Compensation numérique pour convertisseur large bande hautement parallélisé.

Time-interleaved analog-to-digital converters (TIADC) seem to be the holy grail of analog-to-digital conversion. Theoretically, their sampling speed can be increased, very simply, by duplicating the sub-converters. The real world is different because mismatches between the converters strongly reduce the TIADC performance, especially when trying to push forward the sampling speed, or the resolution of the converter. Using background digital mismatch calibration can alleviate this limitation. The first part of the thesis is dedicated to studying the sources and effects of mismatches in a TIADC. Performance metrics such as the SNDR and the SFDR are derived as a function of the mismatch levels. In the second part, new background digital mismatch calibration techniques are presented. They are able to reduce the offset, gain, skew and bandwidth mismatch errors. The mismatches are estimated by using the statistical properties of the input signal and digital filters are used to reconstruct the correct output samples. In the third part, a 1.6 GS/s TIADC circuit, implementing offset, gain and skew mismatch calibration, demonstrates a reduction of the mismatch spurs down to a level of -70 dBFS, up to an input frequency of 750 MHz. The circuit achieves the lowest level of mismatches among TIADCs in the same frequency range, with a reasonable power and area, in spite of the overhead caused by the calibration.Les convertisseurs analogique-numérique à entrelacement temporel (TIADC) semblent être une solution prometteuse dans le monde de la conversion analogique-numérique. Leur fréquence d’échantillonnage peut théoriquement être augmentée en augmentant le nombre de convertisseurs en parallèle. En réalité, des désappariements entre les convertisseurs peuvent fortement dégrader les performances, particulièrement à haute fréquence d’échantillonnage ou à haute résolution. Ces défauts d’appariement peuvent être réduits en utilisant des techniques de calibration en arrière-plan. La première partie de cette thèse est consacrée à l’étude des sources et effets des différents types de désappariements dans un TIADC. Des indicateurs de performance tels que le SNDR ou la SFDR sont exprimés en fonction du niveau des désappariements. Dans la deuxième partie, des nouvelles techniques de calibration sont proposées. Ces techniques permettent de réduire les effets des désappariements d’offset, de gain, d’instant d’échantillonnage et de bande passante. Les désappariements sont estimés en se basant sur des propriétés statistiques du signal et la reconstruction des échantillons de sortie se fait en utilisant des filtres numériques. La troisième partie démontre les performance d’un TIADC fonctionnant a une fréquence d’échantillonnage de 1.6 GE/s et comprenant les calibration d’offset, de gain et d’instant d’échantillonnage proposées. Les raies fréquentielles dues aux désappariements sont réduites à un niveau de -70dBc jusqu’à une fréquence d’entrée de 750 MHz. Ce circuit démontre une meilleure correction de désappariements que des circuits similaires récemment publiés, et ce avec une augmentation de puissance consommée et de surface relativement faible

A Statistic-Based Calibration Method for TIADC System

Time-interleaved technique is widely used to increase the sampling rate of analog-to-digital converter (ADC). However, the channel mismatches degrade the performance of time-interleaved ADC (TIADC). Therefore, a statistic-based calibration method for TIADC is proposed in this paper. The average value of sampling points is utilized to calculate offset error, and the summation of sampling points is used to calculate gain error. After offset and gain error are obtained, they are calibrated by offset and gain adjustment elements in ADC. Timing skew is calibrated by an iterative method. The product of sampling points of two adjacent subchannels is used as a metric for calibration. The proposed method is employed to calibrate mismatches in a four-channel 5 GS/s TIADC system. Simulation results show that the proposed method can estimate mismatches accurately in a wide frequency range. It is also proved that an accurate estimation can be obtained even if the signal noise ratio (SNR) of input signal is 20 dB. Furthermore, the results obtained from a real four-channel 5 GS/s TIADC system demonstrate the effectiveness of the proposed method. We can see that the spectra spurs due to mismatches have been effectively eliminated after calibration

