Practical Non-Uniform Channelization for Multistandard Base Stations

A Multistandard software-defined radio base station must perform non-uniform channelization of multiplexed frequency bands. Non-uniform channelization accounts for a significant portion of the digital signal processing workload in the base station receiver and can be difficult to realize in a physical implementation. In non-uniform channelization methods based on generalized DFT filter banks, large prototype filter orders are a significant issue for implementation. In this paper, a multistage filter design is applied to two different non-uniform generalized DFT-based channelizers in order to reduce their filter orders. To evaluate the approach, a TETRA and TEDS base station is used. Experimental results show that the new multistage design reduces both the number of coefficients and operations and leads to a more feasible design and practical physical implementation

Practical Non-Uniform Channelization for Multistandard Base Stations

A Multistandard software-defined radio base station must perform non-uniform channelization of multiplexed frequency bands. Non-uniform channelization accounts for a significant portion of the digital signal processing workload in the base station receiver and can be difficult to realize in a physical implementation. In non-uniform channelization methods based on generalized DFT filter banks, large prototype filter orders are a significant issue for implementation. In this paper, a multistage filter design is applied to two different non-uniform generalized DFT-based channelizers in order to reduce their filter orders. To evaluate the approach, a TETRA and TEDS base station is used. Experimental results show that the new multistage design reduces both the number of coefficients and operations and leads to a more feasible design and practical physical implementation

Channelization for Multi-Standard Software-Defined Radio Base Stations

As the number of radio standards increase and spectrum resources come under more pressure, it becomes ever less efficient to reserve bands of spectrum for exclusive use by a single radio standard. Therefore, this work focuses on channelization structures compatible with spectrum sharing among multiple wireless standards and dynamic spectrum allocation in particular. A channelizer extracts independent communication channels from a wideband signal, and is one of the most computationally expensive components in a communications receiver. This work specifically focuses on non-uniform channelizers suitable for multi-standard Software-Defined Radio (SDR) base stations in general and public mobile radio base stations in particular. A comprehensive evaluation of non-uniform channelizers (existing and developed during the course of this work) shows that parallel and recombined variants of the Generalised Discrete Fourier Transform Modulated Filter Bank (GDFT-FB) represent the best trade-off between computational load and flexibility for dynamic spectrum allocation. Nevertheless, for base station applications (with many channels) very high filter orders may be required, making the channelizers difficult to physically implement. To mitigate this problem, multi-stage filtering techniques are applied to the GDFT-FB. It is shown that these multi-stage designs can significantly reduce the filter orders and number of operations required by the GDFT-FB. An alternative approach, applying frequency response masking techniques to the GDFT-FB prototype filter design, leads to even bigger reductions in the number of coefficients, but computational load is only reduced for oversampled configurations and then not as much as for the multi-stage designs. Both techniques render the implementation of GDFT-FB based non-uniform channelizers more practical. Finally, channelization solutions for some real-world spectrum sharing use cases are developed before some final physical implementation issues are considered

