Longicornis. Certains aspectsfondamentaux du processus peuvent être décrits par un problème de recherche sur un graphe en présence d’un certain typed’indications bruitées à chaque noeud. Ces indications représentent de courtes traces de phéromones déposées devant l’objettransporté afin de faciliter la navigation. Dans cette thèse, nous donnons une analyse complète du problème lorsque le graphesous-jacent est un arbre, une hypothèse pertinente dans un cadre informatique. En particulier, notre modèle peut être vucomme une généralisation de la recherche binaire aux arbres, en présence de bruit. De manière surprenante, lescomportements des algorithmes optimaux dans ce cadre diffèrent suivant le type de garantie que l’on étudie : convergence enmoyenne ou avec grande probabilité.Le deuxième modèle présenté dans cette thèse a été conçu pour décrire la dissémination d’informations au sein de fourmis dudésert. Dans notre modèle, les échanges ont lieu uniformément au hasard, et sont sujets à du bruit. Nous prouvons une borneinférieure sur le nombre d’interactions requis en fonction de la taille du groupe. La borne, de même que les hypothèses dumodèle, semblent compatible avec les données expérimentales.Une conséquence théorique de ce résultat est une séparation dans ce cadre des variantes PUSH et PULL pour le problème du broadcast avec bruit. Nous étudions aussi une version du problème avec des garanties de convergence plus fortes. Dans cecas, le problème peut-être résolu efficacement, même si les échanges d’information au cours de chaque interaction sont très limitésThis thesis is built around two series of works, each motivated by experiments on ants. We derive and analyse new models,that use computer science concepts and methodology, despite their biological roots and motivation.The first model studied in this thesis takes its inspiration in collaborative transport of food in the P. Longicornis species. Wefind that some key aspects of the process are well described by a graph search problem with noisy advice. The advicecorresponds to characteristic short scent marks laid in front of the load in order to facilitate its navigation. In this thesis, weprovide detailed analysis of the model on trees, which are relevant graph structures from a computer science standpoint. Inparticular our model may be viewed as a noisy extension of binary search to trees. Tight results in expectation and highprobability are derived with matching upper and lower bounds. Interestingly, there is a sharp phase transition phenomenon forthe expected runtime, but not when the algorithms are only required to succeed with high probability.The second model we work with was initially designed to capture information broadcast amongst desert ants. The model usesa stochastic meeting pattern and noise in the interactions, in a way that matches experimental data. Within this theoreticalmodel, we present in this document a strong lower bound on the number of interactions required before information can bespread reliably. Experimentally, we see that the time required for the recruitment process of even few ants increases sharplywith the group size, in accordance with our result. A theoretical consequence of the lower bound is a separation between theuniform noisy PUSH and PULL models of interaction. We also study a close variant of broadcast, without noise this time butunder more strict convergence requirements and show that in this case, the problem can be solved efficiently, even with verylimited exchange of information on each interaction
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Chakrabarty, Soumitro [Verfasser], Emanuel [Akademischer Betreuer] Habets, and Emanuel [Gutachter] Habets. "Robust Direction-of-Arrival estimation and spatial filtering in noisy and reverberant environments / Soumitro Chakrabarty ; Gutachter: Emanuel Habets ; Betreuer: Emanuel Habets." Erlangen : Friedrich-Alexander-Universität Erlangen-Nürnberg (FAU), 2020. http://d-nb.info/1211822036/34.
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Alali, Khaled Ahmed. "Azimuthal Localization and Detection of Vehicular Backup Alarms Under Electronic and Non-Electronic Hearing Protection Devices in Noisy and Quiet Environments." Diss., Virginia Tech, 2011. http://hdl.handle.net/10919/26890.
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Objective assessment for the effect of hearing protectors, background noise levels, and backup alarm acoustic features on listeners' abilities to localize backup alarm signals in the horizontal dimension, as well as on their ability to detect backup alarm signals in the distance dimension, is lacking in the acoustics and safety literature. Accordingly, two research experiments were conducted for this dissertation.
In the first experiment, the effect of seven hearing protectors, two background pink noise levels (60 dBA and 90 dBA), and two backup alarm signals (standard and spectrally-modified) on the ability of normal hearing listeners to localize backup alarm signals in the horizontal dimension was investigated. Results indicated that a diotic sound transmission earmuff significantly degraded localization accuracy as compared to all other hearing protectors and the open ear condition. In addition, no significant difference existed between the open ear condition and the other hearing protectors in localization accuracy in most of the conditions tested. However, the E-A-R/3M HiFiTM earplug was advantageous in localization performance since it provided a significantly higher percentage correct localization than the Moldex foam earplug, the diotic earmuff, and the dichotic earmuff in 90 dBA pink noise. As for main effects of the other independent variables, the 90 dBA pink noise significantly degraded localization performance as compared to the quiet condition of 60 dBA, and a spectrally-modified backup alarm significantly improved localization performance as compared to the standard (narrowband) backup alarm. Potential application of these results includes the revision of backup alarm standards. In addition, these results provide clear advice for safety professionals to avoid the application of diotic sound transmission earmuffs for workers if localizing backup alarms is important.
