Dr Enzo De Sena
Enzo De Sena received the B.Sc. in 2007 and M.Sc. (cum laude) in 2009, both from the Università degli Studi di Napoli "Federico II” in Telecommunication Engineering. In 2013, he received the Ph.D. degree in Electronic Engineering from King’s College London, where he was also a Teaching Fellow from 2012 to 2013. Between 2013 and 2016 he was a Postdoctoral Research Fellow at the Katholieke Universiteit Leuven. Since 2016 he is a Lecturer in Audio at the Institute of Sound Recording at the University of Surrey. He is a former Marie Curie Fellow and he held visiting positions at Stanford University, Aalborg University and Imperial College London.
University roles and responsibilities
- International Relations Officer for the Department of Music and Media
His current research interests include room acoustics modelling, multichannel audio, microphone beam forming and binaural modelling.
Courses I teach on
spatio-temporal representation of the sound field, IEEE/ACM Transactions on Audio, Speech, and Language Processing 25 (10) pp. 1929-1941 IEEE
using a set of microphones and a loudspeaker. When
RIRs spanning a large volume are needed, many microphone
measurements must be used to spatially sample the sound field.
In order to reduce the number of microphone measurements,
RIRs can be spatially interpolated. In the present study, RIR
interpolation is formulated as an inverse problem. This inverse
problem relies on a particular acoustic model capable of representing
the measurements. Two different acoustic models are
compared: the plane wave decomposition model and a novel
time-domain model that consists of a collection of equivalent
sources creating spherical waves. These acoustic models can
both approximate any reverberant sound field created by a far
field sound source. In order to produce an accurate RIR interpolation,
sparsity regularization is employed when solving the
inverse problem. In particular, by combining different acoustic
models with different sparsity promoting regularizations, spatial
sparsity, spatio-spectral sparsity and spatio-temporal sparsity are
compared. The inverse problem is solved using a matrix-free large
scale optimization algorithm. Simulations show that the best RIR
interpolation is obtained when combining the novel time-domain
acoustic model with the spatio-temporal sparsity regularization,
outperforming the results of the plane wave decomposition model
even when far fewer microphone measurements are available.
jointly as an inverse problem. The inverse problem
consists in the interpolation of the sound field measured by a
set of microphones by matching the recorded sound pressure
with that of a particular acoustic model. This model is based
on a collection of equivalent sources creating either spherical
or plane waves. In order to achieve meaningful results, spatial,
spatio-temporal and spatio-spectral sparsity can be promoted
in the signals originating from the equivalent sources.
The inverse problem consists of a large-scale optimization
problem that is solved using a first order matrix-free optimization
algorithm. It is shown that once the equivalent source
signals capable of effectively interpolating the sound field are
obtained, they can be readily used to localize a speech sound
source in terms of Direction of Arrival (DOA) and to perform
dereverberation in a highly reverberant environment.
Echolocation by Untrained Sighted Persons, Applied Acoustics 139 pp. 82-92 Elsevier
typically known as echolocation and it can be acquired by humans, who can
profit from it in the absence of vision. We investigated the ability of twentyone
untrained sighted participants to use echolocation with self-generated oral
clicks for aligning themselves within the horizontal plane towards a virtual wall,
emulated with an acoustic virtual reality system, at distances between 1 and 32
m, in the absence of background noise and reverberation. Participants were able
to detect the virtual wall on 61% of the trials, although with large diµerences
across individuals and distances. The use of louder and shorter clicks led to an
increased performance, whereas the use of clicks with lower frequency content
allowed for the use of interaural time diµerences to improve the accuracy of
reflection localization at very long distances. The distance of 2 m was the most
difficult to detect and localize, whereas the furthest distances of 16 and 32 m
were the easiest ones. Thus, echolocation may be used eµectively to identify
large distant environmental landmarks such as buildings.
Low-order IIR Parametric Equalizers, Journal of the Audio Engineering Society 66 (11) pp. 935-952 Audio Enginering Society
of its response from a desired target response using parametric digital filters.
An optimization procedure is presented for the automatic design of a low-order equalizer
using parametric infinite impulse response (IIR) filters, specifically second-order
peaking filters and first-order shelving filters. The proposed procedure minimizes the
sum of square errors (SSE) between the system and the target complex frequency
responses, instead of the commonly used difference in magnitudes, and exploits a
previously unexplored orthogonality property of one particular type of parametric
filter. This brings a series of advantages over the state-of-the-art procedures, such as
an improved mathematical tractability of the equalization problem, with the possibility
of computing analytical expressions for the gradients, an improved initialization
of the parameters, including the global gain of the equalizer, the incorporation of
shelving filters in the optimization procedure, and a more accentuated focus on
the equalization of the more perceptually relevant frequency peaks. Examples of
loudspeaker and room equalization are provided, as well as a note about extending
the procedure to multi-point equalization and transfer function modeling.
correlated with the ratio between depth and width of the reflector?s spatial features.
obtained, they can be readily used to localize a moving sound source in terms of direction of arrival (DOA) and to perform dereverberation in a highly reverberant environment. Results from simulation experiments and from real measurements show that the proposed algorithm is robust against both localized and diffuse noise exhibiting a noise reduction in the dereverberated signals.
- E. De Sena, H. Hacıhabiboğlu, Z. Cvetković, and J. O. Smith III "Efficient Synthesis of Room Acoustics via Scattering Delay Networks," IEEE/ACM Trans. on Audio Speech Language Process., vol. 23, no. 9, pp 1478 - 1492, Sept. 2015.
- E. De Sena, Niccoló Antonello, Marc Moonen, and Toon van Waterschoot, "On the modeling of rectangular geometries in room acoustic simulations," IEEE/ACM Trans. on Audio Speech Language Process., vol. 23, no. 4, Apr. 2015.
