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David Yuheng Zhao

dblp:07/3116 · DBLP profile ↗
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7ranked-venue papers
5as first author
0since 2021 · last 2008
—ORCID · none

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

Artificial intelligence and machine learning · 5 · 3 first-authorGraphics, computer vision, multimedia, augmented reality and games · 3 · 2 first-authorComputer networks · 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.

Artificial intelligence
2 papers
Speech recognition and synthesis · 100%
Computer graphics and multimedia
1 paper
Audio and music processing · 100%
Computer networks
1 paper
Network management and operations · 100%

Topics — the 6 heaviest of 6, each with the papers that count most for it

TopicWeightPapersLastEvidence papers
Natural language and speech › Speech recognition and synthesis
speech enhancement
0.222008
Online Noise Estimation Using Stochastic-Gain HMM for Speech Enhancement · IEEE Trans. Speech Audio Process. 2008
HMM-Based Gain Modeling for Enhancement of Speech in Noise · IEEE Trans. Speech Audio Process. 2007
Natural language and speech › Speech recognition and synthesis › speech enhancement
noise estimation
0.112008
Online Noise Estimation Using Stochastic-Gain HMM for Speech Enhancement · IEEE Trans. Speech Audio Process. 2008
Natural language and speech › Speech recognition and synthesis › speech enhancement
noise power spectral density estimation
0.112008
Online Noise Estimation Using Stochastic-Gain HMM for Speech Enhancement · IEEE Trans. Speech Audio Process. 2008
Audio and music processing › speech quality assessment
non-intrusive speech quality assessment
0.112006
Low-Complexity, Nonintrusive Speech Quality Assessment · IEEE Trans. Speech Audio Process. 2006
Audio and music processing
speech quality assessment
0.112006
Low-Complexity, Nonintrusive Speech Quality Assessment · IEEE Trans. Speech Audio Process. 2006
Network management and operations
network monitoring
0.012006
Low-Complexity, Nonintrusive Speech Quality Assessment · IEEE Trans. Speech Audio Process. 2006

