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In this paper we propose low-delay techniques for coding 7 kHz speech using subband coding together with code-excited linear predictive coding (CELP). We compare the use of separate as well as joint index codebooks. Specifically, the joint index subband CELP (JISBC) algorithm is found to provide good quality with processing delay in the range of 2.375 ms to 3.375 ms at corresponding bitrates of 16 kbit/s to 8 kbits/s.

The performance of a synchronised call handoff algorithm called the 'hierarchical optimisation handoff algorithm' (HOHA) is analysed. New results are presented with the derivation of an analytical model which describes the behaviour of the HOHA and provides an upper bound on its performance.

An approximate analysis for a dynamic-channel-assignment (DCA) scheme with handoffs is proposed. In a DCA scheme, there is no fixed relationship between channels and cells. Channel usage in a cell is subject to cochannel-interference constraints owing to the shared use of the same channel in nearby cells. This causes exact analysis of DCA schemes to be highly complex even in the absence of handoffs. An approximate analysis for a DCA scheme, called the ordered dynamic-channel-assignment (ODCAR) scheme, is proposed which takes into consideration handoff calls. The analysis results are compared with simulation results for a linear-highway microcellular system. The analysis proposed is simple and shows good approximation to the simulation results for the system parameters used. It is shown, also, that the ODCAR scheme performs close to a full-access system.

The paper examines several pitch estimation and adaptation algorithms in forward and backward adaptive CELP systems. The authors compare open-loop and closed-loop pitch estimation structures, the use of integer and non-integer delay pitch prediction, and the implementation of block adaptive and sample recursive pitch tracking algorithms. Forward adaptive systems are found to perform better with a closed-loop structure in conjunction of non-integer delay pitch prediction. However, temporal and noise feedback problems suggest that open-loop structures with hybrid pitch tracking are better suited for backward adaptive systems

An analytical model to describe the statistical multiplexing performance in a variable partition duplexing (VPD) multiple access scheme is described. The analytical and simulation results are shown to agree closely for a variety of conditions.

The effect of queuing on an ordered dynamic channel assignment with reassignment (QODCAR) scheme in a highway microcellular environment are studied. Results show that queuing improves channel use by approximately 9-43%, and significantly lowers the probability of call failure. >

A hybrid approach in determining the excitation vector in a low-delay code excited linear predictive coder is proposed. By a judicious division of the composite excitation vector into long-term and short-term components, and the use of switched quantisation, substantial improvement in coding quality is obtained.

The authors propose an ‘integrated packet reservation multiple access’ (IPRMA) protocol for transmitting both speech and data information. While speech users are allowed contend for reservation slots on a frame-by-frame basis, data users may reserve multiple slots across a frame to increase throughput. The protocol includes a priority mechanism which ensures that speech users have greater access to idle slots since speech packets have a more demanding delay constraint. The proposed scheme is shown to provide equitable access to channel resources for both types of users yielding improvements in overall system performance while significantly increasing data throughput compared to a system without data packet reservation.

In a highway microcellular structure, accidents and road works result in localised congestions contributing to asymmetrical teletraffic load situations. A dynamic load-sharing scheme based on a reconfigurable overlapped cellular coverage is investigated for various teletraffic conditions. System performance in terms of the probabilities of new calls blocked and handoff failure are improved significantly with this scheme.

The performance of a low bit-rate digital image codec based on a product vector quantiser (VQ) is examined in a noisy transmission channel environment. The VQ codec is a separating gain/shape type providing good data compression at about 1.25 bits per pixel. For small image subblock sizes, there is sufficient interblock correlation in the mean, standard deviation and shape information to facilitate crude error detection mechanisms to be implemented. We reinforce these error detection capabilities with a subblock boundary continuity testing algorithm. Erroneous subblocks are then estimated by extrapolating intensity contours from neighbouring subblocks. Using such an error detection and correction technique, noisy recovered VQ images can be considerably improved. At a bit error rate of 0.01, we obtained an improvement in the normalised mean square error of about 0.013, or equivalently about 6 dB in signal-to-noise ratio.

