Voice Communication System Research Articles

Fundamental concepts for practical software architecture

Architecture of software is a collection of design decisions that are expensive to change. How to identify which design decisions are expensive to change? What are architecture views and which views are needed to adequately describe the architecture of a specific system? How to create and manage software architecture for a product family? This tutorial offers answers to these and other questions that arise in the context of complex software development. We introduce a system of concepts useful in order to understand, design, and evaluate architecture of software intensive systems and system families. Our approach utilizes different software structures in order to control important system qualities related to its development, performance, and evolution. We draw our experience primarily from software embedded in voice and data communication systems. However the same principles can be applied to software architecture in other domains. This tutorial should be useful to engineers and technical managers involved in construction or evaluation of complex software.

A study of evacuation from large retail stores

This paper presents the results obtained from unannounced evacuations of four Marks and Spencer retail stores. By unannounced it is meant that the public and staff had no prior knowledge of the evacuations and were not informed during the evacuations by voice communication systems. The four retail stores participating in the study were chosen to represent three-storey city centre stores and single-storey out-of-town stores in different locations in the United Kingdom. The evacuations were video taped using in-house closed-circuit television cameras (CCTV) orientated to cover pre-determined fields of view and additional portable cameras which were positioned and concealed when the stores were closed for business on the evenings before the evacuations. The evacuating customers were asked to complete a questionnaire by teams of researchers positioned outside, but not impeding, each emergency exit. Data sufficient for occupant characterisation and occupancy profiling for Marks and Spencer stores were obtained. Analysis of the video tapes also yielded information on total evacuation times and pre-movement times for each store. This research provided useful information, which informs the discussion in this paper of many issues relevant to the development of performance based codes, the profession and practice of fire safety engineering.

Validating voice communication requirements using lightweight formal methods

To show that lightweight approaches can facilitate the technological transfer of formal development methods, the authors report on their experience using VDM++ to specify a safety-critical air traffic control voice communication system. Their approach raised both the quality of the informal system specification and the efficiency of the system test suites they used.

Occupant response during a residential highrise fire

On 31 January 1997, a fire occurred in a 25-storey highrise apartment building located in Ottawa, Ontario, Canada. The building contains 296 condominium apartments which were occupied mainly by senior citizens. The fire, which started at 4:30 pm in an apartment on the 6th floor, rapidly burned through the apartment corridor door and spread into the corridor. The fire department arrived promptly on location and extinguished the fire within 10 min of the notification call. On the orders of the fire department, a live announcement was issued, from the building voice communication system, advising all residents to evacuate the building immediately. Many occupants (83%) attempted to evacuate while some (17%) decided to stay in their apartments and initiate protect-in-place activities. During their evacuation, all occupants above the fire floor encountered smoke conditions. Only 54% who attempted to evacuate managed to escape. The others had to return to their apartment (25%) or seek refuge (21%) in a neighbour's apartment. Four of the evacuees had to be treated for smoke inhalation; two seniors suffered heart attacks and one of them died 10 days after the fire. A second person from the apartment of fire origin died 2 months later. This paper discusses the occupants' response and their actions during this fire. Copyright © 1999 Crown in the right of Canada.

Voice quality in new telephony systems

During the development of the wired analog network, engineering teams recognized from the start that voice quality was a primary design driver. As such, projects aimed at understanding speech mechanisms and the human auditory system were a key component of the telephony research executed in the first half of this century. The recent migration from analog to digital (and wired to wireless) voice communication systems has given rise to a new set of signal degradations, and these impairments can have severe perceptual effects. While the ideal system would transmit voice at high bit-rates and with no signal degradation, voice-over-data (VODN) and wireless systems require the use of signal compression in the edge equipment, and the transmission paths are far from ideal. With regard to voice transmission, mainstream communications systems can no longer be thought of as linear and time invariant. The development community is struggling to systematically evaluate the effects of these new impairments, and is beginning to understand that a connection must be made between objective specifications and subjective response. The talk will present a broad overview of the key issues, and will present some thoughts regarding how the communications industry is responding to these interesting problems.

A Multimode and Variable‐Rate Voice Communications System with Source‐Matched Error Protection for Mobile Communications

AbstractThis paper presents a performance evaluation of the multimode Fast Variable Rate CELP coder developed by CSELT S.p.A. in the UMTS Code Division Testbed (CODIT) Project [1] for applications involving channels with time‐varying error characteristics. The radio channel is statistically modelled by one Rician or Rayleigh‐distributed desired signal and several uncorrelated Rayleigh plus log‐normally shadowed interfering signals, propagating according to two different laws: a dual path loss law with a turning point (microcellular environment) and a single path loss law (macrocellular environment). For channels with these error characteristics, each mode of the Fast Variable Rate CELP coder has a particular allocation of bits between source and channel coding stages, leading to an unequal error protection code tailored to suit the different error sensitivities of the source bits. It is shown that, on the basis of the sensitivity evaluation of source bits to errors, according to two different error sensitivity measures. Forward Error Correction whose power is matched to the error sensitivity of the source bits can be used. Increased robustness of source bits to errors can thus be obtained while satisfying bandwidth requirements.

