This paper presents a method to control the stereo stage width to provide additional functionality to a stereophonic sound-field recording/reproduction system that uses two microphones and two loudspeakers. The proposed method interpolates or extrapolates the time-frequency representations of the left and right stereo signals according to the desired scaling magnification. A mathematical analysis is introduced that clarifies the scaling ability of the proposed method. Listening tests using a stereo signal consisting of a mixture of two speech signals with different time-lags are performed. The results indicate that the two stereo signals yield almost the same aural impression when the scaling magnification is under two and the following two requirements are satisfied. 1) The time-lags in the stereo signal are scaled strictly before mixing. 2) The time-lags are scaled approximately by the proposed method after mixing. The listening tests also clarified the superiority of the proposed method in terms of the sound quality with respect to conventional methods.

It is well known that the transfer functions for out-of-head sound localization differ with the listener. Our goal is to find a method that easily realizes excellent sound localization without measuring the individual transfer functions. This paper proposes an out-of-head sound localization system that uses adaptive inverse filtering without any measuring preprocesses. This system estimates the inverse ear canal transfer function (ECTF), and can adaptively obtain the transfer function to fit the listener in real-time. This paper also proposes an adaptive inverse filtering method for out-of-head sound localization that differs from the filtered-x method. In addition, the relationship between convergence time and initial value is assessed. It is clarified that the proposed method is more effective, in terms of convergence, if the initial value is the average of many listeners' impulse responses.

The International Telegraph and Telephone Consultative Committee (CCITT) has recommended an algorithm to code wideband speech and music (i.e. 7-kHz audio) with 64 kb/s as an international standard (recommendation G.722). A multimedia multipoint teleconference system using the 7-kHz audio coding standard is discussed. System requirements for teleconferencing are examined, and control procedures for the system are examined. A system using digital leased circuits is presented, and the audio-bridging technique used in this system is discussed.>

Acoustic echo cancellers utilizing adaptive finite impulse response (FIR) filters conventionally incorporate learning identification (LI) also called the normalized least mean square (NLMS) adaptation algorithm. This echo cancellers have a serious problem in that colored signal input and incremental tap length degrade their echo cancellation characteristics. In this paper, a new adaptation algorithm combining reductive correlation and variable loop gain methods is proposed. The proposed algorithm is shown in the simulation to have good convergence characteristics compared to conventional methods. Furthermore, a sub-band echo canceller is proposed in which the input signal is separated into several sub-band signals in the frequency domain. Using this echo canceller, the computation quantity per sampling period in a processing module can be greatly reduced. Finally, the relationship between the overall echo cancellation performance and the characteristics of the sub-band filters is clarified.

This paper describes stabitity and insertion loss to prevent howling or singing from effecting an audio teleconference network. The necessity for studying stability in a Voice Summing Circuit(VSC) implemented an Audio Conference Bridge(ACB) is first discussed. A basic analytic model of the VSC using a digital signal processing technique and two applications of the model are proposed. It is shown through experimentation that the analytic models used in this presentation are valid. The necessary insertion loss in designing an audio teleconference bridge is studied theoretically and experimentally.

Digital data services have been offered by NTT since 1978. The largest part of the entire digital data network cost is attributable to subscriber loop transmission that includes digital service units (DSU's) and office channel units (OCU's). Therefore, it is important to reduce DSU and OCU costs for economizing on digital data services. In order to achieve this purpose, new LSI DSU and OCU have been developed. The LSI is used for a digital signal processing unit, which occupies most of the equipment cost and size. The LSI can be applied to all bearer rates from 3.2 to 64 kbits/s, for NTT digital data services (leased circuit service, circuit switched service and packet switched service). It is highly significant that adoption of the LSI in DSU and OCU makes the following estimations feasible: 1) About 30 to 40 percent Cost reduction 2) About 25 percent size reduction 3) About 30 to 50 percent failure rate reduction 4) About 50 to 70 percent power consumption reduction. Summary The most expensive part of the network that carries NTT Digital Data Services is the subscriber loop transmission system, which includes DSU and OCU. This paper describes the design philosophy for introducing LSI technology in the DSU and OCU for cost reduction. Then the function, construction and evaluation are presented for the newly developed LSI DSU and OCU, based on the above design concept.