Speech can be modeled as short bursts of vocal energy separated by silence gaps. During typical conversation, talkspurts comprise only 40% of each party's speech and remaining 60% is silence. Communication systems can achieve spectral gain by disconnecting the users from the spectral resource during silence periods. This letter develops a simple and efficient Voice Activity Detection (VAD) algorithm to work in a mobile environment exhibiting dynamically varying background noise. The VAD uses a classification method involving the full-band energy, ratio of low-band energy to full-band energy, zero-crossing rate, and peakiness measure.

In this paper, we propose a dynamic TDMA with priority-based request packet transmission scheme (D-TDMA/PRPTS) which applies priority-based request packet transmission scheme instead of slotted ALOHA (S-ALOHA). D-TDMA/PRPTS can avoid collisions between voice request packets and data request packets and transmit voice request packets preferentially. This makes D-TDMA/PRPTS enlarge the system capacity for voice users with SAD. We analyze voice packet dropping probability and channel utilization for voice traffic by using an appropriate Markov model. We also present simulation results to verify the analysis and to investigate data performances as well, with the voice-data integrated scenario.

In this paper, we consider an integrated voice and data system over CDMA Slotted-ALOHA (CDMA S-ALOHA). We investigate its performance when multi-code CDMA (MC-CDMA) is applied as a multi-rate scheme to support users which require transmission with different bit rates. Two different classes of data users are transmitted together with voice. Performance measurement is obtained in respect of throughput for data and outage probability for voice. Moreover, we consider the Modified Channel Load Sensing Protocol (MCLSP) as a traffic control to improve the throughput of data. As a result, we show that the MC-CDMA technique is an effective one to obtain good throughput for data users at an acceptable voice outage probability. Furthermore, we show that with MCLSP, the throughput of data can be improved to reach a constant value even at a high offered load of data users.

In this paper, we propose a preemptive priority handoff scheme for integrated voice/data cellular mobile systems. In our scheme, calls are divided into three different classes: handoff voice calls, originating voice calls, and data calls. In each cell of the system there is a queue only for data calls. Priority is given to handoff voice calls over the other two kinds of calls. That is, the right to preempt the service of data is given to a handoff voice call if on arrival it finds no idle channels. The interrupted data call returns to the queue. The system is modeled by a two-dimensional Markov chain. We apply the Successive Over-Relaxation (SOR) method to obtain the equilibrium state probabilities. Blocking and forced termination probabilities for voice calls are obtained. Moreover, average queue length and average transmission delay of data calls are evaluated. The results are compared with another handoff scheme for integrated voice/data cellular mobile systems where some numbers of channels are reserved for voice handoff calls. It is shown that, when the data traffic is not very light, the new scheme can provide lower blocking probability for originating voice calls, lower forced termination probability for ongoing voice calls, and shorter average queue length and less average transmission delay for data calls.

Hierarchical macrocell/microcell architectures have been proposed for future cellular mobile communication. The performance analysis for the hierarchical cellular system becomes an important issue. In this paper, extending the analytical methods from[1][2][8], assuming that the imperfect power control follows log-normal statistics, and employing different attenuation models for macrocells and microcells, the capacity plane and outage probability of the system are examined and quantified with and without perfect sectorization. From the numerical results of parameters of IS-95 protocol, the high user capacity and lower outage probability may be expected in the case of relatively tight power control and narrower overlap between sectors. These results are compared with the previously published CDMA nonhierarchical cellular system estimation. When we employ the hierarchical cellular system, we can increase the user capacity 2.3 times with the same bandwidth 1.25 MHz than the one of the nonhierarchical cellular system.

