Analog‐to‐Digital Conversion for Cognitive Radio: Subsampling, Interleaving, and Compressive Sensing

This chapter explores different analog-to-digital conversion techniques that are suitable to be implemented in cognitive radio receivers. This chapter details the fundamentals, advantages, and drawbacks of three promising techniques: subsampling, interleaving, and compressive sensing. Due to their major maturity, subsampling- and interleaving-based systems are described in further detail, whereas compressive sensing-based systems are described as a complement of the previous techniques for underutilized spectrum applications. The feasibility of these techniques as part of software-defined radio, multistandard, and spectrum sensing receivers is demonstrated by proposing different architectures with reduced complexity at circuit level, depending on the application requirements. Additionally, the chapter proposes different solutions to integrate the advantages of these techniques in a unique analog-to-digital conversion process

Time-Interleaved Analog-to-Digital-Converters: Modeling, Blind Identification and Digital Correction of Frequency Response Mismatches

Analog-to-digital-conversion enables utilization of digital signal processing (DSP) in many applications today such as wireless communication, radar and electronic warfare. DSP is the favored choice for processing information over analog signal processing (ASP) because it can typically offer more flexibility, computational power, reproducibility, speed and accuracy when processing and extracting information. Software defined radio (SDR) receiver is one clear example of this, where radio frequency waveforms are converted into digital form as close to the antenna as possible and all the processing of the information contained in the received signal is extracted in a configurable manner using DSP. In order to achieve such goals, the information collected from the real world signals, which are commonly analog in their nature, must be converted into digital form before it can be processed using DSP in the respective systems. The common trend in these systems is to not only process ever larger bandwidths of data but also to process data in digital format at ever higher processing speeds with sufficient conversion accuracy. So the analog-to-digital-converter (ADC), which converts real world analog waveforms into digital form, is one of the most important cornerstones in these systems.The ADC must perform data conversion at higher and higher rates and digitize ever-increasing bandwidths of data. In accordance with the Nyquist-Shannon theorem, the conversion rate of the ADC must be suffcient to accomodate the BW of the signal to be digitized, in order to avoid aliasing. The conversion rate of the ADC can in general be increased by using parallel ADCs with each ADC performing the sampling at mutually different points in time. Interleaving the outputs of each of the individual ADCs provides then a higher digitization output rate. Such ADCs are referred to as TI-ADC. However, the mismatches between the ADCs cause unwanted spurious artifacts in the TI-ADC’s spectrum, ultimately leading to a loss in accuracy in the TI-ADC compared to the individual ADCs. Therefore, the removal or correction of these unwanted spurious artifacts is essential in having a high performance TI-ADC system.In order to remove the unwanted interleaving artifacts, a model that describes the behavior of the spurious distortion products is of the utmost importance as it can then facilitate the development of efficient digital post-processing schemes. One major contribution of this thesis consists of the novel and comprehensive modeling of the spurious interleaving mismatches in different TI-ADC scenarios. This novel and comprehensive modeling is then utilized in developing digital estimation and correction methods to remove the mismatch induced spurious artifacts in the TI-ADC’s spectrum and recovering its lost accuracy. Novel and first of its kind digital estimation and correction methods are developed and tested to suppress the frequency dependent mismatch spurs found in the TI-ADCs. The developed methods, in terms of the estimation of the unknown mismatches, build on statistical I/Q signal processing principles, applicable without specifically tailored calibration signals or waveforms. Techniques to increase the analog BW of the ADC are also analyzed and novel solutions are presented. The interesting combination of utilizing I/Q downconversion in conjunction with TI-ADC is examined, which not only extends the TI-ADC’s analog BW but also provides flexibility in accessing the radio spectrum. Unwanted spurious components created during the ADC’s bandwidth extension process are also analyzed and digital correction methods are developed to remove these spurs from the spectrum. The developed correction techniques for the removal of the undesired interleaving mismatch artifacts are validated and tested using various HW platforms, with up to 1 GHz instantaneous bandwidth. Comprehensive test scenarios are created using measurement data obtained from HW platforms, which are used to test and evaluate the performance of the developed interleaving mismatch estimation and correction schemes, evidencing excellent performance in all studied scenarios. The findings and results presented in this thesis contribute towards increasing the analog BW and conversion rate of ADC systems without losing conversion accuracy. Overall, these developments pave the way towards fulfilling the ever growing demands on the ADCs in terms of higher conversion BW, accuracy and speed

