1,721,112 research outputs found
Modeling of moving sound sources based on array measurements
No full textWhen auralizing moving sound sources in Virtual Reality (VR) environments, the two main input parameters are the location and radiated signal of the source. An array measurement-based model is developed to characterize moving sound sources regarding the two parameters in this thesis. This model utilizes beamforming, i.e. delay and sum beamforming (DSB) and compressive beamforming (CB) to obtain the locations and signals of moving sound sources. A spiral and a pseudorandom microphone array are designed for DSB and CB, respectively, to yield good localization ability and meet the requirement of CB. The de-Dopplerization technique is incorporated in the time-domain DSB to address moving source problems. Time-domain transfer functions (TDTFs) are calculated in terms of the spatial locations within the steering window of the moving source. TDTFs then form the sensing matrix of CB, thus allowing CB to solve moving source problem. DSB and CB are further extended to localize moving sound sources, and the reconstructed signals from the beamforming outputs are investigated to obtain the source signals. Moreover, localization and signal reconstruction are evaluated through varying parameters in the beamforming procedures, i.e. steering position, steering window length and source speed for a moving periodic signal using DSB, and regularization parameter, signal to noise ratio (SNR), steering window length, source speed, array to source motion trajectory and mismatch for a moving engine signal using CB. The parameter studies show guidelines of parameter selection based on the given situations in this thesis for modeling moving source using beamforming. Both algorithms are able to reconstruct the moving signals in the given scenarios. Although CB outperforms DSB in terms of signal reconstruction under particular conditions, the localization abilities of the two algorithms are quite similar. The practicability of the model has been applied on pass-by measurements of a moving loudspeaker using the designed arrays, and the results can match the conclusions drawn above from simulations. Finally, a framework on how to apply the model for moving source auralization is proposed
Going Beyond Counting First Authors in Author Co-citation Analysis
Full text linkThe present study examines one of the fundamental aspects of author co-citation analysis (ACA) - the way co-citation
counts are defined. Co-citation counting provides the data on which all subsequent statistical analyses and mappings
are based, and we compare ACA results based on two different types of co-citation counting - the traditional type that
only counts the first one among a cited work's authors on the one hand and a non-traditional type that takes into
account the first 5 authors of a cited work on the other hand. Results indicate that the picture produced through this non-traditional author co-citation counting contains more coherent author groups and is therefore considerably clearer. However, this picture represents fewer specialties in the research field being studied than that produced through the traditional first-author co-citation counting when the same number of top-ranked authors is selected and analyzed. Reasons for these effects are discussed
Anthropometric individualization of head-related transfer functions analysis and modeling
Full text linkHuman sound localization helps to pay attention to spatially separated speakers using interaural level and time differences as well as angle-dependent monaural spectral cues. In a monophonic teleconference, for instance, it is much more difficult to distinguish between different speakers due to missing binaural cues. Spatial positioning of the speakers by means of binaural reproduction methods using head-related transfer functions (HRTFs) enhances speech comprehension. These HRTFs are influenced by the torso, head and ear geometry as they describe the propagation path of the sound from a source to the ear canal entrance. Through this geometry-dependency, the HRTF is directional and subject-dependent. To enable a sufficient reproduction, individual HRTFs should be used. However, it is tremendously difficult to measure these HRTFs. For this reason this thesis proposes approaches to adapt the HRTFs applying individual anthropometric dimensions of a user. Since localization at low frequencies is mainly influenced by the interaural time difference, two models to adapt this difference are developed and compared with existing models. Furthermore, two approaches to adapt the spectral cues at higher frequencies are studied, improved and compared. Although the localization performance with individualized HRTFs is slightly worse than with individual HRTFs, it is nevertheless still better than with non-individual HRTFs, taking into account the measurement effort
Variations on the Author
Full text link“Variations on the Author” discusses two of Eduardo Coutinho’s recent films (Um Dia na Vida, from 2010, and Últimas Conversas, posthumously released in 2015) and their contribution to the general question of documentary authorship. The director’s filmography is characterized by a consistent yet self-effacing form of authorial self-inscription: Coutinho often features as an interviewer that rather than express opinions propels discourses; an interviewer that is good at listening. This mode of self-inscription characterizes him as an author who is not expressive but who is nonetheless markedly present on the screen. In Um Dia na Vida, however, Coutinho is completely absent form the image, while Últimas Conversas, on the contrary, includes a confessional prologue that moves the director from the margins to the center of his films. This article examines the ways in which these works stand out in the filmography of a director who offers new insights into the notion of cinematic authorship
Fast measurement of individual head-related transfer functions
Full text linkWhile binaural technology applications gained in popularity in recent years, the majority of applications still use non-individual Head-Related Transfer Functions (HRTFs) from artificial heads. These datasets enable a reasonably good spatial localization which works especially well when using an additional visual cue. However, certain applications, for example research of spatial hearing or hearing attention, require an physically exact and realistic binaural signal. Moreover, it was shown in many experiments that there is a substantial gain from the use of individual HRTFs, for example in localization tasks. The limiting factor that prohibits the widespread use of individual HRTFs is the acquisition of such data. A substantial hardware requirement obstructs a more universal usage. Even for institutions that allow individual measurements, the measurement time that is required, and that the subjects are required to remain motionless made most measurements unfeasible in the past. This time requirement has recently been reduced by the use of parallelization in the measurement signal which lead to the development of fast measurement systems capable of acquiring individual and spatially dense HRTF. This thesis provides a objective and subjective evaluation of such a system that is designed with the goal of little disturbance of the measurements in mind. The construction is detailed, followed by both an objective and subjective evaluation. A detailed investigation into additional distortion of the sound field introduced by the system itself is presented and it is shown that the system performs comparably to a conventional system in terms of sound source localization. Furthermore, a method is introduced and evaluated to further reduce the measurement time by using continuous rotation during the measurement. This method is used to reduced the measurement duration from eight minutes to three minutes without audible differences. The introduced methods are also used to reducing additional errors from subject movement. It is shown that this movement can be reduced by a visual feedback system to a level that can be compensated efficiently
