An adaptive channel access control method for CDMA/PRMA protocol is proposed. The proposed method utilizes a load and backlog based access control. Dynamic optimal channel loads by which the required packet loss probability can be satisfied are obtained. The number of contending terminals is also estimated more accurately, using statistical characteristics of source models. Permission probability is then calculated based on the dynamic optimal channel load and the estimated number of contending terminals such that the mean channel load (the mean number of packets transmitted in each time slot) can be maintained at the optimal channel load. By maintaining the mean channel load at the dynamic optimal channel load, the radio channel can be very effectively utilized, satisfying the required packet loss probability. A backlog based data transmission using a mixed mode of contention and reservation mode is also proposed to reduce redundant contention and corruption. Simulations are carried out in an isolated cell environment and a cellular environment. The simulation results show that the system capacity can be improved significantly by the proposed method compared with the conventional permission control methods.

In this letter, an analysis of bit error probability of 4-state soft decision Viterbi decoding is presented. The bit error probability of recursive systematic convolutional code is also derived.

A duration modeling technique is proposed for the HMM based connected digit recognizer. The proposed duration modeling technique uses a cumulative duration probability. The cumulative duration probability is defined as the partial sum of the duration probabilities which can be estimated from the training speech data. Two approaches of using it are presented. First, the cumulative duration probability is used as a weighting factor to the state transition probability of HMM. Second, it replaces the conventional state transition probability. In both approaches, the cumulative duration probability is combined directly to the Viterbi decoding procedure. A modified Viterbi decoding procedure is also presented. One of the advantages of the proposed duration modeling technique is that the cumulative duration probability rules the transitions of states and words at each frame. Therefore, an additional post-procedure is not required. The proposed technique was examined by recognition experiments on Korean connected digit. Experimental results showed that two approach achieved almost same performances and that the average recognition accuracy was enhanced from 83.60% to 93.12%.

In this paper, we show that a priori probabilities of information bits can be incorporated into metrics for syndrome decoding. Then it is confirmed that soft-in/soft-out decoding is also possible for syndrome decoding in the same way as for Viterbi decoding. The derived results again show that the two decoding algorithms are dual to each other.

We have recently developed a programmable composite noise generator (P-CNG) which can easily control noise parameters such as average power, time-based amplitude probability distribution (APD), crossing rate distribution, occurrence frequency distribution and burst duration. Two applications of the P-CNG are demonstrated to show its usefulness. For the first application, Middleton's Class A noise is simulated. A method of setting parameters for Class A noise is demonstrated. The APD of P-CNG output is in good agreement with that of true Class A noise. In the second application, the P-CNG is used for subjective evaluation test (opinion test) of TV picture degradation. Five simple composite noise models with two kinds of APD are used. Other parameters such as average power are kept constant. Experimental results show that the envelope and APD of composite noises do not greatly influence the subjective evaluation. Finally the capabilities of the P-CNG are shown.

In evaluating the teletraffic of mobile communication networks, it is important to model the motion of terminals. In the previous migration model, mobility characteristics of terminals, such as cell dwell time, have been expressed by a single probability distribution. In this paper, we discuss the modeling of the cell dwell time of terminals in each cell. Using measured data we show that cell dwell time differs from cell to cell and follows log-normal distributions rather than conventional exponential distributions.

It have been one open and significant topic for real-time applications to enhance the processing-speed of Boltzmann machines for long time. One effective way of solution of this problem is the augmentation of probability of neurons' state move. In this paper, a novel method, called a rejectionless method, was proposed and introduced into the Boltzmann machines for this augmentation. This method has a feature of independence on the ratio of neurons' state move. The efficiency of this method for speed-up was confirmed with the experiments of TSP and graph problem.

Soft bit speech decoding, as a new approach of error concealment, is applied to the mixed excitation linear prediction (MELP) algorithm. Average residual redundancies of the quantized parameters are exploited in the error concealment process as an a priori knowledge of the source. Results show a significant SNR improvement for parameters decoded using the error concealment scheme.

