GaAs digital IC for FM demodulator

A middle-scale GaAs digital IC for a pulse-counting FM demodulator has been designed and fabricated for use in microwave communication systems. The IC is constructed with the normally-on (depletion) MESFET logic with epitaxial active layers. It exhibits a high-speed pulse train with a rise time of 150 ps and with an output voltage of 2.3 V into a 300-Ω load. Output pulsewidth can be altered by the external voltage for tuning the IC to give the highest demodulation sensitivity. By merely implementing a low-pass filter at the output of the chip, the pulse-counting FM demodulator can be realized at high frequencies where the use of Si IC's has never before been possible. We estimate the demodulation sensitivity to be 20 mV/MHz at the microwave IF band.     

全数字高阶QAM解调系统的设计

给出了一种适合高阶256QAM的全数字QAM解调方案，提出自适应改变环路带宽的极性判决载波恢复算法以及基于CMA算法和SGA算法的盲均衡算法，并且结合内插滤波器对整个解调系统进行了优化。算法用FPGA实现并成功应用到高阶QAM接收机中。     

A compact digital beamforming SMILE array for mobile communications

A compact spatial multiplexing of local elements (SMILE) scheme smart antenna array with adaptive beamforming is presented. Low-noise amplifiers are implemented as switching elements to maintain a low system noise figure and allow fast switching. The switching scheme effectively reduces N RF channels to one, reducing the amount of costly RF hardware by a factor of N. The sampling rate must be higher than the signal bandwidth based on the Nyquist criterion to ensure proper restoration of the original signal. Measured data for destination of arrival estimation, beamforming, and digital data recovery demonstrate the capability and benefits of digital beamforming with this architecture.     

A novel algorithm and architecture for adaptive digital beamforming

A novel algorithm and architecture are described which have specific application to high performance, digital, adaptive beamforming. It is shown how a simple, linearly constrained adaptive combiner forms the basis for a wide range of adaptive antenna subsystems. The function of such an adaptive combiner is formulated as a recursive least squares minimization operation and the corresponding weight vector is obtained by means of theQ-Rdecomposition algorithm using Givens rotations. An efficient pipelined architecture to implement this algorithm is also described. It takes the form of a triangular systolic/wavefront array and has many desirable features for very large scale integration (VLSI) system design.     

A CMOS TV tuner/demodulator IC with digital image rejection

Superheterodyne TV tuners have been implemented in discrete forms using tunable RF and SAW IF filters. Integrating TV tuners in CMOS technology without them is a challenging task to cope with technical issues such as harmonic mixing and image. The image rejection in low- or zero-IF systems has been limited to 30-40 dB by analog imperfections such as I/Q path gain and phase mismatches. A single-chip low-IF TV tuner solution is proposed so that the image can be suppressed digitally using an image cancellation technique based on a complex one-tap LMS signal decorrelation algorithm. Programmable digital filtering and video/sound demodulation make a multistandard TV tuner feasible in the 48-860 MHz VHF/UHF band. The chip has a maximum gain of 63 dB and an input automatic gain control (AGC) range from -15 to 25 dB with 0.85-dB steps. It achieves an image and IF rejection of 60 dB, a peak carrier-to-noise ratio (CNR) of 55 dB, and a peak sound signal-to-noise ratio (SNR) of 44 dB without frequency modulation (FM) de-emphasis. The prototype occupies 6/spl times/6 mm/sup 2/ in 0.25-/spl mu/m CMOS and consumes 1 W at 2.5 V.     

A hybrid adaptive blind equalization algorithm for QAM signals in wireless communications

A hybrid adaptive channel equalization technique for quadrature amplitude modulation (QAM) signals is proposed. The proposed algorithm, which is referred to as the modified constant modulus algorithm (MCMA), minimizes an error cost function that includes both amplitude and phase of the equalizer output. In addition to the amplitude-dependent term that is provided by the conventional constant modulus algorithm (CMA), the cost function includes an additive signal constellation matched error (CME) term. This term can be designed to satisfy a set of desirable properties. The MCMA is compared with the CMA for blind equalization. The performance is measured for wireless channels using both transient and steady-state behavior of the mean square error (MSE). It is shown that MCMA is superior and more robust in low signal-to-noise ratio (SNR) environments. Simulation results demonstrate that using MCMA improves adaptive channel equalization by increasing the convergence rate and decreasing the steady-state mean square error.     

