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IEEE Communications Magazine | 1990

Echo cancellation and applications

Kazuo Murano; Shigeyuki Unagami; Fumio Amano

Practical echo cancellation techniques, in particular, those used in telecommunications, are reviewed. The various situations in which echoes are generated are examined. Echo path modeling techniques and adaptive algorithms for coefficient control are reviewed. Current international standardization activities are discussed, and echo canceler implementation considerations are set forth. These include echo cancelers for telephone circuits, echo cancelers for full-duplex data transmission over voice channels, acoustic echo cancelers, and echo cancelers for ISDN digital loop transmission.<<ETX>>


IEEE Journal on Selected Areas in Communications | 1985

A High Performance LSI Digital Signal Processor for Communication

Yukou Mochida; Kazuo Murano; Toshitaka Tsuda; Hirohisa Gambe; Shigeru Fujii

This paper describes a newly developed CMOS LSI DSP (FDSP3). It has a powerful multiplier, which operates at a rate of 10 M-operations/s, almost twice as fast as the fastest existing LSI DSPs. Some key techniques to attain these high performance characteristics of the DSP are described. Suitable memory capacities of the RAM and ROM were analyzed as a function of the operational capability of the DSP. These were reflected in the design of the FDSP3. To assist in the development of the program, support tools have been developed. A Pascal based cross compiler and an on-line debugging tool are described in some detail.


IEEE Transactions on Communications | 1978

LSI Processor for Digital Signal Processing and Its Application to 4800 Bit/s Modem

Kazuo Murano; Shigeyuki Unagami; Toshitaka Tsuda

This paper describes a fast data processing LSI unit tailored to the digital signal processing (DSP) applications in the field of electrical communications. The results of successful application to the 4800 bit/s modem are also given. The LSI processor discussed here adopts a firmware control scheme to enhance the flexibility and freedom of application and extensively utilizes the pipeline processing technique to attain high speed data handling capability. The various operations encountered in DSP systems are unified into one operation of the type A \times B + C \rightarrow D and the LSI processor is designed to continuously perform this operation, while the data to be operated are transferred sequentially into the processor controlled by exterior firmware. The developed LSI handles 8 bit data at the clock frequency of 1.152 MHz and manages 144 K operations per second (6.9 μs cycle time). The LSI is an N-MOS chip containing 1500 gates and packaged in a 40 pin DIP. The automatic equalizer for 4800 bit/s modem was implemented using two of the developed LSI processors and about 4 K ROM and 1 K RAM memory chips. The measurement on this modem gave the error rate of 10-5at S/N = 17.6 dB and error free phase jitter allowance of 55° p-p. Application of the LSI processor to digital filters for roll-off spectrum shaping and timing signal extraction is also described.


international conference on acoustics, speech, and signal processing | 1986

Frequency domain noise canceller: Frequency bin adaptive filtering (FBAF)

M. Asharif; T. Takebayashi; T. Chujo; Kazuo Murano

In this paper, we are concerned about the frequency domain noise canceller to decrease the number of calculation for its realization. In order to treat the long delay and multi-reflection impulse responses of room, we propose a new method of Frequency Bin Adaptive Filtering (FBAF). Each frequency bin of the FFT data is fed in an independent Tapped Delay Line (TDL) to be processed by the extended complex LMS algorithm. The computer simulation results indicate that the proposed FBAF method has better performance than a simple frequency domain adaptive filter. Especially, when a long term delay is considered the robustness, fast convergency and stability in performance will be obvious. Another merit of FBAF method is the short delay between input and output of the system as a result of utilization of short length window function. The number of calculation is comparable with the one of frequency domain filter and much less than the one of time domain adaptive filter. So that, it is convenient to realize the FBAF method by using a few chips of DSP.


IEEE Communications Magazine | 1990

Technologies towards broadband ISDN

Kazuo Murano; Koso Murakami; Eisuke Iwabuchi; Toshio Katsuki; Hiroshi Ogasawara

It is argued that it is vital to form a sound strategy for developing broadband integrated services digital network (BISDN) technology and services and deploying them in the field, so that smooth migration from the existing network is accelerated. Three major steps in this context are to expand fiber networks into the subscriber loop area to provide broadband capabilities everywhere, to construct a universal digital network that facilitates smooth evolution from the existing network to the broadband network of the future through deployment of (SDH) synchronous digital hierarchy transmission systems, and to integrate both services and network components through introduction of asynchronous transfer mode (ATM) technologies. Development efforts now being carried out at Fujitsu and Fujitsu Laboratories along this line are described, and the impact on network construction and service offerings is indicated.<<ETX>>