In the first experiment, listeners' feeling of comfort for each hearing protector was assessed subjectively by using a comfort rating scale. In addition, a subjective assessment for listeners' confidence in their localization decisions was established. Results indicated no significant difference between the hearing protectors in terms of comfort. However, in terms of listeners' confidence in localization decisions, their confidence was significantly degraded when they were fitted with the diotic earmuff. By contrast, they showed significantly more confidence in their localization decisions when they were fitted with the E-A-R/3M HiFi™ earplug as compared to when they were fitted with the Moldex foam earplug, the E-A-R/3M Ultrafit™ earplug, and the Bilsom passive earmuff.
In the second experiment, listeners' performance in detecting a stationary backup alarm signal, including both a standard (narrowband) and broadband (pulsed white noise) alarm, was determined while they were equipped with various passive and electronic hearing protection devices. Listeners' performance was quantified by detection distance, which was defined as the distance between the stationary backup alarm device and the position where the listener detected the backup alarm signal. The resultant data demonstrated that normal hearing listeners detected a standard (narrowband) backup alarm signal at significantly longer distances as compared to the broadband (Brigade™) backup alarm signal, thus indicating the earlier forewarning by the standard alarm. In addition, passive hearing protection devices characterized with high attenuation significantly reduced the detection distance. These results may be applied to assist safety professionals in selecting hearing protectors and backup alarm signals that provide on-foot workers with ample time to react to an approaching backing vehicle, thus improving their safety.Ph. D.
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Sidorenko, Juri [Verfasser], Urs [Akademischer Betreuer] Hugentobler, Urs [Gutachter] Hugentobler, Christian [Gutachter] Schindelhauer, and Johann [Gutachter] Dambeck. "Self-calibration of time-based localization systems in noisy environments with nonlinear optimization / Juri Sidorenko ; Gutachter: Urs Hugentobler, Christian Schindelhauer, Johann Dambeck ; Betreuer: Urs Hugentobler." München : Universitätsbibliothek der TU München, 2021. http://d-nb.info/1231434600/34.
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Sandström, Rasmus, and Jonas Renngård. "En undersökning och jämförelse av två röststyrningsramverk för Android i bullriga miljöer." Thesis, Högskolan Dalarna, Informatik, 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:du-25579.
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Voice control is a technology that most people encounter or use on a daily basis. The voice control technology can be used to interpret voice commands and execute tasks based on the command pronounced. According to previous studies problems arise with the precision when the voice control technologies are used in noisy environments. This study has been conducted as an experiment where the precision in two voice control frameworks for Android has been examined. The purpose with this study is to examine the precision in these two frameworks to assist a decision making for an organisation who has developed an application which will be used by midwives in low and middle income countries. Two prototypes was developed using the two voice control frameworks PocketSphinx and iSpeech. The precision of these frameworks was tested in three different surroundings. The surroundings the frameworks was tested in had the decibel levels 25, 60, and 80. The result shows that the number of correctly registered voice commands reduces considerably depending on which sound level the frameworks are being tested in. The framework who got the most voice commands correctly registered was PocketSphinx, but even this framework had a big margin of error.Röststyrning är idag en teknologi som de flesta människor någon gång stöter på eller använder sig av dagligen. Röststyrningsteknologin kan användas för att tolka vissa kommandon som sedan utför en uppgift baserat på det kommando som uttalas. Enligt tidigare studier uppkommer det problem med precisionen hos de röststyrningsramverk som används i bullriga miljöer. Denna studie har utförts som ett experiment där precisionen hos två stycken röststyrningsramverk för Android har undersökts. Syftet med denna studie var att undersöka precisionen hos dessa ramverk för att bistå med underlag till en organisation som utvecklat en applikation som används av barnmorskor i låg- och medelinkomstländer. Två stycken prototyper utvecklades med hjälp av röststyrningsramverken PocketSphinx och iSpeech. Dessa ramverks precision testades i tre stycken olika miljöer. De miljöer som prototyperna testades i hade ljudnivåerna 25dB, 60dB samt 80dB. Resultatet påvisar att antalet korrekt registrerade kommandon minskar avsevärt beroende på vilken ljudnivå som ramverken testas i. Det ramverk som korrekt registrerade flest röstkommandon var PocketSphinx men även denna hade en stor felmarginal.
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Occelli, Florian. "Effet d’une exposition à long-terme à un milieu bruité sur l’audiogramme et les propriétés fonctionnelles des neurones du cortex auditif primaire." Thesis, Université Paris-Saclay (ComUE), 2015. http://www.theses.fr/2015SACLS165.