In this paper, we present an approximate analysis method for computing the call blocking probability and Erlang capacity of CDMA systems. The approximated results provide only a few percent difference from the exact values, while reducing the calculation complexity. For CDMA systems with 3 sectors, we also show that the system performances such as call blocking probability and Erlang capacity can be characterized just with two traffic parameters (the traffic load of the most loaded sector and the sum of traffic loads of the other remaining sectors) instead of three sector traffic loads especially when the required call blocking probability is given less than 2e-2, which makes the traffic engineers manage the system more easily.

This paper proposes a new kind of hidden Markov model (HMM) based on multi-space probability distribution, and derives a parameter estimation algorithm for the extended HMM. HMMs are widely used statistical models for characterizing sequences of speech spectra, and have been successfully applied to speech recognition systems. HMMs are categorized into discrete HMMs and continuous HMMs, which can model sequences of discrete symbols and continuous vectors, respectively. However, we cannot apply both the conventional discrete and continuous HMMs to observation sequences which consist of continuous values and discrete symbols: F0 pattern modeling of speech is a good illustration. The proposed HMM includes discrete HMM and continuous HMM as special cases, and furthermore, can model sequences which consist of observation vectors with variable dimensionality and discrete symbols.

For several years, more and more people are joining the Internet and various kind of packets (so called transaction-, block-, and stream-types) have been transmitted in the same network, so that poor network conditions cause loss of the stream-type data packets, such as voices, which request smaller transmission delay time than others. We consider a switching node (router) in a network as an N-series M/G/1-type queueing model and have mainly evaluated the fluctuation of packet delay time and end-to-end delay time, using the two moments matching method with initial value, then define the delay jitter D of a network which consists of jointed N switching nodes. It is clarified that this network is not suitable for voice packets transmission media without measures.

It is well known that interpath interference (IPI) is a major factor that limits the performance of high data rate transmissions over a variable spreading factor wideband-CDMA (W-CDMA) link since the spreading factor is in general small. An optimum combining scheme suppressing IPI was recently proposed for RAKE reception in [1]. The main contribution of this letter is to present a theoretical model for the outage probability and bit error probability of a RAKE receiver utilizing the optimum combining scheme. Analytical and simulation results are closely matched and show that the optimum scheme provides significant performance improvement compared to the conventional maximum ratio combining (MRC) scheme.

In variable-length coding, the probability of codeword length per source letter being above (resp. below) a prescribed threshold is called the overflow (resp. the underflow) probability. In this paper, we show that the infimum achievable threshold given the overflow probability exponent r always coincides with the infimum achievable fixed-length coding rate given the error exponent r, without any assumptions on the source. In the case of underflow probability, we also show the similar results. From these results, we can utilize various theorems and results on the fixed-length coding established by Han for the analysis of overflow and underflow probabilities. Moreover, we generalize the above results to the case with overflow and underflow probabilities of codeword cost.

Near-optimality of subcodes of the cyclic Hamming codes is demonstrated on the binary additive channel whose noise process is the two-state homogeneous Markov chain, which is a model of bursty communication channels.

We have experimentally measured the propagation characteristics of 60-GHz-band millimeter wave between two vehicles to design of inter-vehicle communication (IVC) system in intelligent transport systems (ITS). Received power and bit error rates of 1-Mbps data transmission between a transmitter mounted on a leading vehicle and two receivers attached on a following vehicle were measured. A two-ray propagation model was devised to calculate the instantaneous propagation characteristics, and these estimations agree well with the measured characteristics. The feasibility of 1-Mbps data transmission between the running vehicles on an actual expressway was demonstrated. The cumulative distribution of received power between the two running vehicles when their height from the road surface fluctuated was also determined from the proposed two-ray propagation model and experimental measurements.