IP telephony systems are expected to be deployed worldwide in the near future because of their potential for integrating the multimedia communication infrastructure over IP networks. Phone-to-phone connection over an IP network via IP telephony gateways (IP-GWs) is a key feature of the system. In an IP telephony system, a low-bit-rate voice codec is used to improve bandwidth efficiency. However, due to the packet transfer method over the IP network, it is necessary to add packet headers, including IP, UDP, and RTP headers, which increases the header overhead and thus decreases transfer efficiency. Moreover, because there will be large numbers of short voice packets flowing into the IP network, the load on the Internet will increase. We propose voice stream multiplexing between IP-GWs to solve these problems. In this scheme, multiple voice streams are connected between a pair of IP-GWs, enabling multiplexed voice stream transfer. The voice stream multiplexing mechanism can reduce the header overhead as well as decrease the number of voice packets. The voice stream multiplexing we propose is to concatenate RTP packets destined for the same IP-GW at a multiplexing interval period into a single UDP packet. The advantage of this method is that no new additional header is required and the current well-defined H. 323 and RTP standards can be applied with minimum changes. We implemented and tested the system. The results show that the proposed method is effective at reducing both the header overhead and the number of packets. In a typical case, the bandwidth is cut by 40% for eight G. 723.1-encoded voice streams through header overhead reduction and the number of voice packets is also decreased to 1/8. Furthermore, this method can easily be enhanced to a general RTP packet multiplexing method that is applicable not only to an IP-GW but also to other RTP multiplexing and de-multiplexing applications.

Voice activity detection (VAD) is to determine whether a short time speech frame is voice or silence. VAD is useful in reducing the mean speech coding rate by suppressing transmission during silence periods, and is effective in transmitting speech and other data simultaneously. This letter describes a VAD system that uses a neural network. The neural network gets several parameters by analyzing slices of the speech wave form, and outputs only one scalar value related to voice activity. This output is compared to a threshold to determine whether the slice is voice or silence. The mean code transfer rate can be reduced to less than 50% by using the proposed VAD system.

The development of saturation diving in civil and defense applications has enabled man to work in the sea at great depths and for long periods of time. This advance has resulted, in part, as a consequence of the substitution of helium for nitrogen in breathing gas mixtures. However, utilization of HeO2 breathing mixture at high ambient pressures has caused problems in speech communication; in turn, helium speech enhancement systems have been developed to improve diver communication. These speech unscramblers attempt to process variously the grossly unintelligible speech resulting from the effect of breathing mixtures and ambient pressure, and to reconstruct such signals in order to provide adequate voice communication. It is known that the glottal excitation is quasi-periodic and the vocal tract filter is quasi-stationary. Hence, it is possible to use an auto regressive modelisation to restore speech intelligibility in hyperbaric conditions. Corrections are made on the vocal tract transfer function, either in the frequency domain, or directly on the autocorrelation function. A spectral subtraction or noise reduction may be added to improve speech quality. A new VAD enhanced helium speech unscrambler is proposed for use in adverse conditions or in speech recognition. This system, implementable on single chip DSP of current technology, is capable to work in real time.

The next generation mobile communications systems must support multimedia communications services as well as conventional voice service. DS-CDMA is regarded as the most promising candidate, because it is indispensable to cope with multimedia. The system capacity of DS-CDMA system is limited by the total interference level. As a result, in DS-CDMA systems many users suffer very poor communication quality if the total interference level exceeds this limit. Therefore, this paper considers smoothing interference fluctuation using the difference between voice and data in a type of QoS (quality of service). In other words, voice communication is suitable for a loss system because the quality of voice communication is delay-sensitive. On the other hand, data communication is suitable for a waiting system because the quality of data communication is non-delay-sensitive. This paper focuses on a system that applies a circuit switching method for voice traffic and a reservation type packet switching method for data traffic and proposes a data traffic control method. In this proposed data traffic control method, a base station controls data transmission from a mobile station to utilize unused voice traffic resources. As a result, the proposed method achieves highly efficient use of the radio spectra by smoothing interference fluctuation in DS-CDMA systems. This paper evaluates the performance level of the proposed method from a system capacity standpoint. It is shown that the proposed method achieves higher system capacity in voice/data integrated transmission.

For people with serious disability, it is most significant to be able to use the same communication methods, for instance a telephone and an electronic mail system (e-mail), as ordinary people do in order to get a normal life and communicate with other people for leading a social life. In particular, having communications access to an e-mail is a very effective method of communication that enables them to convey their intention to other people directly while at the same time keep their privacy. However, it takes them much time and effort to input an e-mail text on the computer. They also need much support by their attendants. From this point of view, we propose a multi-modal communication system that is composed of a voice recognizer, a pointing device, and a text composer. This system intend to improve the man-machine interface for people with physical disability. In this system, our voice recognition technology plays a key role in providing a good interface between disabled people and the personal computer. When generating e-mail contents, users access the database containing user keywords, and the guidance menu from which they select the appropriate word by voice. Our experimental results suggest that this communication system improves not only the time efficiency of text composition but also the readiness of disabled people to communicate with other people. In addition, our disabled subject on this paper is not able to move his body, legs and hands due to suffer from muscular dystrophy. And he is able to move only his fingers and speak command words with the assistance of a respirator.