Direct sampling receivers for broadband communications

Today everything tends to be connected in the Internet of Things (IoT) universe, where a broad variety of communication standards and technologies are used for those connected devices. It is always a dream to design a Software-Defined Radio (SDR) supporting different standards solely based on the software configuration. As integrated-circuit (IC) manufacture and design advance, a partial of SDR can be realized. This thesis investigates one of the most important parts in a SDR: the analog design of a direct sampling (DS) receiver, which mainly consists of a broadband RF front end and a wideband ADC. Especially, a DS receiver shows a great flexibility and efficiency for the simultaneous reception of multiple channels comparing with the traditional parallelism of superheterodyne structure. The research contributions of this work include (1) demonstration and comparative analysis of two new architectures of broadband RFPGAs: voltage-mode: RFPGA-V and current-mode: RFPGA-I. RFPGA-V and RFPGA-I utilize an innovative interpolation method and current steering approach, respectively, to achieve a fine gain step of 0.25-dB over 40-dB gain range for several GHz frequency range. Besides, with innovative design, no off-chip inductor is needed for the both RFPGAs. (2) The design of a 5-GS/s 10b time-interleaved SAR. The ADC power efficiency is significantly improved by many design techniques: the low-energy CDAC switching scheme, optimized input common-mode voltage for comparator, optimal reduced radix-2 capacitor ratio for low-power reference buffers and higher conversion speed, etc. The lane-to-lane mismatches in a time-interleave ADC are minimized by using optimal floor plan and then are calibrated digitally. Three prototypes: the broadband RF front ends with RFPGA-V, the broadband RF front ends with RFPGA-I and a 5-GHz ADC, are fabricated to verify the proposed ideas in 28nm CMOS technology.Electrical and Computer Engineerin

Design of high speed folding and interpolating analog-to-digital converter

High-speed and low resolution analog-to-digital converters (ADC) are key elements in the read channel of optical and magnetic data storage systems. The required resolution is about 6-7 bits while the sampling rate and effective resolution bandwidth requirements increase with each generation of storage system. Folding is a technique to reduce the number of comparators used in the flash architecture. By means of an analog preprocessing circuit in folding A/D converters the number of comparators can be reduced significantly. Folding architectures exhibit low power and low latency as well as the ability to run at high sampling rates. Folding ADCs employing interpolation schemes to generate extra folding waveforms are called "Folding and Interpolating ADC" (F&I ADC). The aim of this research is to increase the input bandwidth of high speed conversion, and low latency F&I ADC. Behavioral models are developed to analyze the bandwidth limitation at the architecture level. A front-end sample-and-hold unit is employed to tackle the frequency multiplication problem, which is intrinsic for all F&I ADCs. Current-mode signal processing is adopted to increase the bandwidth of the folding amplifiers and interpolators, which are the bottleneck of the whole system. An operational transconductance amplifier (OTA) based folding amplifier, current mirror-based interpolator, very low impedance fast current comparator are proposed and designed to carry out the current-mode signal processing. A new bit synchronization scheme is proposed to correct the error caused by the delay difference between the coarse and fine channels. A prototype chip was designed and fabricated in 0.35μm CMOS process to verify the ideas. The S/H and F&I ADC prototype is realized in 0.35μm double-poly CMOS process (only one poly is used). Integral nonlinearity (INL) is 1.0 LSB and Differential nonlinearity (DNL) is 0.6 LSB at 110 KHz. The ADC occupies 1.2mm2 active area and dissipates 200mW (excluding 70mW of S/H) from 3.3V supply. At 300MSPS sampling rate, the ADC achieves no less than 6 ENOB with input signal lower than 60MHz. It has the highest input bandwidth of 60MHz reported in the literature for this type of CMOS ADC with similar resolution and sample rate

