Anti-alias Filter Research Articles

Simplified Simulation and Measurement of the Signal Transfer Function of a Continuous-Time Pipelined Analog-to-Digital Converter

The continuous-time pipeline (CTP) analog-to-digital converter is a power- and area-efficient realization of an anti-alias filter + ADC cascade. The signal transfer function (STF), which is the transfer function of the anti-alias filter, is an important property of the CTP ADC. It is conventionally characterized by exciting the ADC with multiple input sinusoids. We demonstrate that a single-tone excitation is enough to determine the STF of the CTP ADC over its useful frequency range. The theory is supported by measurement results from a three-stage CTP ADC designed in a 65 nm CMOS process. The converter, equivalent to a sixth-order anti-alias filter + ADC, achieves 70 dB SNDR in a 100MHz bandwidth while sampling at <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex-math notation="LaTeX">$f_{s}= 800$ </tex-math></inline-formula> MHz.

Auditory brainstem responses to high-resolution audio sounds: Effects of anti-alias filters

Despite the superior technical specifications of high-resolution audio in comparison to conventional compact discs (CDs) and digital versatile discs (DVDs), whether and how high-resolution audio leads to an improved subjective listening experience remains unclear. The sound quality of conventional digital formats is deteriorated by temporal blur, which is caused by a high-cut filter applied to avoid aliasing of a digital sound source, resulting in spurious artificial sound energy before and after the actual sound onset. Using sound materials of high-resolution audio grade, this study investigated the effects of filter-induced temporal blur on the auditory neural pathway from the cochlea through the rostral brainstem by measuring the auditory brainstem responses (ABRs) of participants with normal hearing. The participants (N = 24) listened to 0.1-ms clicks, with and without a high-cut filter of 22 kHz or 11 kHz applied to them. The amplitudes and latencies of five ABR waves (<10 ms) revealed no significant effects caused by filtering. The participants could not distinguish between the original and the 22-kHz filtered click sounds, while they could distinguish between the original and the 11-kHz filtered sounds. These results indicate that a CD-level anti-alias filter does not affect initial auditory processing and the sound difference is not discernible, at least for artificial sounds with a simple acoustic structure.

Wireless Modal Analysis With Backscatter Sensors Subjected to Clock Instability

Modal analysis using many accelerometers is a popular and essential testing of artificial structures and machines. The elimination of wire harness to power and collect data from many accelerometers attached to the object under testing is strongly demanded to reduce the time for testing and also to avoid the disturbances caused by the weights and constraints of the accelerometers and harnesses. The synchronization among acceleration measurement streams is important to derive a frequency response in terms of transfer matrix. This paper examines the practicality of concurrent acceleration data streaming with backscatter sensors to realize wireless and batteryless modal analysis. Concurrent backscatter streaming is possible by using multiple subcarriers, however the major synchronization error caused by the instability of backscatter sensor clock needs to be compensated. Synchronization of the streams by resampling is necessary but a naive implementation may produce artificial delay caused by the difference of anti-alias filters tailored to the individual sample rates of backscatter sensors. In this paper, a set of signal processing to synchronize concurrent backscatter signals is designed and implemented in a prototype system. The prototype system comprises three backscatter sensors, interrogator and modal analyzer. The signal processing in the modal analyzer features a recursive least square regression to recover sampling rate of backscatter sensor and a zero-phase-shift filtering in overlapand-add approach. The proposal is evaluated by comparing the transfer matrices generated from the prototype system and a commercial modal analysis system where wired accelerometers are used. The experimental results show that the proposed signal processing can successfully synchronize concurrent backscatter streams by eliminating the artificial delay.

A 16-Element Fully Integrated 28-GHz Digital RX Beamforming Receiver

A 16-element fully integrated 28-GHz digital beamformer combines with a custom eight-layer LTCC substrate with a 4 $\times \,\,{4}$ patch antenna array for a complete 16-element single-chip 28-GHz millimeter-wave (mm-wave)-to-digital beamforming system. Sixteen-element digital beamforming in a single integrated circuit (IC) represents an excellent tradeoff between die size, signal loss, and I/O routing complexity. Per-element RX slices with an inductor-less mm-wave front end and 4 $\times $ parallel continuous-time bandpass delta–sigma analog-to-digital conversion (ADC) arrays enable compact mm-wave-to-digital conversion. The 4 $\times $ parallel ADC array provides in-built finite-impulse response (FIR) filtering for additional harmonic suppression and anti-alias filtering. ADC sampling of a high (1 GHz) IF facilitates single-phase mm-wave local oscillator (LO) routing and moves the I/Q mixing into the digital domain. Optimum bump and RX slice placement shortens LO and mm-wave signal routing and reduces signal loss. Bit-stream processing (BSP) takes advantage of the narrow bit-width raw outputs of the RX slices to implement digital beamforming with area- and energy-efficient MUXes. The prototype 16-element beamformer generates four independent, simultaneous beams. Over-the-air measurements confirm accurate 3-D beam patterns and indicate a measured overall noise figure of 7 dB and QAM-4 error vector magnitude (EVM) of −18 dB.