IEEE Transactions on Communications | 1982

TDM-FDM Transmultiplexer Using a Digital Signal Processor

K. Wakabayashi; T. Aoyama; Kazuo Murano; Fumio Amano

This paper describes a complete hardware implementation of a 24 channel transmultiplexer employing a digital signal processor. The algorithm for the TDM-FDM translation adopted here is a modification of the FFT and digital polyphase method. The modification is mainly directed towards the reduction of roundoff noise, so that the wordlength of the processor can be minimized. In this case, 16 bits for multiplication with double precision accumulation is found to be sufficient to meet the noise requirements as recommended by CCITT G.792. The above modification enables the implementation of the translation algorithm by a 16 bit parallel type digital signal processor. This processor is designed to be firmware controllable and, thus, has a wide range of applications. The entire 24 channel transmultiplexer comprises 1) four signal processors, each handling 12 channels one way translation including signaling; 2) digital interface part to connect to 1.544 Mbit/s PCM system; and 3) codec and Nyquist filter to connect to two 12 channel FDM systems.


IEEE Journal on Selected Areas in Communications | 1986

Phased Aligned Passive Bus (PAB) Scheme for ISDN User-Network Interface

Hideo Kuwahara; Shigeo Amemiya; Kazuo Murano

This paper proposes an alternative to the passive bus realization of the current CCITT ISDN user-network interface. Our new approach is termed phase aligned passive bus (PAB) scheme. An automatic phase adjusting mechanism using a monitor bit is employed in the terminal equipment. This scheme enables the terminals to operate in complete phase synchronization on a passive bus. The advantages that result from this technique are: 1) Eliminates, in principle, the bus length limitation due to round-trip delay time. If line bit rate of 192 kbit/s is kept, then a practical bus length is about 700 m with no limitation on terminal distribution. 2) Allows various extended bus configurations, such as simple repeatered extention, connection of optical link, and multibranched passive bus. 3) Provides spare bit transmission capability to facilitate intrabus communication and maintenance. As an alternative it keeps the general framework of the current CCITT I-series standard such as B + B + D channel structure and LAP- D protocol. Experiments realizing long passive buses are shown to confirm the feasibility of the proposed scheme. This concept is also shown to be applicable to a 2-wire full duplex user-network interface, advantageous in the existing office environment, employing time compression transmission.


international conference on acoustics, speech, and signal processing | 1983

CMOS LSI DSP and its application to voice band signals

Toshitaka Tsuda; Kazuo Murano; Shigeyuki Unagami; M. Shimada; Hideo Kikuchi; S. Sumi; Y. Miwa

This paper describes a newly developed CMOS LSI DSP and its application to a 32 Kbps ADPCM CODEC and a 4,800 bps data MODEM. The paper first analizes the required memory capacities of ROM and RAM as a function of arithmatic operation capability of DSP. Based on the results, the LSI DSP is developed, which has a proper amount of memory capacities. It has a multiplier which operates at a rate of 1.4 M operations/s. It is also equipped with a versatile I/O interface circuit which is suitable for multi-processors configuration of a system.


IEEE Transactions on Consumer Electronics | 1981

TV Ghost Cancelling by LMS-RAT Digital Filter

Mohammad Reza Asharif; Kazuo Murano; Mitsutoshi Hatori

Recently several methods for automatic TV ghost cancelling are under active research. Generally speaking, these methods are Categorized into two substantial groups: 1) TV ghost cancelling at RF stages including antenna method. 2) Cancellation at video stage.


international conference on acoustics, speech, and signal processing | 1986

A 32kbps ADPCM with improvement in coding characteristics for 9600bps modem signal

Tomoyoshi Takebayashi; Kazuo Murano; K. Yamamoto; H. Mori; Y. Andoh; T. Miyazaki

This paper describes a new 32 kbps ADPCM CODEC which shows high-quality speech coding characteristics and data transmission capability up to 9600 bps. The proposed ADPCM CODEC consists of two coding modes. The first coding mode can code the speech signal and the data transmission signal up to 4800 bps by using the coding algorithm of CCITT recommendation G.721. The second coding algorithm is tuned to handle the 9600 bps modem signal. These two coding modes are automatically switched depending on the input signal to the CODEC. The switching algorithm is based on detection of modem training signal and is found to be highly reliable in practical environments. This algorithm thus provides a means to be compatible with the standard CCITT algorithm for speech and data transmission up to 4800 bps and at the same time be transparent to data transmission at 9600 bps.

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