Design and debugging of multi-step analog to digital converters

With the fast advancement of CMOS fabrication technology, more and more signal-processing functions are implemented in the digital domain for a lower cost, lower power consumption, higher yield, and higher re-configurability. The trend of increasing integration level for integrated circuits has forced the A/D converter interface to reside on the same silicon in complex mixed-signal ICs containing mostly digital blocks for DSP and control. However, specifications of the converters in various applications emphasize high dynamic range and low spurious spectral performance. It is nontrivial to achieve this level of linearity in a monolithic environment where post-fabrication component trimming or calibration is cumbersome to implement for certain applications or/and for cost and manufacturability reasons. Additionally, as CMOS integrated circuits are accomplishing unprecedented integration levels, potential problems associated with device scaling – the short-channel effects – are also looming large as technology strides into the deep-submicron regime. The A/D conversion process involves sampling the applied analog input signal and quantizing it to its digital representation by comparing it to reference voltages before further signal processing in subsequent digital systems. Depending on how these functions are combined, different A/D converter architectures can be implemented with different requirements on each function. Practical realizations show the trend that to a first order, converter power is directly proportional to sampling rate. However, power dissipation required becomes nonlinear as the speed capabilities of a process technology are pushed to the limit. Pipeline and two-step/multi-step converters tend to be the most efficient at achieving a given resolution and sampling rate specification. This thesis is in a sense unique work as it covers the whole spectrum of design, test, debugging and calibration of multi-step A/D converters; it incorporates development of circuit techniques and algorithms to enhance the resolution and attainable sample rate of an A/D converter and to enhance testing and debugging potential to detect errors dynamically, to isolate and confine faults, and to recover and compensate for the errors continuously. The power proficiency for high resolution of multi-step converter by combining parallelism and calibration and exploiting low-voltage circuit techniques is demonstrated with a 1.8 V, 12-bit, 80 MS/s, 100 mW analog to-digital converter fabricated in five-metal layers 0.18-µm CMOS process. Lower power supply voltages significantly reduce noise margins and increase variations in process, device and design parameters. Consequently, it is steadily more difficult to control the fabrication process precisely enough to maintain uniformity. Microscopic particles present in the manufacturing environment and slight variations in the parameters of manufacturing steps can all lead to the geometrical and electrical properties of an IC to deviate from those generated at the end of the design process. Those defects can cause various types of malfunctioning, depending on the IC topology and the nature of the defect. To relive the burden placed on IC design and manufacturing originated with ever-increasing costs associated with testing and debugging of complex mixed-signal electronic systems, several circuit techniques and algorithms are developed and incorporated in proposed ATPG, DfT and BIST methodologies. Process variation cannot be solved by improving manufacturing tolerances; variability must be reduced by new device technology or managed by design in order for scaling to continue. Similarly, within-die performance variation also imposes new challenges for test methods. With the use of dedicated sensors, which exploit knowledge of the circuit structure and the specific defect mechanisms, the method described in this thesis facilitates early and fast identification of excessive process parameter variation effects. The expectation-maximization algorithm makes the estimation problem more tractable and also yields good estimates of the parameters for small sample sizes. To allow the test guidance with the information obtained through monitoring process variations implemented adjusted support vector machine classifier simultaneously minimize the empirical classification error and maximize the geometric margin. On a positive note, the use of digital enhancing calibration techniques reduces the need for expensive technologies with special fabrication steps. Indeed, the extra cost of digital processing is normally affordable as the use of submicron mixed signal technologies allows for efficient usage of silicon area even for relatively complex algorithms. Employed adaptive filtering algorithm for error estimation offers the small number of operations per iteration and does not require correlation function calculation nor matrix inversions. The presented foreground calibration algorithm does not need any dedicated test signal and does not require a part of the conversion time. It works continuously and with every signal applied to the A/D converter. The feasibility of the method for on-line and off-line debugging and calibration has been verified by experimental measurements from the silicon prototype fabricated in standard single poly, six metal 0.09-µm CMOS process

Digital Signal Processing with Signal-Derived Timing: Analysis and Implementation

This work investigates two different digital signal processing (DSP) approaches that rely on signal-derived timing: continuous-time (CT) DSP and variable-rate DSP. Both approaches enable designs of energy-efficient signal processing systems by relating their operation rates to the input activity. The majority of this thesis focuses on CT-DSP, whose operations are completely digital in CT, without the use of a clock. The spectral features of CT digital signals are analyzed first, demonstrating a general pattern of the quantization noise spectrum added in CT amplitude quantization. Then the focus is narrowed to the investigations of the system characteristics and architecture of CT digital infinite-impulse-response (IIR) filters, which are barely studied in the previous work on this topic. This thesis discusses and addresses previously unreported stability issue in CT digital IIR filters with the presence of delay-line mismatches and proposes an innovative method to design high-order CT digital IIR filters with only two tap delays. Introducing an event detector allows the operation rate of a CT digital IIR filter to closely track the input activity even though it is a feedback system. For the first time, the filtered CT digital signal is converted to a synchronous digital signal. This facilitates integrating the CT digital filter and conventional discrete-time systems and expands the applications of the former. This discussion uses a computationally efficient interpolation filter to improve the signal accuracy of the synchronous digital output. On the circuit level, a new delay-cell design is introduced. It ensures low jitter, good matching, robust communication with adjacent circuits and event-independent delay. An integrated circuit (IC) with all these ideas adopted was fabricated in a TSMC 65 nm LP CMOS process. It is the first IC implementation of a CT digital IIR filter. It can process signals with a data rate up to 20 MHz. Thanks to the IIR response and the 16-bit resolution used in the system, the implemented filter can achieve a frequency response much more versatile and accurate than the CT digital filters in prior art. The implemented system features an agile power adaptive to input activity, varying from 2.32mW (full activity) to 40μW (idle) with no power-management circuitry. The second part of the thesis discusses a variable-rate DSP capable of processing samples with a variable sampling rate. The clock rate in the variable-rate DSP tracks the input sampling rate. Compared to a fixed-rate DSP, the proposed system has a lower output data rate and hence is more computationally efficient. A reconstruction filter with a variable cutoff frequency is used to reconstruct the output. The signal-to-noise ratio remains fixed when the sampling rate changes

