The 7th International Conference on
Electronics, Communications and Networks
Nov. 24-27, 2017, National Dong Hwa University, Hualien, Taiwan


Invited Speaker---Prof. Tetsuya Shimamura


Graduate School of Science and Engineering, and Information Technology Center, Saitama University, Japan

Biography: Tetsuya Shimamura is a Professor of Graduate School of Science and Engineering, and Information Technology Center at Saitama University in Japan. He was Dean of Information Technology Center at Saitama University in 2014 and 2015. In 1995 and 1996, he joined Loughborough University, UK, and The Queen’s University of Belfast, UK, respectively, as a visiting Professor. His research interests are in digital signal processing and its applications to speech, audio, image and communication systems. He has published over 90 refereed journal articles and 220 international conference proceedings papers. He is an author or co-author of eight books, and a member of the organizing committee of several international conferences. He has received IEEE Pacific Rim Conference on Communications, Computers and Signal Processing, Gold Paper Award, in 2012, WSEAS International Conference on Multimedia Systems and Signal Processing, Best Paper Award, in 2013, and IEEE IFOST, Best Paper Award, in 2014. Also, he is a recipient of Journal of Signal Processing, Best Paper Award, in 2013, 2015, and 2016.

Speech Title: Recent Modulation Detection and VoIP Speech Communication
Abstract: Two topics related with telecommunications are discussed. In our lab, we focus on the transmitter and receiver, and the media data transmitted over the channel. In my talk, adaptive modulation in wireless communication systems is introduced first. According to noise conditions over the channel, modulation schemes are able to be adjusted to achieve an efficient and effective signal transmission. Some recent techniques are shown with blind modulation detection. Multi- antenna cases are also suggested. Secondly, as one of the applications of signal transmission, internet based speech communication, VoIP, is included in my talk. The quality of the transmitted speech over VoIP is often degraded, sometimes it is difficult for us to distinguish the transmitted sound at the receiver side. That is why packet loss concealment techniques are utilized. The cause of packet loss and remedies for it are discussed, and recent waveform prediction approaches for speech signals are introduced and demonstrated.