A Robust Carrier Frequency Offset Estimation Algorithm in Burst Mode Multicarrier CDMA based Ad Hoc Networks

The future wireless communication systems demand very high data rates, anti-jamming ability and multiuser support. People want large amount of data to be continuously accessible in their personal devices. Direct Sequence (DS) spread spectrum based techniques such as Code Division Multiple Access (CDMA) fulfil these requirements but, at the same time, suffer from the Intersymbol Interference (ISI). Multicarrier CDMA (MC-CDMA) is an emerging technology to be used in mobile devices operating in an ad hoc setting due to its immunity towards ISI and having all the advantages of spread spectrum communication. One of the major problems with MC-CDMA is the high sensitivity towards carrier frequency offsets caused due to the inherent inaccuracy of crystal oscillators. This carrier frequency offset destroys the orthogonality of the subcarriers resulting in Intercarrier Interference (ICI). In this paper, we propose a computationally efficient algorithm based on Fast Fourier Transform (FFT) and biquadratic Lagrange interpolation. The FFT is based on the use of overlapping windows for each frame of the data instead of non-overlapping windows. This gives a coarse estimate of the frequency offset which is refined by the successive application of Lagrange quadratic interpolation to the samples in the vicinity of FFT peak. The proposed algorithm has been applied to the multiuser ad hoc network and simulated in Stanford University Interim (SUI) channels. It has been shown by simulations that the proposed algorithm provides better performance of almost 1~2 dB as compared to the well-known algorithms

Enhanced Blind Maximum Ratio Combining in Broadcasting Systems

We propose an enhanced blind maximum ratio combiner (BMRC) allowing for a transmit signal independent diversity combining in multi-antenna receivers. The underlying Multi-Channel Frequency Least Mean Squares (MCFLMS) algorithm comes with reasonable computational complexity and estimates the channel impulse response for each receive antenna iteratively by means of second order statistics. In literature, the MCFLMS algorithm is mainly applied to audio signals. In this work, we describe several enhancements of this algorithm to ensure its proper convergence with oversampled communication signals which are distorted by frequency-selective fast-fading channels. In addition, we provide BER simulation results for a 1x2 SIMO DVB-T2 system and show that our blind MRC can even outperform conventional pilot-based MRC at the receiver side

The SST Multi-G-Sample/s Switched Capacitor Array Waveform Recorder with Flexible Trigger and Picosecond-Level Timing Accuracy

The design and performance of a multi-G-sample/s fully-synchronous analog transient waveform recorder I.C. ("SST") with fast and flexible trigger capabilities is presented. Containing 4 channels of 256 samples per channel and fabricated in a 0.25 {\mu}m CMOS process, it has a 1.9V input range on a 2.5V supply, achieves 12 bits of dynamic range, and uses ~160 mW while operating at 2 G-samples/s and full trigger speeds. With a standard 50 Ohm input source, the SST's analog input bandwidth is ~1.3 GHz within about +/-0.5 dB and reaches a -3 dB bandwidth of 1.5 GHz. The SST's internal sample clocks are generated synchronously via a shift register driven by an external LVDS oscillator, interleaved to double its speed (e.g., a 1 GHz clock yields 2 G-samples/s). It can operate over 6 orders of magnitude in sample rates (2 kHz to 2 GHz). Only three active control lines are necessary for operation: Reset, Start/Stop and Read-Clock. Each of the four channels integrates dual-threshold discrimination of signals with ~1 mV RMS resolution at >600 MHz bandwidth. Comparator results are directly available for simple threshold monitoring and rate control. The High and Low discrimination can also be AND'd over an adjustable window of time in order to exclusively trigger on bipolar impulsive signals. Trigger outputs can be CMOS or low-voltage differential signals, e.g. 1.2V CMOS or positive-ECL (0-0.8V) for low noise. After calibration, the imprecision of timing differences between channels falls in a range of 1.12-2.37 ps sigma at 2 G-samples/s.Comment: 9 pages, 16 figures, 1 tabl

Generalized Fast-Convolution-based Filtered-OFDM: Techniques and Application to 5G New Radio