OFDM under Oscillator Phase Noise : Contributions to Analysis and Estimation Methods

Most modern transmitters and receivers involve an analog front-end unit and a digital back-end unit. The digital back-end is responsible for information processing which involves thefollowing: redundancy removal from information; information representation; improvinginformation resilience; and information correction. The analog front-end is responsible forinformation transmission and reception. The information processing algorithms developedand implemented in the digital back-end assume a linear and noiseless analog front-end which,in reality, is not the case. This renders some of the information processing algorithms to be lesseffective in practice. The focus of this thesis is on orthogonal frequency-division multiplexing(OFDM) systems under the influence of oscillator phase noise. OFDM is an efficientinformation representation technique used in many communication systems. On the otherhand, phase noise is one type of undesired noise that occurs in the oscillator device used in theanalog front-end. It arises due to the imperfect task of frequency conversion, performed by theoscillator device, between baseband and radio frequency.  This thesis contributes to the areas of analysis and estimation in OFDM systems under theinfluence of oscillator phase noise. With regard to analysis, this thesis contributes by derivingthe channel capacity assuming a Gaussian input alphabet. The aim here is to show bothquantitatively and qualitatively the degradation in performance of the OFDM system in thepresence of phase noise. The analysis is conducted for phase noise processes that occur in bothfree-running and phase-locked loop based oscillators and also extended to include the effect ofcarrier frequency offset. With regard to estimation, two new phase noise estimation algorithmsare proposed in this thesis. In particular, these algorithms restrict the search space to a specific subset, where the desired phase noise parameter of interest is shown to lie. For example, in the first estimation method, possible subspaces in which the desired phase noise spectral vector may lie are used in the estimation step. In the second method, the geometry of the desired phase noise spectral vector is used in the estimation step. Specifically, this geometry corresponds to a non-convex set described by a set of quadratic forms that involve permutation matrices. By restricting the search space to this set, the accuracy of phase noise estimation can be improved

Circuit techniques for low-voltage and high-speed A/D converters

The increasing digitalization in all spheres of electronics applications, from telecommunications systems to consumer electronics appliances, requires analog-to-digital converters (ADCs) with a higher sampling rate, higher resolution, and lower power consumption. The evolution of integrated circuit technologies partially helps in meeting these requirements by providing faster devices and allowing for the realization of more complex functions in a given silicon area, but simultaneously it brings new challenges, the most important of which is the decreasing supply voltage. Based on the switched capacitor (SC) technique, the pipelined architecture has most successfully exploited the features of CMOS technology in realizing high-speed high-resolution ADCs. An analysis of the effects of the supply voltage and technology scaling on SC circuits is carried out, and it shows that benefits can be expected at least for the next few technology generations. The operational amplifier is a central building block in SC circuits, and thus a comparison of the topologies and their low voltage capabilities is presented. It is well-known that the SC technique in its standard form is not suitable for very low supply voltages, mainly because of insufficient switch control voltage. Two low-voltage modifications are investigated: switch bootstrapping and the switched opamp (SO) technique. Improved circuit structures are proposed for both. Two ADC prototypes using the SO technique are presented, while bootstrapped switches are utilized in three other prototypes. An integral part of an ADC is the front-end sample-and-hold (S/H) circuit. At high signal frequencies its linearity is predominantly determined by the switches utilized. A review of S/H architectures is presented, and switch linearization by means of bootstrapping is studied and applied to two of the prototypes. Another important parameter is sampling clock jitter, which is analyzed and then minimized with carefully-designed clock generation and buffering. The throughput of ADCs can be increased by using parallelism. This is demonstrated on the circuit level with the double-sampling technique, which is applied to S/H circuits and a pipelined ADC. An analysis of nonidealities in double-sampling is presented. At the system level parallelism is utilized in a time-interleaved ADC. The mismatch of parallel signal paths produces errors, for the elimination of which a timing skew insensitive sampling circuit and a digital offset calibration are developed. A total of seven prototypes are presented: two double-sampled S/H circuits, a time-interleaved ADC, an IF-sampling self-calibrated pipelined ADC, a current steering DAC with a deglitcher, and two pipelined ADCs employing the SO technique.reviewe

