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Zhaozhang Jin

dblp:37/7051 · DBLP profile ↗
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10ranked-venue papers
6as first author
0since 2021 · last 2011
—ORCID · none

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

Artificial intelligence and machine learning · 6 · 3 first-authorGraphics, computer vision, multimedia, augmented reality and games · 6 · 3 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
3 papers
Audio and music processing · 100%

Topics — the 5 heaviest of 5, each with the papers that count most for it

TopicWeightPapersLastEvidence papers
Audio and music processing
computational auditory scene analysis
0.222011
Reverberant Speech Segregation Based on Multipitch Tracking and Classification · IEEE Trans. Speech Audio Process. 2011
A Supervised Learning Approach to Monaural Segregation of Reverberant Speech · IEEE Trans. Speech Audio Process. 2009
Audio and music processing › source separation
reverberant source separation
0.222011
Reverberant Speech Segregation Based on Multipitch Tracking and Classification · IEEE Trans. Speech Audio Process. 2011
A Supervised Learning Approach to Monaural Segregation of Reverberant Speech · IEEE Trans. Speech Audio Process. 2009
Audio and music processing › source separation › speech separation
single-channel speech separation
0.222011
Reverberant Speech Segregation Based on Multipitch Tracking and Classification · IEEE Trans. Speech Audio Process. 2011
A Supervised Learning Approach to Monaural Segregation of Reverberant Speech · IEEE Trans. Speech Audio Process. 2009
Audio and music processing › music transcription
multipitch tracking
0.112011
HMM-Based Multipitch Tracking for Noisy and Reverberant Speech · IEEE Trans. Speech Audio Process. 2011
Audio and music processing › speech processing
pitch estimation
0.112011
HMM-Based Multipitch Tracking for Noisy and Reverberant Speech · IEEE Trans. Speech Audio Process. 2011

