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Jaeuk Byun

dblp:296/4703 · DBLP profile ↗
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4ranked-venue papers
2as first author
4since 2021 · last 2023
0000-0002-2974-7058ORCID · corroborated

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

Graphics, computer vision, multimedia, augmented reality and games · 3 · 1 first-author · 3 since 2021Artificial intelligence and machine learning · 1 · 1 first-author · 1 since 2021

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
1 paper
Speech recognition and synthesis · 100%

Topics — the 4 heaviest of 4, each with the papers that count most for it

TopicWeightPapersLastEvidence papers
Natural language and speech › Speech recognition and synthesis › speech separation
monaural speech separation
0.512021
Monaural Speech Separation Using Speaker Embedding From Preliminary Separation · IEEE ACM Trans. Audio Speech Lang. Process. 2021
Natural language and speech › Speech recognition and synthesis › speaker recognition
speaker embedding
0.512021
Monaural Speech Separation Using Speaker Embedding From Preliminary Separation · IEEE ACM Trans. Audio Speech Lang. Process. 2021
Natural language and speech › Speech recognition and synthesis
speaker recognition
0.512021
Monaural Speech Separation Using Speaker Embedding From Preliminary Separation · IEEE ACM Trans. Audio Speech Lang. Process. 2021
Natural language and speech › Speech recognition and synthesis
speech separation
0.512021
Monaural Speech Separation Using Speaker Embedding From Preliminary Separation · IEEE ACM Trans. Audio Speech Lang. Process. 2021

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

speaker embedding · 0.5conv-tasnet · 0.5affine transformation · 0.5SuDoRM-RF · 0.5
YearPublicationVenuePosition
2023 An Empirical Study on Speech Restoration Guided by Self-Supervised Speech Representation
abstract
Enhancing speech quality is an indispensable yet difficult task as it is often complicated by a range of degradation factors. In addition to additive noise, reverberation, clipping, and speech attenuation can all adversely affect speech quality. Speech restoration aims to recover speech components from these distortions. This paper focuses on exploring the impact of self-supervised speech representation learning on the speech restoration task. Specifically, we employ speech representation in various speech restoration networks and evaluate their performance under complicated distortion scenarios. Our experiments demonstrate that the contextual information provided by the self-supervised speech representation can enhance speech restoration performance in various distortion scenarios, while also increasing robustness against the duration of speech attenuation and mismatched test conditions.
Jaeuk Byun, Youna Ji, Soo-Whan Chung, Soyeon Choe, Min-Seok Choi
ICASSP1
2023 Diffusion-Based Generative Speech Source Separation
abstract
We propose DiffSep, a new single channel source separation method based on score-matching of a stochastic differential equation (SDE). We craft a tailored continuous time diffusion-mixing process starting from the separated sources and converging to a Gaussian distribution centered on their mixture. This formulation lets us apply the machinery of score-based generative modelling. First, we train a neural network to approximate the score function of the marginal probabilities of the diffusion-mixing process. Then, we use it to solve the reverse time SDE that progressively separates the sources starting from their mixture. We propose a modified training strategy to handle model mismatch and source permutation ambiguity. Experiments on the WSJ0_2mix dataset demonstrate the potential of the method. Furthermore, the method is also suitable for speech enhancement and shows performance competitive with prior work on the VoiceBank-DEMAND dataset.
Robin Scheibler, Youna Ji, Soo-Whan Chung, Jaeuk Byun, Soyeon Choe, Min-Seok Choi
ICASSP4
2021 Time-Domain Speaker Verification Using Temporal Convolutional Networks
abstract
Recently, speaker verification systems using deep neural networks have been widely studied. Many of them utilize hand-crafted features such as mel-filterbank energies, mel-frequency cepstral coefficients, and magnitude spectrograms, which are not designed specifically for the speaker verification task and may not be optimal. Recent releases of the large datasets such as VoxCeleb enable us to extract the task-specific features in a data-driven way. In this paper, we propose a speaker verification system that takes the time-domain raw waveforms as inputs, which adopts a learnable encoder and temporal convolutional networks (TCNs) that have shown impressive performance in speech separation. Moreover, we have applied the squeeze and excitation networks after each TCN block to apply channel-wise attention. Our experiments on the VoxCeleb1 dataset demonstrate that the speaker verification system utilizing the proposed feature extraction model outperforms previously proposed time-domain speaker verification systems.
Sangwook Han, Jaeuk Byun, Jong Won Shin
ICASSP2
2021 Monaural Speech Separation Using Speaker Embedding From Preliminary Separation
abstract
In speech separation, the identities of the speakers may be an important cue to discriminate speeches in the mixture and separate them better. A few recent researches used the speaker embedding as an additional information, but they often require prior information about the target speaker or used noisy speaker embedding extracted from the mixture signal. In this article, we propose monaural speech separation that utilizes the speaker embedding in the later separator blocks, which is extracted from the intermediate separated results obtained by the early stages of the separator network. The later blocks in the separator networks consisting of repeated blocks such as the fully-convolutional time-domain audio separation network (Conv-TasNet) or the successive downsampling and resampling of multi-resolution features (SuDoRM-RF) are modified to take the speaker information as a form of affine transformation or addition to the original input tensor. The experimental results showed that the proposed methods significantly improved the performances of existing separation systems with a moderate number of additional parameters.
Jaeuk Byun, Jong Won Shin
IEEE ACM Trans. Audio Speech Lang. Process.1