This paper presents novel multimedia ATM networks which are capable of transmitting voice data efficiently and unify the switching methods among heterogeneous traffic. Fully ATMized multimedia networks are using fellow cell switches. The proposed assembly method can pack plural calls which have different virtual channel connection (VCC) into one cell. Every call in cells is able to be dynamically rearranged by the fellow cell switch to achieve an efficient use of network resources. The switching functions are supported by shared virtual channel identifier (VCI) cells and fellow cells in it. The fellow cell switch for 622 Mbps links is integrated into a single chip. The multimedia ATM networks including voice transmission can be constructed by the fellow cell switches being attached to the standard ATM switches.

The ATM switching system accommodating the public switched telephone network (PSTN) and narrowband ISDN (N-ISDN) subscribers should ensure the continued support of existing services and applications and guarantee the same quality of voice services for the telephone users. The voice message connection control discussed in this paper is one of the various technical issues for voice services in the interworking function unit, IWF between asynchronous transfer mode (ATM) node and existing synchronous transfer mode (STM) node [2]. We describe the technical points for the implementation of the voice message connection control with the consideration of the development time and cost. And then we discuss several technical problems such as mapping pulse code modulation PCM coded voice data into an ATM cell, different switching operation, keeping performance of the ATM-PSTN interworking system and then present benefits of the voice message connection control processing from the hardware/software point of views.

In this paper, we focus on an integrated voice and data system over a CDMA Unslotted ALOHA and investigate the effect of transmission control for data traffic. We consider Channel Load Sensing Protcol (CLSP) as a transmission control protocol and investigate the effect of the thresholds, which may differ due to the requirement of each medium. As a result, we find that the threshold assigned for data is very effective to improve both performances of the throughput for data and the Erlang capacity for voice users, and also, to correspond to the priority for both media. Consequently, we obtain an optimum threshold for data to make the best use of the total channel capacity.

In this paper, an efficient voice/data integrated access algorithm for future personal communication networks (PCNs) is proposed and analyzed based on an equilibrium point analysis (EPA) method. A practical wireless communication channel may be impaired by noise and multipath distortion, and thus corrupted real-time packets have to recompete immediately in order to be transmitted within the stringent delay constraint. Also, real-time traffic users have to transmit their packets irrespective of the amount of non real-time data messages so that heavy non real-time traffic does not degrade the quality of real-time traffic. In the proposed algorithm, request subslots are distributed in the beginning of every slot to reduce access delay of real-time traffic. Moreover, slots are assigned to real-time traffic first and the remaining idle slots are assigned later to non real-time traffic by using the scheme of contention separation. We analyze the throughput and delay characteristics of this system based on an EPA mothod, and validate their performances by simulations. This scheme can support different quality of services (QoSs) imposed by different services efficiently and show good quality of real-time traffic, especially voice packets, no matter how heavy non real-time traffic is.

This paper proposes a new emphasizing three-dimensional pointing device considering user friendliness and lack of cable clutter. The proposed method utilizes five degrees of freedom via the medium of non-verbal voice of human. That is, the spatial direction of the sound source, the type of the voice phoneme and the tone of the voice phoneme are utilized. The input voice is analyzed regarding the above factors and then taking proper effects as previously defined for human interface. In this paper the estimated spatial direction is used for three-dimensional movement for the virtual object as three degrees of freedom. Both of the type and the tone of the voice phoneme are used for remaining two degrees of freedom. Since vocalization of nonverbal human voice is an everyday task, and the intonation of the voice can be quite easily and intentionally controlled by human vocal ability, the proposed scheme is a new three-dimensional spatial interaction medium. In this sense, this paper realizes a cost-effective and handy nonverbal interface scheme without any artificial wearing materials which might give a physical and psychological fatigue. By using the prototype the authors evaluate the performance of the scheme from both of static and dynamic points of view and show some advantages of look and feel, and then prospect possibilities of the application for the proposed scheme.

