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

Novel Systematic Phase Noise Reduction Techniques for Phase Interpolator Clock and Data Recovery

This work focused on high-speed source-synchronous clock and multi-channel data receivers for inter-chip communications. Designs of inter-chip communication are becoming increasingly difficult with the rise in clock rates and the reduction in voltage supplies. Data transmissions at rates of gigabits per second require a fast and accurate clock and data recovery system on the front end of receivers. Many designs allow for source-synchronous clocking architectures, but this work focused on a dual-loop with a phase-locked loop for frequency tracking and phase integrators for tracking each individual data lane. Limitations with the phase interpolator architecture cause systematic jitter, reducing the data eye. Various techniques exist that aim to reduce or eliminate this systematic jitter from phase interpolator architectures. A technique based on digital lock detection was developed for this work that eliminates the phase interpolator systematic jitter

Correction of errors and harmonic distortion in pulse-width modulation of digital signals

Article number 153991Pulse-Width (PW) modulation is widely used in those applications where an analog or digital signal has to be encoded in the time domain as a binary stream, such as switched-mode power amplifiers in transmitters of modern telecommunication standards, high-resolution digital signal conversion using single-bit digital-to-analog converters, and many others. Due to the fact that digital signals are sampled in the time domain, the quality of the resulting PW modulated waveforms is worsened by harmonic distortion. Multilevel PW modulation has been proposed to reduce these adverse effects, but the modulated waveform is no longer binary. In this paper, the mechanisms by which harmonic distortion is produced are analyzed. As a result, the distortion terms are mathematically quantified and used to correct the errors. Note that a correction network based on a simple subtraction of the distortion terms from the PW modulated signal would produce a waveform that would no longer be binary. The proposed correction network is implemented in the digital domain and, by means of a sigma-delta modulator, preserves the binary feature of the PW modulated output.Ministerio de Ciencia, Innovación y Universidades (España) RTI201- 099189-B-C2

Iterative Receiver for MIMO-OFDM System with ICI Cancellation and Channel Estimation

As a multi-carrier modulation scheme, Orthogonal Frequency Division Multiplexing (OFDM) technique can achieve high data rate in frequency-selective fading channels by splitting a broadband signal into a number of narrowband signals over a number of subcarriers, where each subcarrier is more robust to multipath. The wireless communication system with multiple antennas at both the transmitter and receiver, known as multiple-input multiple-output (MIMO) system, achieves high capacity by transmitting independent information over different antennas simultaneously. The combination of OFDM with multiple antennas has been considered as one of most promising techniques for future wireless communication systems. The challenge in the detection of a space-time signal is to design a low-complexity detector, which can efficiently remove interference resulted from channel variations and approach the interference-free bound. The application of iterative parallel interference canceller (PIC) with joint detection and decoding has been a promising approach. However, the decision statistics of a linear PIC is biased toward the decision boundary after the first cancellation stage. In this thesis, we employ an iterative receiver with a decoder metric, which considerably reduces the bias effect in the second iteration, which is critical for the performance of the iterative algorithm. Channel state information is required in a MIMO-OFDM system signal detection at the receiver. Its accuracy directly affects the overall performance of MIMO-OFDM systems. In order to estimate the channel in high-delay-spread environments, pilot symbols should be inserted among subcarriers before transmission. To estimate the channel over all the subcarriers, various types of interpolators can be used. In this thesis, a linear interpolator and a trigonometric interpolator are compared. Then we propose a new interpolator called the multi-tap method, which has a much better system performance. In MIMO-OFDM systems, the time-varying fading channels can destroy the orthogonality of subcarriers. This causes serious intercarrier interference (ICI), thus leading to significant system performance degradation, which becomes more severe as the normalized Doppler frequency increases. In this thesis, we propose a low-complexity iterative receiver with joint frequency- domain ICI cancellation and pilot-assisted channel estimation to minimize the effect of time-varying fading channels. At the first stage of receiver, the interference between adjacent subcarriers is subtracted from received OFDM symbols. The parallel interference cancellation detection with decision statistics combining (DSC) is then performed to suppress the interference from other antennas. By restricting the interference to a limited number of neighboring subcarriers, the computational complexity of the proposed receiver can be significantly reduced. In order to construct the time variant channel matrix in the frequency domain, channel estimation is required. However, an accurate estimation requiring complete knowledge of channel time variations for each block, cannot be obtained. For time- varying frequency-selective fading channels, the placement of pilot tones also has a significant impact on the quality of the channel estimates. Under the assumption that channel variations can be approximated by a linear model, we can derive channel state information (CSI) in the frequency domain and estimate time-domain channel parameters. In this thesis, an iterative low-complexity channel estimation method is proposed to improve the system performance. Pilot symbols are inserted in the transmitted OFDM symbols to mitigate the effect of ICI and the channel estimates are used to update the results of both the frequency domain equalizer and the PICDSC detector in each iteration. The complexity of this algorithm can be reduced because the matrices are precalculated and stored in the receiver when the placement of pilots symbols is fixed in OFDM symbols before transmission. Finally, simulation results show that the proposed MIMO-OFDM iterative receiver can effectively mitigate the effect of ICI and approach the ICI-free performance over time-varying frequency-selective fading channels