This paper proposes a generalized model and methods for fast-convolution (FC)-based waveform generation and processing with specific applications to fifth generation new radio (5G-NR). Following the progress of 5G-NR standardization in 3rd generation partnership project (3GPP), the main focus is on subband-filtered cyclic prefix (CP) orthogonal frequency-division multiplexing (OFDM) processing with specific emphasis on spectrally well localized transmitter processing. Subband filtering is able to suppress the interference leakage between adjacent subbands, thus supporting different numerologies for so-called bandwidth parts as well as asynchronous multiple access. The proposed generalized FC scheme effectively combines overlapped block processing with time- and frequency-domain windowing to provide highly selective subband filtering with very low intrinsic interference level. Jointly optimized multi-window designs with different allocation sizes and design parameters are compared in terms of interference levels and implementation complexity. The proposed methods are shown to clearly outperform the existing state-of-the-art windowing and filtering-based methods.Comment: To appear in IEEE Transactions on Signal Processin

Continuous-Time and Companding Digital Signal Processors Using Adaptivity and Asynchronous Techniques

The fully synchronous approach has been the norm for digital signal processors (DSPs) for many decades. Due to its simplicity, the classical DSP structure has been used in many applications. However, due to its rigid discrete-time operation, a classical DSP has limited efficiency or inadequate resolution for some emerging applications, such as processing of multimedia and biological signals. This thesis proposes fundamentally new approaches to designing DSPs, which are different from the classical scheme. The defining characteristic of all new DSPs examined in this thesis is the notion of "adaptivity" or "adaptability." Adaptive DSPs dynamically change their behavior to adjust to some property of their input stream, for example the rate of change of the input. This thesis presents both enhancements to existing adaptive DSPs, as well as new adaptive DSPs. The main class of DSPs that are examined throughout the thesis are continuous-time (CT) DSPs. CT DSPs are clock-less and event-driven; they naturally adapt their activity and power consumption to the rate of their inputs. The absence of a clock also provides a complete avoidance of aliasing in the frequency domain, hence improved signal fidelity. The core of this thesis deals with the complete and systematic design of a truly general-purpose CT DSP. A scalable design methodology for CT DSPs is presented. This leads to the main contribution of this thesis, namely a new CT DSP chip. This chip is the first general-purpose CT DSP chip, able to process many different classes of CT and synchronous signals. The chip has the property of handling various types of signals, i.e. various different digital modulations, both synchronous and asynchronous, without requiring any reconfiguration; such property is presented for the first time CT DSPs and is impossible for classical DSPs. As opposed to previous CT DSPs, which were limited to using only one type of digital format, and whose design was hard to scale for different bandwidths and bit-widths, this chip has a formal, robust and scalable design, due to the systematic usage of asynchronous design techniques. The second contribution of this thesis is a complete methodology to design adaptive delay lines. In particular, it is shown how to make the granularity, i.e. the number of stages, adaptive in a real-time delay line. Adaptive granularity brings about a significant improvement in the line's power consumption, up to 70% as reported by simulations on two design examples. This enhancement can have a direct large power impact on any CT DSP, since a delay line consumes the majority of a CT DSP's power. The robust methodology presented in this thesis allows safe dynamic reconfiguration of the line's granularity, on-the-fly and according to the input traffic. As a final contribution, the thesis also examines two additional DSPs: one operating the CT domain and one using the companding technique. The former operates only on level-crossing samples; the proposed methodology shows a potential for high-quality outputs by using a complex interpolation function. Finally, a companding DSP is presented for MPEG audio. Companding DSPs adapt their dynamic range to the amplitude of their input; the resulting can offer high-quality outputs even for small inputs. By applying companding to MPEG DSPs, it is shown how the DSP distortion can be made almost inaudible, without requiring complex arithmetic hardware

Time measurements by means of digital sampling techniques: a study case of 100 ps FWHM time resolution with a 100 MSample/s, 12 bit digitizer

Abstract An application of digital sampling techniques is presented which can simplify experiments involving sub-nanosecond time-mark determinations and energy measurements with nuclear detectors, used for Pulse Shape Analysis and Time of Flight measurements in heavy ion experiments. The basic principles of the method are discussed as well as the main parameters that influence the accuracy of the measurements. The method allows to obtain both time and amplitude information with an electronic chain simply consisting of a charge preamplifier and a fast high resolution ADC (in the present application: 100 MSample/s , 12 bit ) coupled to an efficient on-line software. In particular an accurate Time of Flight information can be obtained by mixing a beam related time signal with the output of the preamplifier. Examples of this technique applied to Silicon detectors in heavy-ions experiments involving particle identification via Pulse Shape analysis and Time of Flight measurements are presented. The system is suited for applications to large detector arrays and to different kinds of detectors