Demo Data from SIGIR 2026 / reviewers in the wild / expert
Jae-Mo Yang
dblp:22/11067
· DBLP profile ↗
2ranked-venue papers
2as first author
0since 2021 · last 2014
—ORCID · none
Domains — the database's venue-derived domains; a paper can count in several
Artificial intelligence and machine learning · 1 · 1 first-authorGraphics, computer vision, multimedia, augmented reality and games · 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% |
Topics — the 5 heaviest of 5, each with the papers that count most for it
| Topic | Weight | Papers | Last | Evidence papers |
|---|---|---|---|---|
Audio and music processing › speech enhancement
dereverberation |
0.2 | 1 | 2014 | Online Speech Dereverberation Algorithm Based on Adaptive Multichannel Linear Prediction · IEEE ACM Trans. Audio Speech Lang. Process. 2014 |
Audio and music processing › speech enhancement › dereverberation
multi-channel linear prediction |
0.2 | 1 | 2014 | Online Speech Dereverberation Algorithm Based on Adaptive Multichannel Linear Prediction · IEEE ACM Trans. Audio Speech Lang. Process. 2014 |
Audio and music processing
speech enhancement |
0.2 | 1 | 2014 | Online Speech Dereverberation Algorithm Based on Adaptive Multichannel Linear Prediction · IEEE ACM Trans. Audio Speech Lang. Process. 2014 |
Audio and music processing
acoustic signal processing |
0.1 | 1 | 2014 | Online Speech Dereverberation Algorithm Based on Adaptive Multichannel Linear Prediction · IEEE ACM Trans. Audio Speech Lang. Process. 2014 |
Audio and music processing › acoustic signal processing › audio signal reconstruction › audio restoration
blind dereverberation |
0.1 | 1 | 2014 | Online Speech Dereverberation Algorithm Based on Adaptive Multichannel Linear Prediction · IEEE ACM Trans. Audio Speech Lang. Process. 2014 |
Methods — techniques the papers use, named apart from their topics
linear prediction · 0.2covariance matrix inversion · 0.2adaptive filtering · 0.2
| Year | Publication | Venue | Position |
|---|---|---|---|
| 2014 | An Efficient Multichannel Linear Prediction-Based Blind Equalization Algorithm in Near Common Zeros ConditionabstractThis letter proposes an efficient multichannel acoustic channel equalization method under insufficient channel diversity conditions. To overcome an ill-posed problem caused by near common zeros (NCZs) conditions between different channels, a regularization method that restricts the filter norm has been investigated. However, direct application of this method to the linear-predictive multi-input equalization (LIME) method is not effective. To address this situation, this letter puts forth a novel method to disregard the erroneous term of the LIME solution matrix and to increase forced channel diversity (FCD). The accuracy of the proposed equalization filter is compared to that of the conventional regularization method. Experimental results confirm that the NCZs problem can be solved by adopting the proposed methods. Jae-Mo Yang, Hong-Goo Kang |
IEEE Signal Process. Lett. | 1 |
| 2014 | Online Speech Dereverberation Algorithm Based on Adaptive Multichannel Linear PredictionabstractThis paper proposes a real-time acoustic channel equalization method that uses an adaptive multichannel linear prediction technique. In general, multichannel equalization algorithms can eliminate reverberation if they meet the following specific conditions including: the co-primeness between channels and sufficient filter length. It also requires the characteristic of correct channel information, however, it is difficult to estimate accurate acoustic channels in a practical system. The proposed method utilizes a theoretically perfect channel equalization algorithm and considers problems that may arise in the actual system. Linear-predictive multi-input equalization (LIME) is also an appropriate attempt at blind dereverberation by assuring the theoretical basis. However, a huge computational cost is incurred by calculating the large dimensions of a covariance matrix and its inversion. The proposed equalizer is developed as a multichannel linear prediction (MLP) oriented structure with a new formula that is optimized to time-varying acoustical room environments. Moreover, experimental results show that the proposed method works well even if the channel characteristics of each microphone are similar. The results of experiments using various room impulse response (RIR) models, including both the synthesized and real room environments, show that the proposed method is superior to conventional methods. Jae-Mo Yang, Hong-Goo Kang |
IEEE ACM Trans. Audio Speech Lang. Process. | 1 |