Two Loop Adaptive Delta Modulation Systems

Two adaptive delta modulation (ADM) coders, one for A/D conversion of speech and the other for fiat spectrum signals, have been developed by incorporating a second adaptive feedback loop around the basic delta modulation (DM) coder. The dynamic range of the coder for integrated spectrum has been extended to more than 35 dB with a signal-to-noise ratio (SNR) > 28 dB, for a pulse-repetition frequency (PRF) of 32 kbits/s only, by using an adaptive circuit consisting of digital/analog slope detector and an FET expander. For flat spectrum signals, similar results with constant SNR over the band are obtained at a bit rate of 40 kbits/s. Thus the coders have performances equivalent to those of a 7-digit PCM, using only 32 and 40 kbits/s, respectively.     

A Performance Study of Delta Modulation Systems for Voiceband Data Signals

In this paper the performance of delta modulation (DM) systems have been studied by computer simulation when the input signal to the DM coder is a voiceband data signal. First, the parameter values of three DM systems, linear DM (LDM), constant factor DM (CFDM), and continuously variable slope DM (CVSD) has been optimized for 4800 bits/s differential phase shift keying (DPSK) signal. Then, the performance of the three DM systems have been studied for ideal and noisy channels. It has been found that the peak signal-to-quantization-noise ratio (SQNR) is nearly the same regardless of coding scheme used, but CFDM yields the widest dynamic range. In a noisy channel, however, CFDM is very sensitive to channel bit errors. Considering the overall performance, CVSD appears to be the best among the three DM systems studied. Also, the performances of DM's have been compared with those of PCM and DPCM systems. In addition, we have studied the effect of DM quantization noise on modem bit error rate by the Monte Carlo simulation method. It is possible to transmit a 4800 bits/s DPSK signal at a bit error rate below 10^-5by CVSD with the rate of 32 kbits/s.     

Quantization Noise in Adaptive Delta Modulation Systems

New analytical formulas for quantization noise of two widely known adaptive delta modulation (ADM) systems, continuously variable slope delta modulation (CVSD) and Jayant's firstorder constant factor delta modulation (CFDM), have been obtained. These formulas are derived based upon the previously obtained results for linear delta modulation (LDM) and adaptive differential pulse code modulation (ADPCM). To verify the results, computer simulation has been done using band-limited Gaussian signals. Close agreements between the formulas and the simulation results have been obtained in the wide range of the input signal level.     

Adaptive Delta Modulation Systems for Video Encoding

This paper describes several adaptive delta modulators designed to encode video signals. One- and two-dimensional ADM algorithms are discussed and compared. Results are shown for bit rates of 2 bits/pixel, 1 bit/pixel and 0.5 bits/pixel. Pictures showing the difference between the encodeddecoded pictures and the original pictures are presented. Results are also presented to illustrate the effect of channel errors on the reconstructed picture. A two-dimensional ADM using interframe encoding is also presented. This system operates at the rate of 2 bits/pixel and produces excellant quality pictures when there is little motion. We also describe and illustrate the effect of large amounts of motion on the reconstructed picture.     

A Discretely Adaptive Delta Modulation Codec

A discretely variable slope delta modulation (DVSD) codec is described, which is suitable for integrated circuit realization. The step size is varied by a pulse number modulation method that does not require a precision digital-to-analog conversion circuit. An adaptation algorithm is discussed, taking into consideration the effect of transmission errors. The quantizer and integrator portion has been fabricated on a monolithic chip using MOS technology. Results obtained from an experimental 32 kbit/s codec demonstrate its excellent performance.     

A Multidigit Adaptive Delta Modulation (ADM) System

A multidigit adaptive delta modulation (ADM) system has been proposed where the error signal, between the input and the approximated signal produced by ADM coder, is coded in an auxiliary encoder. The error in the auxiliary coder is processed by another ADM and so on. The bit rate of each of these coders isf_{r}/Nwhere fris the overall transmission rate andNis the number of coders used. The bit streams are interleaved for transmission and at the receiver they are separated and decoded, and these signals are added and filtered. It is shown that for a given transmission rate, each coder operates at a basic sampling rate of frBsuch thatN_{opt} = f_{r}/f_{rB}gives the optimum number of coders to be used for maximum signal-to-noise ratio (SNR). A bound is derived for the maximum SNR of such a system and is compared with the bounds derived for other predictive coders. The experimental results of a two-digit ADM are presented. An average SNR of 30 dB is obtained with a dynamic range of 32 dB at fr= 32 kbits/s for band-limited noise signals. The SNR increases with the sampling rate at 15 dB/octave, as against 9 dB for a single-digit ADM. The frequency response is good and the variation of SNR with the message frequency of the delta coding system has been improved. The effect of channel errors has also been studied and the performance of the system is found satisfactory.     