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

supervised classification · 0.1multipitch tracking · 0.1likelihood ratio test · 0.1hidden markov model · 0.1channel selection · 0.1auditory front-end · 0.1supervised learning · 0.1segmentation and grouping · 0.1pitch-based auditory features · 0.1
YearPublicationVenuePosition
2011 HMM-Based Multipitch Tracking for Noisy and Reverberant Speech
abstract
Multipitch tracking in real environments is critical for speech signal processing. Determining pitch in reverberant and noisy speech is a particularly challenging task. In this paper, we propose a robust algorithm for multipitch tracking in the presence of both background noise and room reverberation. An auditory front-end and a new channel selection method are utilized to extract periodicity features. We derive pitch scores for each pitch state, which estimate the likelihoods of the observed periodicity features given pitch candidates. A hidden Markov model integrates these pitch scores and searches for the best pitch state sequence. Our algorithm can reliably detect single and double pitch contours in noisy and reverberant conditions. Quantitative evaluations show that our approach outperforms existing ones, particularly in reverberant conditions.
Zhaozhang Jin, DeLiang Wang
IEEE Trans. Speech Audio Process.1
2011 Reverberant Speech Segregation Based on Multipitch Tracking and Classification
abstract
Room reverberation creates a major challenge to speech segregation. We propose a computational auditory scene analysis approach to monaural segregation of reverberant voiced speech, which performs multipitch tracking of reverberant mixtures and supervised classification. Speech and nonspeech models are separately trained, and each learns to map from a set of pitch-based features to a grouping cue which encodes the posterior probability of a time-frequency (T-F) unit being dominated by the source with the given pitch estimate. Because interference may be either speech or nonspeech, a likelihood ratio test selects the correct model for labeling corresponding T-F units. Experimental results show that the proposed system performs robustly in different types of interference and various reverberant conditions, and has a significant advantage over existing systems.
Zhaozhang Jin, DeLiang Wang
IEEE Trans. Speech Audio Process.1
2010 A multipitch tracking algorithm for noisy and reverberant speech
abstract
Determining multiple pitches in noisy and reverberant speech is an important and challenging task. We propose a robust multipitch tracking algorithm in the presence of both background noise and room reverberation. A new channel selection method is utilized in conjunction with an auditory front-end to extract periodicity features in the time-frequency space. These features are combined to formulate frame level conditional probabilities given each pitch state. A hidden Markov model is then applied to integrate these probabilities and search for the most likely pitch state sequences. The proposed approach can reliably detect up to two simultaneous pitch contours in noisy and reverberant conditions. Quantitative evaluations show that our system significantly outperforms existing ones, particularly in reverberant environments.
Zhaozhang Jin, DeLiang Wang
ICASSP1
2010 A computational auditory scene analysis system for speech segregation and robust speech recognition
Soundararajan Srinivasan, Zhaozhang Jin, DeLiang Wang
Comput. Speech Lang.3
2009 Learning to maximize signal-to-noise ratio for reverberant speech segregation
abstract
Monaural speech segregation in reverberant environments is a very difficult problem. We develop a supervised learning approach by proposing an objective function that directly relates to the computational goal of maximizing signal-to-noise ratio. The model trained using this new objective function yields significantly better results for time-frequency unit labeling. In our segregation system, a segmentation and grouping framework is utilized to form reliable segments under reverberant conditions and organize them into streams. Systematic evaluations show very promising results.
Zhaozhang Jin, DeLiang Wang
ICASSP1
2009 An auditory-based feature for robust speech recognition
abstract
A conventional automatic speech recognizer does not perform well in the presence of noise, while human listeners are able to segregate and recognize speech in noisy conditions. We study a novel feature based on an auditory periphery model for robust speech recognition. Specifically, gammatone frequency cepstral coefficients are derived by applying a cepstral analysis on gammatone filterbank responses. Our evaluations show that the proposed feature performs considerably better than conventional acoustic features. We further demonstrate that integrating the proposed feature with a computational auditory scene analysis system yields promising recognition performance.
Zhaozhang Jin, DeLiang Wang, Soundararajan Srinivasan
ICASSP2
2009 Monaural segregation of voiced speech using discriminative random fields
Rohit Prabhavalkar, Zhaozhang Jin, Eric Fosler-Lussier
INTERSPEECH2
2009 A Supervised Learning Approach to Monaural Segregation of Reverberant Speech
abstract
A major source of signal degradation in real environments is room reverberation. Monaural speech segregation in reverberant environments is a particularly challenging problem. Although inverse filtering has been proposed to partially restore the harmonicity of reverberant speech before segregation, this approach is sensitive to specific source/receiver and room configurations. This paper proposes a supervised learning approach to monaural segregation of reverberant voiced speech, which learns to map from a set of pitch-based auditory features to a grouping cue encoding the posterior probability of a time-frequency (T-F) unit being target dominant given observed features. We devise a novel objective function for the learning process, which directly relates to the goal of maximizing signal-to-noise ratio. The models trained using this objective function yield significantly better T-F unit labeling. A segmentation and grouping framework is utilized to form reliable segments under reverberant conditions and organize them into streams. Systematic evaluations show that our approach produces very promising results under various reverberant conditions and generalizes well to new utterances and new speakers.
Zhaozhang Jin, DeLiang Wang
IEEE Trans. Speech Audio Process.1
2007 A Supervised Learning Approach to Monaural Segregation of Reverberant Speech
abstract
Room reverberation degrades speech signals and poses a major challenge to current monaural speech segregation systems. Previous research relies on inverse filtering as a front-end for partially restoring the harmonicity of the reverberant signal. We show that the inverse filtering approach is sensitive to different room configurations, hence undesirable in general reverberation conditions. We propose a supervised learning approach to map a set of harmonic features into a pitch based grouping cue for each time-frequency (T-F) unit. We use a speech segregation method to estimate an ideal binary T-F mask which retains the reverberant mixture in a local T-F unit if and only if the energy of target is stronger than interference energy. Results show that our approach improves the segregation performance considerably.
Zhaozhang Jin, DeLiang Wang
ICASSP (4)1
2006 A computational auditory scene analysis system for robust speech recognition
abstract
We present a computational auditory scene analysis system for separating and recognizing target speech in the presence of competing speech or noise. We estimate, in two stages, the ideal binary time-frequency (T-F) mask which retains the mixture in a local T-F unit if and only if the target is stronger than the interference within the unit. In the first stage, we use harmonicity to segregate the voiced portions of individual sources in each time frame based on multipitch tracking. Additionally, unvoiced portions are segmented based on an onset/offset analysis. In the second stage, speaker characteristics are used to group the T-F units across time frames. The resulting T-F masks are used in conjunction with missing-data methods for recognition. Systematic evaluations on a speech separation challenge task show significant improvement over the baseline performance. Index Terms: speech segregation, computational auditory scene analysis, binary time-frequency mask, robust speech recognition.
Soundararajan Srinivasan, Zhaozhang Jin, DeLiang Wang
INTERSPEECH3