Identification of cubically nonlinear systems excited by bandpass inputs

Identification of cubically nonlinear systems excited by bandpass inputs using discrete-time samples of the input and output signals is investigated. When excited by a bandpass signal, a nonlinear system may generate an output signal which occupies multiple frequency bands. Therefore, although the input signal can be bandpass sampled without causing aliasing by following the well known bandpass sampling theorem, the output signal must obey more sophisticated bandpass sampling criteria to avoid aliasing. Specifically, when a cubically nonlinear system is excited by a bandpass input signal whose highest frequency is larger than three times its bandwidth, a proper bandpass sampling frequency without causing aliasing in the sampled output signal must be at least 18 times the bandwidth of the input signal. First, a method for identifying the cubically nonlinear system using properly bandpass sampled data is developed. Then, a novel method is proposed, which allows identification of the cubically nonlinear system using data sampled at about twice the bandwidth of the input signal, even though aliasing exists in the sampled output under these circumstances. Compared to conventional methods, the proposed methods have the advantage of requiring a lower sampling rate. This makes the proposed methods highly appreciated in situations where high-speed sampling is unattainable.     

Understanding linearity in wireless communication amplifiers

This paper investigates the linearity of active devices and amplifiers that have modulated input signals. We describe an implementation of a mathematical technique for calculating spectral regrowth due to the nonlinear amplification of modulated signals typically used in wireless communication systems. This technique only requires knowledge of the single-tone gain and phase data as a function of input power and can be applied to any modulated signal given (a) the device or amplifier characteristics do not change significantly over the bandwidth of the input signal and (b) the modulation frequencies are much less than the carrier frequency. Verification of the mathematical technique is presented using examples of measured and calculated spectra for π/4-differential quadrature phase shift keying (DQPSK) personal handy phone system (PHS) and code division multiple access (CDMA) Offset-QPSK personal communication system (PCS) modulation schemes. This calculation technique is essential in determining fundamental tradeoffs between device performance (linearity, output power, gain, and power-added efficiency) and device bias conditions and load impedance     

Blind channel estimation for single‐input multiple‐output OFDM systems: zero padding based or cyclic prefix based?

Orthogonal frequency division multiplexing (OFDM) transmission equipped with multiple receive antennas constitutes a single‐input multiple‐output (SIMO) OFDM system. SIMO‐OFDM systems have been widely used in wireless communications. Compared to those approaches using training sequences, blind channel estimation methods for SIMO‐OFDM systems have the advantage of saving bandwidth and improving energy efficiency and system throughput. As far as blind channel identification is concerned, it is known that zero padding (ZP)‐based single‐input single‐output (SISO)‐OFDM systems have desirable features compared to conventional cyclic prefix (CP)‐based SISO‐OFDM systems. However, it is yet unknown whether ZP‐ or CP‐based SIMO‐OFDM systems are favourable for blind channel estimation. To investigate this problem, we first propose a short‐data effective method for blind channel estimation for ZP‐based SIMO‐OFDM systems. Then we analyse a number of issues surrounding blind channel estimation for ZP‐ and CP‐based SIMO‐OFDM systems. The issues brought up in the paper have not been discussed in the existing research. The significance of our investigation is that it provides a deep insight into blind channel estimation for ZP‐ and CP‐based SIMO‐OFDM systems. Copyright © 2011 John Wiley & Sons, Ltd.     

Digital interpolation beamforming for low-pass and bandpass signals

Digital time-domain beamforming requires that samples of the sensor signals be available at a sufficient rate to realize accurate time delays for beam steering. For many applications, this input rate, which may be significantly higher than the Nyquist rate required for waveform reconstruction, places stringent requirements on A/D converter hardware and transmission cable bandwidth. Recently, a technique referred to as digital interpolation beamfonning was introduced which greatly relaxes the sampling requirement and provides substantial hardware savings through more flexible design options. In this approach, the sensor channels need only be sampled at a rate which satisfies aliasing requirements. The vernier beam-delay increments are then synthesized using digital interpolation which can be implemented at the beamformer input or output to minimize digital processing complexity. Previously, this concept was presented for the case of "low-pass" signals. This paper extends this work by examining the relationship between interpolation and beamforming for the important class of "bandpass" signals. Specifically, sampling methods are discussed whereby the original waveform can be reconstructed from samples taken at a rate consistent with the bandwidth of the bandpass signal. Beamformer implementations are presented which utilize these bandwidth-sampling techniques in conjunction with interpolation and which compute beam output points at the generally low rate dictated by the signal bandwidth. The interpolation beamformer achieves time-delay quantization (beam-steering accuracy) independent of both the input and output sampling rates. This approach generally requires less hardware than conventional procedures. Interpolation falter characteristics dictated by the bandwidth-sampling procedure are described and efficient methods of implementation employing nonrecursive digital bandpass and low-pass filters are presented.     