QAM解调器自动增益控制的设计

提出了一种针对数字电视DVB—C系统QAM解调器自动增益控制模块的设计方案，该设计通过前端AGC模块控制高频头和中频放大器增益。补偿信号衰减，通过解调器内部数字AGC精确调整信号电平。     

VLSI circuits for adaptive digital beamforming in ultrasound imaging

For phased-array ultrasound imaging, alternative beamforming techniques and their VLSI circuits are studied to form a fully digital receive front-end hardware. In order to increase the timing accuracy in beamforming, a computationally efficient interpolation scheme to increase the sampling rate is examined. For adaptive beamforming, a phase aberration correction method with very low computational complexity is described. Image quality performance of the method is examined by processing the non-aberrated and aberrated phased-array experimental data sets of an ultrasound resolution phantom. A digital beamforming scheme based on receive focusing at the raster focal points is examined. The sector images of the resolution phantom, reconstructed from the phased-array experimental data by beamforming at the radial and raster focal points, are presented for comparison of the image resolution performances of the two beamforming schemes. VLSI circuits and their implementations for the proposed techniques are presented.     

用于自适应数字波束形成的稳健子阵异步RLS算法

提出了一种修正的递归最小二乘自适应算法--稳健子阵异步递推最小二乘算法(MSARLS)--用于自适应数字波束形成.该算法综合运用稳健估计和子阵异步递推技术.改进后的算法,不但大大减少了运算量,而且增强了算法抗突发强干扰的性能.另还给出了计算的理论分析和计算机仿真结果.     

高性能数字电视QAM初始解调器的VLSI结构设计

本文设计了一种适用于高清晰数字电视（HDTV）接收芯片的全数字正交幅度（QAM）解调器结构。该解调器由数字混频器、数字频率合成器（DDFS）和低通滤波器构成。提出采用最佳平方逼近算法优化DDFS的输出正余弦波形，使得最差情况下的无杂散动态范围（SFDR）小于-75dBc。同时，对低通滤波器的实现方式采用了正交信号复用和系数扩展的方法，优化了滤波器结构，降低了所需电路的面积。     

大型线阵自适应数字波束形成超低副瓣技术

自适应数字波束形成技术是现代阵列天线系统必须采用的关键技术。为了对付强有源干扰,现代相控阵雷达都必须具有自适应的干扰抑制能力。除了对抗有源干扰外,大部分雷达还要求具有强杂波背景下检测目标的能力,这就需要雷达天线具有低或超低副瓣电平。本文针对大型线阵,结合数字波束形成,讨论了在保证自适应干扰置零的前提下,如何控制自适应波束的副瓣电平,从而实现阵列系统的超低副瓣性能。     

Antenna and receiver system with digital beamforming for satellite navigation and communications

To demonstrate and investigate the possibilities of active antennas with digital beamforming, a compact breadboard model operating in C-band with planar patches as radiators is fabricated and measured. It was originally designed as a technology demonstrator for navigation purposes, but will be extended for future use in combined navigation and communication systems. Several parameters for beam position, sidelobe reduction, and ing are set and controlled by a suitable user interface. A special calibration and error-correction procedure is developed to compensate for drift of active elements, mutual coupling of the patches, and finite-ground plane effects. The excellent results demonstrate the versatile features of digital beamforming for satellite navigation and communication systems.     

A frame synchronization method for adaptive array antennas in digital mobile communications

This paper proposes a frame synchronization method for an adaptive array antenna (AAA) used in digital mobile communications. The proposed scheme, which is based on the maximum likelihood estimation (MLE), calculates error signals by subtracting a training sequence convolved with an estimated channel impulse response from the AAA outputs and searches for the timing that minimizes the mean squared errors. Because the proposed scheme can effectively exploit delayed paths of the desired signal, it can improve the synchronization performance on frequency-selective fading channels. Computer simulations show that the proposed scheme can operate properly even in interference-rich fading environments and that it can significantly improve the synchronization performance.     