Dirty RF Signal Processing for Mitigation of Receiver Front-end Non-linearity

Moderne drahtlose Kommunikationssysteme stellen hohe und teilweise gegensätzliche Anforderungen an die Hardware der Funkmodule, wie z.B. niedriger Energieverbrauch, große Bandbreite und hohe Linearität. Die Gewährleistung einer ausreichenden Linearität ist, neben anderen analogen Parametern, eine Herausforderung im praktischen Design der Funkmodule. Der Fokus der Dissertation liegt auf breitbandigen HF-Frontends für Software-konfigurierbare Funkmodule, die seit einigen Jahren kommerziell verfügbar sind. Die praktischen Herausforderungen und Grenzen solcher flexiblen Funkmodule offenbaren sich vor allem im realen Experiment. Eines der Hauptprobleme ist die Sicherstellung einer ausreichenden analogen Performanz über einen weiten Frequenzbereich. Aus einer Vielzahl an analogen Störeffekten behandelt die Arbeit die Analyse und Minderung von Nichtlinearitäten in Empfängern mit direkt-umsetzender Architektur. Im Vordergrund stehen dabei Signalverarbeitungsstrategien zur Minderung nichtlinear verursachter Interferenz - ein Algorithmus, der besser unter "Dirty RF"-Techniken bekannt ist. Ein digitales Verfahren nach der Vorwärtskopplung wird durch intensive Simulationen, Messungen und Implementierung in realer Hardware verifiziert. Um die Lücken zwischen Theorie und praktischer Anwendbarkeit zu schließen und das Verfahren in reale Funkmodule zu integrieren, werden verschiedene Untersuchungen durchgeführt. Hierzu wird ein erweitertes Verhaltensmodell entwickelt, das die Struktur direkt-umsetzender Empfänger am besten nachbildet und damit alle Verzerrungen im HF- und Basisband erfasst. Darüber hinaus wird die Leistungsfähigkeit des Algorithmus unter realen Funkkanal-Bedingungen untersucht. Zusätzlich folgt die Vorstellung einer ressourceneffizienten Echtzeit-Implementierung des Verfahrens auf einem FPGA. Abschließend diskutiert die Arbeit verschiedene Anwendungsfelder, darunter spektrales Sensing, robuster GSM-Empfang und GSM-basiertes Passivradar. Es wird gezeigt, dass nichtlineare Verzerrungen erfolgreich in der digitalen Domäne gemindert werden können, wodurch die Bitfehlerrate gestörter modulierter Signale sinkt und der Anteil nichtlinear verursachter Interferenz minimiert wird. Schließlich kann durch das Verfahren die effektive Linearität des HF-Frontends stark erhöht werden. Damit wird der zuverlässige Betrieb eines einfachen Funkmoduls unter dem Einfluss der Empfängernichtlinearität möglich. Aufgrund des flexiblen Designs ist der Algorithmus für breitbandige Empfänger universal einsetzbar und ist nicht auf Software-konfigurierbare Funkmodule beschränkt.Today's wireless communication systems place high requirements on the radio's hardware that are largely mutually exclusive, such as low power consumption, wide bandwidth, and high linearity. Achieving a sufficient linearity, among other analogue characteristics, is a challenging issue in practical transceiver design. The focus of this thesis is on wideband receiver RF front-ends for software defined radio technology, which became commercially available in the recent years. Practical challenges and limitations are being revealed in real-world experiments with these radios. One of the main problems is to ensure a sufficient RF performance of the front-end over a wide bandwidth. The thesis covers the analysis and mitigation of receiver non-linearity of typical direct-conversion receiver architectures, among other RF impairments. The main focus is on DSP-based algorithms for mitigating non-linearly induced interference, an approach also known as "Dirty RF" signal processing techniques. The conceived digital feedforward mitigation algorithm is verified through extensive simulations, RF measurements, and implementation in real hardware. Various studies are carried out that bridge the gap between theory and practical applicability of this approach, especially with the aim of integrating that technique into real devices. To this end, an advanced baseband behavioural model is developed that matches to direct-conversion receiver architectures as close as possible, and thus considers all generated distortions at RF and baseband. In addition, the algorithm's performance is verified under challenging fading conditions. Moreover, the thesis presents a resource-efficient real-time implementation of the proposed solution on an FPGA. Finally, different use cases are covered in the thesis that includes spectrum monitoring or sensing, GSM downlink reception, and GSM-based passive radar. It is shown that non-linear distortions can be successfully mitigated at system level in the digital domain, thereby decreasing the bit error rate of distorted modulated signals and reducing the amount of non-linearly induced interference. Finally, the effective linearity of the front-end is increased substantially. Thus, the proper operation of a low-cost radio under presence of receiver non-linearity is possible. Due to the flexible design, the algorithm is generally applicable for wideband receivers and is not restricted to software defined radios

Time-Interleaved Analog-to-Digital-Converters: Modeling, Blind Identification and Digital Correction of Frequency Response Mismatches

