1,101 research outputs found

    Adaptive Estimation and Compensation of the Time Delay in a Periodic Non-uniform Sampling Scheme

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    High sampling rate Analog-to-Digital Converters (ADCs) can be obtained by time-interleaving low rate (and thus low cost) ADCs into so-called Time-Interleaved ADCs (TI-ADCs). Nevertheless increasing the sampling frequency involves an increasing sensibility of the system to desynchronization between the different ADCs that leads to time-skew errors, impacting the system with non linear distortions. The estimation and compensation of these errors are considered as one of the main challenge to deal with in TI-ADCs. Some methods have been previously proposed, mainly in the field of circuits and systems, to estimate the time-skew error but they mainly involve hardware correction and they lack of flexibility, using an inflexible uniform sampling reference. In this paper, we propose to model the output of L interleaved and desynchronized ADCs with a sampling scheme called Periodic Non-uniform Sampling of order L (PNSL). This scheme has been initially proposed as an alternative to uniform sampling for aliasing cancellation, particularly in the case of bandpass signals. We use its properties here to develop a flexible on-line digital estimation and compensation method of the time delays between the desynchronized channels. The estimated delay is exploited in the PNSL reconstruction formula leading to an accurate reconstruction without hardware correction and without any need to adapt the sampling operation. Our method can be used in a simple Built-In Self-Test (BIST) strategy with the use of learning sequences and our model appears more flexible and less electronically expensive, following the principles of “Dirty Radio Frequency” paradigm: designing imperfect analog circuits with subsequently digital corrections of these imperfections

    Adaptive Estimation and Compensation of the Time Delay in a Periodic Non-uniform Sampling Scheme

    Get PDF
    High sampling rate Analog-to-Digital Converters (ADCs) can be obtained by time-interleaving low rate (and thus low cost) ADCs into so-called Time-Interleaved ADCs (TI-ADCs). Nevertheless increasing the sampling frequency involves an increasing sensibility of the system to desynchronization between the different ADCs that leads to time-skew errors, impacting the system with non linear distortions. The estimation and compensation of these errors are considered as one of the main challenge to deal with in TI-ADCs. Some methods have been previously proposed, mainly in the field of circuits and systems, to estimate the time-skew error but they mainly involve hardware correction and they lack of flexibility, using an inflexible uniform sampling reference. In this paper, we propose to model the output of L interleaved and desynchronized ADCs with a sampling scheme called Periodic Non-uniform Sampling of order L (PNSL). This scheme has been initially proposed as an alternative to uniform sampling for aliasing cancellation, particularly in the case of bandpass signals. We use its properties here to develop a flexible on-line digital estimation and compensation method of the time delays between the desynchronized channels. The estimated delay is exploited in the PNSL reconstruction formula leading to an accurate reconstruction without hardware correction and without any need to adapt the sampling operation. Our method can be used in a simple Built-In Self-Test (BIST) strategy with the use of learning sequences and our model appears more flexible and less electronically expensive, following the principles of “Dirty Radio Frequency” paradigm: designing imperfect analog circuits with subsequently digital corrections of these imperfections

    Concepts for smart AD and DA converters

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    This thesis studies the `smart' concept for application to analog-to-digital and digital-to-analog converters. The smart concept aims at improving performance - in a wide sense - of AD/DA converters by adding on-chip intelligence to extract imperfections and to correct for them. As the smart concept can correct for certain imperfections, it can also enable the use of more efficient architectures, thus yielding an additional performance boost. Chapter 2 studies trends and expectations in converter design with respect to applications, circuit design and technology evolution. Problems and opportunities are identfied, and an overview of performance criteria is given. Chapter 3 introduces the smart concept that takes advantage of the expected opportunities (described in chapter 2) in order to solve the anticipated problems. Chapter 4 applies the smart concept to digital-to-analog converters. In the discussed example, the concept is applied to reduce the area of the analog core of a current-steering DAC. It is shown that a sub-binary variable-radix approach reduces the area of the current-source elements substantially (10x compared to state-of-the-art), while maintaining accuracy by a self-measurement and digital pre-correction scheme. Chapter 5 describes the chip implementation of the sub-binary variable-radix DAC and discusses the experimental results. The results confirm that the sub-binary variable-radix design can achieve the smallest published current-source-array area for the given accuracy (12bit). Chapter 6 applies the smart concept to analog-to-digital converters, with as main goal the improvement of the overall performance in terms of a widely used figure-of-merit. Open-loop circuitry and time interleaving are shown to be key to achieve high-speed low-power solutions. It is suggested to apply a smart approach to reduce the effect of the imperfections, unintentionally caused by these key factors. On high-level, a global picture of the smart solution is proposed that can solve the problems while still maintaining power-efficiency. Chapter 7 deals with the design of a 500MSps open-loop track-and-hold circuit. This circuit is used as a test case to demonstrate the proposed smart approaches. Experimental results are presented and compared against prior art. Though there are several limitations in the design and the measurement setup, the measured performance is comparable to existing state-of-the-art. Chapter 8 introduces the first calibration method that counteracts the accuracy issues of the open-loop track-and-hold. A description of the method is given, and the implementation of the detection algorithm and correction circuitry is discussed. The chapter concludes with experimental measurement results. Chapter 9 introduces the second calibration method that targets the accuracy issues of time-interleaved circuits, in this case a 2-channel version of the implemented track-and-hold. The detection method, processing algorithm and correction circuitry are analyzed and their implementation is explained. Experimental results verify the usefulness of the method