We present three methods of weighting N-bit μ-law Pulse Code Modulation (PCM) with binary modulation, and evaluate their performances theoretically for transmission over Rayleigh fading and Gaussian channels. The digital modulation methods considered are noncoherent frequency shift keying and coherent phase shift keying. Ideal selection combining and ideal maximal ratio combining diversity techniques are employed when the transmission is over fading channels. Weighting System 1 is the conventional weighted pulse code modulated system, where the bits in every PCM word have the same weighting profile. Weighting System 2 has 2N unique weighting profiles, while System 3 has an addendum to System 2, where every bit in a particular word is further weighted by a unique multiplicative factor. For Rayleigh fading channels and the encoder operating at an input level that provided maximum signal-to-quantization noise-ratio, we obtained gains in overall signal-to-noise ratio (s/n) over unweighted μ-law PCM of 3, 4.5, and 6 dB for Systems 1, 2, and 3, respectively. When the systems were used in conjunction with Gaussian channels, the corresponding gains were 10, 12, and 17 dB, for a channel s/n of 10 dB. In addition to the theoretical results, we conducted computer simulations using four concatenated speech sentences transmitted via our weighted μ-law PCM systems over mobile radio channels. The simulation performances were in good agreement with our theoretical results, which were based on input signals having an exponential distribution.

In pulse code modulation of speech, transmission errors cause changes in the statistical properties of the received samples. Recently published work describes computer simulations of a difference detection and correction (DDC) system which detect errors on the basis of these changes. The system examines the differences between adjacent received samples. If a particular difference exceeds a threshold, which is dependent on the rms value of a block of 64 differences, the quantized sample responsible for this large difference is deemed to be erroneous. A nonlinear filter of the median type is then introduced as a corrector. We now present a comprehensive description of the hardware realization of this system, and present measured results of signal-to-noise ratio (snr) as a function of bit error rate, input power and the error criterion for narrow-band white noise input signals. An improvement of 6 dB in snr is achieved for error rates between 0.2 and 2.0%. The snr results are consistent with subjective impressions of the quality of speech processed by the DDC system.

Transmission errors in μ-law pulse-code-modulated systems are capable of inflicting considerable distortion in the recovered analog signals. We analyze this distortion when the transmission is over radio channels, and when either noncoherent frequency shift keying or coherent phase shift keying (CPSK) modulation is employed. We consider both Gaussian and Rayleigh fading channels, and for the latter we investigate the effect of bit scrambling prior to transmission and two types of diversity. We determined the gain in overall signal-to-noise ratio due to using the minimum distance code compared to the conventional folded binary code for low values of input signal power. In addition to the theoretical results, we present computer simulations of μ-law pulse-code-modulated encoded speech transmitted over the same channels, using CPSK modulation, bit scrambling, and diversity where appropriate, i.e., for the same conditions as employed in the derivation of our theory.

Packet communication systems cannot, in general, guarantee accurate and prompt delivery of every packet. The effect of network congestion and transmission impairments on data packets is extended delay; in voice communications these problems lead to lost packets. When some speech packets are not available, the simplest response of a receiving terminal is to substitute silence for the missing speech. Here, we explore techniques for replacing missing speech with wave-form segments from correctly received packets in order to increase the maximum tolerable missing packet rate. After presenting a simple formula for predicting the probability of waveform substitution failure as a function of packet duration and packet loss rate, we introduce two techniques for selecting substitution waveforms. One method is based on pattern matching and the other technique explicitly estimates voicing and pitch. Both approaches achieve substantial improvements in speech quality relative to silence substitution. After waveform substitution, a significant component of the perceived distortion is due to discontinuities at packet boundaries. To reduce this distortion, we introduce a simple smoothing procedure.

The teletraffic performance of a highway microcellular digital mobile radio system having an oversailing macrocell that spans many microcells is presented. The microcellular cluster is composed of concatenated segments of the highway where each segment is a microcell, typically 500-2000 m in length, with the base stations located at lamp-post elevations. A narrowband time-division-multiple-access arrangement supporting ten channels per carrier and one carrier per base station is used. The teletraffic analysis assumes there are n-up and n-down lanes, and that the vehicular speeds conform to a truncated Gaussian distribution whose mean speed is 100 or 50 km/h when the vehicular traffic is free-flowing or in traffic-congested conditions, respectively. >

To increase the capacity of blank and burst transmission of control information in cellular communication systems, the authors explore the application of a speech segment reconstruction technique. The transmitter selects a blanking interval on the basis of a segment classification measure, while the receiver replaces the missing speech with an estimate based on previously received speech. Using this approach, approximately 3.2 data bursts per second, each lasting 16 ms may be sent without degrading speech quality.