Transmission latencies in a telemetry-linked brain-machine interface

To be clinically viable, a brain-machine interface (BMI) requires transcutaneous telemetry. Spike-based compression algorithms can be used to reduce the amount of telemetered data, but this type of system is subject to queuing-based transmission delays. This paper examines the relationships between the ratio of output to average input bandwidth of an implanted device and transmission latency and required queue depth. The examination was performed with a computer model designed to simulate the telemetry link. The input to the model was presorted spike data taken from a macaque monkey performing a motor task. The model shows that when the output bandwidth/average input bandwidth is in unity, significant transmission latencies occur. For a 32-neuron system, transmitting 50 bytes of data per spike and with an average neuron firing rate of 8.93 spikes/s, the average maximum delay was approximately 3.2 s. It is not until the output bandwidth is four times the average input bandwidth that average maximum delays are reduced to less than 10 ms. A comparison of neuron firing rate and resulting latencies shows that high latencies result from neuron bursting. These results will impact the design of transcutaneous telemetry in a BMI.     

A 3.3 V, 10 Bits, Clock-Feedthrough Compensated Switched-Current Second Order Sigma-Delta Modulator

This article presents a low-pass sigma-delta modulator for Analogue-to-Digital conversion. The circuit uses a switched-current technique which presents a well known drawback called clock feedthrough. This phenomenon induces an error on the output signal value. In order to cancel the clock feedthrough effect, we use a new method based on a current feedback loop. The circuit is designed in 0.8 μm AMS “Austria Mikro Systems” single poly CMOS process. Measurements of the modulator are performed under A/D converters characterisation system, and show 55 dB dynamic range at 2.048 MHz sampling rate with 8 kHz input frequency bandwidth. These characteristics are suitable for audio applications.     

An efficient baseband modem technique for data transmission overanalog FM links

In mobile radio where data are transmitted over existing analog FM systems, the receive bandpass bandwidth, which is adapted to the analog speech transmission, is larger than would be required by data transmission. This results in poor error performance. A novel baseband modem technique which drastically improves the error performance is proposed and analyzed. A smearing filter is used to convert the compound baseband noise at the limiter discriminator output to approximately Gaussian noise. This optimizes the performance at large carrier-to-noise ratios (CNRs). A baseband click detection and elimination scheme, which improves the performance at small CNR is proposed. Simulation results show that this system requires 3.8 dB less CNR than the conventional digital FM system to achieve a bit error probability of 10^-4. It is concluded that the technique is attractive for data transmission over analog FM links     

石英MEMS陀螺检测模块的测试方法研究

提出了一种石英微机械(MEMS)陀螺检测模块的测试方法和测试系统方案.该方法仅将检测模块的输入/输出端口接入测试系统,无需其他位置的信号注入和引出,即可实现增益、相位、频带、阻尼等电路特性的测试和标定,适用于生产过程对检测模块的快速测试及研制过程对检测模块的性能分析.     

A new wideband regulated cascode amplifier with improved performance and its application

This paper presents a new high frequency Regulated Cascode (RGC) amplifier with improved performance. The split-length compensation technique is used to increase both the bandwidth and output impedance, and decrease the input impedance of the conventional RGC. The bandwidth of the proposed RGC amplifier is 5.81 GHz, which is about 2.7 GHz larger than that of simple one. The improved performance of the introduced circuit is achieved with no additional passive element and DC power dissipation. In the paper, output impedance and bandwidth of the proposed circuit are derived by using small signal analysis and have also been compared with the traditional one. In addition, a wideband high performance current mirror is designed in the work as an application of the proposed RGC structure. The bandwidth extension ratio (BWER) of the modified wideband current mirror is 1.37. The proposed circuits are designed by using TSMC 0.18 µm CMOS process and BSIM3 Level 49 device model. The circuits are simulated on Spectre simulator of Cadence to validate the analytical results obtained in the paper.     

