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Dive into the research topics where Shigenobu Minami is active.

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Featured researches published by Shigenobu Minami.


international solid-state circuits conference | 2000

A 60 MHz 240 mW MPEG-4 video-phone LSI with 16 Mb embedded DRAM

Tsuyoshi Nishikawa; Masafumi Takahashi; Mototsugu Hamada; Toshinari Takayanagi; Hideho Arakida; Noriaki Machida; Hideaki Yamamoto; Toshihide Fujiyoshi; Osamu Yamagishi; T. Samata; Atsushi Asano; Toshihiro Terazawa; Kenji Ohmori; Junya Shirakura; Y. Watanabe; Hiroki Nakamura; Shigenobu Minami; Tadahiro Kuroda; Tohru Furuyama

A 240 mW single-chip MPEG-4 video-phone LSI with a 16 Mb embedded DRAM is fabricated in a 0.25 /spl mu/m CMOS, triple-well, quad-metal technology. The chip integrates a 16 Mb DRAM and three dedicated 16 b RISC processors with dedicated hardware accelerators that serve as an MPEG-4 video codec, a speech codec, and a multiplexer. It also integrates camera, display, and audio interfaces required for a video-phone system. It consumes 240 mW at 60 MHz operation, which is only 22% of the power dissipation of a conventional design. A variable threshold voltage CMOS (VTCMOS) technology is employed to reduce standby leakage current to 26 /spl mu/A, which is only 17% of the conventional CMOS design.


international conference of the ieee engineering in medicine and biology society | 2013

Blood pressure estimation from pulse wave velocity measured on the chest

Sawa Puke; Takuji Suzuki; Kanako Nakayama; Hirokazu Tanaka; Shigenobu Minami

Recently, monitoring of blood pressure fluctuation in the daily life is focused on in the hypertension care area to predict the risk of cardiovascular and cerebrovascular disease events. In this paper, in order to propose an alternative system to the existed ambulatory blood pressure monitoring (ABPM) sphygmomanometer, we have developed a prototype of small wearable device consisting of electrocardiogram (ECG) and photopelthysmograph (PPG) sensors. In addition, it was examined whether blood pressure can be estimated based on pulse wave transit time (PWTT) only by attaching that device on the surface of the chest. We indicated that our system could also sense tendency of time-dependent change of blood pressure by measuring pulse of vessel over the sternum while its propagation distance is short.


Journal of the Acoustical Society of America | 1997

Stereo voice transmission apparatus, stereo signal coding/decoding apparatus, echo canceler, and voice input/output apparatus to which this echo canceler is applied

Shigenobu Minami; Osamu Okada

According to this invention, a stereo voice transmission apparatus for coding and decoding voice signals input from a plurality of input units includes a discriminating means for discriminating a single utterance mode from a multiple simultaneous utterance mode, a first coding means for coding the voice signal when the discriminating means discriminates the single utterance mode, a first decoding means for decoding voice information coded by the first coding means, a plurality of second coding means, arranged in correspondence with the plurality of input units, for coding the voice signals when the discriminating means discriminates the multiple simultaneous utterance mode, and a plurality of second decoding means, arranged in correspondence with the plurality of second coding means, for decoding pieces of voice information respectively coded by the plurality of second coding means.


international symposium on circuits and systems | 2000

A scalable MPEG-4 video codec architecture for IMT-2000 multimedia applications

Masafumi Takahashi; Tsuyoshi Nishikawa; Hideho Arakida; Noriaki Machida; Hideaki Yamamoto; Toshihide Fujiyoshi; Yoko Matsumoto; Osamu Yamagishi; T. Samata; Atsushi Asano; Toshihiro Terazawa; Kenji Ohmori; Junya Shirakura; Yoshinori Watanabe; Hiroki Nakamura; Shigenobu Minami; Tohru Furuyama

A scalable MPEG-4 video codec architecture is proposed to achieve low power consumption and high cost-effectiveness for IMT-2000 multimedia applications. The MPEG-4 video codec consists of a 16-bit multimedia-extended RISC processor and dedicated hardware accelerators, which bring about both low power consumption and programmability. The proposed architecture is extended and applied for the development of two MPEG-4 LSIs. One is an MPEG-4 video codec LSI, which performs an MPEG-4 video encoding and decoding at 15 frames per second with quarter common intermediate format. The other is an MPEG-4 audiovisual LSI, containing three 16-bit RISC processors and a 16-Mbit embedded DRAM, executes the major functions of 3GPP 3G-324M video telephony for IMT-2000 applications. By introducing the optimization of the embedded DRAM configuration, clock gating technique, and low power motion estimation, the MPEG-4 audiovisual LSI consumes only 240 mW when it activates MPEG-4 video SP@L1 codec, the AMR speech codec, and the H.223 annex B multiplex at 60 MHz clock rate.


international conference on acoustics, speech, and signal processing | 1995

A stereophonic echo canceler using single adaptive filter

Shigenobu Minami

The paper proposes a new stereophonic echo canceler which can be realized by using single adaptive filter. Since most of conversations in telecommunication are single talking, the pseudo-stereophonic echo canceler (PST-EC), which is a monaural echo canceler and has only been applied to pseudo-stereophonic speech sound produced by attenuating and delaying monaural sound, is applicable to stereophonic communication. To cope with double talking, stereophonic speech sound is classified into strongly a correlated component (single talking) and a less correlated component (double talking). Therefore, PST-EC and echo suppressors can be applied to the former and the latter component, respectively.


