Low bit-rate speech coders for multimedia communication

The International Telecommunications Union (ITU) has standardized three speech coders which are applicable to low-bit-rate multimedia communications. ITU Rec. G.729 8 kb/s CS-ACELP has a 15 ms algorithmic codec delay and provides network-quality speech. It was originally designed for wireless applications, but is applicable to multimedia communications as well. Annex A of Rec. G.729 is a reduced-complexity version of the CS-ACELP coder. It was designed explicitly for simultaneous voice and data applications that are prevalent in low-bit-rate multimedia communications. These two coders use the same bitstream format and can interoperate. The ITU Rec. G.723.1 6.3 and 5.3 kb/s speech coder for multimedia communications was designed originally for low-bit-rate videophones. Its frame size of 30 ms and one-way algorithmic codec delay of 37.5 ms allow for a further reduction in bit rate compared to the G.729 coder. In applications where low delay is important, the delay of G.723.1 may be too large. However, if the delay is acceptable, G.723.1 provides a lower-complexity alternative to G.729 at the expense of a slight degradation in quality. This article describes the attributes of speech coders such as bit rate, complexity, delay, and quality. Then it discusses the basic concepts of the three new ITU coders by comparing their specific attributes. The second part of this article describes the standardization process for each of these coders     

Undersampled boundary pre-/postfilters for low bit-rate DCT-based block coders.

It has been well established that critically sampled boundary pre-/postfiltering operators can improve the coding efficiency and mitigate blocking artifacts in traditional discrete cosine transform-based block coders at low bit rates. In these systems, both the prefilter and the postfilter are square matrices. This paper proposes to use undersampled boundary pre- and postfiltering modules, where the pre-/postfilters are rectangular matrices. Specifically, the prefilter is a "fat" matrix, while the postfilter is a "tall" one. In this way, the size of the prefiltered image is smaller than that of the original input image, which leads to improved compression performance and reduced computational complexities at low bit rates. The design and VLSI-friendly implementation of the undersampled pre-/postfilters are derived. Their relations to lapped transforms and filter banks are also presented. Two design examples are also included to demonstrate the validity of the theory. Furthermore, image coding results indicate that the proposed undersampled pre-/postfiltering systems yield excellent and stable performance in low bit-rate image coding.     

On selection of system bit-rate in the mobile multipath channel

A simple analytic model of the multipath fading channel is presented. For a given echo density and distance of communication, it is shown that as the system bit-rate is increased, the probability of unacceptable distortion will rapidly increase. As the bit-rate is reduced, flat fading will increase. In between these two cases there exists a bit-rate for which the mnltipath components can be exploited by an adaptive receiver to improve performance. The probability of this event is shown to have a maximum value for a particular bit-rate. It will also be shown how the maximum acceptable bit-rate depends on range, receiver complexity, and propagation characteristics.     

Video rate control using a radial basis function estimator for constant bit-rate MPEG coders

Compressed video is a source of bursty traffic in communication networks whose data rate needs to be controlled within the available channel capacity, particularly, when it is transmitted via a fixed rate channel. Since the video rate is nonstationary and bursty at large-scene variations in a statistical sense, we propose a feed-forward, estimator-based rate control scheme associated with spatio-temporal activity features (STAF) for MPEG video encoders. This information is used to estimate the video rate of input picture frames. The estimated video rate enables the future buffer occupancy to be calculated and permits the encoder to adapt the quantisation step size to limit the increase or decrease in video rate due to dramatic scene variation. The current and future occupancies are used in a nonlinear quantiser control scheme to determine an appropriate quantisation step size depending on them. The novelty of this technique is that the nonlinear prediction and the nonlinear quantiser control are combined to achieve effective feed-forward video rate control, particularly, for realistic video containing various scene variations. In this paper, we highlight the innovative structure of the scheme and evaluate the performance of rate control algorithms with heuristic, linear and nonlinear rate estimators in the framework of the MPEG2 test model 5 video encoder. The performance measures are the occupancy of a two-frame delay buffer and peak signal-to-noise ratio (PSNR) for video quality.     