Kerdock预编码与自适应调制联合优化性能研究

自适应技术在多输入多数出(MIMO)系统中能提高无线通信系统的可靠性和吞吐量,在无线通信系统中有着广阔的应用前景.设计的Kerdock码书与调制方式的自适应MIMO系统通过有限量化反馈信道采用传输信道容量最大化准则实时选择码字、按比特分配算法选择每个发送符号的调制方式,经仿真实验得出该有限量化反馈预编码与调制技术相结合的自适应系统对误码性能大的改善,并在低信噪比时呈现出好的分集增益特性.     

图象信号的三态ADM编码的研究

本文首次提出了三态自适应增量调制编码方法。通过对图象信号进行编码和滤波的模拟研究,证明了这种三态ADM编码的动态范围比二态ADM编码增大6—10dB,信噪比改善约为1.5—3.5dB,并保持了可实现无乘法滤波的特点。     

Performance Evaluation of Admission Control and Adaptive Modulation in OFDMA WiMax Systems

In this paper, we study the performance of multi-cell OFDMA WiMAX systems, in both downlink and uplink. We calculate analytically the number of collisions when the number of users in each cell is known. We then calculate the QoS indicators (e.g., blocking rates, download time and bit error rates) taking into account the physical layer conditions (modulation, propagation and MIMO), the MAC layer techniques (HARQ and radio resource management algorithms) and the traffic characteristics, in a cross-layer approach. We finally evaluate the impact of using adaptive modulation and coding on the overall performance of the system. This analysis allows us to calculate the Erlang capacity of a WiMAX system. Our numerical applications then show how to choose the best admission control and modulation schemes that extend the Erlang capacity region.      

Feedback-Detection Strategies for Adaptive Modulation Systems

This letter analyzes the performance of an adaptive modulation system, taking into account additive noise and fading on the feedback channel. It is shown that these feedback channel imperfections could significantly degrade the throughput gains of adaptive modulation over nonadaptive transmission. Specifically, feedback errors can result in an outage region in the low signal-to-noise ratio region. Two feedback receivers are proposed: one is based on the finite-state Markov channel model; and the other is a generalized Bayesian receiver. These receivers reduce the outage region due to feedback errors, and they can complement or be used as alternatives to error-control coding schemes.     

Linear, Nonlinear, and Adaptive Delta Modulation

The paper treats linear and nonlinear methods that refine the delta modulation process. These methods improve the quality of the decoded signal at a given transmission rate. Two linear processes are analyzed. The first one matches the source signal to the coder for minimum weighted noise in the decoded signal. The second one influences the signal which is fed to the comparator; it shapes the spectrum of the quantization noise desirably. Two nonlinear methods used in the delta modulation process are discussed. The first achieves a best possible prediction of the future signal from past binary decisions and thus a reduction of the error signal. The methods of the second type make the coder adaptive to variations in the power level of the source signal. This results in a large dynamic range of the coding system. A comparison of the various methods is made and a coding system for speech signals, based on a suitable combination of these methods, is described.     

Performance Analysis of Adaptive Modulation for Precoded MIMO Systems with a GMD Zero-Forcing Transceiver

Geometric mean decomposition (GMD) has emerged as an alternative method to design multiple-input multiple-output (MIMO) transceivers. The MIMO-GMD scheme decouples the MIMO channel into multiple independent links with identical gains. The GMD-based system with zero-forcing decision feedback equalizer (ZF-DFE) is known to minimize the bit error rate (BER) for high signal-to-noise ratios (SNRs). In addition, adaptive modulation has been widely used to enhance the average spectral efficiency (ASE) while maintaining a target BER and transmit power. In this paper, we present an analytic study of the adaptive modulation for GMD-ZF-DFE systems under Rayleigh flat fading correlated channels. In order to adjust the constellation size, the SNR at the equalizer output is sent back to the transmitter. The SNR at the DFE output is a function of the determinant of a Wishart complex matrix. The complementary cumulative distribution function (CCDF) is then an important key to our analysis. To evaluate the performance of the considered system, we use some bounds on the CCDF of the determinant and the trace of a Wishart matrix. Closed-form expressions of the BER, the ASE and the outage probability are derived and compared to Monte Carlo simulation results. Furthermore, we analyze the effect of the channel spatial correlation.     

