Demo Data from SIGIR 2026 / reviewers in the wild / expert
Zenton Goh
dblp:34/10824
· DBLP profile ↗
4ranked-venue papers
3as first author
0since 2021 · last 1999
—ORCID · none
Domains — the database's venue-derived domains; a paper can count in several
Artificial intelligence and machine learning · 2 · 2 first-authorGraphics, computer vision, multimedia, augmented reality and games · 2 · 1 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 3 heaviest of 3, each with the papers that count most for it
| Topic | Weight | Papers | Last | Evidence papers |
|---|---|---|---|---|
Audio and music processing
speech enhancement |
0.0 | 2 | 1999 | Kalman-filtering speech enhancement method based on a voiced-unvoiced speech model · IEEE Trans. Speech Audio Process. 1999 Postprocessing method for suppressing musical noise generated by spectral subtraction · IEEE Trans. Speech Audio Process. 1998 |
Audio and music processing › acoustic signal processing › audio signal reconstruction › audio restoration
musical noise suppression |
0.0 | 1 | 1998 | Postprocessing method for suppressing musical noise generated by spectral subtraction · IEEE Trans. Speech Audio Process. 1998 |
Audio and music processing › speech enhancement
spectral subtraction |
0.0 | 1 | 1998 | Postprocessing method for suppressing musical noise generated by spectral subtraction · IEEE Trans. Speech Audio Process. 1998 |
Methods — techniques the papers use, named apart from their topics
minimum mean square error estimation · 0.0kalman filter · 0.0spectral subtraction · 0.0
| Year | Publication | Venue | Position |
|---|---|---|---|
| 1999 | Kalman-filtering speech enhancement method based on a voiced-unvoiced speech modelabstractIn this work, we are concerned with optimal estimation of clean speech from its noisy version based on a speech model we propose. We first propose a (single) speech model which satisfactorily describes voiced and unvoiced speech and silence (i.e., pauses between speech utterances), and also allows for exploitation of the long term characteristics of noise. We then reformulate the model equations so as to facilitate subsequent application of the well-established Kalman filter for computing the optimal estimate of the clean speech in the minimum-mean-square-error sense. Since the standard algorithm for Kalman-filtering involves multiplications of very large matrices and thus demands high computational cost, we devise a mathematically equivalent algorithm which is computationally much more efficient, by exploiting the sparsity of the matrices concerned. Next, we present the methods we use for estimating the model parameters and give a complete description of the enhancement process. Performance assessment based on spectrogram plots, objective measures and informal subjective listening tests all indicate that our method gives consistently good results. As far as signal-to-noise ratio is concerned, the improvements over existing methods can be as high as 4 dB. Zenton Goh, Kah-Chye Tan, B. T. G. Tan |
IEEE Trans. Speech Audio Process. | 1 |
| 1998 | Speech enhancement based on a voiced-unvoiced speech modelabstractIn this work, we attempt to refine the methods based on autoregressive (AR) modeling for speech enhancement (Paliwal and Basu, 1987); Gibson et al., 1991). As a matter of fact, AR modelling, which is a key strategy of the two above methods, is known to be good for representing unvoiced speech but not quite appropriate for voiced speech which is quite periodic in nature. Here, we incorporate a speech model which satisfactorily describes voiced and unvoiced speeches and silence (i.e., pauses between speech utterances) into the enhancement framework developed in the two above methods, and specifically devise an algorithm for computing the optimal estimate of the clean speech in the minimum-mean-square-error sense. We also present the methods we use for estimating the model parameters and give a description of the complete enhancement procedure. Performance assessment based on spectrogram plots, objective measures and informal subjective listening tests all indicate that our method gives consistently good results. Zenton Goh, Kah-Chye Tan, B. T. G. Tan |
ICASSP | 1 |
| 1998 | Postprocessing method for suppressing musical noise generated by spectral subtractionabstractWe investigate whether musical noise, which often exists in speech enhanced using spectral subtraction, can be suppressed. Via exploiting some specific characteristics of human speech, we propose a method that can effectively suppress musical noise without a noticeable effect on speech intelligibility. Performance assessments confirm that our method is effective. Zenton Goh, Kah-Chye Tan, B. T. G. Tan |
IEEE Trans. Speech Audio Process. | 1 |
| 1994 | A construction of arrays free of rank ambiguitiesabstractSchmidt [1981] has established that ambiguity in steering vectors would lead to ambiguous direction-of-arrival (DOA) estimates. In the literature, arrays free of upto rank-k ambiguities, where k/spl ges/3, have not been identified. The present authors construct a class of cross arrays which are free of upto rank-k ambiguities, where k/spl isin/{1, 2...}. Through the study, they derive a theorem for characterizing inherent ambiguities associated with a class of cross and circular arrays. They also provide a counterexample to the conjecture for characterizing higher rank ambiguity, proposed by Lo and Marple [1992].> Kah-Chye Tan, Zenton Goh |
ICASSP (4) | 2 |