Output signal-to-noise ratios of FM correlation systems

An analysis is presented on the output signal-to-noise ratios for the FM correlation systems having an FM detector in each input channel of a conventional correlator. The input consists of a frequency-modulated signal combined additively with stationary narrow-band Gaussian random noise. A general expression is derived for the output signal-to-noise ratio. A detailed calculation is made for the output signal-to-noise ratios when each input signal is a carrier frequency-modulated by a Gaussian random process and the integrating filter is of RC low pass. The dependence of the output signal-to-noise ratios on several parameters is discussed.     

A Single Input-Multiple Output Time Reversal UWB Communication System

Time reversal is a promising technique for the improvement of UWB communication systems. Intersymbol interference (ISI) limits the system performance in such wireless systems. This paper presents a general ISI analysis for time reversal UWB communication systems. The time reversal UWB system gives good performance for rates below the coherence bandwidth but at higher data rates the performance of the system is limited by intersymbol interference and bit error rate saturates even for high signal-to-noise ratio. To mitigate the ISI effects, a single input/multiple output (SIMO) time reversal UWB system is used and its performance is analyzed. It is shown that by using a SIMO TR transceiver, ISI reduces and the system capacity increases. Transmitted signal power at SIMO time reversal decreases, therefore in low data rate SISO performance is better than SIMO, But in high rate scenario, SIMO TR suppresses the ISI better than the SISO TR and its performance is better than SISO TR. It is possible to compensate the reduced power by using a receiver with more sensitivity.     

Design and analysis of DC gain and transconductance boosted recycling folded cascode OTA

A new technique for improving the transconductance and low frequency output impedance of recycling folded cascode (RFC) amplifiers is presented. This enhancement was achieved by using a positive feedback and upgrading the recycling structure. The new structure profits from better transconductance, slew rate, and DC gain in comparison with conventional folded cascode (FC) amplifier. Moreover, the input referred noise is reduced and the phase-margin improved. The enhanced amplifier, simulated in 0.18 μm CMOS technology, exhibits a DC gain enhancement of 16.3 dB as well as 115.5 MHz increase in gain bandwidth compared to conventional FC configuration. The amplifier consumes 360 μW @ 1.2 V which makes it suitable for low-voltage applications.     

H.264编码器中1/4像素精度插值算法的VLSI实现

H.264视频编码标准中引入了1/4像素精度插值算法,大大提高了压缩效率,但同时使运算复杂度增加、存储带宽增大。针对以上问题,从运动估计的角度出发,采用一步插值法和数据复用技术,可使带宽减少26%,处理周期可减少45%;设计了相应的硬件结构:采用了5级流水线实现一步插值算法,通过输入缓冲单元实现了参考数据的复用;针对插值过程中产生的大量数据,采用乒乓操作结构,保证数据及时传递。该结构可以显著降低带宽,提高吞吐率,完全可以应用于实时编码器中。     

High-speed MQW electroabsorption optical modulators integrated withlow-loss waveguides

An MQW electro-absorption optical modulator integrated with low-loss input and output waveguides is proposed to achieve larger modulation bandwidth with a shorter modulation region, keeping the total device length large enough for easy fabrication and packaging. A fabricated 1-mm-long modulator with a modulation-region length of 50 μm shows a low insertion loss (7 dB), low driving voltage (V_{10 dB}=2.6 V), and large modulation bandwidth f_{3 dB} =40 (GHz) extrapolated from measurements up to 20 GHz. This modulator is suitable for application to ultra-high-speed fiber transmission systems     

A LINC-based polar transmitter with reduced envelope bandwidth for wideband communications

This paper presents a polar transmitter with reduced envelope bandwidth and the linear amplifier with nonlinear components (LINC) technique is used to produce constant-envelope signals according to the remaining envelope information. This architecture relaxes the bandwidth requirement for the traditional envelope modulators. Only the low-frequency part of the envelope signal is amplified to provide power supply for the power amplifier (PA) stage. In the LINC path, the remaining envelope information is modulated into the phase signals, which are used as the radio frequency (RF) input to the nonlinear PA pair. At the RF output, the envelope information is retrieved from these two parts by the supply-modulated PA pair. The simulation results show that the envelope bandwidth is reduced to around one third of the original bandwidth by the proposed technique. For 2-level LINC structures, the combining efficiency of the proposed architecture is improved to more than twice as the one of LINC-only structure since the combining angles are reduced.     