Design and development of mobile channel simulators using digital signal processing techniques

A mobile channel simulator can be constructed either in the time domain using a tapped delay line filter or in the frequency domain using the time variant transfer function of the channel. Transfer function modelling has many advantages over impulse response modelling. Although the transfer function channel model has been envisaged by several researchers as an alternative to the commonly employed tapped delay line model, so far it has not been implemented. In this work, channel simulators for single carrier and multicarrier OFDM system based on time variant transfer function of the channel have been designed and implemented using DSP techniques in SIMULINK. For a single carrier system, the simulator was based on Bello's transfer function channel model. Bello speculated that about 10Βτ(_MAX) frequency domain branches might result in a very good approximation of the channel (where в is the signal bandwidth and τ(_MAX) is the maximum excess delay of the multi-path channel). The simulation results showed that 10Bτ(_MAX) branches gave close agreement with the tapped delay line model(where Be is the coherence bandwidth). This number is π times higher than the previously speculated 10Bτ(_MAX).For multicarrier OFDM system, the simulator was based on the physical (PHY) layer standard for IEEE 802.16-2004 Wireless Metropolitan Area Network (WirelessMAN) and employed measured channel transfer functions at the 2.5 GHz and 3.5 GHz bands in the simulations. The channel was implemented in the frequency domain by carrying out point wise multiplication of the spectrum of OFDM time The simulator was employed to study BER performance of rate 1/2 and rate 3/4 coded systems with QPSK and 16-QAM constellations under a variety of measured channel transfer functions. The performance over the frequency selective channel mainly depended upon the frequency domain fading and the channel coding rate

N-bit ΔΣ optical transmitter for Digitized Radio Over Fiber Fronthaul Transmission

According to an estimate by the Global Technology, Media and Telecom (GTMT) team, global mobile data traffic grew 70% in 2012, which was nearly 12 times the size of the entire global Internet in 2000. In the future, the amount of data traffic will grow at a pace never seen before. Many recent forecasts project mobile data traffic to grow more than 24 between 2010 and 2015, and much higher beyond 2015. To catch up with the need and to remain competitive, network operators need to increase the broadband capability of their networks fast. This poses a big challenge for wireless communication system designers. Researchers have been working on innovative systems that will provide several Gbit/s over the air interface. Digitized radio over fiber (DROF) offers the capability to support various current and future wireless standards, independent of wireless system specifics if the carrier frequency falls within the passband of the ROF link. For example, the same ROF links should be able to transmit either time-division multiple access (TDMA), code-division multiple access (CDMA), or orthogonal frequency-division multiple access (OFDMA) radio signals without modification if their carrier frequencies are the same. Properly designed, the ROF link can also carry multiple RF carriers simultaneously in a subcarrier-multiplexing manner and support multi-standard radio. If the ROF link is properly designed, the portable device should be unaware of the existence of fiber in its radio path. Essentially, radio over fiber (ROF) is an analog communications system, and with DROF, the signal it carries is digital. Since nonlinear distortions, limited dynamic range, and cumulating noise are major concerns with the analog ROF backbone; alternative approaches are also investigated by researchers. One approach is to transmit a digitized RF signal over fiber from the base station to the radio access point. The falling cost of high-speed digital-to-analog converter (DAC) and analog-to-digital converter (ADC) converters has led to heightened recent interest in digitized radio over fiber links (DROF). In DROF, the I and Q baseband digital signals immediately after the digital signal processor are converted to optical and transported via the fiber. This means that the remote radio heads can be relatively simple too, consisting of DAC converters, up-converters, and amplifiers in the downlink direction and ADC converters, down-converters, and amplifiers in the uplink direction. Signal processing and modulation functions will take place in the central base station (CBS). Therefore, this architecture also satisfies the requirement that the RAP remains small and relatively simple. Such digital links are uses for current wireless systems (UMTS, WiMAX, and LTE) to connect digital base stations to remote radio heads. In order to use optical fiber to deliver radio signal to remote antennas, methods include the use of an intensity modulator to introduce an RF subcarrier onto the intensity of a CW laser source. This method cannot be extended to millimetre waves due to the limited bandwidth of available modulators. A novel transmitter architecture for the generation and distribution of GHz RF signals is described in this work. One of the principal objectives of this Master Thesis is to present the development of a digitized radio over fiber optical transmission systems under advanced modulation formats. We analyze the impact of chromatic dispersion and nonlinear microwave devices distortions considering one optical subcarrier carrying multiple RF signals

A mixed-signal ASIC for time and charge measurements with GEM detectors

L'abstract è presente nell'allegato / the abstract is in the attachmen

Adaptive compressive telemetry techniques Final report

Fidelity criteria for evaluating compression algorithms, and design data on adaptive data compression system for future manned spacecraf

