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Dive into the research topics where Banu Gunel is active.

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Featured researches published by Banu Gunel.


international conference on acoustics, speech, and signal processing | 2009

Intensity vector direction exploitation for exhaustive blind source separation of convolutive mixtures

Banu Gunel; Huseyin Hacihabiboglu; Ahmet M. Kondoz

This article presents a technique that uses the intensity vector direction exploitation (IVDE) method for exhaustive separation of convolutive mixtures. While only a four-element compact sensor array is used, multiple channels for all possible source directions are produced by exhaustive separation. Singular value decomposition (SVD) is then applied to determine the signal subspace and the directions of the local maxima of the signal energy. This information is then used to select the channels containing the individual sources. While the original IVDE method requires the prior knowledge of the directions of sources for separation, the present method eliminates this need and achieves fully blind separation. Performing SVD at a post-processing stage also improves the sound quality. The method has been tested for convolutive mixtures of up to four sources and typical separation performances are given.


IEEE Transactions on Signal Processing | 2007

Analysis of Root Displacement Interpolation Method for Tunable Allpass Fractional-Delay Filters

H. Hachabiboglu; Banu Gunel; Ahmet M. Kondoz

One of the simplest ways of designing allpass fractional-delay filters with maximally flat group delays is by using the Thiran approximation by which the filter coefficients are calculated using a closed-form equation. However, due to the number of multiplications and divisions involved, the calculation of these coefficients is a computationally costly task and is not suitable for real-time applications. The analysis of a root-displacement-based interpolation method used in allpass tunable fractional delays is presented in this paper. The method allows continuous adjustments of the approximated fractional delay without the explicit calculation of a new set of filter coefficients. The transient error observed at the output due to the change of filter coefficients is analyzed. The direct and cascade implementations are compared with respect to their transient errors. An example application of the proposed method from the field of model-based sound synthesis is given.


international conference on image processing | 2010

A scalable multi-view audiovisual entertainment framework with content-aware distribution

Erhan Ekmekcioglu; Banu Gunel; Maheshi B. Dissanayake; S. Worrall; Ahmet M. Kondoz

Delivery of 3D immersive entertainment to the home remains a highly challenging problem due to the large amount of data involved, and the need to support a wide variety of different displays. Support of such displays may require different numbers of views, delivered over time varying networks. This requires a delivery scheme featuring scalable compression to adapt to varying network conditions, and error resiliency to overcome disturbing losses in the 3D perception. Audio and video attention models can be used in designing an optimal content-aware compression and transmission scheme, by prioritizing the most visually important areas of the video. This paper gives an overview of a content-aware, scalable multi-view audiovisual entertainment delivery framework. Results are shown to evaluate the kinds of error robustness improvements that could be seen using such a system.


Journal of the Acoustical Society of America | 2008

Comparison of Subjective and Objective Evaluation Methods for Audio Source Separation

Josef Kornycky; Banu Gunel; Ahmet M. Kondoz

3937 The evaluation of audio separation algorithms can either be performed objectively by calculation of numerical measures, or subjectively through listening tests. Although objective evaluation is inherently more straightforward, subjective listening tests are still essential in determining the perceived quality of separation. This paper aims to find relationships between objective and subjective results so that numerical values can be translated into perceptual criteria. A generic audio source separation system was modelled which provided varying levels of interference, noise and artifacts. This enabled a full spread of objective measurement values to be obtained. Extensive tests were performed utilising the output synthesised by this separation model. The relationships found were presented and the factors of prime importance were determined.


IEEE Transactions on Audio, Speech, and Language Processing | 2008

Time-Domain Simulation of Directive Sources in 3-D Digital Waveguide Mesh-Based Acoustical Models

Huseyin Hacihabiboglu; Banu Gunel; Ahmet M. Kondoz

Digital waveguide mesh (DWM) models offer a simple, accurate, time-domain, numerical solution of the wave equation. A specific case where such accurate and computationally simple solutions are needed is the acoustical modeling of open or closed volumes. It is possible to model 3-D propagation of waves in enclosures such as rooms using DWM models. Generally, idealized omnidirectional sources are used for obtaining the impulse response in the DWM. However, real-life sound sources are never completely isotropic, causing wavefronts with directional properties. This paper presents two methods to simulate analytical and empirical directivities in 3-D DWM models in the far-field. The first method is based on the direct excitation of the mesh with the velocity component of the directional source and is used to simulate analytical sources. The second method is based on the weighting of velocity components generated by an omnidirectional source at different octave-bands and is used to simulate sources with frequency-dependent empirical directivity functions. A simple interpolation method for obtaining a closed-form description of the directivity function from incomplete directivity data is also proposed. Simulation results are presented for two sources in an acoustical model of a rectangular room.