This paper describes and analyses a new priority CSMA/CD protocol for multichannel local area networks. In this scheme, low priority messages can be transmitted by using the CSMA/CD protocol only when the number of idle channels exceeds a certain value. Following a general description of the proposed priority protocol, this paper analyses the protocol by first formulating the various state transition probabilities of the network. The delay and throughput characteristics of the various priority messages are then calculated and presented. It is found that the performance of this simple scheme is comparable to the priority schemes studies in Reference 8 for CSMA/CD networks. Also, in order not to starve off low priority stations, the largest number of idle channels when the transmission of low priority packets becomes inhibited should not be more than about one third of the total number of channels.

This paper is composed of two parts: firstly to bring home to readers a comprehensive and authoritative introduction of the achievements in the field of video surveillance, specifically, of the software algorithms. We review the state-of-the-art techniques ranging from the preliminary visual processing of foreground detection, to mid-level object tracking, and highlevel action recognition, which play central roles in automated surveillance systems for a wide variety of applications. Secondly, we demonstrate our pioneering surveillance architecture based on IP camera, media server and characterized by scalability and shareability. To our knowledge, such a kind of surveillance framework is introduced for the first time.

A method for embedding data into speech signals without recourse to bandwidth expansion is proposed. Sampled speech is assembled into contiguous blocks of N samples and the Discrete Fourier Transform (DFT) is performed on each block. All the phase components in the message band, or the last J components in this band, are discarded when unvoiced or voiced speech is present, respectively. The data is introduced in the place of these rejected phase components, being +π/2 for a logical 0 and −π/2 for a logical 1. The magnitude of the coefficients associated with the data-carrying phase components are scaled to guard against data errors resulting from channel noise. The inverse DFT yields the transmitted sequence. The receiver performs the inverse process, stripping off the data and replacing it with random phase values. For an average transmission rate of approximately 1 kb/s and a channel signal-to-noise ratio of 30 dB, the bit error rate was 5.5 × 10−4, and the average signal-to-noise ratios for voiced and unvoiced speech were 24 and −3 dB, respectively. However, the unvoiced sounds were perceived with negligible distortion owing to the preservation of their magnitude spectra. Modest error-correction codes can be used to reduce the bit error rate to 10−4 while maintaining the same recovered speech quality, provided the average transmitted bit rate is decreased to ≃500 b/s.

In this paper, we propose DINPeer middleware to overcome limitations in current peer-to-peer (P2P) overlay systems. DINPeer exploits a spiral-ring method to discover an inner ring with relative largest bandwidth to form a DINloop (Data-In-Network loop). DINPeer further integrates DINloop with P2P overlay network via node state and routing algorithm. The key features of DINPeer include using DINloop to replace a multicast rendezvous point and turning DINloop into a cache to achieve data persistency. Simulations show that DINPeer is able to optimize P2P communication in a number of ways, such as when the size of the DINloop is capped within a limit, it can achieve a better performance than native IP multicast and P2P overlay multicast systems.

Packet communication systems cannot, in general, guarantee accurate and prompt delivery of every packet. The effect of network congestion and transmission impairments on data packets is extended delay; in voice communications these problems lead to lost packets. This paper describes techniques for replacing missing speech with waveform segments from correctly received packets in order to increase the maximum tolerable missing packet rate. After presenting a simple formula for predicting the probability of waveform substitution failure as a function of packet duration and packet loss rate, we introduce two techniques for selecting substitution waveforms. One method is based on pattern matching and the other technique explicitly estimates voicing and pitch. Both approaches achieve substantial improvements in speech quality relative to silence substitution.

Micro-sensor human motion capture has shown its potentials because of its ubiquity and low cost. One of the biggest challenges in micro-sensor motion estimation is the drift problem caused by integration of angular rates to obtain orientation. To reduce the drift, existing algorithms make use of gravity and earth magnetic filed measured by accelerometers and magnetometers respectively. Unfortunately, body segment acceleration and environment magnetic disturbance produce strong interferences to the gravity and earth magnetic field measurement respectively. This paper presents a novel sensor fusion algorithm for drift-free orientation estimation, where a quaternion-based complementary Kalman filter is designed. To optimize the performance under interference, this filter fuses gyroscope, accelerometer and magnetometer signals adaptively based on their information confidence, which are evaluated by computing their interference level. The proposed algorithm showed least error compared with the existing methods in the quantitative experiments, and its effectiveness was also verified by the stable and accurate human motion estimation.