A 65-nm CMOS Continuous-Time Pipeline ADC Achieving 70-dB SNDR in 100-MHz Bandwidth

We describe the design principles and circuit details of a three-stage continuous-time pipeline (CTP) ADC that achieves 70-dB SNDR in a 100-MHz bandwidth while sampling at 800 MHz. Implemented in 65-nm CMOS, the ADC is easy to drive and incorporates an inherent anti-alias filter that achieves 60-dB rejection in the first Nyquist band. Each pipeline stage is realized using a second-order Rauch-filter-based residue amplifier that incorporates a 9-level resistive DAC and an RC-delay line. A dummy-switching scheme relaxes DAC reference-buffer requirements. The back-end ADC is a $4\times $ time-interleaved 7-bit SAR converter. The Schreier and Walden FoMs of our ADC are 165.4 dB and 56.1 fJ/level, respectively.

Low pass anti-alias filter for ADC with differential input on base Op-Amp with differential output with a minimum number of capacitors

The new circuit of third order low-pass filter (LPF) on operational amplifier (Op-Amp) with differential output designed to limit the spectrum of signals at the input of an analog-to-digital converter (ADC) is presented. The developed circuit contains one capacitor less than in classical LPF. The basic equations of the new circuit are presented and the results of computer simulation obtained in Micro-Cap CAD, which based on models of THS4130 Op-Amp by Texas Instruments, are shown. The results of computer simulation confirm the efficiency of this LPF scheme and show a close to ideal correspondence between the theoretical and real amplitude-frequency response.

Effects of sampling rate and type of anti-aliasing filter on linear-predictive estimates of formant frequencies in men, women, and children

The purpose of this study was to assess the effect of downsampling the acoustic signal on the accuracy of linear-predictive (LPC) formant estimation. Based on speech produced by men, women, and children, the first four formant frequencies were estimated at sampling rates of 48, 16, and 10 kHz using different anti-alias filtering. With proper selection of number of LPC coefficients, anti-alias filter and between-frame averaging, results suggest that accuracy is not improved by rates substantially below 48 kHz. Any downsampling should not go below 16 kHz with a filter cut-off centered at 8 kHz.

Information-Theoretic Bounds for the Forensic Detection of Downscaled Signals

The detection of rescaling operations represents an important task in multimedia forensics. While many effective heuristics have been proposed, there is no theory on the forensic detectability revealing the conditions of more or less reliable detection. We study the problem of discriminating 1D and 2D genuine signals from signals that have been downscaled with the goal of quantifying the statistical distinguishability between these two hypotheses. This is done by assuming known signal models and deriving the expressions of statistical distances that are linked to the hypothesis testing theory, namely, the symmetrized form of Kullback–Leibler divergence known as the Jeffreys divergence, and the Bhattacharyya divergence. The analysis is performed for varying parameters of both the genuine signal model (variance and one-step correlation) and the rescaling process (rescaling factor, interpolation kernel, grid shift, and anti-alias filter), thus allowing us to reveal the insights on their influence and interplay. In addition to the signal itself, we consider the signal transformations (prefilter and covariance matrix estimators) that are often involved in practical rescaling detectors, showing that they yield similar results in terms of distinguishability. Numerical tests on synthetic and real signals confirm the main observations from the theoretical analysis.

Multiplexed readout of kinetic inductance bolometer arrays

Kinetic inductance bolometer technology is a candidate for passive submillimeter wave and terahertz imaging systems. Its benefits include scalability into large 2D arrays and operation with intermediate cryogenics in the temperature range of 5-10 K. We have previously demonstrated the scalability in terms of device fabrication, optics integration, and cryogenics. In this article, we address the last missing ingredient, the readout. The concept, serial addressed frequency excitation, is an alternative to full frequency-division multiplexing at microwave frequencies conventionally used to read out kinetic inductance detectors. We introduce the concept and analyze the criteria of the multiplexed readout avoiding the degradation of the signal-to-noise ratio in the presence of a thermal anti-alias filter inherent to thermal detectors. We present a practical scalable realization of a readout system integrated into a prototype imager with 8712 detectors. This is used for demonstrating the noise properties of the readout. Furthermore, we present practical detection experiments with a stand-off laboratory-scale imager.