Analog-to-digital-conversion enables utilization of digital signal processing (DSP) in many applications today such as wireless communication, radar and electronic warfare. DSP is the favored choice for processing information over analog signal processing (ASP) because it can typically offer more flexibility, computational power, reproducibility, speed and accuracy when processing and extracting information. Software defined radio (SDR) receiver is one clear example of this, where radio frequency waveforms are converted into digital form as close to the antenna as possible and all the processing of the information contained in the received signal is extracted in a configurable manner using DSP. In order to achieve such goals, the information collected from the real world signals, which are commonly analog in their nature, must be converted into digital form before it can be processed using DSP in the respective systems. The common trend in these systems is to not only process ever larger bandwidths of data but also to process data in digital format at ever higher processing speeds with sufficient conversion accuracy. So the analog-to-digital-converter (ADC), which converts real world analog waveforms into digital form, is one of the most important cornerstones in these systems.The ADC must perform data conversion at higher and higher rates and digitize ever-increasing bandwidths of data. In accordance with the Nyquist-Shannon theorem, the conversion rate of the ADC must be suffcient to accomodate the BW of the signal to be digitized, in order to avoid aliasing. The conversion rate of the ADC can in general be increased by using parallel ADCs with each ADC performing the sampling at mutually different points in time. Interleaving the outputs of each of the individual ADCs provides then a higher digitization output rate. Such ADCs are referred to as TI-ADC. However, the mismatches between the ADCs cause unwanted spurious artifacts in the TI-ADC’s spectrum, ultimately leading to a loss in accuracy in the TI-ADC compared to the individual ADCs. Therefore, the removal or correction of these unwanted spurious artifacts is essential in having a high performance TI-ADC system.In order to remove the unwanted interleaving artifacts, a model that describes the behavior of the spurious distortion products is of the utmost importance as it can then facilitate the development of efficient digital post-processing schemes. One major contribution of this thesis consists of the novel and comprehensive modeling of the spurious interleaving mismatches in different TI-ADC scenarios. This novel and comprehensive modeling is then utilized in developing digital estimation and correction methods to remove the mismatch induced spurious artifacts in the TI-ADC’s spectrum and recovering its lost accuracy. Novel and first of its kind digital estimation and correction methods are developed and tested to suppress the frequency dependent mismatch spurs found in the TI-ADCs. The developed methods, in terms of the estimation of the unknown mismatches, build on statistical I/Q signal processing principles, applicable without specifically tailored calibration signals or waveforms. Techniques to increase the analog BW of the ADC are also analyzed and novel solutions are presented. The interesting combination of utilizing I/Q downconversion in conjunction with TI-ADC is examined, which not only extends the TI-ADC’s analog BW but also provides flexibility in accessing the radio spectrum. Unwanted spurious components created during the ADC’s bandwidth extension process are also analyzed and digital correction methods are developed to remove these spurs from the spectrum. The developed correction techniques for the removal of the undesired interleaving mismatch artifacts are validated and tested using various HW platforms, with up to 1 GHz instantaneous bandwidth. Comprehensive test scenarios are created using measurement data obtained from HW platforms, which are used to test and evaluate the performance of the developed interleaving mismatch estimation and correction schemes, evidencing excellent performance in all studied scenarios. The findings and results presented in this thesis contribute towards increasing the analog BW and conversion rate of ADC systems without losing conversion accuracy. Overall, these developments pave the way towards fulfilling the ever growing demands on the ADCs in terms of higher conversion BW, accuracy and speed

