Methodology for Mismatch Reduction in Time-Interleaved ADCs

This paper presents a methodology to minimize mismatch errors in time-interleaved analog-to-digital converters (ADC) by means of averaging multiple channels. A simple algorithm improving both spurious free dynamic range (SFDR) and signal-to-noise and distortion ratio (SINAD) is demonstrated. The presented technique provides robustness against inaccurately identified mismatch errors and does not require computationally expensive post-processing of the signal

Architectural Improvements Towards an Efficient 16-18 Bit 100-200 MSPS ADC

As Data conversion systems continue to improve in speed and resolution, increasing demands are placed on the performance of high-speed Analog to Digital Conversion systems. This work makes a survey about all these and proposes a suitable architecture in order to achieve the desired specifications of 100-200MS/s with 16-18 bit of resolution. The main architecture is based on paralleled structures in order to achieve high sampling rate and at the same time high resolution. In order to solve problems related to Time-interleaved architectures, an advanced randomization method was introduced. It combines randomization and spectral shaping of mismatches. With a simple low-pass filter the method can, compared to conventional randomization algorithms, improve the SFDR as well as the SINAD. The main advantage of this technique over previous ones is that, because the algorithm only need that ADCs are ordered basing on their time mismatches, the absolute accuracy of the mismatch identification method does not matter and, therefore, the requirements on the timing mismatch identification are very low. In addition to that, this correction system uses very simple algorithms able to correct not only for time but also for gain and offset mismatches

Design of Energy-Efficient A/D Converters with Partial Embedded Equalization for High-Speed Wireline Receiver Applications

As the data rates of wireline communication links increases, channel impairments such as skin effect, dielectric loss, fiber dispersion, reflections and cross-talk become more pronounced. This warrants more interest in analog-to-digital converter (ADC)-based serial link receivers, as they allow for more complex and flexible back-end digital signal processing (DSP) relative to binary or mixed-signal receivers. Utilizing this back-end DSP allows for complex digital equalization and more bandwidth-efficient modulation schemes, while also displaying reduced process/voltage/temperature (PVT) sensitivity. Furthermore, these architectures offer straightforward design translation and can directly leverage the area and power scaling offered by new CMOS technology nodes. However, the power consumption of the ADC front-end and subsequent digital signal processing is a major issue. Embedding partial equalization inside the front-end ADC can potentially result in lowering the complexity of back-end DSP and/or decreasing the ADC resolution requirement, which results in a more energy-effcient receiver. This dissertation presents efficient implementations for multi-GS/s time-interleaved ADCs with partial embedded equalization. First prototype details a 6b 1.6GS/s ADC with a novel embedded redundant-cycle 1-tap DFE structure in 90nm CMOS. The other two prototypes explain more complex 6b 10GS/s ADCs with efficiently embedded feed-forward equalization (FFE) and decision feedback equalization (DFE) in 65nm CMOS. Leveraging a time-interleaved successive approximation ADC architecture, new structures for embedded DFE and FFE are proposed with low power/area overhead. Measurement results over FR4 channels verify the effectiveness of proposed embedded equalization schemes. The comparison of fabricated prototypes against state-of-the-art general-purpose ADCs at similar speed/resolution range shows comparable performances, while the proposed architectures include embedded equalization as well

Multi-band Oversampled Noise Shaping Analog to Digital Conversion

Oversampled noise shaping analog to digital (A/D) converters, which are commonly known as delta-sigma (ΔΣ) converters, have the ability to convert relatively low bandwidth signals with very high resolution. Such converters achieve their high resolution by oversampling, as well as processing the signal and quantization noise with different transfer functions. The signal transfer function (STF) is typically a delay over the signal band while the noise transfer function (NTF) is designed to attenuate quantization noise in the signal band. A side effect of the NTF is an amplification of the noise outside the signal band. Thus, a digital filter subsequently attenuates the out-of-band quantization noise. The focus of this thesis is the investigation of ΔΣ architectures that increase the bandwidth where high resolution conversion can be achieved. It uses parallel architectures exploiting frequency or time slicing to meet this objective. Frequency slicing involves quantizing different portions of the signal frequency spectrum using several quantizers in parallel and then combining the results of the quantizers to form an overall result. Time slicing involves quantizing various groups of time domain signal samples with different quantizers in parallel and then combining the results of the quantizers to form an overall output. Several interesting observations can be made from this general perspective of frequency and time slicing. Although the representation of a signal are completely equivalent in time or frequency, the thesis shows that this is not the case for known frequency and time sliced A/D architectures. The performance of such systems under ideal conditions are compared for PCM as well as for ΔΣ A/D converters. A multi-band frequency sliced architecture for delta-sigma conversion is proposed and its performance is included in the above comparison. The architecture uses modulators which realize different NTFs for different portions of the signal band. Each band is converted in parallel. A bank of FIR filters attenuates the out of-band noise for each band and achieves perfect reconstruction of the signal component. A design procedure is provided for the design of the filter bank with reduced computational complexity. The use of complex NTFs in the multi-band ΔΣ architecture is also proposed. The peformance of real and complex NTFs is compared. Performance evaluations are made for ideal systems as well as systems suffering from circuit implementation imperfections such as finite opamp gain and mismatched capacitor ratios

A self-adaptive frequency response compensation method for a TIADC system

Time interleaving is one of the most efficient techniques employed in high-speed sampling systems. However, the frequency response mismatch among different channels will create distortion tones that degrade the system performance. In this paper, a selfadaptive frequency response mismatch compensation method is presented, where the design of compensation filter is optimized with a self-adapting strategy. This digital postprocessing technique realizes the compensation of frequency response effectively and also the increase of the digital bandwidth of the acquisition system. MATLAB-based simulation and an actual two-channel acquisition system test verify the effectiveness of the algorithm

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