Implementación de un Control Activo de Ruido en Configuración Feedback basado en un Microcontrolador

Este trabajo describe la implementación de un sistema de control activo de ruido sobre la plataforma EDU-CIAA-NXP con hardware reducido y de bajo costo. Se hace una revisión del algoritmo filtered-X LMS en configuraciones feedforward y feedback, y se detalla el diseño de filtros anti-alias y de reconstrucción. Se describe el software que corre sobre la plataforma, y se muestran resultados de ensayos con la configuración feedback. Usando un tono puro como señal perturbadora se logró una atenuación de 30dB, que es adecuada según requerimientos de performance en aplicaciones de control activo de ruido.

Alias-Free Products of Signals Near Nyquist Rate

Products of time-series signals have found wide-spread application in many fields of signal processing. For example, they are often used in modeling and compensating distortions generated by analog and mixed-signal components. Without excess bandwidth, the products of time-series signals will produce aliased artifacts that do not represent the physical phenomenology of the components being modeled. To ameliorate the effects of these potentially unwanted, aliased distortion products, we compare two multidimensional filters: the first is a simple one-dimensional (1-D) polyphase filter that is used to generate excess bandwidth; the second is a multidimensional (M-D) filter that is convolved with the time-series product. We demonstrate that the 1-D polyphase filter, although computationally simple, still produces unwanted aliasing when applied to signals near Nyquist rate, which can adversely affect modeling and/or compensation performance. By contrast, the M-D filter is computationally expensive, but is capable of most closely matching the ideal antialias filter response.

Dynamic Phasor Estimation Through DSTKF Under Transient Conditions

This paper presents a double suboptimal-scaling-factor adaptive strong tracking Kalman filter (DSTKF)-based phasor measurement unit algorithm which can meet the accuracy requirement of the IEEE standard C37.118.1 under the dynamic condition. This method uses a $k$ th Taylor polynomial to linearize the complex exponential of the signal model, and estimates the dynamic phasor using DSTKF. The antialias filter unit for DSTKF algorithm is included to suppress high-order harmonics which will augment computing burden. By applying double suboptimal-scaling factors, DSTKF adjusts the process noise covariance matrix and the prediction covariance matrix in real time. Thereafter, amplitude and frequency measurement performances under various dynamic conditions are assessed in detail. Theoretical analysis and comprehensive experimental results validate the effectiveness of the proposed method.

Fully-Digital Blind Compensation of Non-Linear Distortions in Wideband Receivers

Digital post-distortion is becoming an increasingly attractive solution to compensate for the non-linearities of RF receivers implemented in deep submicron CMOS technologies. A very promising technique which provides flexibility and robustness is the frequency spreading based digital post-distortion. This paper focuses on the analysis of this approach for wideband receivers and its main limitations, such as the anti-alias filter (AAF) impact. As a matter of fact, distortions filtered by the AAF can degrade the correction performance. To overcome this problem, a novel solution based on mimicking the AAF frequency response in the digital domain is proposed. Aspects regarding the estimation and the algorithm convergence are also studied: the noise effect and the choice of the free frequency band. Furthermore, two other techniques are developed to improve the modeling and the correction of the memory effect using, respectively, a bank of filters and two-step architecture. The proposed techniques are demonstrated by simulations and measurements on a multi-channel receiver suited for DVB-T applications. The receiver is based on a 1-GHz bandwidth RF front-end followed by a 2.7-GHz 13-bit analog-to-digital converter.

Correcting EM system bandwidth limitations

All EM transmitters, sensors and data acquisition systems have bandwidth limitations. Transmitters have upper bandwidth limitations due to finite slew rate issues, and systems have a lower bandwidth set by the base frequency used. Receivers and data acquisition systems ideally should have a flat bandwidth response that spans the transmitted signal bandwidth. The data acquisition system should sample fast enough to capture the highest frequencies of interest, with anti-alias filters to prevent data contamination from unwanted signals. Sensors however may have physical bandwidth limitations, for example fluxgates and feedback MT sensors may have an upper corner frequency of a few kHz, and ARMIT and feedback MT sensors have lower corner frequencies in the sub 1 Hz range. In many cases, the sensor corner frequency can be mathematically described as a single- or multi-pole response. In this case, it is possible to exactly deconvolve the data to exactly correct for the sensor imperfection. A limitation of this process is that noise as well as signal may be amplified in this correction process. Without correction, data may be incorrectly modelled or interpreted. This paper illustrates the correction of fluxgate (mostly a time delay of hundred or more microseconds), ARMIT 2 (where a significant but exact correction is required), ANT23 feedback and 3D3 dBdt data.