Circuit Techniques for Multiple and Wideband Beamforming

University of Minnesota Ph.D. dissertation.June 2018. Major: Electrical Engineering. Advisor: Ramesh Harjani. 1 computer file (PDF); x, 102 pages.This thesis presents different architectures with regard to multiple beamforming and wideband phased array transceiver. Three different designs are implemented in TSMC 65nm RF CMOS to demonstrate different solutions. The design in this thesis have included major RF blocks in state-of-art wireless transceiver: RF receiver, local oscillator, and RF transmitter. First, a RF/analog FFT based four-channel four-beam receiver with progressive partial spatial ltering is proposed. This architecture is particularly well suited for MIMO systems where multiple beams are used to increase throughput. Like the FFT, the proposed architecture reuses computations for multi-beam systems. In particular, the proposed architecture redistributes the computations so as to maximize the reuse of the structure that already exist in a receiver chain. In many fashions the architecture is quite similar to a Butler matrix but unlike the Butler matrix it does not use large passive components at RF. Further, we exploit the normally occurring quadrature down-conversion process to implement the tap weights. In comparison to traditional MIMO architectures, that effectively duplicate each path, the distributed computations of this architecture provide partial spatial ltering before the final stage, improving interference rejection for the blocks between the LNA and the ADC. Additionally, because of the spatial ltering prior to the ADC, a single interferer only jams a single beam allowing for continued operation though at a lower combined throughput. The four-beam receiver core prototype in 65nm CMOS implements the basic FFT based architecture but does not include an LNA or extensive IF stages. This four-channel design consumes 56mW power and occupies an active area of 0:65mm2 excluding pads and test circuits. Second, a wideband phased array receiver architecture with simultaneous spectral and spatial filtering by sub-harmonic injection oscillators is presented. The design avoids using expensive delay elements by many conventional wideband phased array. Different from prior art of channelization which cannot solve beam-squinting issue among the sub-channels, we use sub-harmonic injection locking scheme, which make the center frequencies of all sub-channels point to the same spatial direction to overcome beam-squinting issue. The low frequency, low power and narrowband phase shifters are placed at LO in comparison to conventional way of placing delay elements or phase shifters in the signal path. This avoids receiver performance degradation from delay elements or phase shifters. The simultaneous spectral and spatial ltering dictates less ADC dynamic range requirement and further reduces power. The injection locking scheme reduces the phase noise contribution from the oscillators. The two-band prototype design realized in 65nm GP CMOS is centered at 9GHz, provides 4GHz instantaneous bandwidth, reduces beam-squinting by half, consumes 31.75mW/antenna and occupies 2.7mm2 of chip area. In the third work, a steerable RF/analog FFT based four-beam transmitter architecture is presented. This work is based on the idea of FFT based multiple beamforming in 1st work, but extended to the transmitter and make the all beams steerable. Due to the reciprocity between receiver and transmitter, decimation-in-frequency (DIF) FFT is utilized in the transmitter. All the beams are steered simultaneously by front-end phase shifters, while keep each of the beams is independent of the others. The steerability of FFT based multiple beamforming scheme makes this proposed prototype could tackle more complicated portable wireless environment. The first and second proposed architecture have been silicon veried, and the design of the third has been finished and ready for tapeout

Epälineaarisen signaaliriippuvan akustisen keilanmuodostajan reaaliaikaimplementaatio

A real-time acoustical beamforming system incorporating the cross pattern coherence (CroPaC) post filtering method is implemented in this thesis. The real-time implementation consists of a signal-independent beamformer that is used for spatial discrimination of a sound field. The signal of the beamformer is post filtered by modulating it with a parameter that is derived from the cross-spectrum of two directional microphone signals. The post filter is implemented to enhance performance of beamforming (increase in signal-to-noise ratio), because beamformers are not efficient in environments with high level of reverberation. The post filtering method has been previously implemented in MATLAB for non-real-time use, and this system is the first real-time implementation of an acoustical beamforming system utilizing it. The implementation is programmed in the programming language C for the graphical signal processing program Max developed by Cycling '74. It utilizes a time-frequency domain processing, and the spherical Fourier transform for a decomposition of a sound field into spherical harmonic signals. The implementation can be used with microphone arrays with maximum of 32 microphone capsules, which are laid over rigid sphere with uniform or nearly-uniform arrangements. The real-time implementation can be utilized in many applications, which require algorithm to work in real-time, such as teleconferencing and acoustical cameras.Tässä diplomityössä implementoidaan reaaliaikainen akustinen keilanmuodostusjärjestelmä signaalien väliseen koherenssiin perustuvalla (CroPaC) jälkisuodatuksella. Reaaliaikaimplementaatio koostuu signaaliriippumattomasta keilanmuodostajasta, jota käytetään äänikentän spatiaaliseen suodatukseen. Keilanmuodostajan signaalia jälkisuodatetaan moduloimalla sitä parametrilla, joka johdetaan kahden suuntamikrofonin signaalin välisestä koherenssista. Jälkisuodatus implementoidaan keilanmuodostajan suorituskyvyn parantamiseksi (signaali-kohina-suhteen kasvu), sillä keilanmuodostajat eivät ole tehokkaita kaiuntaisissa ympäristöissä. Jälkisuodatusmetodi on aikaisemmin implementoitu MATLABissa ei-reaaliaikakäyttöä varten. Tämän työn implementaatio on ensimmäinen reaaliaikainen akustinen keilanmuodostusjärjestelmä, joka hyödyntää CroPaC-jälkisuodatusta. Implementaatio on ohjelmoitu C-ohjelmointikielellä graafiselle signaalinprosessointityökalulle Max, jonka on kehittänyt Cycling '74. Prosessointi tapahtuu aika-taajuustasossa ja siinä hyödynnetään äänikentän dekompositiota palloharmonisiin signaaleihin. Implementaatiota voidaan käyttää mikrofoniryhmällä, jossa on korkeintaan 32 mikrofonikapselia, jotka on asetettu jäykän pallon päälle tasavälein tai lähes tasavälein. Reaaliaikaimplementaatiota voidaan hyödyntää lukuisissa sovelluksissa, jotka edellyttävät algoritmin reaaliaikaista toimintaa, esimerkiksi puhelinkokouksissa ja akustisissa kameroissa