IEEE Transactions on Audio, Speech, and Language Processing | 2010

Simulation of Directional Microphones in Digital Waveguide Mesh-Based Models of Room Acoustics

Hueseyin Hacihabiboglu; Banu Gunel; Zoran Cvetkovic

Digital waveguide mesh (DWM) models are time-domain numerical methods providing computationally simple solutions for wave propagation problems. They have been used in various acoustical modeling and audio synthesis applications including synthesis of musical instrument sounds and speech, and modeling of room acoustics. A successful model of room acoustics should be able to account for source and receiver directivity. Methods for the simulation of directional sources in DWM models were previously proposed. This paper presents a method for the simulation of directional microphones in DWM-based models of room acoustics. The method is based on the directional weighting of the microphone response according to the instantaneous direction of incidence at a given point. The direction of incidence is obtained from instantaneous intensity that is calculated from local pressure values in the DWM model. The calculation of instantaneous intensity in DWM meshes and the directional accuracies of different mesh topologies are discussed. An intensity-based formulation for the response of a directional microphone is given. Simulation results for an actual microphone with frequency-dependent, non-ideal directivity function are presented.


IEEE Signal Processing Letters | 2008

On the Accuracy of First-Order Numerical Derivatives in Multidimensional Digital Waveguide Mesh Topologies

H. Hachabiboglu; Banu Gunel; Ahmet M. Kondoz

Digital waveguide mesh (DWM) models are numerical solvers for the wave equation in -dimensions. They are used for obtaining the traveling-wave solution in practical acoustical modeling applications. Although unstructured meshes can be used with DWMs, regular mesh topologies are traditionally used due to their implementation simplicity. This letter discusses the accuracy of first-order approximations to numerical derivatives on more general unstructured mesh topologies. The results are applied to structured, regular mesh topologies as used in DWM modeling. A comparison of 2-D and 3-D DWM topologies with respect to the accuracy of first-order approximations to numerical derivatives is presented.


workshop on applications of signal processing to audio and acoustics | 2005

Head-related transfer function filter interpolation by root displacement

Huseyin Hacihabiboglu; Banu Gunel; Ahmet M. Kondoz

Head-related transfer function (HRTF) filters are used in virtual auditory displays for the binaural synthesis of the direction of a sound source over headphones. Once low-order HRTF filters are designed, the interpolation of these filters becomes an important issue for the synthesis of moving sound sources. An HRTF filter interpolation method based on the displacement of HRTF filter roots is proposed. It is possible to obtain a minimum-phase interpolated filter given that the original filters are also minimum-phase. The computational complexity of the method is the lower than that of the linear interpolation of magnitude responses.


workshop on applications of signal processing to audio and acoustics | 2003

Loudspeaker localization using B-format recordings

Banu Gunel

Positions of loudspeakers with respect to the listener play an important role in loudspeaker reproduction. When the listener is outside the sweet spot, one of the loudspeakers becomes dominant, decreasing the reproduction fidelity. Compensation of the time and level differences between the loudspeakers is possible if the positions of the loudspeakers can be detected automatically. The paper proposes a method to determine the loudspeaker positions with respect to the listener using B-format recordings captured at the position of the listener. Loudspeaker localization is based on calculating impulse responses using the original played sounds and altering the pointing direction of the cardioids generated from B-format recordings. The analysis provides very accurate distance estimation and direction estimation with an error of 12/spl deg/ on average.


multimedia signal processing | 2010

Spatial synchronization of audiovisual objects by 3D audio object coding

Banu Gunel; Erhan Ekmekcioglu; Ahmet M. Kondoz

Free viewpoint video enables the visualisation of a scene from arbitrary viewpoints and directions. However, this flexibility in video rendering provides a challenge in 3D media for achieving spatial synchronicity between the audio and video objects. When the viewpoint is changed, its effect on the perceived audio scene should be considered to avoid mismatches in the perceived positions of audiovisual objects. Spatial audio coding with such flexibility requires decomposing the sound scene into audio objects initially, and then synthesizing the new scene according to the geometric relations between the A/V capturing setup, selected viewpoint and the rendering system. This paper proposes a free viewpoint audio coding framework for 3D media systems utilising multiview cameras and a microphone array. A real-time source separation technique is used for object decomposition followed by spatial audio coding. Binaural, multichannel sound systems and wave field synthesis systems are addressed. Subjective test results shows that the method achieves spatial synchronicity for various viewpoints consistently, which is not possible by conventional recording techniques.

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Huseyin Hacihabiboglu

Middle East Technical University

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Ferda Özdemir Sönmez

Middle East Technical University

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Pinar Senkul

Middle East Technical University

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