基于自适应增量调制的语音延时电路设计

传统的脉冲编码调制需要较长的码,其实现结构复杂。文中介绍了增量调制(DM)的原理,说明了它的优缺点,引入了对它改进后的调制方式——自适应增量调制(ADM)。自适应增量调制不仅集成了DM调制实现结构简单的特点,而且大大降低了DM调制中的量化噪声和斜率过载噪声,性能更为优良。根据ADM的原理,设计了采用自适应增量调制原理的语音延时电路,通过流片验证了其优异的性能,实测延时后的噪声电压为-88dBV,总谐波噪声+失真<0.5%。     

A Hierarchical Digital Modulation Classification Algorithm for Adaptive Wireless Communication Systems

Blind modulation classification has been applied in adaptive transmission, non-cooperative communications, and interference identification. This paper propose a novel hierarchical classifier for four digital modulation schemes. The hierarchical classifier is based on quasi-log-likelihood ratio. Its performance is verified via extensive simulations. Simulation results show that the hierarchical algorithm is effective for classification of four SQAM signals. FENG Xiang received the B.S. and M.S. degrees in Electronics Engineering from The Missile College of the Air Force PLA, China, in 1991 and 1994, respectively. He is currently working toward the Ph.D degree at Xidian University, Xi'an, China. His research interests include digital communication systems, digital signal processing, adaptive transmission, and multicarrier modulation. LI Jiandong was graduated from Xidian University with Bachelor Degree, Master Degree and Ph.D in communications and Electronic System respectively in 1682, 1985 and 1991. He is Professor and Dean of School of Telecommunications Engineering, Xidian University. He is a senior member of IEEE and CIE and the fellow of CIC. His research interests include Broadband Wireless Communications, Ad hoc Networks, and Software Radio.     

无线协同中继系统中的自适应调制策略

协同中继技术由于其较易实现及在提高系统性能方面的有效性,成为未来移动通信系统中一项很有竞争力的技术。目前的自适应调制研究局限在放大转发系统的自适应,或者只针对部分链路进行自适应。本研究针对译码转发系统进行了自适应调制策略的设计,并且综合考虑各条链路的信道状况进行发送端与中继端调制方式的自适应选择。通过互信息的方式对目的端不同调制方式的合并信号进行性能预测,以用于调制方式的选择。仿真结果表明本文提出的自适应调制策略能够有效的提高系统吞吐量。     

On Practical Issues of Adaptive Modulation in OFDM Systems

1　Introduction　 Orthogonal Frequency Division Multiplexing(OFDM)[1~7] is a communications technique that divides acommunications channel into a number of equally spaced fre quency bands. A subcarrier carrying a portion of the user in formation is transmitted in each band. Each subcarrier is or thogonal with every other subcarrier, which differentiatingOFDM from the commonly used Frequency Division Multi plexing (FDM). Over the past decade, OFDM has been ap plied in a…     

Voice-Band Data Modem Performance Over Companded Delta Modulation Channels

This paper reports upon the results of tests of the transmission of data over single- and multiple-hop companded delta modulation (DM) systems. The DM coder-decoder (CODEC) was optimized for voice transmission. Modem bit error rate (BER) achievable over the range of 1200-9600 bits/s is presented. A comparison with the performance of pulse code modulation (PCM) is included. The comparison indicates that the two systems are comparable for error-free digital lines but favors the DM system for lines with errors.     

A Modified Adaptive Delta Modulator

This paper presents the results of a study in data compression of adaptive delta modulated video signals. The Song mode ADM is first investigated at a sampling rate of 16 Mbits/s and shown not to produce enough redundancy to warrant entropy encoding. We then present a modified adaptive delta modulator algorithm that operates at a sampling rate of 16 Mbits/s and does produce sufficient redundancy to yield a 40-50 percent data compression by using a simple code on 4 bit data blocks. Other techniques such as field interpolation and direct substitution are shown to increase the possible data compression further without noticeable degradation in the two input images used in this investigation. The effects of channel errors in the transmission of packet video over a computer network are considered. A leaky integrator is used to reduce the effects of channel errors in the data bits. We show that the effects of channel errors can be reduced by field interpolating those packets that can be shown to contain errors.     

A Three-Level Adaptive Delta Modulator

A three-level adaptive delta modulator for image encoding is presented. Its advantage is better tracking of signal levels. The characteristic constants of this modulator are designed to satisfy certain convergence and error properties. Behavioral curves of the modulator for different choices of the characteristic constants are included as well as the reconstructed versions of pictures that were encoded by this modulator.