Quality of service and bandwidth efficiency of cellular mobile radio with variable bit-rate speech transmission

The performance of an adjustable source/channel codec in a cellular mobile-radio environment is investigated. The speech transmission rate and the amount of forward error correction change in response to changing channel conditions. The channel rate is constant at 32 kb/s, and when the channel is good all of these bits are used for speech transmission. In intermediate and poor channels the speech rate is 24 or 16 kb/s, and the remaining channel symbols are used for forward error correction. Relative to conventional transmission this approach offers an improved grade of service. For example, the outage rate (the proportion of "poor or worse" communications) goes from nine percent with fixed-rate to three percent with variable-rate transmission. Alternatively, this improved grade of service can be exchanged for higher bandwidth efficiency. The fixed-rate system (with nine percent outage) has 23 users per cell. With 52 users per cell the outage of the variable-rate system is only six percent.     

Evaluation of a diversity system for u.h.f. mobile radio

A single-receiver diversity system for use in the lower u.h.f. mobile-radio band has been built. The results of field trials show the usefulness of such a system in reducing the effects of fading in builtup areas.     

A switched diversity receiving system for mobile radio

A low cost switched diversity receiving system has been developed for use in UHF-FM mobile radio. The input of a single receiver is switched back and forth between two antennas upon command from a signal level sensing logic circuit. The system has been measured on simulated Rayleigh fading channels and has been found to give a significant improvement to both voice and data signals.     

Frequency allocation for a mobile radio telephone system

Several ways of allocating frequencies efficiently are suggested that are based on the use of Latin squares. A Latin square of order n is a square array of n rows and n columns, and involving n symbols each of which occurs n times within the square in such a way that all the different symbols occur once in each row and once in each column. This approach provides two alternative ways of arranging the transmitters for a mobile radio telephone system, one rectangular and the other close-packed hexagonal. In both cases optimal patterns for frequency allocation result. A solution using the least-possible number of frequencies for each case is given, as well as one that is ideal in the sense that all the transmitters that surround any particular one are provided with distinct frequencies     

An error-correcting system for mobile radio data transmission

An error-correcting system for mobile radio data transmission with improved reliability and simple implementation is presented here. The new rate one-half code absolutely corrects two errors within 12 consecutive bits, while the (15, 7, 2) Bose-Chaudhuri-Hocquenquem (BCH) code corrects two errors within 15 bits and Hagelbarger's code corrects two errors within 14 bits. Error propagation in the feedback majority logic decoder is discussed, and it is proved empirically that the new code does not propagate infinite errors. In order to correct burst errors, a 12-column interleaving is proposed for fading channels.     

Design study for a CDMA-based third-generation mobile radio system

This paper focuses on a CDMA design study for future third-generation mobile and personal communication systems such as FPLMTS and UMTS. In the design study, a rigorous top down approach is adopted starting from the most essential objectives and requirements of universal third-generation mobile systems. Emphasis is laid on high flexibility with respect to the implementation of a wide range of services and service bit rates including variable rate and packet services. Flexibility in frequency and radio resource management, system and service deployment, and easy operation in mixed-cell and multioperator scenarios are further important design goals. The system concept under investigation is centered around an open and flexible radio interface architecture based on asynchronous direct-sequence CDMA with three different chip rates of approximately 1, 5, and 20 Mchip/s. The presented CDMA system concept forms the basis for an experimental test system (testbed) which is currently under development. This experimental system concept has been jointly established by the partners in the European RACE project R2020 (CODIT). The paper describes the radio transmission scheme and appropriate receiver principles and presents first performance results based on simulations     