Nonlinear models and algorithms for RF systems digital calibration

Focusing on the receiving side of a communication system, the current trend in pushing the digital domain ever more closer to the antenna sets heavy constraints on the accuracy and linearity of the analog front-end and the conversion devices. Moreover, mixed-signal implementations of Systems-on-Chip using nanoscale CMOS processes result in an overall poorer analog performance and a reduced yield. To cope with the impairments of the low performance analog section in this "dirty RF" scenario, two solutions exist: designing more complex analog processing architectures or to identify the errors and correct them in the digital domain using DSP algorithms. In the latter, constraints in the analog circuits' precision can be offloaded to a digital signal processor. This thesis aims at the development of a methodology for the analysis, the modeling and the compensation of the analog impairments arising in different stages of a receiving chain using digital calibration techniques. Both single and multiple channel architectures are addressed exploiting the capability of the calibration algorithm to homogenize all the channels' responses of a multi-channel system in addition to the compensation of nonlinearities in each response. The systems targeted for the application of digital post compensation are a pipeline ADC, a digital-IF sub-sampling receiver and a 4-channel TI-ADC. The research focuses on post distortion methods using nonlinear dynamic models to approximate the post-inverse of the nonlinear system and to correct the distortions arising from static and dynamic errors. Volterra model is used due to its general approximation capabilities for the compensation of nonlinear systems with memory. Digital calibration is applied to a Sample and Hold and to a pipeline ADC simulated in the 45nm process, demonstrating high linearity improvement even with incomplete settling errors enabling the use of faster clock speeds. An extended model based on the baseband Volterra series is proposed and applied to the compensation of a digital-IF sub-sampling receiver. This architecture envisages frequency selectivity carried out at IF by an active band-pass CMOS filter causing in-band and out-of-band nonlinear distortions. The improved performance of the proposed model is demonstrated with circuital simulations of a 10th-order band pass filter, realized using a five-stage Gm-C Biquad cascade, and validated using out-of-sample sinusoidal and QAM signals. The same technique is extended to an array receiver with mismatched channels' responses showing that digital calibration can compensate the loss of directivity and enhance the overall system SFDR. An iterative backward pruning is applied to the Volterra models showing that complexity can be reduced without impacting linearity, obtaining state-of-the-art accuracy/complexity performance. Calibration of Time-Interleaved ADCs, widely used in RF-to-digital wideband receivers, is carried out developing ad hoc models because the steep discontinuities generated by the imperfect canceling of aliasing would require a huge number of terms in a polynomial approximation. A closed-form solution is derived for a 4-channel TI-ADC affected by gain errors and timing skews solving the perfect reconstruction equations. A background calibration technique is presented based on cyclo-stationary filter banks architecture. Convergence speed and accuracy of the recursive algorithm are discussed and complexity reduction techniques are applied

A New Wideband, Fully Steerable, Decametric Array at Clark Lake

A fully steerable, decametric array for radio astronomy is under construction at the Clark Lake Radio Observatory near Borrego Springs, California. This array will be a T of 720 conical spiral antennas (teepee-shaped antennas, hence the array is called the TPT), 3.0 km by 1.8 km capable of operating between 15 and 125 MHz. Both its operating frequency and beam position will be adjustable in less than one millisecond, and the TPT will provide a 49-element picture around the central beam position for extended source observations. Considerable experience was gained in the operation of completed portions of the array, and successful operation of the final array is assured. The results are described of the tests which were conducted with the conical spirals, and the planned electronics and data processing systems are described