- E. De Sena, H. Hacıhabiboğlu, and Z. Cvetković - “Analysis and Design of Multichannel Systems for Perceptual Sound Field Reconstruction,” IEEE Trans. on Audio, Speech and Language Process., vol. 21 , no. 8, pp 1653-1665, Aug. 2013.
- E. De Sena, H. Hacihabiboglu and Z. Cvetkovic - "On the design and implementation of higher-order differential microphones," IEEE Trans. on Audio, Speech and Language Process., vol. 20, no. 1, pp 162-174, Jan. 2012.
- G. Vairetti, E. De Sena, M. Catrysse, S. H. Jensen, M. Moonen, and T. van Waterschoot, “Multichannel Identification of Room Acoustic Systems with Adaptive Filters based on Orthonormal Basis Functions,” IEEE Int. Conf. on Acoust. Speech and Signal Process. (ICASSP-16), Mar. 2016.
- N. Antonello, E. De Sena, M. Moonen, P. A. Naylor, T. van Waterschoot, "Sound field control in a reverberant room using the Finite Difference Time Domain method" in AES 60th Int. Conf., Leuven, Belgium, Feb. 2016.
- G. Vairetti, E. De Sena, M. Catrysse, S. H. Jensen, M. Moonen, T. van Waterschoot, "Room acoustic system identification using orthonormal basis function models," in AES 60th Int. Conf., Leuven, Belgium, Feb. 2016.
- C. S. J. Doire, M. Brookes, P. A. Naylor, E. De Sena, T. van Waterschoot, S. H. Jensen, “Acoustic Environment Control: Implementation of a Reverberation Enhancement System,” in AES 60th Int. Conf., Leuven, Belgium, Feb. 2016.
- E. De Sena, N. Kaplanis, P. A. Naylor, T. van Waterschoot, “Large-scale auralised sound localisation experiment,” in AES 60th Int. Conf., Leuven, Belgium, Feb. 2016.
- G. Vairetti, E. De Sena, T. van Waterschoot, M. Moonen, M. Catrysse, N. Kaplanis and S.H. Jensen, "A Physically-motivated Parametric Model for Compact Representation of Room Impulse Responses based on Orthonormal Basis Functions," in Proc. 10th European Congress and Exposition on Noise Control Engineering Maastricht, The Netherlands, June 2015.
- E. De Sena and Z. Cvetković, "A Computational Model for the Estimation of Localisation Uncertainty," IEEE Int. Conf. on Acoust. Speech and Signal Process. (ICASSP-13), May 2013, Vancouver, Canada.
- H. Hacıhabiboğlu, E. De Sena, and Z. Cvetković, "Frequency-Domain Scattering Delay Networks for Simulating Room Acoustics in Virtual Environments " in proceedings of the 7th ACM/IEEE International Conference on Signal Image Tech. and Internet-based Syst. (SITIS'11), Dijon, France, November 2011.
- E. De Sena, H. Hacıhabiboğlu and Z. Cvetković - “A Generalized Design Method for Directivity Patterns of Spherical Microphone Arrays”, in proceedings of IEEE International Conference on Acoustic, Speech and Signal Processing (ICASSP-11), May 2011, Prague, Czech Republic.
- E. De Sena, H. Hacıhabiboğlu and Z. Cvetković - “Scattering Delay Network: an Interactive Reverberator for Computer Games”, in AES 41st Int. Conf., February 2011, London, UK.
- H. Hacıhabiboğlu, E. De Sena and Z. Cvetković - “Design of a Circular Microphone Array for Panoramic Audio Recording and Reproduction: Microphone Directivity”, presented at the 128th Audio Engineering Society Convention, May 2010, London, UK.
- E. De Sena, H. Hacıhabiboğlu and Z. Cvetković - “Design of a Circular Microphone Array for Panoramic Audio Recording and Reproduction: Array Radius”, presented at the 128th Audio Engineering Society Convention, May 2010, London, UK.
- E. De Sena, H. Hacıhabiboğlu and Z. Cvetković - “Perceptual Evaluation of a Circularly Symmetric Microphone Array for Panoramic Recording of Audio”, in proceedings of the 2nd Int. Symposium on Ambisonics and Spherical Acoustics, May 2010, Paris, France.
- E. Giordano, E. De Sena, G. Pau and M. Gerla - “Vergilius: A Scenario Generator for Vanet”, in proceedings of IEEE 71st Vehicular Technology Conference (VTC), May 2010, Taipei, Taiwan.
- G. Marfia, G. Pau, E. Giordano, E. De Sena, M. Gerla – “VANET: On Mobility Scenarios and Urban Infrastructure. A Case Study”, in proceedings of MOVE Workshop in conjunction with IEEE INFOCOM 2007, May 2007, Alaska, USA.
- G. Marfia, G. Pau, E. De Sena, E. Giordano, M. Gerla – “Evaluating Vehicle Network Strategies for Downtown Portland: opportunistic infrastructure and the importance of realistic mobility models”, in proceedings of the First International Workshop on Mobile Opportunistic Networking ACM/SIGMOBILE MobiOpp 2007, in conjunction with MobiSys 2007, June 2007, Puerto Rico, USA.
- E. De Sena, H. Hacıhabiboğlu, and Z. Cvetković, inventors; King's College London, assignee, "Electronic Device with Digital Reverberator and Method", USPTO Patent n. 8,908,875, filed 2/2/2012, granted 09/12/2014.
- H. Hacıhabiboğlu, E. De Sena, and Z. Cvetković, inventors; King's College London, assignee, "Microphone array", USPTO Patent n. 8,976,977, filed 15/10/2010, granted 10/3/2015.