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

speech-coding parameter features · 0.1objective quality modeling · 0.1stochastic-gain hidden markov model · 0.1recursive expectation-maximization · 0.1maximum likelihood · 0.1recursive EM · 0.1hidden markov model · 0.1expectation-maximization · 0.1
YearPublicationVenuePosition
2008 Online Noise Estimation Using Stochastic-Gain HMM for Speech Enhancement
abstract
We propose a noise estimation algorithm for single-channel noise suppression in dynamic noisy environments. A stochastic-gain hidden Markov model (SG-HMM) is used to model the statistics of nonstationary noise with time-varying energy. The noise model is adaptive and the model parameters are estimated online from noisy observations using a recursive estimation algorithm. The parameter estimation is derived for the maximum-likelihood criterion and the algorithm is based on the recursive expectation maximization (EM) framework. The proposed method facilitates continuous adaptation to changes of both noise spectral shapes and noise energy levels, e.g., due to movement of the noise source. Using the estimated noise model, we also develop an estimator of the noise power spectral density (PSD) based on recursive averaging of estimated noise sample spectra. We demonstrate that the proposed scheme achieves more accurate estimates of the noise model and noise PSD, and as part of a speech enhancement system facilitates a lower level of residual noise.
David Yuheng Zhao, W. Bastiaan Kleijn, Alexander Ypma, Bert de Vries
IEEE Trans. Speech Audio Process.1
2007 HMM-Based Gain Modeling for Enhancement of Speech in Noise
abstract
Accurate modeling and estimation of speech and noise gains facilitate good performance of speech enhancement methods using data-driven prior models. In this paper, we propose a hidden Markov model (HMM)-based speech enhancement method using explicit gain modeling. Through the introduction of stochastic gain variables, energy variation in both speech and noise is explicitly modeled in a unified framework. The speech gain models the energy variations of the speech phones, typically due to differences in pronunciation and/or different vocalizations of individual speakers. The noise gain helps to improve the tracking of the time-varying energy of nonstationary noise. The expectation-maximization (EM) algorithm is used to perform offline estimation of the time-invariant model parameters. The time-varying model parameters are estimated online using the recursive EM algorithm. The proposed gain modeling techniques are applied to a novel Bayesian speech estimator, and the performance of the proposed enhancement method is evaluated through objective and subjective tests. The experimental results confirm the advantage of explicit gain modeling, particularly for nonstationary noise sources
David Yuheng Zhao, W. Bastiaan Kleijn
IEEE Trans. Speech Audio Process.1
2006 HMM-Based Speech Enhancement using Explicit Gain Modeling
abstract
We propose a hidden Markov model (HMM) based speech enhancement method using explicit modeling of speech and noise gains. The gains are considered to be stochastic variables in an HMM framework. The speech gain models the energy variations of speech phones, typically due to differences in pronunciation and/or different vocalizations of individual speakers. The noise gain helps to improve the tracking of the time-varying energy of non-stationary noise. The time-varying parameters of the gain models are estimated on-line using the recursive expectation maximization (EM) algorithm. The performance of the proposed enhancement system is evaluated through both objective and subjective tests. The experimental results confirm the advantage of explicit gain modeling, particularly for non-stationary noise sources
David Yuheng Zhao, W. Bastiaan Kleijn
ICASSP (1)1
2006 Non-intrusive speech quality assessment with low computational complexity
abstract
We describe an algorithm for monitoring subjective speech quality without access to the original signal that has very low computational and memory requirements. The features used in the proposed algorithm can be computed from commonly used speechcoding parameters. Reconstruction and perceptual transformation of the signal are not performed. The algorithm generates quality assessment ratings without explicit distortion modeling. The simulation results indicate that the proposed non-intrusive objective quality measure performs better than the ITU-T P.563 standard despite its very low computational complexity. Index Terms: non-intrusive quality assessment, quality of service. 1.
Volodya Grancharov, David Yuheng Zhao, Jonas Lindblom, W. Bastiaan Kleijn
INTERSPEECH2
2006 Low-Complexity, Nonintrusive Speech Quality Assessment
abstract
Monitoring of speech quality in emerging heterogeneous networks is of great interest to network operators. The most efficient way to satisfy such a need is through nonintrusive, objective speech quality assessment. In this paper, we describe a low-complexity algorithm for monitoring the speech quality over a network. The features used in the proposed algorithm can be computed from commonly used speech-coding parameters. Reconstruction and perceptual transformation of the signal is not performed. The critical advantage of the approach lies in generating quality assessment ratings without explicit distortion modeling. The results from the performed experiments indicate that the proposed nonintrusive objective quality measure performs better than the ITU-T P.563 standard
Volodya Grancharov, David Yuheng Zhao, Jonas Lindblom, W. Bastiaan Kleijn
IEEE Trans. Speech Audio Process.2
2005 On noise gain estimation for HMM-based speech enhancement
abstract
To address the variation of noise level in non-stationary noise signals, we study the noise gain estimation for speech enhancement using hidden Markov models (HMM). We consider the noise gain as a stochastic process and we approximate the probability density function (PDF) to be log-normal distributed. The PDF parameters are estimated for every signal block using the past noisy signal blocks. The approximated PDF is then used in a Bayesian speech estimator minimizing the Bayes risk for a novel cost function, that allows for an adjustable level of residual noise. As a more computationally efficient alternative, we also derive the maximum likelihood (ML) estimator, assuming the noise gain to be a deterministic parameter. The performance of the proposed gain-adaptive methods are evaluated and compared to two reference methods. The experimental results show significant improvement under noise conditions with time-varying noise energy.
David Yuheng Zhao, W. Bastiaan Kleijn
INTERSPEECH1
2004 Multiple-description vector quantization using translated lattices with local optimization [joint source/channel coding]
abstract
Multiple-description coding is a joint source- and channel coding technique suitable for real-time multimedia transmission over erasure channels. This work improves the previous methods of multiple-description vector quantization using lattice structured codebooks by introducing translated lattices in the single-description codebooks. The quantizer can easily adapt to the current channel conditions, using the locally optimized combined-description codebooks, assuming that channel statistics are available at the encoder. Compared to previous methods, the central distortion is greatly reduced for noisy channels, without a significant effect on complexity.
David Yuheng Zhao, W. Bastiaan Kleijn
GLOBECOM1