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Depuis quelques années, des recherches décrivent des effets alarmants de l’exposition à des environnements acoustiques artificiels sur les propriétés fonctionnelles des neurones du système auditif. L’objectif de ce projet était de déterminer si une exposition à très long terme à une intensité sonore, qui n’est pas reconnue par les législations pour provoquer des pertes permanentes ou temporaires (80dB SLP 8h/jour), induisait ou pas des changements au niveau des audiogrammes et des propriétés fonctionnelles des neurones du cortex auditif primaire.Des rattes adultes (Sprague Dawley) ont été exposées entre 3 mois à 18 mois (selon les groupes) à un milieu acoustique mimant les environnements sonores quotidiens de la majorité de la population et dont les effets n’ont jamais été étudiés sur de telles durées. L’originalité de ce projet réside dans l’analyse des effets à tous les niveaux du système auditif depuis le niveau périphérique (ABRs) jusqu’au niveau central (électrophysiologie corticale) ainsi que les conséquences possibles au niveau comportemental. Une tâche d’apprentissage perceptif inédite a été mise au point afin d’évaluer les effets de l’exposition. Au cours du vieillissement, nos données montrent une baisse des performances comportementales, une atteinte progressive des seuils ABRs et des atteintes de certains paramètres des réponses neuronales comme (i) la latence, (ii) la durée, (iii) la détection de silence dans une vocalisation, (iv) le suivit d’une modulation d’amplitude, (v) la reproductibilité des réponses à une vocalisation. Le principal effet de l’exposition à un environnement bruité est l’apparition d’un TTS après 6 à 12 mois d’exposition (qui disparait complètement en 3 semaines), sans que cela ait, de façon très surprenante, la moindre conséquence notable sur les seuils ABRs, l’activité évoquée corticale, ou les performances de discrimination des animaux. Ces résultats nous incitent à la prudence sur la généralisation des conclusions à tirer des expositions à des environnements bruités artificielsOver the last few years, studies have described alarming effects of exposure to artificial acoustic environments on the functional properties of neurons in the auditory system. The aim of this project was to determine if long-lasting exposure at a sound intensity which is not recognized by the legislation to cause permanent or temporary hearing loss (80 dB SLP 8h/ day) induced, or not, changes in the audiograms and functional properties of neurons in theprimary auditory cortex. Adult female rats (Sprague Dawley) were exposed over 3 to 18 months (depending on the group) to an acoustic environment mimicking daily sound environments surrounding a large part of the population, and whose effects have never been studied on such durations. The originality of this project lies in analyzing the effects at alllevels of the auditory system from peripheral (via ABRs) to central levels (cortical electrophysiology) and also the possible consequences at the behavioral level. A new perceptual learning task has been developed to assess the effects of exposure. During aging, our data showed a decrease in behavioral performance, a gradual impairment of ABRs thresholds as well as an impairment in parameters of the neural responses such as (i) the response latency, (ii) response duration, (iii) the ability to detect silence in a vocalization (iv) or to follow an amplitude modulation, (v) the reproducibility of response to vocalization. The main effect of exposure to a noisy environment is the appearance of a Temporary Threshold Shift (TTS) after 6 to 12 months of exposure (which completely disappears in three weeks). Surprisingly, this long lasting TTS had apparently no e ffect on ABRs thresholds, the evokedcortical activity, or the animal’s discrimination performance. These results encourage us to be quite cautious in generalizing the conclusions to be drawn from exposures to artificial noisyenvironments
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Fatoorechi, Mohsen. "Electroencephalogram signal acquisition in unshielded noisy environment." Thesis, University of Sussex, 2015. http://sro.sussex.ac.uk/id/eprint/55034/.
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Researchers have used electroencephalography (EEG) as a window into the activities of the brain. High temporal resolution coupled with relatively low cost compares favourably to other neuroimaging techniques such as magnetoencephalography (MEG). For many years silver metal electrodes have been used for non-invasive monitoring electrical activities of the brain. Although these electrodes provide a reliable method for recording EEG they suffer from noise, such as offset potentials and drifts, and usability issues, e.g. skin prepa- ration and short circuiting of adjacent electrodes due to gel running. Low frequency noise performance is the key indicator in determining the signal to noise ratio of an EEG sensor. In order to tackle these issues a prototype Electric Potential Sensor (EPS) device based on an auto-zero operational amplifier has been developed and evaluated. The absence of 1/f noise in these devices makes them ideal for use with signal frequencies ~10Hz or less. The EPS is a novel active electrode electric potential sensor with ultrahigh input impedance. The active electrodes are designed to be physically and electrically robust and chemically and biochemically inert. They are electrically insulated (anodized) and scalable. These sensors are designed to be immersed in alcohol for sterilization purposes. A comprehensive study was undertaken to compare the results of EEG signals recorded by the EPS with different commercial systems. These studies comprised measurements of both free running EEG and Event Related Potentials. Strictly comparable signals were observed with cross correlations of higher than 0.9 between the EPS and other systems.
