Xampling: Signal Acquisition and Processing in Union of Subspaces

We introduce Xampling, a unified framework for signal acquisition and processing of signals in a union of subspaces. The main functions of this framework are two. Analog compression that narrows down the input bandwidth prior to sampling with commercial devices. A nonlinear algorithm then detects the input subspace prior to conventional signal processing. A representative union model of spectrally-sparse signals serves as a test-case to study these Xampling functions. We adopt three metrics for the choice of analog compression: robustness to model mismatch, required hardware accuracy and software complexities. We conduct a comprehensive comparison between two sub-Nyquist acquisition strategies for spectrally-sparse signals, the random demodulator and the modulated wideband converter (MWC), in terms of these metrics and draw operative conclusions regarding the choice of analog compression. We then address lowrate signal processing and develop an algorithm for that purpose that enables convenient signal processing at sub-Nyquist rates from samples obtained by the MWC. We conclude by showing that a variety of other sampling approaches for different union classes fit nicely into our framework.Comment: 16 pages, 9 figures, submitted to IEEE for possible publicatio

Oversampling PCM techniques and optimum noise shapers for quantizing a class of nonbandlimited signals

We consider the efficient quantization of a class of nonbandlimited signals, namely, the class of discrete-time signals that can be recovered from their decimated version. The signals are modeled as the output of a single FIR interpolation filter (single band model) or, more generally, as the sum of the outputs of L FIR interpolation filters (multiband model). These nonbandlimited signals are oversampled, and it is therefore reasonable to expect that we can reap the same benefits of well-known efficient A/D techniques that apply only to bandlimited signals. We first show that we can obtain a great reduction in the quantization noise variance due to the oversampled nature of the signals. We can achieve a substantial decrease in bit rate by appropriately decimating the signals and then quantizing them. To further increase the effective quantizer resolution, noise shaping is introduced by optimizing prefilters and postfilters around the quantizer. We start with a scalar time-invariant quantizer and study two important cases of linear time invariant (LTI) filters, namely, the case where the postfilter is the inverse of the prefilter and the more general case where the postfilter is independent from the prefilter. Closed form expressions for the optimum filters and average minimum mean square error are derived in each case for both the single band and multiband models. The class of noise shaping filters and quantizers is then enlarged to include linear periodically time varying (LPTV)M filters and periodically time-varying quantizers of period M. We study two special cases in great detail

Delay Flip-Flop (DFF) Metastability Impact on Clock and Data Recovery (CDR) and Phase-Locked Loop (PLL) Circuits

Modeling delay flip-flops for binary (e.g., Alexander) phase detectors requires paying close attention to three important timing parameters: setup time, hold time, and clock edge-to-output (or briefly C2Q time). These parameters have a critical role in determining the status of the system on the circuit level. This study provided a guideline for designing an optimum DFF for an Alexander phase detector in a clock and data recovery circuit. Furthermore, it indicated DFF timing requirements for a high-speed phase detector in a clock and data recovery circuit. The CDR was also modeled by Verilog-A, and the results were compared with Simulink model achievements. Eventually designed in 45 nm CMOS technology, for 10 Gbps random sequence, the recovered clock contained 0.136 UI and 0.15 UI peak-to-peak jitter on the falling and rising edges respectively, and the lock time was 125 ns. The overall power dissipation was 21 mW from a 1 V supply voltage. Future work includes layout design and manufacturing of the proposed design

Discrete-Time Chaotic-Map Truly Random Number Generators: Design, Implementation, and Variability Analysis of the Zigzag Map

In this paper, we introduce a novel discrete chaotic map named zigzag map that demonstrates excellent chaotic behaviors and can be utilized in Truly Random Number Generators (TRNGs). We comprehensively investigate the map and explore its critical chaotic characteristics and parameters. We further present two circuit implementations for the zigzag map based on the switched current technique as well as the current-mode affine interpolation of the breakpoints. In practice, implementation variations can deteriorate the quality of the output sequence as a result of variation of the chaotic map parameters. In order to quantify the impact of variations on the map performance, we model the variations using a combination of theoretical analysis and Monte-Carlo simulations on the circuits. We demonstrate that even in the presence of the map variations, a TRNG based on the zigzag map passes all of the NIST 800-22 statistical randomness tests using simple post processing of the output data.Comment: To appear in Analog Integrated Circuits and Signal Processing (ALOG

