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Felipe Orduña-Bustamante

dblp:11/11337 · DBLP profile ↗
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5ranked-venue papers
2as first author
1since 2021 · last 2026
0000-0001-9026-0727ORCID · verified

Domains — the database's venue-derived domains; a paper can count in several

Artificial intelligence and machine learning · 3 · 1 since 2021Graphics, computer vision, multimedia, augmented reality and games · 2 · 2 first-author

Expertise — from the expertise taxonomy: the topics of the expert's papers under the CCF categories. A weight counts papers with recency: 1 for a paper about the topic, 0.3 when the topic is its context, halved every five years.

Computer graphics and multimedia
2 papers
Audio and music processing · 100%

Topics — the 2 heaviest of 2, each with the papers that count most for it

TopicWeightPapersLastEvidence papers
Audio and music processing › spatial audio
sound field reproduction
0.011998
Fast deconvolution of multichannel systems using regularization · IEEE Trans. Speech Audio Process. 1998
Audio and music processing › audio reproduction
multichannel sound reproduction
0.011995
Inverse filter design and equalization zones in multichannel sound reproduction · IEEE Trans. Speech Audio Process. 1995

Methods — techniques the papers use, named apart from their topics

regularization · 0.0finite impulse response filter design · 0.0fast fourier transform · 0.0multiple-input/output inverse filtering theorem · 0.0multiple error LMS algorithm · 0.0
YearPublicationVenuePosition
2026 Automatic speech recognition in the presence of babble noise and reverberation compared to human speech intelligibility in Spanish
Carlos Mena, Ana Laura Padilla-Ortíz, Felipe Orduña-Bustamante
Comput. Speech Lang.3
2018 Binaural speech intelligibility through personal and non-personal HRTF via headphones, with added artificial noise and reverberation
Felipe Orduña-Bustamante, Ana Laura Padilla-Ortíz, Edgar Augusto Torres Gallegos
Speech Commun.1
2001 Prediction and measurement of acoustic crosstalk cancellation robustness
abstract
The condition number of the matrix of electro-acoustic head-related transfer functions (HRTF) in a two-channel sound reproduction system has been used as a measure of robustness of the Atal-Schroeder (1962) crosstalk canceler. A comparative study has been made using results produced by computer simulations and HRTFs measured in an anechoic chamber by means of a dummy head. It has been found that acoustic scattering by the head has a very important and beneficial influence on robustness, specially for large loudspeaker separations. For narrow loudspeaker separations of less than about 40 degrees it is found that crosstalk cancellation exhibits a large variation of alternating very low and very high robustness. Also, simulations and measurements have been made of the natural channel separation under the same conditions. Scattering by the head is seen to provide a good level of natural channel separation at high frequencies and large loudspeaker angles. At low frequencies or small loudspeaker angles natural channel separation is poor.
Felipe Orduña-Bustamante, José J. López, Alberto González 0001
ICASSP1
1998 Fast deconvolution of multichannel systems using regularization
abstract
A very fast deconvolution method, which is based on the fast Fourier transform (FFT), can be used to control the outputs from a multichannel plant comprising any number of control sources and error sensors. The result is a matrix of causal finite impulse response filters whose performance is optimized at a large number of discrete frequencies. The paper is particularly aimed at multichannel sound reproduction and more specifically reproducing the sound field from a set of loudspeakers.
Ole Kirkeby, Philip Arthur Nelson, Hareo Hamada, Felipe Orduña-Bustamante
IEEE Trans. Speech Audio Process.4
1995 Inverse filter design and equalization zones in multichannel sound reproduction
abstract
A discussion is given of two techniques for designing inverse filters for use in multichannel sound reproduction systems. The first is the multiple-input/output inverse filtering theorem, which uses direct inversion of a matrix containing the coefficients of filters used to specify the electroacoustic transmission paths. The second is an adaptive technique based on the multiple error LMS algorithm. The theory presented reconciles the two approaches and furthermore, derives explicit conditions which must be fulfilled if an exact inverse is to exist. A formula is derived which gives the number of coefficients required in the inverse filters in terms of the number of coefficients used to represent the transmission paths. Some numerical examples are also presented which illustrate the dependence of the mean square error on both the choice of modeling delay and the number of coefficients in the inverse filters. Finally, the results of some simulations are given which demonstrate the acoustical possibilities associated with these filtering techniques.>
Philip Arthur Nelson, Felipe Orduña-Bustamante, Hareo Hamada
IEEE Trans. Speech Audio Process.2