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Afiomah, Stephen U. "A decision-directed-detection scheme for PCM systems in a noisy environment." Ohio : Ohio University, 1986. http://www.ohiolink.edu/etd/view.cgi?ohiou1183125570.
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Georgiadis, Apostolos T. "Adaptive equalisation for impulsive noise environments." Thesis, University of Edinburgh, 2001. http://hdl.handle.net/1842/429.
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This thesis addresses the problem of adaptive channel equalisation in environments where the interfering noise exhibits non–Gaussian behaviour due to impulsive phenomena. The family of alpha-stable distributions has proved to be a suitable and flexible tool for the modelling of signals with impulsive nature. However,non–Gaussian alpha–stable signals have infinite variance, and signal processing techniques based on second order moments are meaningless in such environments. In order to exploit the flexibility of the stable family and still take advantage of the existing signal processing tools, a novel framework for the integration of the stable model in a communications context is proposed, based on a finite dynamic range receiver. The performance of traditional signal processing algorithms designed under the Gaussian assumption may degrade seriously in impulsive environments. When this degradation cannot be tolerated, the traditional signal processing methods must be revisited and redesigned taking into account the non–Gaussian noise statistics. In this direction, the optimum feed–forward and decision feedback Bayesian symbol–by–symbol equalisers for stable noise environments are derived. Then, new analytical tools for the evaluation of systems in infinite variance environments are presented. For the centers estimation of the proposed Bayesian equaliser, a unified framework for a family of robust recursive linear estimation techniques is presented and the underlying relationships between them are identified. Furthermore, the direct clustering technique is studied and robust variants of the existing algorithms are proposed. A novel clustering algorithm is also derived based on robust location estimation. The problem of estimating the stable parameters has been addressed in the literature and a variety of algorithms can be found. Some of these algorithms are assessed in terms of efficiency, simplicity and performance and the most suitable is chosen for the equalisation problem. All the building components of an adaptive Bayesian equaliser are then put together and the performance of the equaliser is evaluated experimentally. The simulation results suggest that the proposed adaptive equaliser offers a significant performance benefit compared with a traditional equaliser, designed under the Gaussian assumption. The implementation of the proposed Bayesian equaliser is simple but the computational complexity can be unaffordable. However, this thesis proposes certain approximations which enable the computationally efficient implementation of the optimum equaliser with negligible loss in performance.
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Wilson, John Michael. "Robust communication in a time-varying noisy environment." Thesis, Virginia Polytechnic Institute and State University, 1987. http://hdl.handle.net/10919/80062.
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Matched filter detectors are used to detect known signal waveforms transmitted under noisy conditions. Moving-average matched filters (MAMF's) are a class of digital filters whose performance is measured in terms of Signal to Noise Ratio (SNR). The overall performance of a MAMF is described by the SNR Improvement (SNRI) which is the ratio of Output SNR (OSNR) to Input SNR (ISNR). The OSNR and ISNR are the SNR at the output and input of the MAMF respectively. SNRI is maximized by maximizing OSNR since ISNR is fixed for a received signal and noise. The OSNR of a MAMF is a function of the noise autocorrelation sequence and the transmitted signal vector. The maximum OSNR of a MAMF is produced when the signal vector is the eigenvector associated with the smallest eigenvalue of the Toeplitz matrix formed from the noise autocorrelation sequence. If the noise autocorrelation is not known in advance of transmission, or not stationary, then it must be estimated at the receiver. Since autocorrelation estimators derive their estimates from noise samples, i.e. a random process, the estimates are probabilistic. In a practical implementation wherein the signal vector is fixed, the noise is stationary over short periods of time, and the noise autocorrelation sequence is estimated, the SNRI or performance of the MAMF varies and can even become less than unity if either the estimates are poor or the noise characteristics differ from those expected when the signal vectors were selected. A SNRI less than unity is highly undesirable as processing, which is done with the objective of obtaining higher OSNR than ISNR, i.e. a SNRI greater than unity, has become counterproductive.
This thesis proposes a variation to the classical MAMF communication system and investigates the performance of the resulting MAMF. In the classical MAMF communication system the N-dimensional signal vector is treated as a single vector. In the proposed MAMF communication system, the N-dimensional signal vector is composed of two or more linearly independent basis vectors spanning a signal vector subspace of dimension M. By combining the linearly independent basis vectors in the receiver, one can effectively change the transmitted signal vector to any signal vector in the signal vector subspace to maximize OSNR. The OSNR of a MAMF is a function of the autocorrelation of the noise as well as the signal vector. The autocorrelation of the noise is estimated in both the classical and proposed systems. For relatively few noise samples, the estimated autocorrelation of the noise deviates from the actual autocorrelation. The proposed system is formed from the classical system by proceeding the MAMF with a processor that extracts the received linearly independent basis vectors with additive colored Gaussian noise from the received transmission and combines them to yield maximum OSNR assuming the estimated autocorrelation of the noise is exact. Since the autocorrelation of the noise is estimated from the random noise process, the autocorrelations themselves are probabilistic and hence the maximum OSNR is too. As the estimated noise autocorrelation approaches the actual noise autocorrelation, the OSNR approaches the absolute maximum OSNR for the M-dimensional system. The theoretical aspects of both the classical and proposed MAMF communication systems are developed in this thesis.