Transmitter multiplexing system in UHF mobile radio

At a base station of a UHF mobile radio telephone system in a large city, many transmitters are used. In such a case it is not practical to use an individual antenna for each transmitter from the viewpoint of system economy and interference. Common use of one antenna for many radio-frequency channels is suitable and transmitter multiplexing devices should be used for this purpose. A transmitter multiplexer has been developed which can be used for multiplexing more than twelve 100-W transmitters in the 450-MHz band with a channel spacing of 150 kHz. The multiplexer consists of circulators, cavity resonators, and junction boxes. The principal features of this multiplexer are H101mode cavity resonators which are ultrastable with respect to the variation of environment conditions such as temperature and humidity. The resonators are temperature compensated to maintain the variation of resonant frequency within 5 kHz at the ambient temperature variation of 40°C. The voltage standing-wave ratio (VSWR) of the multiplexer, looking from the transmitter side, is less than 2.0 and the insertion loss is less than 3 dB at the operating frequency. Design objectives, over-all performance, and characteristics of cavity resonators and circulators are also presented.     

Computer simulation of a mobile packet radio system

A software simulator for a mobile packet radio system is described. Bitwise coherent addition of interfering DPSK-packets can be expected, resulting in a smaller throughput than achieved with incoherent addition, but higher than in a nonfading channel     

Data transmission for mobile radio

During the last few years data communication has become important in the mobile radio services. This tutorial paper introduces the reader to the basic concepts of data transmission. Typical transmission channel configurations, various modulation methods, signal spectra, and bit error probabilities when transmitting data over a mobile radio channel are described.     

Single-receiver diversity system for reducing the effect of fast fading in mobile radio

A 2-aerial diversity system suitable for combating fast fading is described. It is easily applied to a conventional v.h.f. or u.h.f. area-coverage receiver, and is therefore of particular interest in the mobile-radio field. Some initial results from laboratory tests are given.     

Evaluation of 16 kbit/s digital voice transmission for mobile radio

In this paper 16 kbit/s digital voice transmission with conventional channel spacing of 25 kHz, employing a 16 kbit/s adaptive delta modulation (ADM) coder-decoder (CODEC) is evaluated. The main characteristics of narrow-band digital FM modulation schemes, such as tamed FM, Gaussian filtered minimum shift keying (GMSK), four-level FM and phase locked loop-quaternary phase shift keying (PLL-QPSK), are compared by laboratory tests. Digitized voice quality in a digital channel incorporating a 16 kbit/s ADM CODEC and GMSK coherent detection was compared with voice quality of a conventional analog FM channel. Bit error ratio (BER) performance is shown to depend primarily on demodulation schemes. Digital voice quality is inferior to that of analog voice with an opinion score difference of about 0.5 in fading environments. This kind of digital voice transmission will be applicable for those systems that require high security at an expense of speech quality.     

Capture probability in a mobile packet radio system

The probability of q_i of successful packet reception when i users transmit simultaneously in a mobile packet radio system is shown to decrease monotonically with i for a number of commonly used capture and spatial distribution models, with no fading. Examples of both noiseless and noisy systems in which q_i is not monotonically decreasing with i are also given     

A dynamic space division multiplex mobile radio system

The dispersed array mobile radio system represents a radical new approach to the problem of spectrum congestion in the land-mobile radio service. This new approach provides a very substantial improvement in spectrum utilization over the present mode of operation and at a cost to the user of about 20 percent less. Spectrum efficiency is achieved by a dynamic combination of space, time, and frequency diversity. Cost reduction is achieved because the mobile radio unit is a compact low-power short-range all solid-state transceiver.     

An integrated voice/data system for mobile indoor radio networks

A centralized, integrated voice/data radio network for fading multipath indoor radio channels is proposed and analyzed. The packets of voice and data are integrated through a movable boundary method. The uplink channel access uses a framed-polling protocol whereas the downlink uses a time-division multiple-access (TDMA) scheme. This system dynamically switches between two transmission rates and uses multiple antennas to maximize the throughput in the fading multipath indoor environment. Throughput and delay characteristics of the system are analyzed using four different techniques. The results are compared with those of Monte Carlo computer simulations. A simple relationship between the number of voice terminals and the throughput of the data traffic are derived for an upper bound of 10-ms delay for the data packets