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한국인을 위한 외국어 발음 교정 시스템의 개발 및 성능 평가
김무중,김효숙,김선주,김병기,하진영,권철홍,Kim Mu Jung,Kim Hyo Sook,Kim Sun Ju,Kim Byoung Gi,Ha Jin-Young,Kwon Chul Hong 대한음성학회 2003 말소리 Vol.46 No.-
In this paper, we present an English pronunciation correction system for Korean speakers and show some of experimental results on it. The aim of the system is to detect mispronounced phonemes in spoken words and to give appropriate correction comments to users. There are several English pronunciation correction systems adopting speech recognition technology, however, most of them use conventional speech recognition engines. From this reason, they could not give phoneme based correction comments to users. In our system, we build two kinds of phoneme models: standard native speaker models and Korean's error models. We also design recognition network based on phonemes to detect Koreans' common mispronunciations. We get 90% detection rate in insertion/deletion/replacement of phonemes, but we cannot get high detection rate in diphthong split and accents.
Convolution Block Attention Module 을 적용한 단일 이미지 생성 모델의 성능 비교 실험
김무중(Kim Mu-Jung),김민정(Kim-Minjeong),유지상(Yoo Ji-Sang),권순철(Kwon Soonchul) 한국통신학회 2022 한국통신학회 학술대회논문집 Vol.2022 No.6
본 논문은 단일 이미지만으로 학습하여 이미지를 생성하여 주목받은 딥러닝 기반 생성모델 SinGAN 의 훈련시간과 파라미터 수가 매우 크다는 한계를 개선시킨 ConSinGAN 모델의 생성기와 판별기의 CNN 구조에 각각 또는 동시에 Convolutional Block Attention Module (CBAM)을 추가하여 학습한 뒤 생성된 이미지를 바탕으로 모델의 성능을 비교하였다. 본 실험은 기존 ConSinGAN 에 CBAM 을 적용하여 생성한 이미지에 대한 정량적 평가를 통해 모델 성능의 개선 여부와 한계를 확인한다.
인터넷 폰 시스템에서 음성 패킷 손실 보상을 위한 동적 부가 전송 알고리즘의 성능 분석
김무중,권철홍 대전대학교 산업기술연구소 2001 산업기술연구소 論文集 Vol.12 No.1
In ITU H.323 teleconference system, the RTP/RTCP protocol is offered to transfer real-time multimedia stream. Both sender and receiver have experience in packet loss and jitter which result from network congestion over Internet. Audio quality over Internet depends on the number of lost packets and on jitter between successive packets. The goal of our study is to improve the speech quality over Internet by adapting the packet loss characteristics of the network and adopting the buffer control management mechanism at receiver. We suggest a dynamic redundant audio transmission mechanism which examines the packet loss rate and uses the feedback information through RTCP.
HMM 및 보정 알고리즘을 이용한 자동 음성 분할 시스템
김무중,권철홍 한국음성과학회 2002 음성과학 Vol.9 No.4
In this paper we propose an automatic segmentation system that outputs the time alignmetn information of phoneme boundary using Viterbi search with HMM (Hidden Markov Model) and corrects these results by an UVS (unvoiced/voiced/silence) classifica-tion algorithm. We selecte a set of 39 monophones and a set of 647 extended phones of HMM models. For the UVS classification we use the feature parameters such as ZCR (Zero Crossing Rate), log energy, spectral distribution. The result of forced alignment using the etended phone set is 11% better than that of the monophone set. The UVS classification algorithm shows high performance to correct the segmentation results.