Finline Integrated Trans-receiver at Ka-Band

This paper describes the design and development of Integrated Finline Trans-receiver at Ka-Band. Integrated trans-receiver has been designed for Simplex Voice Communication System. The trans-receiver includes a modulator, duplexer and a mixer integrated together on a single substrate RT Duroid 5880. Two Millimeter Wave Gunn sources, one as transmit source and the other one as Local Oscillator source, in Waveguide are connected externally. The system has been realised and tested with the integrated trans-receiver unit for ASK voice Communication System for a range of 2.5 KM LOS. The system is highly useful for short-range communication systems such as Hand-held Millimeter Wave Radio.

Sound modification for use in simultaneous voice and data communications

A simultaneous voice and data (SVD) communications system includes an SVD modem at either endpoint of the system. Each SVD modem provides an SVD signal to the opposite endpoint, where this SVD signal includes both a voice channel and a data channel. One, or both, of these SVD modems allows the respective user to alter either the voice signal that is provided to that user, or the voice signal that is transmitted from that user, via the voice channel of the SVD signal, to the opposite SVD endpoint.

Noise suppression in digital voice communications systems

Noise is suppressed during pauses and silent periods in conversation, and voiced signals are freely passed in a mobile communications system thereby improving the quality of the transmitted speech. Noise suppression is implemented in such a way that a low level of noise is still allowed to pass to provide presence of the remote speaker; that is, the line is not made completely silent as this may falsely indicate that the connection has been interrupted. The noise suppression has an added feature of decreasing the background noise fractionally when voice is no longer detected. This provides perceptually improved quality of the communication.

Combination head-protective helmet and communications system

A generally hands-free, voice communication system in combination with a head-protective helmet. The helmet protects the wearer's head and the communications system permits voice communications between journeyman personnel, e.g. firefighters, police, military, industrial, hazardous material handling personnel, in relatively close proximity with each other and between journeymen and a group leader; and the group leader's communication system to also permit relatively long-range communication between the group leader and a relatively distant communications center such as a fire engine or distant fire company, base station or repeater.

Performance of Reed–Solomon codes with dependent symbol errors

Reed-Solomon (RS) error-correcting codes are often proposed for communication systems requiring burst-error-correction capabilities. The performance of RS codes is examined for the case when symbol errors are dependent and therefore not random. This situation arises when interleaving is not possible due to delay constraints, such as in most voice-communication systems. The performance is compared with that obtained using a random-error assumption.

CONTROL AND COMMUNICATIONS.

This paper describes the inter-related control and communications systems designed and developed for the Channel Tunnel, where road, railway shuttle, and through train transport services are provided. In addition to the main transport services, there are complex mechanical and electrical systems in the tunnel, and terminal areas, whose operation must also be integrated into the whole transport system operation. The operational control of the tunnel operations is achieved by three inter-related systems: (1) the rail traffic management system (RTM); (2) the terminal road traffic management system (TIM); and (3) the engineering management system (EMS). The paper describes the operational control of these three systems, and then the voice communications and data/voice transmission systems in the tunnel. The voice communication systems are: (1) telephone systems; (2) radio/radiotelephone systems; (3) radio paging systems; and (4) public address systems. They also interface, directly or indirectly, with: (1) the French and UK national public telephone networks; and (2) other external telecommunication networks. There is also a data transmission system (DTS), which is an infrastructure for transmitting telecommunication signals through the tunnel. It has three subsystems, and is designed to survive partial system failure. For the covering abstract see IRRD 877203.

Scientific bases of human-machine communication by voice.

The scientific bases for human-machine communication by voice are in the fields of psychology, linguistics, acoustics, signal processing, computer science, and integrated circuit technology. The purpose of this paper is to highlight the basic scientific and technological issues in human-machine communication by voice and to point out areas of future research opportunity. The discussion is organized around the following major issues in implementing human-machine voice communication systems: (i) hardware/software implementation of the system, (ii) speech synthesis for voice output, (iii) speech recognition and understanding for voice input, and (iv) usability factors related to how humans interact with machines.