FPGA Implementation of Channel Mismatch Calibration in TIADCs for Signals in Any Nyquist Bands

This paper presents a fully digital background calibration technique of the gain and timing mismatches in undersampling Time-Interleaved Analog-to-Digital Converters for the wideband bandlimited input signal at any Nyquist bands. The proposed technique does not require an additional reference channel nor a pilot input. The channel mismatch parameters are estimated based on the mismatch frequency band. The experimental results shows the efficiency of the proposed mitigation technique with the SNDR improvement of 16dB for 4-channel 60dB SNR TIADC clocked at 2.7GHz given a multi-tone input occupied at the third Nyquist band. The hardware architecture of the proposed technique is designed and validated on Altera FPGA DE4 board. The synthesized design utilizes a very little amount of the hardware resource in the FPGA chip and works correctly on a Hardware-In-the-Loop emulation framework

Nonlinear models and algorithms for RF systems digital calibration

Focusing on the receiving side of a communication system, the current trend in pushing the digital domain ever more closer to the antenna sets heavy constraints on the accuracy and linearity of the analog front-end and the conversion devices. Moreover, mixed-signal implementations of Systems-on-Chip using nanoscale CMOS processes result in an overall poorer analog performance and a reduced yield. To cope with the impairments of the low performance analog section in this "dirty RF" scenario, two solutions exist: designing more complex analog processing architectures or to identify the errors and correct them in the digital domain using DSP algorithms. In the latter, constraints in the analog circuits' precision can be offloaded to a digital signal processor. This thesis aims at the development of a methodology for the analysis, the modeling and the compensation of the analog impairments arising in different stages of a receiving chain using digital calibration techniques. Both single and multiple channel architectures are addressed exploiting the capability of the calibration algorithm to homogenize all the channels' responses of a multi-channel system in addition to the compensation of nonlinearities in each response. The systems targeted for the application of digital post compensation are a pipeline ADC, a digital-IF sub-sampling receiver and a 4-channel TI-ADC. The research focuses on post distortion methods using nonlinear dynamic models to approximate the post-inverse of the nonlinear system and to correct the distortions arising from static and dynamic errors. Volterra model is used due to its general approximation capabilities for the compensation of nonlinear systems with memory. Digital calibration is applied to a Sample and Hold and to a pipeline ADC simulated in the 45nm process, demonstrating high linearity improvement even with incomplete settling errors enabling the use of faster clock speeds. An extended model based on the baseband Volterra series is proposed and applied to the compensation of a digital-IF sub-sampling receiver. This architecture envisages frequency selectivity carried out at IF by an active band-pass CMOS filter causing in-band and out-of-band nonlinear distortions. The improved performance of the proposed model is demonstrated with circuital simulations of a 10th-order band pass filter, realized using a five-stage Gm-C Biquad cascade, and validated using out-of-sample sinusoidal and QAM signals. The same technique is extended to an array receiver with mismatched channels' responses showing that digital calibration can compensate the loss of directivity and enhance the overall system SFDR. An iterative backward pruning is applied to the Volterra models showing that complexity can be reduced without impacting linearity, obtaining state-of-the-art accuracy/complexity performance. Calibration of Time-Interleaved ADCs, widely used in RF-to-digital wideband receivers, is carried out developing ad hoc models because the steep discontinuities generated by the imperfect canceling of aliasing would require a huge number of terms in a polynomial approximation. A closed-form solution is derived for a 4-channel TI-ADC affected by gain errors and timing skews solving the perfect reconstruction equations. A background calibration technique is presented based on cyclo-stationary filter banks architecture. Convergence speed and accuracy of the recursive algorithm are discussed and complexity reduction techniques are applied