    Digital Background Self-Calibration Technique for Compensating Transition Offsets in Reference-less Flash ADCs

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    This Dissertation focusses on proving that background calibration using adaptive algorithms are low-cost, stable and effective methods for obtaining high accuracy in flash A/D converters. An integrated reference-less 3-bit flash ADC circuit has been successfully designed and taped out in UMC 180 nm CMOS technology in order to prove the efficiency of our proposed background calibration. References for ADC transitions have been virtually implemented built-in in the comparators dynamic-latch topology by a controlled mismatch added to each comparator input front-end. An external very simple DAC block (calibration bank) allows control the quantity of mismatch added in each comparator front-end and, therefore, compensate the offset of its effective transition with respect to the nominal value. In order to assist to the estimation of the offset of the prototype comparators, an auxiliary A/D converter with higher resolution and lower conversion speed than the flash ADC is used: a 6-bit capacitive-DAC SAR type. Special care in synchronization of analogue sampling instant in both ADCs has been taken into account. In this thesis, a criterion to identify the optimum parameters of the flash ADC design with adaptive background calibration has been set. With this criterion, the best choice for dynamic latch architecture, calibration bank resolution and flash ADC resolution are selected. The performance of the calibration algorithm have been tested, providing great programmability to the digital processor that implements the algorithm, allowing to choose the algorithm limits, accuracy and quantization errors in the arithmetic. Further, systematic controlled offset can be forced in the comparators of the flash ADC in order to have a more exhaustive test of calibration

    High-Speed Low-Power Analog to Digital Converter for Digital Beam Forming Systems

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    abstract: Time-interleaved analog to digital converters (ADCs) have become critical components in high-speed communication systems. Consumers demands for smaller size, more bandwidth and more features from their communication systems have driven the market to use modern complementary metal-oxide-semiconductor (CMOS) technologies with shorter channel-length transistors and hence a more compact design. Downscaling the supply voltage which is required in submicron technologies benefits digital circuits in terms of power and area. Designing accurate analog circuits, however becomes more challenging due to the less headroom. One way to overcome this problem is to use calibration to compensate for the loss of accuracy in analog circuits. Time-interleaving increases the effective data conversion rate in ADCs while keeping the circuit requirements the same. However, this technique needs special considerations as other design issues associated with using parallel identical channels emerge. The first and the most important is the practical issue of timing mismatch between channels, also called sample-time error, which can directly affect the performance of the ADC. Many techniques have been developed to tackle this issue both in analog and digital domains. Most of these techniques have high complexities especially when the number of channels exceeds 2 and some of them are only valid when input signal is a single tone sinusoidal which limits the application. This dissertation proposes a sample-time error calibration technique which bests the previous techniques in terms of simplicity, and also could be used with arbitrary input signals. A 12-bit 650 MSPS pipeline ADC with 1.5 GHz analog bandwidth for digital beam forming systems is designed in IBM 8HP BiCMOS 130 nm technology. A front-end sample-and-hold amplifier (SHA) was also designed to compare with an SHA-less design in terms of performance, power and area. Simulation results show that the proposed technique is able to improve the SNDR by 20 dB for a mismatch of 50% of the sampling period and up to 29 dB at 37% of the Nyquist frequency. The designed ADC consumes 122 mW in each channel and the clock generation circuit consumes 142 mW. The ADC achieves 68.4 dB SNDR for an input of 61 MHz.Dissertation/ThesisDoctoral Dissertation Electrical Engineering 201

    Time-Interleaved Analog-to-Digital Converter (TIADC) Compensation Using Multichannel Filters

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    Published methods that employ a filter bank for compensating the timing and bandwidth mismatches of an M-channel time-interleaved analog-to-digital converter (TIADC) were developed based on the fact that each sub-ADC channel is a downsampled version of the analog input. The output of each sub-ADC is filtered in such a way that, when all the filter outputs are summed, the aliasing components are minimized. If each channel of the filter bank has N coefficients, the optimization of the coefficients requires computing the inverse of an MN times MN matrix if the weighted least squares (WLS) technique is used as the optimization tool. In this paper, we present a multichannel filtering approach for TIADC mismatch compensation. We apply the generalized sampling theorem to directly estimate the ideal output of each sub-ADC using the outputs of all the sub-ADCs. If the WLS technique is used as the optimization tool, the dimension of the matrix to be inversed is N times N. For the same number of coefficients (and also the same spurious component performance given sufficient arithmetic precision), our technique is computationally less complex and more robust than the filter-bank approach. If mixed integer linear programming is used as the optimization tool to produce filters with coefficient values that are integer powers of two, our technique produces a saving in computing resources by a factor of approximately (100.2N(M- 1)/(M-1) in the TIADC filter design.published_or_final_versio
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