1306-km 20x1248-Gb/s PM-64QAM Transmission over PSCF with Net SEDP 11,300 (b∙km)/s/Hz using 115 samp/symb DAC

We demonstrated the transmission of a Nyquist-WDM signal based on PM-64QAM modulation in an EDFA-only submarine configuration composed of 54.4 km-long fiber spans: 20 channels at 124.8-Gb/s were propagated over 1306 km of low-loss pure-silica-core fiber (PSCF). Thanks to an aggressive digital spectral shaping, we achieved a raw spectral efficiency (SE) of 10.4 b/s/Hz, corresponding to 8.67 b/s/Hz net SE when considering a 20% FEC overhead. Transmitter DACs are operated at a record-low 1.15 samples/symbol, enabled by the insertion of advanced anti-alias filters. The achieved SE-times-distance product was 11,327 (b ∙ km)/(s ∙ Hz), the highest reported so far for PM-64QAM. Combining the experimental results with the performance predictions obtained using an analytical model of nonlinear propagation in uncompensated coherent optical systems (the so-called "GN-model"), we show that PM-64QAM is a realistic option for ultra-high capacity systems in the 1,000 km range, carrying up 40 Tb/s in the C-band.

A Reconfigurable Direct RF Receiver With Jitter Analysis and Applications

In typical direct radio frequency (RF) sampling receivers, the sample clock jitter results in SNR degradation on received signals that increases log linearly with RF signal frequency. De-jittering received signals is computationally expensive, especially for wideband communications signals, because the induced jitter on received signals varies as a function of RF signal frequency. This paper considers a reconfigurable direct RF receiver (RDRFR) that has a stair-step jitter pattern as a function of RF signal frequency. The RDRFR uses a two-stage sampling circuit. In the first stage, the RF input is bandpass filtered and pulse sampled without quantizing the signal. The discrete-time analog signal is then interpolated with a continuous time low-pass filter, then sampled and quantized by a traditional analog-to-digital converter. With this two-stage sampling circuit, the induced signal jitter from the RF sampling is identical for all in-band signals and is the same to within an integer scale factor for all signals, in-band or out-of-band. This key circuit property means that it is possible to remove jitter with very low computational complexity in the RDRFR. Furthermore, this result can be exploited to improve system performance with non-ideal anti-alias filters by identifying non-desired signals caused by out-of-band leakage.

A Gm-C continuous-time anti-alias filter for UWB analog front-end

A proposed constant drain-source transconductor topology is designed to keep linearity at high frequency. By using the proposed operational transconductance amplifier as a building block, a fourth-order low-pass filter is realized. The filter was fabricated in 0.18 μm CMOS technology and feathers a 250 MHz cutoff frequency. The measured IM3 performance is ?36 dB at 0.6 Vpp input swing and the power consumption is 22 mW.

Generating dithering noise for maximum likelihood estimation from quantized data

The Quantization Theorem I (QT I) implies that the likelihood function can be reconstructed from quantized sensor observations, given that appropriate dithering noise is added before quantization. We present constructive algorithms to generate such dithering noise. The application to maximum likelihood estimation (mle) is studied in particular. In short, dithering has the same role for amplitude quantization as an anti-alias filter has for sampling, in that it enables perfect reconstruction of the dithered but unquantized signal’s likelihood function. Without dithering, the likelihood function suffers from a kind of aliasing expressed as a counterpart to Poisson’s summation formula which makes the exact mle intractable to compute. With dithering, it is demonstrated that standard mle algorithms can be re-used on a smoothed likelihood function of the original signal, and statistically efficiency is obtained. The implication of dithering to the Cramér–Rao Lower Bound (CRLB) is studied, and illustrative examples are provided.

Influence of readout process on presampled modulation transfer function in computed radiography system

In a computed radiography (CR) system, there are a number of different mechanisms that cause blurring. The aim of this study was to analyze intrinsic system factors of CR systems that could affect the modulation transfer function (MTF), such as afterglow during readout and the anti-alias filter before analog to digital converter. Methods and Materials: Mathematical slit and edge images were generated from analytical functions. These images were arranged perpendicularly to the laser scan and the plate scan directions, respectively. The influence of afterglow and the anti-alias filter was simulated by using Microsoft Excel. The MTF values calculated from those simulation images were compared with the theoretical MTF values obtained analytically. Results: MTF values in the laser scan direction measured with the slit and edge methods were significantly lower than MTF values in the plate scan direction. The degree of influence on MTF with respect to afterglow and the anti-alias filters was different depending on the measurement method and the scan directions of CR systems. The influence of the anti-alias filters mainly contributed to the differences between MTF values with the slit and edge methods in the laser scan direction.

A 2.4 GHz Fractional-N Frequency Synthesizer With High-OSR ΔΣ Modulator and Nested PLL

This paper presents a nested-PLL architecture for a low-noise wide-bandwidth fractional-N frequency synthesizer. In order to reduce the quantization noise, operating frequency of ΔΣ modulator (DSM) is increased by using an intermediate output of feedback divider. A PLL which serves as an anti-alias filter is added to suppress noise aliasing caused by the divider. Prototype implemented in a 0.13 μm CMOS using ring VCOs achieves 26.3 dB of quantization noise suppression while consuming 15.2 mW and occupying 0.17 mm <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">2</sup> .