international conference on acoustics, speech, and signal processing | 1990

Stereophonic ADPCM voice coding method

Shigenobu Minami; Osamu Okada

A new stereophonic voice coding method is presented, where a conventional ADPCM method is applied after reducing correlated signals between right and left microphone output signals by an adaptive filter called a cross-channel correlation canceller. To evaluate the method, computer simulations were carrier out by using actual stereophonic voice sound. Simulation results showed that about 6-10-dB improvement of segmental S/N was attainable by reducing cross-channel correlation using the cross-channel correlation canceller. The method was actually implemented as a stereophonic ADPCM codec by using two Toshibas DSP chips. By means of the codec, 3.4-kHz stereophonic voice sound is transferred at 64 kbps. Subjective tests of the codec were carried out on 33 listeners. The test results showed that the proposed method provides more listener satisfaction than double-channel conventional 32 k-ADPCM (CCITT G.723).<<ETX>>


electronic components and technology conference | 2014

Heterogeneous system integration pseudo-SoC technology for Smart-health-care Intelligent Life Monitor Engine & Eco-system (Silmee)

Hiroshi Yamada; Yasuhiro Sato; Nobuhiro Ooshima; Hiroyuki Hirai; Takuji Suzuki; Shigenobu Minami

Heterogeneous devices integration technologies constitute one of most promising driving forces of system integration as they enhance the system performance of electronics products. Currently, many technologies for heterogeneous devices integration have been reported [1]-[2]. The heterogeneous devices integration technologies that have been reported are mainly implemented for SoC (System on Chip) by applying the advantages of process compatibility between the devices. However, it has been impossible to integrate them in the case that the processes are incompatible. Also, many integration technologies applying SiP (System in Package) technology with the interposer substrate have been reported. However, SiP technology, it has been impossible to achieve high integration density comparable to that of monolithic integrated SoC because the interposer substrate occupies a large area in SiP. Accordingly, development of an advanced heterogeneous devices integration technology is required. In the previous work, pseudo-SoC realized by a wafer-level heterogeneous devices system integration technology, which incorporates MEMS and CMOS-LSI, has been studied to verify the validity [3]-[6]. In the work, the flexible pseudo-SoC for mobile terminal application was also verified [5]. Furthermore, in the previous work, high-density pseudo-SoC of AFE (Analog Front End) for Smart-health-care Intelligent Life Monitor Engine & Eco-system (SilmeeTM) application was developed [7]-[8]. In the application, 2 kinds of pseudo-SoC of AFEs which are for ECG (Electrocardiogram) and for Pulse were developed. However, developed pseudo-SoC of AFE has not sufficient density because the 2 kinds of pseudo-SoC of AFE occupy a large area in the Silmee engine. This paper describes the pseudo-SoC which integrated AFE circuit of ECG and Pulse in one microchip for Silmee application that incorporates several types of heterogeneous devices.


international conference on communications | 1992

CCITT H.261 compatible mixed bit rate coding of video for ATM networks

Shigenobu Minami

A mixed bit rate (MBR) video coding method is presented for asynchronous transfer mode (ATM) networks which has communication compatibility between constant bit rate (CBR) video codecs and variable bit rate (VBR) codecs. In MBR video coding, essential information is encoded at a CBR by using a conventional CBR coding method such as H.261 and enhancement information is encoded at a VBR to attain constant image quality. To realize the MBR coding of video, the conventional embedded pulse code modulation (PCM) quantization is extended by applying the requantization method only to the least significant bit (LSB) of the quantizer output. In the EX-EMB PCM method, the most significant part and least significant part (MSP/LSP) separation of embedded PCM and an additional threshold for the requantizer are dynamically determined to attain a desired distortion. The MBR codec with EX-EMB PCM was evaluated by computer simulations. The simulation results showed that the proposed method was effective not only for improving video quality but also it prevented the degradations of image quality at the scene change by increasing the bit rate for the VBR channel.<<ETX>>


brazilian symposium on computer graphics and image processing | 2012

Multi-Modal Acoustic Echo Canceller for Video Conferencing Systems

Mario Gazziro; Guilherme Almeida; Paulo Matias; Hirokazu Tanaka; Shigenobu Minami

In video conferencing, if two people talk at the same time howling noises can appear due to failures in the echo cancellation system. In order to reduce these noise situations, a novel AEC (Acoustic Echo Cancellation) system is proposed by using audio and video information. Through image processing techniques we have developed a new DTD (Double Talk Detector) based on video information (video-DTD). A novel Acoustic Echo Cancellation system was successfully developed, integrating audio and video information. In simulated results using data from a real scenary the misalignment of the new AEC (with video-DTD) was smaller than a traditional AEC.


international symposium on intelligent signal processing and communication systems | 2009

WARP-AEC: A stereo acoustic echo canceller based on W-Adaptive filters for Rapid Projection

Shigenobu Minami; Hirokazu Tanaka; Miki Haseyama

It is difficult for stereo acoustic echo cancellers to converge toward real stereo echo path characteristics in the case of strong cross-channel correlation in right and left channel signals. To overcome this problem, we propose the WARP (W-Adaptive filters for Rapid Projection) method, which uses a monaural adaptive filter in parallel with a stereo adaptive filter. In this method, stereo echo path estimation is realized by projecting from two monaural estimated echo path characteristics at a far-end talker change. Stereo echo path estimation error, which is gradually decreased in conventional methods, is quickly improved at a far-end talker change by adopting the WARP method. This method has been evaluated by using 5 talker positions stereo signals with/without 10% level drift in right and left channels. The results showed that there was a 15–20dB improvement in estimation error at a far-end talker change compared with estimation errors in conventional methods, such as MC-NLMS and Affine projection method with p=3 in the case of WGN, and that there were more than 10 dB echo path estimation error improvement and 20–30 dB residual echo level improvement in the case of voice source.

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