Satoshi Miyaji
KDDI
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Publication
Featured researches published by Satoshi Miyaji.
Smpte Journal | 1999
Takahiro Hamada; Satoshi Miyaji; Shuichi Matsumoto
Video compression technologies, such as MPEG-2, are now standardized, and a number of video codecs based on these standards are being developed in many organizations, for use in digital television broadcast and transmission services. The degradation in picture quality produced by video compression is so complicated for human eyes to evaluate that neither a simple signal-to-noise ratio (SNR) nor detailed objective assessment parameters are accurate for this measurement. Studies on picture quality assessment have been conducted based on human visual perception models, which have led to an applicable modeling of the visual process.
international conference on multimedia and expo | 2007
Emi Myodo; Koichi Takagi; Satoshi Miyaji; Yasuhiro Takishima
This paper proposes a novel visual cryptography method that embeds a natural secret image into two natural carrier images with high quality at low computational complexity. The secret image is decoded visually by superposing the two carrier images. For some applications such as entertainment use, all three images should be high quality natural images and low computational complexity is also required. However, in the conventional methods, picture quality and processing speed have been tradeoff relations. To solve this problem, we propose a method which is based on conventional visual cryptography using error diffusion halftoning and introduce an additional feedback mechanism to embed the high quality natural image. Experimental results show that the proposed method can achieve higher quality natural images of a superposed image and carrier images while computational complexity is maintained low when compared with the conventional methods.
international conference on image processing | 2005
Satoshi Miyaji; Yasuhiro Takishima; Yoshinori Hatori
This paper proposes a H.264 rate control method considering picture characteristics and buffer constraints. Selecting the optimum buffer size corresponding to each applications requirement gives better compression quality and functionality. However, conventional methods allocate the number of bits equally to each GOP and do not explicitly consider the buffer size, where picture quality might not be optimal and the buffer size is not optimally determined for applications. This paper provides a bit allocation method that reflects the picture characteristic of each frame while keeping the buffer transition compliant within a given size. In the proposed method, the number of bits to be generated is estimated from the complexity of each frame, and it is adjusted by a scaling coefficient newly introduced in this paper. Thus, the allocated number of bits reflects picture characteristics and fully complies with buffer constraints. Experimental results show that the proposed method improves picture quality and suppresses quality fluctuation.
multimedia signal processing | 2006
Koichi Takagi; Satoshi Miyaji; Shigeyuki Sakazawa; Yasuhiro Takishima
In this paper, we propose an efficient algorithm for conversion of an MP3 stream into AAC. Generally, this kind of conversion, transcoding, requires full decoding and re-encoding. However, the re-encoding based transcoding process may result in degradation of quality and take longer than encoding from a PCM signal. This paper proposes a method to inherit the frame structure and quantization scale from MP3 to AAC. This enables a reduction in the iteration process, which requires the most time in the AAC encoding process, without incurring degradation in quality. Experimental results show that the proposed method can perform bitstream domain transcoding at high speed while maintaining a high level of audio quality
international conference on multimedia and expo | 2007
Koichi Takag; Satoshi Miyaji; Yasuhiro Takishima
This paper proposes a high speed bit rate reduction technique on the compressed domain for AAC audio. There are many kinds of AAC audio devices such as PCs and mobile terminals. In order to adapt to various requirements on audio quality and storage capacity, a fast audio transcoding method from a higher to a lower bit rate would be beneficial for the efficient use of master AAC contents. Although the scalability function of AAC or baseband transcoding, which converts a bitstream by decoding and re-encoding, can be employed for this purpose, lower coding efficiency or higher computation load is induced. To resolve these problems, this paper derives the relationship among MDCT, quantization and scaling, and achieves a fast and high quality bit rate reduction by introducing (1) an adaptive scalefactor control, (2) MDCT gain control and 3) the reuse of side information. Experimental results show that the proposed method is about 1.4 times faster than baseband transcoding while achieving almost the equivalent quality.
Archive | 1997
Satoshi Miyaji; Takahiro Hamada; Shuichi Matsumoto
Archive | 2001
Junichi Kimura; Satoshi Miyaji; Shigeyuki Sakasawa; Norihiro Suzuki; Yasuhiro Takishima; Masahiro Wada; Toru Yokoyama; 正裕 和田; 悟史 宮地; 淳一 木村; 徹 横山; 康弘 滝嶋; 茂之 酒澤; 教洋 鈴木
Archive | 1998
Satoshi Miyaji; Shuichi Matsumoto
Archive | 2006
Satoshi Miyaji; 悟史 宮地
Archive | 2005
Satoshi Miyaji; Yasuhiro Takishima