Subsampling receivers with applications to software defined radio systems

Este trabajo de tesis propone la utilización sistemas basados en submuestreo como una alternativa para la implementación de la etapa de down-conversion de los receptores de radio frecuencia (RF) empleados para aplicaciones multi-estándar y SDR (Software Defined Radio). El objetivo principal será el de optimizar el diseño en cuanto a flexibilidad y simplicidad, las cuales son propiedades inherentes en los sistemas basados en submuestreo. Por tanto, como reducir el número de componentes al mínimo es clave cuando un mismo receptor procesa diferentes estándares de comunicación, las arquitecturas basadas en submuestreo han sido seleccionadas, donde la reusabilidad de los componentes empleados es posible, así como la reducción de los costes totales de los receptores de comunicación y de los equipos de certificación que emplean estas arquitecturas. Un motivo adicional por el que los sistemas basados en submuestreo han sido seleccionados es el concerniente a la topología del receptor. Como la idea de la tecnología SDR es implementar todas las funcionalidades del receptor (filtrado, amplificación) en el dominio digital, el convertidores analógico-digital (ADC) deberá estar localizado en la cadena de recepción lo más cerca posible a la antena, siendo el objetivo final el convertir la señal directamente de RF a digital. Sin embargo, con los actuales ADC no es posible implementar esta idea debido al alto ancho de banda que necesitarían sin perder resolución para cubrir las especificaciones de los estándares de comunicaciones inalámbricas. Por tanto, los sistemas basados en submuestreo se presentan como la opción más adecuada para implementar este tipo de sistemas debido a que pueden muestrear la señal de entrada por debajo de la tasa de Nyquist, si se cumplen ciertas restricciones en cuanto a la elección de la frecuencia de muestreo. De este modo, los requerimientos del ADC serán relajados ya que, usando estas arquitecturas, este componente procesará la señal a frecuencias intermedias. Una vez se han introducido los conceptos principales de las técnicas de submuestreo, esta tesis doctoral presenta el diseño de una tarjeta de adquisición de datos basada en submuestreo con la finalidad de ser implementada como un receptor de test y certificación de banda ancha. El sistema propuesto proporciona una alta resolución para un elevado ancho de banda, a partir del uso de un S&H de bajo jitter y de un convertidor analógico digital ADC que trabaja a frecuencias intermedias. El sistema es implementado usando dispositivos comerciales en una placa de circuito impreso diseñada y fabricada, y cuya caracterización experimental muestra una resolución de más 8 bits para un ancho de banda analógico de 20 MHz. Concretamente, la resolución medida será mayor de 9 bits hasta una frecuencia de entrada de 2.9 GHz y mayor de 8 bits para una frecuencia de entrada de hasta 6.5 GHz, lo cual resulta suficiente para cubrir los requerimientos de la mayor parte de los actuales estándares de comunicaciones inalámbricas (GPS, GSM, GPRS, UMTS, Bluetooth, Wi-Fi, WiMAX). Sin embargo, los receptores basados en submuestreo presentan algunos importantes inconvenientes, como son adicionales fuentes de ruido (jitter y plegado de ruido térmico) y una dificultad añadida para implementarlo en escenarios multi-banda y no lineales. Acerca del plegado de ruido en la banda de interés, esta tesis propone el uso de una técnica basada en una arquitectura de reloj múltiple con el objetivo de aumentar la resolución y cubrir un número mayor de estándares para su test y certificación. Empleando una frecuencia de muestreo mayor para el caso del S&H, se conseguirá reducir este efecto, aumentando la resolución en aproximadamente 0.5-1 bit respecto al caso de sólo usar una fuente de reloj. Las expresiones teóricas de esta mejora son desarrolladas y presentadas en esta tesis, siendo posteriormente corroboradas de modo experimental. Por otra parte, esta tesis también propone novedosas técnicas para la aplicación de estos sistemas de submuestreo en entornos multi-banda y no lineales, los cuales presentan desafíos adicionales por el hecho de existir la posibilidad de solapamiento entre la señal de interés y los otros canales de comunicación, así como de solapamiento con sus armónicos. De este modo, esta tesis extiende el uso de los sistemas basados en submuestreo para este tipo de entornos, proponiendo técnicas para la elección de la frecuencia óptima de muestreo que evitan el solapamiento entre señales, a la vez que consiguen incrementar la resolución del receptor. Finalmente, se presentará la optimización en cuanto a características de ruido de un receptor concreto para aplicaciones de banda dual en entornos no lineales. Dicho receptor estará basado en las técnicas de reloj múltiple presentadas anteriormente y en una estructura de multi-filtro entre el S&H y el ADC. El sistema diseñado podrá emplearse para diversas aplicaciones a ambos lados de la cadena de comunicación, tal como en receptores de detección de espectro para radio cognitiva, o implementando el bucle de realimentación de un transmisor para la linealización de amplificadores de potencia. Por tanto, la presente tesis doctoral cuenta con tres contribuciones diferenciadas. La primera de ellas es la dedicada al diseño de un prototipo de recepción multi-estándar basado en submuestreo para aplicaciones de test y certificación. La segunda aportación es la dedicada a la optimización de las especificaciones de ruido a partir de las técnicas presentadas basadas en reloj múltiple. Por último, la tercera contribución principal es la relacionada con la extensión de este tipo de técnicas a sistemas multi-banda en entornos no lineales. Todas estas contribuciones han sido estudiadas teóricamente y experimentalmente validadas