The off‐grid frequency selective millimeter wave channel estimation

Broadband millimeter wave (mmW) systems are a promising pioneer of cellular communication for next generation which is utilizing the hybrid baseband/analog beamforming structures along with the miniature massive antenna arrays at both sides of the communication link. mmW channel with an available unlicensed spread spectrum is frequency selective because the signal bandwidth can be larger than the coherence bandwidth. Due to the sparse nature of mmW channel, extracting compressive sensing model of the system is preferable. In fact, exploiting the sparse structure will lead to the reduction of the computational complexity, because there is a reduction in the channel training length compared with the conventional methods such as least square estimation. Most of the prior works have considered on‐grid quantized departure/arrival angles in the input/output antennas to obtain a sparse virtual channel model. However, the sparse angles in the physical channel model are continuous where this continuity indicates a mismatch between the physical angles and the on‐grid angles. Such a mismatch will contribute to unwanted components in the virtual channel model. Given these extra components, the conventional compressive sensing tools are unable to recover the channel. In this paper, we propose two solutions for overcoming the problem caused by off‐grid angle selection. The first is based on the vector shaping, and the second one is based on the sparse total least square concepts. Simulation results demonstrate that the proposed methods both could obtain an adequate channel recovery and are preferable regarding computational complexity concerning the newly developed surrogate method.     

Optimization of Driver Preemphasis for On-Chip Interconnects

In modern digital systems, on-chip interconnects have become the system bottleneck, limiting the performance of high-speed clock distributions and data communications in terms of speed and power dissipation. An inverse signaling analysis is developed to optimize the driving signal waveforms for lossy interconnects. By specifying the performance parameters, i.e., the signal swing and edge rate of the interconnect output signal, the corresponding input signals can be derived analytically. The result can be used to guide and optimize the design of interconnect preemphasis drivers. Numerical examples are shown for both lossy RC and RLC distributed lines. Analysis shows that optimized driving voltage and current can increase the interconnect bandwidth without voltage overshoot at the output. The significance of an interconnect inductance is also evaluated with this technique.     

New type of time-varying octave filter

Double quadrature-modulation realisations of lowpass and highpass filters were described recently by Saraga. Extensions to these methods lead to the design of an octave filter using one carrier oscillator rather than the two needed for an equivalent filter designed by the technique referred to above. If the output frequencies are not required to be identical to the input frequencies, this filter also uses fewer modulators. By employing two carrier oscillators, any desired bandwidth may be obtained; the bandwidth and passband position may then be independently varied.     

A clocked high-pass-filter-based offset cancellation technique for high-gain biomedical amplifiers

In this article, a simple offset cancellation technique based on a clocked high-pass filter with extremely low output offset is presented. The configuration uses the on-resistance of a complementary metal oxide semiconductor (CMOS) transmission gate (X-gate) and tunes the lower 3-dB cut-off frequency with a matched pair of floating capacitors. The results compare favourably with the more complex auto-zeroing and chopper stabilisation techniques of offset cancellation in terms of power dissipation, component count and bandwidth, while reporting inferior output noise performance. The design is suitable for use in biomedical amplifier systems for applications such as ENG-recording. The system is simulated in Spectre Cadence 5.1.41 using 0.6 μm CMOS technology and the total block gain is ～83.0 dB while the phase error is <5°. The power consumption is 10.2 mW and the output offset obtained for an input monotone signal of 5 μVpp is 1.28 μV. The input-referred root mean square noise voltage between 1 and 5 kHz is 26.32 nV/√Hz.     

Head-positioning control using resonant modes in hard disk drives

The best way to enhance the input/output (I/O) performance of a hard disk drive is by increasing the spindle speed. Therefore, the effect of windage vibrations caused by the airflow increases as the spindle speed increases. The servo bandwidth is limited by the primary resonant frequency of the mechanical system. However, the frequencies of the windage vibrations are higher than the primary resonant frequency. Accordingly, these frequencies are also above the servo bandwidth and are too high to be controlled by a conventional control system. In response to this problem, we have developed two methods for designing a servo control system that can suppress the windage vibrations. One method uses a stable mechanical resonant mode, and the other uses a stable resonant mode created by a digital filter. By using these methods, the head-positioning system can control the vibrations above the frequency of the primary resonant mode and the servo bandwidth. Application of these methods to actual hard disk drives showed that they can greatly decrease the windage vibrations, in which the peak frequency is about six times the open-loop gain 0-dB crossover frequency.