It has been reported that IP packet traffic exhibits the self-similar nature and causes the degradation of network performance. Therefore it is crucial for the appropriate buffer design of routers and switches to predict the queueing behavior with self-similar input. It is well known that the fitting methods based on the second-order statistics of counts for the arrival process are not sufficient for predicting the performance of the queueing system with self-similar input. However recent studies have revealed that the loss probability of finite queuing system can be well approximated by the Markovian input models. This paper studies the time-scale impact on the loss probability of MMPP/D/1/K system where the MMPP is generated so as to match the variance of the self-similar process over specified time-scales. We investigate the loss probability in terms of system size, Hurst parameters and time-scales. We also compare the loss probability of resulting MMPP/D/1/K with simulation. Numerical results show that the loss probability of MMPP/D/1/K are not significantly affected by time-scale and that the loss probability is well approximated with resulting MMPP/D/1/K.

This paper presents a new approach to modeling speech spectra and pitch for text-independent speaker identification using Gaussian mixture models based on multi-space probability distribution (MSD-GMM). MSD-GMM allows us to model continuous pitch values of voiced frames and discrete symbols for unvoiced frames in a unified framework. Spectral and pitch features are jointly modeled by a two-stream MSD-GMM. We derive maximum likelihood (ML) estimation formulae and minimum classification error (MCE) training procedure for MSD-GMM parameters. The MSD-GMM speaker models are evaluated for text-independent speaker identification tasks. The experimental results show that the MSD-GMM can efficiently model spectral and pitch features of each speaker and outperforms conventional speaker models. The results also demonstrate the utility of the MCE training of the MSD-GMM parameters and the robustness for the inter-session variability.

In code division multiple access (CDMA) systems with adaptive antennas, the direction of terminals must be considered when controlling new call admission. This paper proposes a data access control algorithm based on estimated signal-to-interference-plus-noise ratio (SINR) at the output of adaptive antennas. The algorithm estimates SINR for new data call using a response vector of the request packet to determine acceptance or blocking of the new data call. Numerical results show that the combination of transmission technology of adaptive antennas and proposed data access control can effectively increase the capacity of CDMA systems.

In this letter, performance evaluation of a system that combines between Code-Division Multiple Access (CDMA) and ALOHA protocol in multimedia networks is presented. In our analysis, we compare the performance between the two basic techniques of ALOHA protocol, i.e., Slotted-ALOHA (S-ALOHA) and Unslotted-ALOHA (U-ALOHA), when combined with CDMA scheme to support voice and data users operating in same CDMA channel. The quality of service (QoS) required for voice and data media is completely taken care of. We obtain the throughput of data media, and the outage probability for voice considering both voice and data offered loads. Throughput performance of S-ALOHA technique is almost twice of that of U-ALOHA. However, we show in this letter that when we combine CDMA with the two basic techniques of ALOHA to accomplish multimedia transmission, both techniques have almost same performance. And, thus, CDMA U-ALOHA can be a good candidate for multimedia networks.

An appropriate call admission control in the next generation wireless networks is expected to make efficient use of scarce wireless resource and improve quality-of-service, while supporting multimedia services. On one hand, blocking handoff calls is normally more annoying than blocking new calls. On the other hand, blocking new calls reduces resource utilization. More importantly, handoff call arrival rate depends strongly on call holding time. A novel Call-Holding-Time-Based Random Early Blocking (CHTREB) scheme is proposed to achieve the aforesaid goals in a two-tier cellular voice/data network. With CHTREB, new calls are accepted according to some acceptance probability taking into account the call hloding time difference between voice and data calls. An iterative algorithm is developed to calculate performance measures of interest, i.e., new call blocking probability and forced termination probability. First, simulation results are shown to verify analytical results. Then, numerical results are presented to show the robustness of CHTREB. It is found that CHTREB outperforms TR and CHTREB-FAP under both stationary and nonstationary traffic scenarios. Last but not least, the studied 2-tier system is compared with 1-tier counterpart. It is shown that 2-tier system performs better in terms of average number of handoffs per data call.