The performance of the proposed MAMF communication system is investigated for a practical implementation wherein the signal vector is composed of two linearly independent basis vectors and the noise characteristics vary over time. The performance of the proposed system is first compared to that of the classical system with both systems using various signal vectors, over various noise colors, and with the exact noise autocorrelation given. The performance comparison between the classical and proposed systems is then repeated with the noise autocorrelation, as in a practical implementation, estimated using either the classical biased or Burg estimator. The performance is measured by SNRI and the results are tabulated and graphed.
Finally, the proposed system is implemented and its performance measured by bit error rates as a function of ISNR. This will show whether SNRI performance is a good prediction of bit error rate performance. The color of the stationary Gaussian noise is kept constant during transmission of a particular bit. The color of the stationary Gaussian noise is changed between bit transmissions to observe the robustness of the system under different colored noise conditions while maintaining the same signal vectors, or signal subspace. The results are again tabulated and graphed.Master of Science
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Peng, Dandan, and 彭丹丹. "Compact environmental noise absorber." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 2014. http://hdl.handle.net/10722/209491.
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With the development of the industry and the society, noise pollution becomes increasingly severe, especially in large cities. Generally, there are three major noise sources, namely industrial noise, traffic noise and community noise. In response, several measurements have been developed to achieve effective noise control. Examples of noise control methods are reduction of noise at source and abatement of noise during the transmission. Since noise sources are more difficult to control, and sometimes already fixed, noise control during sound transmission is more broadly applied. Traditional passive noise control techniques include Helmholtz resonators and noise absorption materials like felt, glass mineral. The sound absorption materials (SAM) are found to be efficient in attenuating noise in high frequency domain, but their performance at low frequencies is quite poor. On the other hand, the Helmholtz resonator works perfect at any target frequency but suffers from its narrow absorption bandwidth, so that it’s only effective within a limited frequency band.
As an innovative solution to passive noise control problem, micro-perforated panel absorber (MPPA) has attracted great interest in recent years. It turns out to be a competitive alternative to sound absorption materials and Helmholtz resonator. The parallel and series arrangement of MPPAs backed with cavities of different depths allows them to obtain decent noise absorption performance over a relatively broad frequency range. However, the performance of MPPA is restricted by its volume, as large volume is demanded for decent low-frequency absorption, which is also the case for noise absorption materials.
In this thesis, a potential way to improve the low-frequency performance of the MPPA without occupying extra volume is proposed and implemented to tests. The focus is the adjustment of speed of sound and it is beneficial in different applications such as the following. In low frequency noise control, the size of the absorber in at least one dimension is often related to the wavelength and it is often too long. With a reduced speed of sound one can reduce this size while keeping the overall volume constant. Along this line of thinking, the effect of cavity configuration on its acoustic properties is investigated by two steps. Firstly, the property of a waveguide consisting of several identical elements is studied. The number of element is chosen to magnify the effect of the configuration. It turns out the irregularity of the duct shape can slow down the speed of sound of the plane wave by increasing the acoustic mass. Secondly, the absorption performance of an MPPA backed with an irregular cavity is evaluated. The shape of the cavity is the same as the element in the first step. In advance, the parallel arrangement of two MPPAs backed with irregular cavities is investigated, in order to look into the effect of cavity shape on inter-resonator interaction. The final results indicate that cavity design is an effective method to enhance the noise absorption performance of the MPPA arrays in the low-frequency domain.published_or_final_version
Mechanical Engineering
Master
Master of Philosophy
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SANTOS, DEBORA ANDREA DE OLIVEIRA. "SPEECH RECOGNITION IN NOISE ENVIRONMENT." PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO, 2001. http://www.maxwell.vrac.puc-rio.br/Busca_etds.php?strSecao=resultado&nrSeq=1987@1.