Video bandwidth allocation for multimedia teleconferences

To ensure the quality of a multimedia teleconference, it is essential that sufficient bandwidth be allocated for its use. In this paper a conference traffic model is formulated and link level and conference level congestion measures are derived. Motivated by the advantages of sharing transmission resources in TASI related voice communication systems, an analogous transmission policy for conference videos is proposed. The quantification of conference traffic also enables us to set an admission policy so that the network can accommodate as many conferences as possible without violating conference quality constraints. >

Applications for GaAs and silicon integrated circuits in next generation wireless communication systems

Emerging applications for portable wireless voice and data communications systems are requiring increased data rates and functionality. Meeting cost and performance goals requires careful attention to system level design and partitioning such that appropriate technologies are employed in cost-effective solutions. New circuit designs and techniques are required to meet size, power, and regulatory restrictions. This provides an exciting opportunity for GaAs, silicon, and passive component technologies. This review paper will discuss factors influencing the choice of which technology is best suited to a particular application and present several system level architectures of radio-based communication systems. The paper will illustrate appropriate applications of GaAs, silicon, and passive integrated circuit technologies. A summary is given that highlights the relative strengths and weaknesses of each technology to date. >

An intelligent multimode voice communications system for indoor communications

A novel high-quality, low-complexity dual-rate 4.7 and 6.5 kbits/s algebraic code excited linear predictive codec is proposed for adaptive multi-mode speech communicators, which can drop their source rate and speech quality under network control in order to invoke a more error resilient modem amongst less favorable channel conditions. Source-matched binary Bose-Chaudhuri-Hocquenghem (BCH) codecs combined with unequal protection diversity- and pilot-assisted 16and 64-level quadrature amplitude modulation (16-QAM, 64-QAM) are employed in order to accommodate both the 4.7 and the 6.5 kbits/s coded speech bits at a signaling rate of 3.1 kBd. Assuming an excess bandwidth of 100%, in a bandwidth of 200 kHz 32 time slots can be created, which allows us to support in excess of 50 users, when employing packet reservation multiple access (PRMA). Good communications quality speech is delivered in an equivalent bandwidth of 4 kHz, if the channel signal-to-noise ratio (SNR) and signal-to-interference ratio (SIR) of the benign indoors cordless channel are in excess of about 15 and 25 dB for the lower and higher speech quality 16-QAM and 64-QAM systems, respectively, and the PRMA time-slots are sufficiently uninterfered due to using time-slot classification algorithms and due to the attenuation of partitioning walls and ceilings.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Anxiety During MRI-Reply

In Reply. —We are grateful to Dr Persili and colleagues for drawing our attention to three additional published articles concerning anxiety-related reactions during MRI examinations. We apologize for not having captured these references with our search strategy. The 12% incidence of motor unrest or claustrophobia reported by Oberstein et al 1 is within the range found in our review. Of special interest was the correlation shown between final diagnosis and claustrophobic reactions in 21 633 subjects undergoing MRI examinations. This observation may be helpful in screening for subjects at risk. The article by Nozzolillo et al, 2 based on 28 subjects, is also in keeping with findings reported in the publications reviewed in our article. The article by Ellis et al 3 describes a handheld air bellows communication device that may be useful to facilitate patient communication during the test, though modern scanners have built-in voice communication systems.

Digital air/ground communications for air traffic control

The early 1990's communications for air traffic control (ATC) uses analog single channel radios with conventional amplitude modulation (AM) in the very high frequency (VHF) band. To overcome eventual saturation of the current system, a sample next generation ATC communications system has been designed to increase capabilities and provide a graceful transition from the current system. The new ATC communication system must address problems with the modulation format and a balance between increased channel capacity and overall cost. The controller/pilot workloads can be reduced in that the information segments allow for either semi-automatic or fully automatic handoff or frequency change. The principal performance factor is the addition of data, fully integrated with voice, while offering an increase in throughput. The architecture is structured to put priority on the uplink voice messages while offering significant information capacity capabilities for external data sources. When digital data and voice communication systems mature in the ATC environment, a natural evolution to more data traffic and less voice will occur. At that time, a simple restructuring of the channel assignments and priorities could offer increased throughput for connection to ground based data sources such as high capacity routers. >

Underwater voice communications using digital techniques

A digital signal processor (DSP) based voice communication system for divers with a 2400 bit/s transmission rate is described. Each speech signal frame is analysed using linear predictive coding (LPC) and the essential parameters such as amplitude, pitch period and digital filter coefficients are extracted. These parameters are transmitted through a water channel at a rate of 2400 bit/s by employing digital pulse position modulation (DPPM)

Character voice communication system

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