Direct digital synthesizers : theory, design and applications

Traditional designs of high bandwidth frequency synthesizers employ the use of a phase-locked-loop (PLL). A direct digital synthesizer (DDS) provides many significant advantages over the PLL approaches. Fast settling time, sub-Hertz frequency resolution, continuous-phase switching response and low phase noise are features easily obtainable in the DDS systems. Although the principle of the DDS has been known for many years, the DDS did not play a dominant role in wideband frequency generation until recent years. Earlier DDSs were limited to produce narrow bands of closely spaced frequencies, due to limitations of digital logic and D/A-converter technologies. Recent advantages in integrated circuit (IC) technologies have brought about remarkable progress in this area. By programming the DDS, adaptive channel bandwidths, modulation formats, frequency hopping and data rates are easily achieved. This is an important step towards a "software-radio" which can be used in various systems. The DDS could be applied in the modulator or demodulator in the communication systems. The applications of DDS are restricted to the modulator in the base station. The aim of this research was to find an optimal front-end for a transmitter by focusing on the circuit implementations of the DDS, but the research also includes the interface to baseband circuitry and system level design aspects of digital communication systems. The theoretical analysis gives an overview of the functioning of DDS, especially with respect to noise and spurs. Different spur reduction techniques are studied in detail. Four ICs, which were the circuit implementations of the DDS, were designed. One programmable logic device implementation of the CORDIC based quadrature amplitude modulation (QAM) modulator was designed with a separate D/A converter IC. For the realization of these designs some new building blocks, e.g. a new tunable error feedback structure and a novel and more cost-effective digital power ramp generator, were developed.reviewe

K-Delta-1-Sigma Modulators for Wideband Analog-to-Digital Conversion

As CMOS technology scales, the transistor speed increases enabling higher speed communications and more complex systems. These benefits come at the cost of decreasing inherent device gain, increased transistor leakage currents, and additional mismatches due to process variations. All of these drawbacks affect the design of high-resolution analog-to-digital converters (ADCs) in nano-CMOS processes. To move towards an ADC topology useful in these small processes a first-order K-Delta-1-Sigma (KD1S) modulator-based ADC was proposed. The KD1S topology employs inherent time-interleaving with a shared integrator and K-quantizing feedback paths and can potentially achieve significantly higher conversion bandwidths when compared to the traditional switched-capacitor delta-sigma ADCs. The shared integrator in the KD1S modulator settles over a half the clock period and the op-amp is designed to operate at the base clock frequency. In this dissertation, the first-order KD1S modulator topology is analyzed for the effects of the non-idealities introduced by the K-path operation of the switched-capacitor integrator. Then, the concept of KD1S modulator is extended to higher-order modulators in order to achieve superior noise-shaping performance. A systematic synthesis method has been developed to design and simulate higher-order KD1S modulators at the system level. In order to demonstrate the developed theory, a prototype second-order KD1S modulator has been designed and fabricated in a 500-nm CMOS technology. The second-order KD1S modulator exhibits wideband noise-shaping with an SNDR of 42.7 dB or 6.81 bits in resolution for Kpath = 8 paths, an effective sampling rate of ƒs,new=800 MHz, effective oversampling ratio Kpath•OSR=64 and a signal bandwidth of 6.25 MHz. The second-order KD1S modulator consumes an average current of 3.0 mA from the 5 V supply and occupies an area of 0.55 mm2