This paper deals with the method of integration of voice and data in wireless communication systems. By applying the DSV (Data Steal into Voice) technique to D-TDMA (Dynamic Time Division Multiple Access) systems, this paper presents an MAC (Media Access Control) method of integration of voice and data for the systems such as cellular radios and cordless phones. After a brief review of the D-TDMA scheme and the DSV technique, the protocol called D-TDMA/DSV is described. Then, a static analysis to derive the channel capacity and a dynamic analysis to evaluate the throughput and delay performance are presented. An extension of TFA (Transient Fluid Approximation) analysis is employed in the dynamic analysis. With the same system parameters, the capacity of D-TDMA/DSV is compared with that of the traditional D-TDMA. Under the limitation of the blocking probability required for cellular radios, some numerical examples of dynamic analysis are given to show the throughput and delay performance of the system.

In this paper, we propose DSVMA (Data Steal into Voice Multiple Access) scheme for integration of voice and data in wireless information networks. By using speech activity detectors and effective downstream control signals, DSVMA enables data terminals to transmit multi-packet messages when voice terminals are in silent periods. The S-G (throughput versus offered load) performance of the DSVMA system and the blocking probabilities of both the second generation systems and the DSVMA systems are evaluated by the static analysis. A dynamic analysis of a system with finite number of terminals is also presented using an approximate Markov analysis method. Some numerical examples are given in the paper. As a result, it is shown that DSVMA can improve the channel utility efficiency of a circuit-switched TDMA (Time Division Multiple Access) wireless communication system and is directly applicable for second generation wireless information systems.

Personal Handy Phone (PHP), the Japanese digital cordless telephone system is being developed. The 32kbits/s ADPCM (Adaptive Differential Pulse Code Modulation) codec has been standardized for PHP. This paper describes firstly, the advanced algorithms of a Voice Activity Detection (VAD) function that reduces power dissipation of a digital cordless telephone terminal, secondly, a comfort noise generator operates in conjunction with the VAD and finally, a transmission error control based on the use of the prediction coefficients generated in the ADPCM codec. These proposed algorithms function in the low signal-to-noise ratio (SNR) environment of personal radio communications. The quality of the reconstructed speech after the process is influenced by the VAD decision errors (false detection when no voice is present, or no detection when voice is present) , the similarity of the generated comfort noise to the actual background noise, and the transmission quality. The simulation results of the performance achieved by these algorithms are shown and required loading of the computation are also given.

In our preceding paper, I-ISMA (Idle Signal Multiple Access for Integrated services), a combination of ISMA and time reservation technique, was proposed to transmit an integrated voice and data traffic in third generation wireless communication networks. There, the channel capacity of I-ISMA was evaluated by the static analysis. To fully estimate performance of contention-based channel access protocols, however, we also need dynamic analysis to evaluate stability, delay, etc. Particularly, in systems concerning real-time voice transmission, delay is one of the most important performance measures. A six-mode model to describe an I-ISMA system is set up. With some assumptions for simplification, the dynamic behavior of the system is approximated by a Markov process so that the EPA (Equilibrium Point Analysis), a fluid approximation method, can be applied to the analysis. Then, numerical and simulation results are obtained for some examples. By means of the same analysis method and under the same conditions, the performance of PRMA is evaluated and compared briefly with that of I-ISMA.

Corresponding to the development of B-ISDN, integrated services for data, voice, etc. are imperatively required for the so called third generation wireless communication networks. In this paper, I-ISMA (Idle Signal Multiple Access for Integrated services) is proposed to transmit integrated voice and data traffic from dispersed terminals to a base station. In the system, data packets and the first packets of talkspurts of conversational speeches are transmitted using ISMA protocol over a shared channel while subsequent packets of talkspurts are sent with time reservation technique. The channel capacity of I-ISMA is evaluated and compared with that of PRMA. The region in which I-ISMA has larger capacity than PRMA is figured out. Generally speaking, I-ISMA has larger capacity than PRMA when the duration for transmitting and detecting an idle signal is not too long and the channel is not too congested by the reserved voice transmissions. When we concern real time voice transmission, delay is one of the most important performance measures. Only is a qualitative discussion on delay performance given here. The quantitative evaluation is obtained by the dynamic analysis in our succeeding paper.