Equalization-Based Digital Background Calibration Technique for Pipelined ADCs

In this paper, we present a digital background calibration technique for pipelined analog-to-digital converters (ADCs). In this scheme, the capacitor mismatch, residue gain error, and amplifier nonlinearity are measured and then corrected in digital domain. It is based on the error estimation with nonprecision calibration signals in foreground mode, and an adaptive linear prediction structure is used to convert the foreground scheme to the background one. The proposed foreground technique utilizes the LMS algorithm to estimate the error coefficients without needing high-accuracy calibration signals. Several simulation results in the context of a 12-b 100-MS/s pipelined ADC are provided to verify the usefulness of the proposed calibration technique. Circuit-level simulation results show that the ADC achieves 28-dB signal-to-noise and distortion ratio and 41-dB spurious-free dynamic range improvement, respectively, compared with the noncalibrated ADC

Further Discussion on Modeling of Measuring Process via Sampling of Signals

In this paper, we continue a topic of modeling measuring processes by perceiving them as a kind of signal sampling. And, in this respect, note that an ideal model was developed in a previous work. Whereas here, we present its nonideal version. This extended model takes into account an effect, which is called averaging of a measured signal. And, we show here that it is similar to smearing of signal samples arising in nonideal signal sampling. Furthermore, we demonstrate in this paper that signal averaging and signal smearing mean principally the same, under the conditions given. So, they can be modeled in the same way. A thorough analysis of errors related to the signal averaging in a measuring process is given and illustrated with equivalent schemes of the relationships derived. Furthermore, the results obtained are compared with the corresponding ones that were achieved analyzing amplitude quantization effects of sampled signals used in digital techniques. Also, we show here that modeling of errors related to signal averaging through the so-called quantization noise, assumed to be a uniform distributed random signal, is rather a bad choice. In this paper, an upper bound for the above error is derived. Moreover, conditions for occurrence of hidden aliasing effects in a measured signal are given

Multichannel Sampling of Pulse Streams at the Rate of Innovation

We consider minimal-rate sampling schemes for infinite streams of delayed and weighted versions of a known pulse shape. The minimal sampling rate for these parametric signals is referred to as the rate of innovation and is equal to the number of degrees of freedom per unit time. Although sampling of infinite pulse streams was treated in previous works, either the rate of innovation was not achieved, or the pulse shape was limited to Diracs. In this paper we propose a multichannel architecture for sampling pulse streams with arbitrary shape, operating at the rate of innovation. Our approach is based on modulating the input signal with a set of properly chosen waveforms, followed by a bank of integrators. This architecture is motivated by recent work on sub-Nyquist sampling of multiband signals. We show that the pulse stream can be recovered from the proposed minimal-rate samples using standard tools taken from spectral estimation in a stable way even at high rates of innovation. In addition, we address practical implementation issues, such as reduction of hardware complexity and immunity to failure in the sampling channels. The resulting scheme is flexible and exhibits better noise robustness than previous approaches

A Simple Technique for Fast Digital Background Calibration of A/D Converters

A modification of the background digital calibration procedure for A/D converters by Li and Moon is proposed, based on a method to improve the speed of convergence and the accuracy of the calibration. The procedure exploits a colored random sequence in the calibration algorithm, and can be applied both for narrowband input signals and for baseband signals, with a slight penalty on the analog bandwidth of the converter. By improving the signal-to-calibration-noise ratio of the statistical estimation of the error parameters, our proposed technique can be employed either to improve linearity or to make the calibration procedure faster. A practical method to generate the random sequence with minimum overhead with respect to a simple PRBS is also presented. Simulations have been performed on a 14-bit pipeline A/D converter in which the first 4 stages have been calibrated, showing a 15 dB improvement in THD and SFDR for the same calibration time with respect to the original technique