Appropriate Similarity Measures for Author Cocitation Analysis
Full text linkWe provide a number of new insights into the methodological discussion about author cocitation analysis. We first argue that the use of the Pearson correlation for measuring the similarity between authors’ cocitation profiles is not very satisfactory. We then discuss what kind of similarity measures may be used as an alternative to the Pearson correlation. We consider three similarity measures in particular. One is the well-known cosine. The other two similarity measures have not been used before in the bibliometric literature. Finally, we show by means of an example that our findings have a high practical relevance.information science;Pearson correlation;cosine;similarity measure;author cocitation analysis
Beamforming in modalen Schallfeldern von Fahrzeuginnenräumen
No full textPhysically the conventional beamforming method (CBF) is based on a decomposition of the incident wave field into plane or spherical elementary waves whose relative phases are interpreted to mean that the directions of incidence and source locations can be identified and classified. A central assumption of the CBF is its application in the acoustic free field. This assumption is particularly invalid in confined spaces such as a vehicle interior, in which to a certain frequency range (below the Schroeder frequency) modal influences dominate the sound field. Under these conditions the CBF only shows a poor localization ability; up to a complete localization failure. However, to apply beamforming under these conditions and in this frequency range, in this thesis two approaches are being pursued in parallel to minimize the room acoustic influences on the Beamforming and thus improve the localization result particularly in a vehicle interior. A first approach is based on the pre-processing of cross-spectral matrix (CSM) using the so-called generalized cross-correlation. A second approach uses “macro modeling” to approximate the pole frequencies of the room transfer functions. Based on the determined pole frequencies an inverse filter is automatically generated, which also applied to the CSM minimizes modal influences before the actual beamforming. Both approaches are as well discussed theoretically as practically examined and evaluated in consecutive steps. For this purpose, based on principle studies on defined reflection surfaces, the behavior of both approaches inside a scale model room are shown, analyzed and evaluated. Subsequently, both approaches are evaluated under the real conditions of the vehicle interior. A digression regarding the combination of beamforming with arbitrary transfer functions for the further development of the beamforming beyond the free field assumption concludes the investigation work. Finally, this thesis is concluded by the chapter “Zusammenfassung und Ausblick”
Dispelling the Myths Behind First-author Citation Counts
Full text linkWe conducted a full-scale evaluative citation analysis study of scholars in the XML research field to explore just how different from each other author rankings resulting from different citation counting methods actually are, and to demonstrate the capability of emerging data and tools on the Web in supporting more realistic citation counting methods. Our results contest some common arguments for the continued
use of first-author citation counts in the evaluation of scholars, such as high correlations between author rankings by first-author citation counts and other citation
counting methods, and high costs of using more realistic citation counting methods that are not well-supported by the ISI databases. It is argued that increasingly available digital full text research papers make it possible for citation analysis studies to go beyond what the ISI databases have directly supported and to employ more
sophisticated methods
Investigation on autoencoder models for online system identification
No full textSpeech communication devices are indispensable in our daily work and personal lives. Using them in hands free mode can create an echo signal, which, if no action is taken, would disturb the speaker. However, the echo signal can be predicted, when the impulse response between loudspeaker and microphone is known. For this task, system identification algorithms exist, such as the Least-Mean-Square (LMS) algorithm, the Normalized-Least-Mean-Square (NLMS)algorithm, and the Kalman filter. They work well in general, but face difficulties when confronted with high correlation input signals, high noise levels, or rapidly changing impulse responses over time. This thesis aims to explore whether prior knowledge about the impulse response can improve system identification. The key approach is to utilize the manifold hypothesis, which has shown promising results in previous works in mapping acoustic room impulse responses toa lower dimensional subspace. These approaches require training data of impulse responses. This thesis investigates how well affine subspace models can represent impulse response with a limited number of subspace components compared to the same number of components in the time domain. One well known way to find an optimal affine subspace is by Principal-Component-Analysis (PCA). It is shown that the affine subspace model can have the same achievable system mismatch with significantly less number of subspace components, when the loudspeaker and the microphone are constrained in their positions. The manifold LMS algorithm, the manifold NLMS algorithm and the manifold Kalman filter are proposed in this thesis, which can utilise general non linear manifolds for the acoustic echo compensation task. For the manifold LMS and NLMS algorithm in the case of white noise excitation and an affine manifold, the expected convergence speed and the expected steady state system mismatch are derived theoretically and are shown to accurately describe the algorithms behaviour in simulations. For scenarios with constrained loudspeaker and microphone positions it is shown that the manifold NLMS algorithm significantly out performs the time domain NLMS algorithm. The manifold Kalman filter is compared to the time domain Kalman filter and another subspace approach from literature. The manifold Kalman filter shows faster initial convergence speed in simulations when the achievable steady state system mismatch is set to be the same for all approaches, which can be explained by its larger step size compared to the reference approaches. Further research could include the evaluation of the proposed algorithms for non line Armani folds, which can be obtained by neural autoencoders or locally affine subspaces. The latter approach includes the search for an optimal distance measure to select the nearest neighbours of an impulse response in the training data
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