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COORDENAÇÃO DE APERFEIÇOAMENTO DO PESSOAL DE ENSINO SUPERIOREste trabalho apresenta um estudo comparativo de três técnicas de melhoria das taxas de reconhecimento de voz em ambiente adverso, a saber: Normalização da Média Cepestral (CMN), Subtração Espectral e Regressão Linear no Sentido da Máxima Verossimilhança (MLLR), aplicadas isoladamente e em concomitância, duas a duas. Os testes são realizados usando um sistema simples: reconhecimento de palavras isoladas (dígitos de zero a nove, e meia), modo dependente do locutor, modelos ocultos de Markov do tipo contínuo, e vetores de atributos com doze coeficientes cepestrais derivados da análise de predição linear. São adotados três tipos de ruído (gaussiano branco, falatório e de fábrica) em nove razões sinal-ruído diferentes. Os resultados experimentais demonstram que o emprego isolado das técnicas de reconhecimento robusto é, em geral, vantajoso, pois nas diversas razões sinal-ruído para as quais os testes são efetuados, quando as taxas de reconhecimento não sofrem um acréscimo, mantém-se as mesmas obtidas quando não se aplica nenhum método de aumento da robustez. Analisando-se comparativamente as implementações isoladas e simultânea das técnicas, constata-se que a simultânea nem sempre é atraente, dependendo da dupla empregada. Apresentam-se, ainda, os resultados decorrentes do uso de modelos ruidosos, observando-se que, embora sejam inegavelmente melhores, sua utilização é inviável na prática. Das técnicas implementadas, a que representa resultados mais próximos ao emprego de modelos ruidosos é a MLLR, seguida pela CMN, e por último pela Subtração Espectral. Estas últimas, embora percam em desempenho para a primeira, apresentam como vantagem a simplicidade e a generalidade. No que concerne as técnicas usadas concomitantemente, a dupla Subtração Espectral e MLLR é a considerada de melhor performance, pois mostra-se conveniente em relação ao emprego isolado de ambos os métodos, o que nem sempre ocorre com o uso de outras combinações das técnicas individuais.
This work presents a comparative study of three techniques for improving the speech recognition rates in adverse environment, namely: Cepstral Mean Normalization (CMN), Spectral Subtraction and Maximum Likelihood Linear Regression (MLLR). They are implemented in two ways: separately and in pairs. The tests are carried out on a simple system: recognition of isolated words (digits from zero to nine, and the word half), speaker-dependent mode, continuous hidden Markov models, and speech feature vectors with twelve cepstral coefficients derived from linear predictive analysis. Three types of noise are considered (the white one, voice babble and from factory) at nine different signal-to-noise ratios. Experimental result demonstrate that it is worth using separately the techniques of robust recognition. This is because for all signal-to-noise conditions when the recognition accuracy is not improved it is the same one obtained when no method for increasing the robustness is applied. Analyzing comparatively the isolated and simultaneous applications of the techniques, it is verified that the later is not always more attractive than the former one. This depends on the pair of techniques. The use of noisy models is also considered. Although it presents better results, it is not feasible to implement in pratical situations. Among the implemented techniques, MLLR presents closer results to the ones obtaneid with noisy models, followed by CMN, and, at last, by Spectral Subtraction. Although the two later ones are beaten by the first, in terms of recognition accuracy, their advantages are the simplicity and the generality. The use of simultaneous techniques reveals that the pair Spectral Subtraction and MLLR is the one with the best performance because it is superior in comparison with the individual use of both methods. This does not happen with other combination of techniques.
Este trabajo presenta un estudio comparativo de tres técnicas de mejoría de las tasas de reconocimiento de voz en ambiente adverso, a saber: Normalización de la Media Cepextral (CMN), Substracción Espectral y Regresión Lineal en el Sentido de la Máxima Verosimilitud (MLLR), aplicadas separada y conjuntamente, dos a dos. Las pruebas son realizados usando un sistema simple: reconocimiento de palabras aisladas (dígitos de cero al nueve, y media), de modo dependiente del locutor, modelos ocultos de Markov de tipo contínuo, y vectores de atributos con doce coeficientes cepextrales derivados del análisis de predicción lineal. Se adoptan tres tipos de ruido (gausiano blanco, parlatorio y de fábrica) en nueve razones señal- ruido diferentes. Los resultados experimentales demuestran que el empleo aislado de las técnicas de reconocimiento robusto es, en general, ventajoso, pues en las diversas relaciones señal ruido para las cuales las pruebas son efetuadas, cuando la tasa de reconocimiento no aumenta, manteniendo las mismas tasas cuando no se aplica ningún método de aumento de robustez. Analizando comparativamente las implementaciones aisladas y simultáneas de las técnicas, se constata que no siempre la simultánea resulta atractiva, dependiendo de la dupla utilizada. Se presentan además los resultados al utilizar modelos ruidosos, observando que, aunque resultan mejores, su utilización em la práctica resulta inviable. De las técnicas implementadas, la que presenta resultados más próximos al empleo de modelos ruidosos es la MLLR, seguida por la CMN, y por último por la Substracción Espectral. Estas últimas, aunque tienen desempeño peor que la primera, tienen como ventaja la simplicidad y la generalidad. En lo que se refiere a las técnicas usadas concomitantemente, la dupla Substracción Espectral y MLLR es la de mejor performance, pues se muestra conveniente en relación al empleo aislado de ambos métodos, lo que no siempre ocurre con el uso de otras combinaciones de las técnicas individuales.