Design of Analog-to-Digital Converters with Embedded Mixing for Ultra-Low-Power Radio Receivers

In the field of radio receivers, down-conversion methods usually rely on one (or more) explicit mixing stage(s) before the analog-to-digital converter (ADC). These stages not only contribute to the overall power consumption but also have an impact on area and can compromise the receiver’s performance in terms of noise and linearity. On the other hand, most ADCs require some sort of reference signal in order to properly digitize an analog input signal. The implementation of this reference signal usually relies on bandgap circuits and reference buffers to generate a constant, stable, dc signal. Disregarding this conventional approach, the work developed in this thesis aims to explore the viability behind the usage of a variable reference signal. Moreover, it demonstrates that not only can an input signal be properly digitized, but also shifted up and down in frequency, effectively embedding the mixing operation in an ADC. As a result, ADCs in receiver chains can perform double-duty as both a quantizer and a mixing stage. The lesser known charge-sharing (CS) topology, within the successive approximation register (SAR) ADCs, is used for a practical implementation, due to its feature of “pre-charging” the reference signal prior to the conversion. Simulation results from an 8-bit CS-SAR ADC designed in a 0.13 μm CMOS technology validate the proposed technique

Reconfigurable Receiver Front-Ends for Advanced Telecommunication Technologies

The exponential growth of converging technologies, including augmented reality, autonomous vehicles, machine-to-machine and machine-to-human interactions, biomedical and environmental sensory systems, and artificial intelligence, is driving the need for robust infrastructural systems capable of handling vast data volumes between end users and service providers. This demand has prompted a significant evolution in wireless communication, with 5G and subsequent generations requiring exponentially improved spectral and energy efficiency compared to their predecessors. Achieving this entails intricate strategies such as advanced digital modulations, broader channel bandwidths, complex spectrum sharing, and carrier aggregation scenarios. A particularly challenging aspect arises in the form of non-contiguous aggregation of up to six carrier components across the frequency range 1 (FR1). This necessitates receiver front-ends to effectively reject out-of-band (OOB) interferences while maintaining high-performance in-band (IB) operation. Reconfigurability becomes pivotal in such dynamic environments, where frequency resource allocation, signal strength, and interference levels continuously change. Software-defined radios (SDRs) and cognitive radios (CRs) emerge as solutions, with direct RF-sampling receivers offering a suitable architecture in which the frequency translation is entirely performed in digital domain to avoid analog mixing issues. Moreover, direct RF- sampling receivers facilitate spectrum observation, which is crucial to identify free zones, and detect interferences. Acoustic and distributed filters offer impressive dynamic range and sharp roll off characteristics, but their bulkiness and lack of electronic adjustment capabilities limit their practicality. Active filters, on the other hand, present opportunities for integration in advanced CMOS technology, addressing size constraints and providing versatile programmability. However, concerns about power consumption, noise generation, and linearity in active filters require careful consideration.This thesis primarily focuses on the design and implementation of a low-voltage, low-power RFFE tailored for direct sampling receivers in 5G FR1 applications. The RFFE consists of a balun low-noise amplifier (LNA), a Q-enhanced filter, and a programmable gain amplifier (PGA). The balun-LNA employs noise cancellation, current reuse, and gm boosting for wideband gain and input impedance matching. Leveraging FD-SOI technology allows for programmable gain and linearity via body biasing. The LNA's operational state ranges between high-performance and high-tolerance modes, which are apt for sensitivityand blocking tests, respectively. The Q-enhanced filter adopts noise-cancelling, current-reuse, and programmable Gm-cells to realize a fourth-order response using two resonators. The fourth-order filter response is achieved by subtracting the individual response of these resonators. Compared to cascaded and magnetically coupled fourth-order filters, this technique maintains the large dynamic range of second-order resonators. Fabricated in 22-nm FD-SOI technology, the RFFE achieves 1%-40% fractional bandwidth (FBW) adjustability from 1.7 GHz to 6.4 GHz, 4.6 dB noise figure (NF) and an OOB third-order intermodulation intercept point (IIP3) of 22 dBm. Furthermore, concerning the implementation uncertainties and potential variations of temperature and supply voltage, design margins have been considered and a hybrid calibration scheme is introduced. A combination of on-chip and off-chip calibration based on noise response is employed to effectively adjust the quality factors, Gm-cells, and resonance frequencies, ensuring desired bandpass response. To optimize and accelerate the calibration process, a reinforcement learning (RL) agent is used.Anticipating future trends, the concept of the Q-enhanced filter extends to a multiple-mode filter for 6G upper mid-band applications. Covering the frequency range from 8 to 20 GHz, this RFFE can be configured as a fourth-order dual-band filter, two bandpass filters (BPFs) with an OOB notch, or a BPF with an IB notch. In cognitive radios, the filter’s transmission zeros can be positioned with respect to the carrier frequencies of interfering signals to yield over 50 dB blocker rejection