Efficient algorithms for arbitrary sample rate conversion with application to wave field synthesis

Arbitrary sample rate conversion (ASRC) is used in many fields of digital signal processing to alter the sampling rate of discrete-time signals by arbitrary, potentially time-varying ratios. This thesis investigates efficient algorithms for ASRC and proposes several improvements. First, closed-form descriptions for the modified Farrow structure and Lagrange interpolators are derived that are directly applicable to algorithm design and analysis. Second, efficient implementation structures for ASRC algorithms are investigated. Third, this thesis considers coefficient design methods that are optimal for a selectable error norm and optional design constraints. Finally, the performance of different algorithms is compared for several performance metrics. This enables the selection of ASRC algorithms that meet the requirements of an application with minimal complexity. Wave field synthesis (WFS), a high-quality spatial sound reproduction technique, is the main application considered in this work. For WFS, sophisticated ASRC algorithms improve the quality of moving sound sources. However, the improvements proposed in this thesis are not limited to WFS, but applicable to general-purpose ASRC problems.Verfahren zur unbeschränkten Abtastratenwandlung (arbitrary sample rate conversion,ASRC) ermöglichen die Änderung der Abtastrate zeitdiskreter Signale um beliebige, zeitvarianteVerhältnisse. ASRC wird in vielen Anwendungen digitaler Signalverarbeitung eingesetzt.In dieser Arbeit wird die Verwendung von ASRC-Verfahren in der Wellenfeldsynthese(WFS), einem Verfahren zur hochqualitativen, räumlich korrekten Audio-Wiedergabe, untersucht.Durch ASRC-Algorithmen kann die Wiedergabequalität bewegter Schallquellenin WFS deutlich verbessert werden. Durch die hohe Zahl der in einem WFS-Wiedergabesystembenötigten simultanen ASRC-Operationen ist eine direkte Anwendung hochwertigerAlgorithmen jedoch meist nicht möglich.Zur Lösung dieses Problems werden verschiedene Beiträge vorgestellt. Die Komplexitätder WFS-Signalverarbeitung wird durch eine geeignete Partitionierung der ASRC-Algorithmensignifikant reduziert, welche eine effiziente Wiederverwendung von Zwischenergebnissenermöglicht. Dies erlaubt den Einsatz hochqualitativer Algorithmen zur Abtastratenwandlungmit einer Komplexität, die mit der Anwendung einfacher konventioneller ASRCAlgorithmenvergleichbar ist. Dieses Partitionierungsschema stellt jedoch auch zusätzlicheAnforderungen an ASRC-Algorithmen und erfordert Abwägungen zwischen Performance-Maßen wie der algorithmischen Komplexität, Speicherbedarf oder -bandbreite.Zur Verbesserung von Algorithmen und Implementierungsstrukturen für ASRC werdenverschiedene Maßnahmen vorgeschlagen. Zum Einen werden geschlossene, analytischeBeschreibungen für den kontinuierlichen Frequenzgang verschiedener Klassen von ASRCStruktureneingeführt. Insbesondere für Lagrange-Interpolatoren, die modifizierte Farrow-Struktur sowie Kombinationen aus Überabtastung und zeitkontinuierlichen Resampling-Funktionen werden kompakte Darstellungen hergeleitet, die sowohl Aufschluss über dasVerhalten dieser Filter geben als auch eine direkte Verwendung in Design-Methoden ermöglichen.Einen zweiten Schwerpunkt bildet das Koeffizientendesign für diese Strukturen, insbesonderezum optimalen Entwurf bezüglich einer gewählten Fehlernorm und optionaler Entwurfsbedingungenund -restriktionen. Im Gegensatz zu bisherigen Ansätzen werden solcheoptimalen Entwurfsmethoden auch für mehrstufige ASRC-Strukturen, welche ganzzahligeÜberabtastung mit zeitkontinuierlichen Resampling-Funktionen verbinden, vorgestellt.Für diese Klasse von Strukturen wird eine Reihe angepasster Resampling-Funktionen vorgeschlagen,welche in Verbindung mit den entwickelten optimalen Entwurfsmethoden signifikanteQualitätssteigerungen ermöglichen.Die Vielzahl von ASRC-Strukturen sowie deren Design-Parameter bildet eine Hauptschwierigkeitbei der Auswahl eines für eine gegebene Anwendung geeigneten Verfahrens.Evaluation und Performance-Vergleiche bilden daher einen dritten Schwerpunkt. Dazu wirdzum Einen der Einfluss verschiedener Entwurfsparameter auf die erzielbare Qualität vonASRC-Algorithmen untersucht. Zum Anderen wird der benötigte Aufwand bezüglich verschiedenerPerformance-Metriken in Abhängigkeit von Design-Qualität dargestellt.Auf diese Weise sind die Ergebnisse dieser Arbeit nicht auf WFS beschränkt, sondernsind in einer Vielzahl von Anwendungen unbeschränkter Abtastratenwandlung nutzbar