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Renault, Raphael. "Detection of Fast Moving Pulses in a Noisy Environment." Thesis, Virginia Tech, 2000. http://hdl.handle.net/10919/31041.
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We develop and analyze a combination of techniques to improve timing measurement accuracy of systems processing Gaussian pulses distorted by noise. The approach involves M/N detection, integration, and either correlation or threshold timing measurement techniques. The gain of this process is an increase of the detection capabilities of the system: improvement of the detection probability and decrease in false alarm probability, reduction in pulse distortion, and increase of the accuracy of time delay measurements between pulses using either threshold or correlation measurement methods. Each element of the proposed architecture is studied separately, and modeled analytically. As a result, a design method is proposed in order to develop an appropriate solution to any system requiring accurate time delay measurements in noisy environments. This general method is then applied to a real system, and the results in terms of detection improvement and rms timing error of the method meet expectations: the signal to noise ratio (SNR) operating point of the system is lowered by 10dB, and correlation proves to generate 2dB less rms timing error than threshold.
Master of Science
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Ledoux, Christelle Michelle. "Robust speech filtering in impulsive noise environments." Thesis, Virginia Tech, 1999. http://hdl.handle.net/10919/46325.
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This thesis presents a new robust filtering technique that suppresses impulsive noise in speech signals. The method makes use of Projection Statistics based on medians to detect segments of speech with impulses. The autoregressive model employed to smooth out the speech signal is identified by means of a robust nonlinear estimator known as the Schweppe-type Huber GM-estimator. Simulation results are presented that demonstrate the effectiveness of the filter. Another contribution of the work is the development of a robust version of the Kalman filter based on the Huber M-estimator. The performances of this filter are evaluated for a simple autoregressive process.Master of Science
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Rybing, Peter. "Active Noise Control in Home Environment." Thesis, KTH, Reglerteknik, 2003. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-109486.
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Disturbing noise is a growing problem in the society. Also in the home environment noise making devices exist, for example the vacuum cleaner. A simple way to decrease the annoyance from a vacuum cleaner is to use personal passive ear defenders. A problem with passive ear defenders is that they also attenuate wanted signals, such as speech or music. In this thesis a pair of prototype active ear defenders for vacuum cleaner noise attenuation have been developed and evaluated. Active noise control technology was used, which solved the problem with wanted signal attenuation. A measured noise reference was used with a pair of open earphones as actuator. The overall cancellation performance of the prototype system was quite low for vacuum cleaner noise. Due to that the coherence between the measured noise reference and the unwanted noise was low. Wanted signals were shown to be just slightly affected by the prototype system.
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Miksis-Olds, Jennifer L. "Manatee response to environmental noise /." View online ; access limited to URI, 2006. http://0-wwwlib.umi.com.helin.uri.edu/dissertations/dlnow/3225323.
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Li, Qi. "Advanced morphological filters for processing transient signals in noisy environment." Thesis, University of Liverpool, 2004. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.406637.
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BORGES, FRED BERKOWICZ. "LOW RATE CODECS OPERATING IN NOISY ENVIRONMENT AND IP NETWORKS." PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO, 2005. http://www.maxwell.vrac.puc-rio.br/Busca_etds.php?strSecao=resultado&nrSeq=6358@1.
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CONSELHO NACIONAL DE DESENVOLVIMENTO CIENTÍFICO E TECNOLÓGICOEste trabalho examina o impacto da quantização vetorial das LSFs sobre a qualidade de voz em codecs a baixas taxas operando em redes IP e em diversos ambientes ruidosos. São considerados diferentes esquemas de quantização vetorial (QV) multiestágio com busca em árvore envolvendo QV sem memória e QV preditiva chaveada com 2 e 4 classes. A distribuição de perda de quadros em redes IP foi modelada de acordo com o Modelo de Gilbert e a avaliação de desempenho foi realizada tanto em termos das distorções espectrais como da qualidade de voz resultante de codecs a baixas taxas. Ainda neste trabalho, foi avaliada a qualidade da voz codificada após a utilização de uma técnica de supressão de ruído baseada em transformadas wavelets (Wavelet Denoising).
This work investigates the impact of LSF vector quantisation over the voice quality in low rate codecs operating in IP networks. Tree-structured multistage vector quantisation (VQ) schemes involving memoryless VQ and switched-predictive VQ with 2 and 4 classes are considered. The packet loss frame distribution in IP networks was modelled according to the Gilbert Model and the performance was carried out both in terms of spectral distortions and the speech quality at the out put of low rate codecs. In this work, we also evaluated the quality of the coded speech after employing Wavelet Denoising.