Analog-to-digital interface design in wireless receivers

As one of the major building blocks in a wireless receiver, the Analog-to-Digital Interface (ADI) provides link and transition between the analog Radio Frequency (RF) frontend and the baseband Digital Signal Processing (DSP) module. The rapid development of the radio technologies raises new design challenges for the receiver ADI implementation. Requirements, such as power consumption optimization, multi-standard compatibility, fast settling capability and wide signal bandwidth capacity, are often encountered in a low voltage ADI design environment. Previous research offers ADI design schemes that emphasize individual merit. A systematic ADI design methodology is, however, not suffciently studied. In this work, the ADI design for two receiver systems are employed as research vehicles to provide solutions for different ADI design issues. A zero-crossing demodulator ADI is designed in the 0.35µm CMOS technology for the Bluetooth receiver to provide fast settling. Architectural level modification improves the process variation and the Local Oscillation (LO) frequency offset immunity of the demodulator. A 16.2dB Signal-to-Noise Ratio (SNR) at 0.1% Bit Error Rate (BER) is achieved with less than 9mW power dissipation in the lab measurement. For ADI in the 802.11b/Bluetooth dual-mode receiver, a configurable time-interleaved pipeline Analog-to-Digital-Converter (ADC) structure is adopted to provide the required multi-standard compatibility. An online digital calibration scheme is also proposed to compensate process variation and mismatching. The prototype chip is fabricated in the 0.25µm BiCMOS technology. Experimentally, an SNR of 60dB and 64dB are obtained under the 802.11b and Bluetooth receiving modes, respectively. The power consumption of the ADI is 20.2mW under the 802.11b receiving mode and 14.8mW under the Bluetooth mode. In this dissertation, each step of the receiver ADI design procedure, from system level optimization to the transistor level implementation and lab measurement, is illustrated in detail. The observations are carefully studied to provide insight on receiver ADI design issues. The ADI design for the Ultra-Wide Band (UWB) receiver is also studied at system level. Potential ADI structure is proposed to satisfy the wide signal bandwidth and high speed requirement for future applications

System demonstration of an optically-sampled, wavelength-demultiplexed photonic analog-to-digital converter

Thesis (S.M.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2011.Cataloged from PDF version of thesis.Includes bibliographical references (p. 84-86).The performance of electronic analog-to-digital converters (ADCs) at high sampling rates is fundamentally limited by the timing jitter of electronic clocks. To circumvent this limitation, one method is to exploit the orders-of-magnitude lower timing jitter of mode-locked lasers and implement optical sampling as a front-end for electronic ADCs. The optical-sampling, wavelength-demultiplexing approach to A/D conversion, which is explored in this thesis, offers key benefits such as ease of scalability to higher aggregate sampling rates via passive wavelength-division demultiplexing (WDM) filters and potential for full integration via silicon photonics platform for chip-scale signal processing applications. This thesis will first cover the design issues for each stage in the optically-sampled, wavelength-demultiplexed photonic ADC architecture, followed by experimental results from two system demonstrations. Digitization of a 41-GHz signal with 7.0 effective bits at a sampling rate of 2 GSa/s was demonstrated with a discrete-component photonic ADC, which corresponds to 15 fs of jitter, a 4-5 times improvement over state-of-the-art electronic ADCs. On the way towards an integrated photonic ADC, a silicon chip with core photonic components was fabricated and used to digitize a 10-GHz signal with 3.5 effective bits. Drop-port transmission measurements of an integrated 20-channel WDM filter bank are included to show potential for high sampling rate operation with 10 effective bits.by Michael Yung Peng.S.M