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Abbas Keshavarz

dblp:64/8327 · DBLP profile ↗
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1ranked-venue papers
1as first author
0since 2021 · last 2012
—ORCID · none

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

Artificial intelligence and machine learning · 1 · 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
1 paper
Audio and music processing · 100%
Artificial intelligence
1 paper
Speech recognition and synthesis · 100%

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

TopicWeightPapersLastEvidence papers
Audio and music processing › room acoustics › room acoustic parameter estimation
reverberation time estimation
0.112012
Speech-Model Based Accurate Blind Reverberation Time Estimation Using an LPC Filter · IEEE Trans. Speech Audio Process. 2012
Natural language and speech › Speech recognition and synthesis › speech enhancement
speech dereverberation
0.012012
Speech-Model Based Accurate Blind Reverberation Time Estimation Using an LPC Filter · IEEE Trans. Speech Audio Process. 2012

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

maximum likelihood estimation · 0.3linear predictive coding · 0.3
YearPublicationVenuePosition
2012 Speech-Model Based Accurate Blind Reverberation Time Estimation Using an LPC Filter
abstract
In this paper, we propose a speech-model based method using the linear predictive (LP) residual of the speech signal and the maximum-likelihood (ML) estimator proposed in “Blind estimation of reverberation time,” (R. Ratnam , J. Acoust. Soc. Amer., 2004) to blindly estimate the reverberation time (RT60). The input speech is passed through a low order linear predictive coding (LPC) filter to obtain the LP residual signal. It is proven that the unbiased autocorrelation function of this LP residual has the required properties to be used as an input to the ML estimator. It is shown that this method can successfully estimate the reverberation time with less data than existing blind methods. Experiments show that the proposed method can produce better estimates of RT60, even in highly reverberant rooms. This is because the entire input speech data is used in the estimation process. The proposed method is not sensitive to the type of input data (voiced, unvoiced), number of gaps, or window length. In addition, evaluation using white Gaussian noise and recorded babble noise shows that it can estimate RT60in the presence of (moderate) background noise.
Abbas Keshavarz, Saeed Mosayyebpour, Mehrzad Biguesh, T. Aaron Gulliver, Morteza Esmaeili
IEEE Trans. Speech Audio Process.1