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Sun, Deqiang. "Landau-Zener transitions in noisy environment and many-body systems." [College Station, Tex. : Texas A&M University, 2009. http://hdl.handle.net/1969.1/ETD-TAMU-2009-05-773.
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Makhubela, J. K. "Visual simultaneous localization and mapping in a noisy static environment." Thesis, Vaal University of Technology, 2019. http://hdl.handle.net/10352/462.
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M. Tech. (Department of Information and Communication Technology, Faculty of Applied and Computer Sciences), Vaal University of TechnologySimultaneous Localization and Mapping (SLAM) has seen tremendous interest amongst the research community in recent years due to its ability to make the robot truly independent in navigation. Visual Simultaneous Localization and Mapping (VSLAM) is when an autonomous mobile robot is embedded with a vision sensor such as monocular, stereo vision, omnidirectional or Red Green Blue Depth (RGBD) camera to localize and map an unknown environment. The purpose of this research is to address the problem of environmental noise, such as light intensity in a static environment, which has been an issue that makes a Visual Simultaneous Localization and Mapping (VSLAM) system to be ineffective. In this study, we have introduced a Light Filtering Algorithm into the Visual Simultaneous Localization and Mapping (VSLAM) method to reduce the amount of noise in order to improve the robustness of the system in a static environment, together with the Extended Kalman Filter (EKF) algorithm for localization and mapping and A* algorithm for navigation. Simulation is utilized to execute experimental performance. Experimental results show a 60% landmark or landfeature detection of the total landmark or landfeature within a simulation environment and a root mean square error (RMSE) of 0.13m, which is minimal when compared with other Simultaneous Localization and Mapping (SLAM) systems from literature. The inclusion of a Light Filtering Algorithm has enabled the Visual Simultaneous Localization and Mapping (VSLAM) system to navigate in an obscure environment.
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Huang, Jiehui. "Generation of entanglement and its decay in a noisy environment." HKBU Institutional Repository, 2007. http://repository.hkbu.edu.hk/etd_ra/897.
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Volchansky, Nadezhda V. "Identifying sleep-disruptive noise factors in healthcare environments." Greensboro, N.C. : University of North Carolina at Greensboro, 2007. http://libres.uncg.edu/edocs/etd/1504Volchansky/umi-uncg-1504.pdf.
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Thesis (M.S.)--University of North Carolina at Greensboro, 2007.Title from PDF t.p. (viewed Feb. 28, 2008). Directed by Kenneth Gruber; submitted to the School of Human Environmental Sciences. Includes bibliographical references (p. 67-70).
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Moeller, Michael M. Jr. "Noise environment characterization in military treatment facilities." Thesis, Georgia Institute of Technology, 2012. http://hdl.handle.net/1853/48995.
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Hospital sound environments are complex and hard to understand. One of the most important factors in these environments is the effective communication between staff members in regards to patient care and successful communication depends in part on the hospital’s sound environment. In this study, objective sound measurements as well as occupant perceptive data were collected at three hospitals. Sound pressure levels; including maximum, peak, minimum and equivalent levels were recorded in these hospitals, in addition to active impulse response measurements. Acoustic descriptors of the sound environment such as spectral content, level distributions, energy decay and temporal patterns were examined. The perception of the hospital soundscape (sound environment) was evaluated through surveys of the staff, patients and visitors to units. It was found that noise levels in all patient rooms and work areas were significantly higher than guidelines laid out in previous literature and by professional organizations. This work contributes to the field by broadening the metrics used to quantify hospital acoustic environments. In addition, this work added to the field by providing the most rigorous acoustic field measurement set published to date. This was done to create an accurate portrayal of the hospital soundscape environment.
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Alberola, Javier. "Predicting variability in environmental noise measurements." Thesis, University of Southampton, 2005. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.414693.
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Krishnan, Venkatesh. "A framework for low bit-rate speech coding in noisy environment." Diss., Available online, Georgia Institute of Technology, 2005, 2005. http://etd.gatech.edu/theses/available/etd-04042005-182043/unrestricted/krishnan%5Fvenkatesh%5F200505%5Fphd.pdf.
Full textAbstract:
Thesis (Ph. D.)--Electrical and Computer Engineering, Georgia Institute of Technology, 2005.Anderson, David, Committee Chair ; Barnwell-III, Thomas, Committee Member ; Clements, Mark, Committee Member ; Truong, Kwan, Committee Member ; Basu, Saugata, Committee Member. Vita. Includes bibliographical references.
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Myers, M. Parker (Marion Parker). "Improving production testing of RF products in a noisy measurement environment." Thesis, Massachusetts Institute of Technology, 1996. http://hdl.handle.net/1721.1/10942.
Full textAbstract:
Thesis (M.S.)--Massachusetts Institute of Technology, Sloan School of Management, 1996, and Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1996.Includes bibliographical references (leaf 60).
by M. Parker Myers.
M.S.