A robust speech communications system over a digital mobile radio channel is considered. We examine a system using 12 Kbit/s embedded subband coding of speech and rate compatible punctured convolutional (RCPC) codes with generalised Viterbi decoding for combined error detection and correction. System performance is improved by exploiting the frame-toframe redundancy in the subband energy profile used for adaptive bit allocation in the subband coder. Unreliable subband side information about the energy profile is identified through a two-stage process comprising the channel error detector and a source-based subband energy profile error detector. The subband side information deemed unreliable is replaced with the estimates based on frame-to-frame redundancy. This improved error detection capability in the subband side information allows a judicious reassignment of channel error protection bits to the subband main information, leading to an overall improvement in system performance. Using a realistic simulation model of a digital mobile radio communication system, an improvement of up to 4.4 dB in recovered speech segmental SNR is obtained at a channel SNR of 6 dB, compared with a system with only channel error protection.

Source-matched digital transmission techniques are proposed to maximize robustness, recovered signal quality, as well as bandwidth efficiency. Miki et al. (1993) proposed a pseudo-analog transmission technique where adaptive differential pulse code modulation (ADPCM) samples were transmitted using nonsymmetric skewed differential phase shift keying (SDPSK). The SDPSK transmission format linearly maps the ADPCM samples to the signal constellation points so that transmission errors due to adverse channel conditions result in proportional distortion on the recovered source signal. The performance of such a system was evaluated using computer simulations without a comprehensive supporting theory. In this paper we analyze the performance of transmitting PCM samples using skewed phase shift keying. Both theoretical and simulation results are presented which are found to corroborate well. Optimal skewing angles are obtained for various quantizer resolutions for both the Gaussian channel and for Rician fading channels. The source-matched digital transmission schemes considered in this paper have possible applications in a subclass of wireless personal communication systems (PCS) where low delay is an important requirement. This implies operating at relatively high channel signal-to-noise ratios without or with limited channel coding and interleaving.

A new scheme to transmit data and analogue speech simultaneously over an analogue transmission line is presented. The transmitter simply combines speech and modulated voice-band data signals together while the receiver uses an adaptive cancelling technique to recover these signals from the received composite. The scheme works for phase-dispersive channels and without bandwidth expansion.

A sequency difference detection and correction (s.d.d.c.) system is described which enables the partial correction of transmission errors in a Walsh-Hadamard transform image to be achieved without channel coding. Using a first-order two-dimensional random Gaussian Markov field as the image, the percentage mean-square error in the recovered signal is reduced with the aid of the s.d.d.c. system by two orders of magnitude for transmission error rates

International audience; In this article, we provide mechanisms enabling the backup path computation to be performed on-line and locally by the Points of Local Repair (PLRs), in the context of the MPLS-TE fast reroute. To achieve a high degree of bandwidth sharing, the Backup Path Computation Entities (BPCEs), running on PLRs, require the knowledge and maintenance of a great quantity of bandwidth information (non aggre-gated link information or per path information) which is undesirable in distributed environments. To get around this problem, we propose a distributed PLR (Point of Local Repair)-based heuristic (DPLRH) which aggregates and noticeably decreases the size of the bandwidth information advertised in the network while maintaining the bandwidth sharing high. DPLRH is scalable, easy to be deployed and balances equitably computations on the network routers. Simulations show that with the transmission of a small quantity of aggregated information per link, the ratio of rejected backup paths is low and close to the ideal.

International audience; We propose and discuss an overlay architecture relying on a mobile ad hoc network, called Arigatoni on wheels (Ariwheels for short). More specifically, Ariwheels is a virtual network organization that is designed for a vehicular network underlay environment. It provides efficient and transparent service advertising and retrieves services carried by on-board and roadside nodes. The paper outlines application scenarios for Ariwheels and evaluates them through simulation in a realistic vehicular environment.