A dual-tone reference digital demodulator for mobile digitalcommunication

A demodulator applicable to the dual-pilot tone modulation technique is presented. Examination of orthogonal and ISI (intersymbol interference) free pulse shapes leads to the novel demodulator structure. This demodulator is advantageous because the processing is strictly linear, the pilot channel and data channel ideally do not interfere or create self-noise, no differential encoding is required since no phase ambiguity is created with the pilot tone processing, and near ideal bit error probability performance is achieved. This demodulator offers an alternative to the transmitted tone-in-band and the pilot symbol assisted modulation techniques. The author considers π/4-QPSK modulation with 33% excess bandwidth in the isotropic time-varying fading channel as a design example     

A new structure for adaptive broad-band beamforming

The performance characteristics of adaptive array systems with derivative constraints have been investigated by several authors. As a result of the derivative constraints, the array beamwidth in the look direction could be made as broad as desired; however, this increased beamwidth was achieved at the price of reducing array gain. A structure is proposed for an adaptive broadband beamforming system in a manner that can produce a broader beamwidth in the look direction without any derivative constraints imposed on the adaptive processor. Compared to the conventional Frost beamformer with derivative constraints, the computer simulations illustrate the effectiveness of the proposed structure, which broadens the beamwidth in the look direction with only slight sacrifice of array gain     

A projection approach for robust adaptive beamforming

It is well known that calibration errors can seriously degrade performance in adaptive arrays, particularly when the input signal-to-noise ratio is large. The effect is caused by the perturbation of the presumed steering vector from its optimal value. Although it is not as widely known, similar degradation occurs in sampled matrix inversion processing when the covariance matrix is estimated while the desired signal is present in the snapshot data. Under these conditions, performance loss is due to errors in the estimated covariance matrix and occurs even when the steering vector is known exactly. In the paper, a new method based of modification of the steering vector is proposed to overcome both the problems of perturbation and of sample covariance errors. The method involves projection of the presumed steering vector onto the observed signal-plus-interference subspace. An analysis is also presented illustrating that the sample covariance errors can be viewed as a particular type of perturbation error and a simple approximation is derived for the expected beamformer performance as a function of the number of data snapshots. Both analytical and experimental results are presented that illustrate the advantages of the proposed method     

A 5-MHz IF digital FM demodulator

Key requirements for digital frequency-modulation (FM) demodulators are wide spurious-free dynamic range in the intermediate-frequency quantizer, linear-phase passband filtering, and accurate frequency discrimination. The proposed FM demodulator implemented digitally achieves high linearity by numerical differentiation performed at a 112× oversampling rate, suppresses adjacent channels by placing zeros of the SINC function on them, and rejects amplitude-modulation (AM) components by numerical division. A 5-MHz FM demodulator integrated with a fourth-order bandpass delta-sigma front end exhibits 74.7 dB signal-to-noise ratio, -80.7 dB total harmonic distortion, and 61 dB AM rejection within a 9-kHz message bandwidth. The 0.65-μm CMOS chip occupies 3.5×3.5 mm² of active area and consumes 180 mW with 4-V supply when clocked at 20 MHz     

Performance of multi-level QAM with post-detection maximal ratiocombining space diversity for digital land-mobile radio communications

A new type of post-detection maximal ratio combining for space diversity receivers in digital land-mobile communications is proposed. The combining scheme is designed for multilevel quadrature amplitude modulation (QAM) and employs a pilot-symbol-aided fading estimator. It is realized at baseband using a digital signal processing (DSP) technique to simplify the receiver hardware. Computer simulation and laboratory experiments confirm that the proposed diversity scheme achieves a performance very close to that of ideal maximal ratio combining     

基于压缩感知的圆阵自适应数字波束形成算法

根据目标在空间的稀疏性,在圆形面阵的接收端,提出了一种基于压缩感知的自适应数字波束形成算法.该算法在不改变波束性能与天线口径的前提下,可以大大减少实际的阵元数目,是一种新的稀布阵方法.在阵元稀布的情况下,根据压缩感知的压缩采样理论,先用重构算法恢复缺失通道的回波信息,然后利用恢复得到的信号计算自适应权系数,得到理想的自适应数字波束方向图.不同信噪比和干噪比情况下的仿真结果验证了所提算法的正确性和有效性.     

A multilevel QAM demodulator VLSI with wideband carrier recoveryand dual equalizing mode

A 16-/64-/256-quadrature amplitude modulation (QAM) demodulator LSI with an all-digital carrier-recovery loop including a novel phase detector and a fractionally/symbol-spaced equalizer is described. The phase detector, which decides the transmitted symbol from received signal power, detects phase errors up to ±45° and enables the loop to internally eliminate the ±80 kHz carrier-frequency offset. The fractionally spaced equalizer is implemented at the same clock rate as the symbol-spaced equalizer by increasing only multiplexers and flip-flops, though the former theoretically requires an operation twice as fast as the latter. A large scale integrator (LSI) operating at a symbol rate up to 8 MBaud is successfully implemented. The 0.5-μm CMOS three-metal technology is used to implement 880 000 transistors in a die size of 12.87×12.39 mm²