On prefilters for digital FIR filter design

A new family of digital prefilter structures is introduced, based on the Dolph-Chebyshev function. These prefilters can be combined with appropriately designed "equalizer" filters based on equiripple methods, leading to efficient FIR digital filter designs. Design examples are included, demonstrating the simplicity of the resulting designs, as compared to conventional equiripple designs

Computer-Aided Design of Switched-Capacitor Filters

This thesis describes a series of computer methods for the design of switched-capacitor filters. Current software is greatly restricted in the types of transfer function that can be designed and in the range of filter structures by which they can be implemented. To solve the former problem, several new filter approximation algorithms are derived from Newton's method, yielding the Remez algortithm as a special case (confirming its convergency properties). Amplitude responses with arbitrary passband shaping and stopband notch positions are computed. Points of a specified degree of tangency to attenuation boundaries (touch points) can be placed in the response, whereby a family of transfer functions between Butterworth and elliptic can be derived, offering a continuous trade-off in group delay and passive sensitivity properties. The approximation algorithms have also been applied to arbitrary group delay correction by all-pass functions. Touch points form a direct link to an iterative passive ladder design method, which bypasses the need for Hurwitz factorisation. The combination of iterative and classical synthesis methods is suggested as the best compromise between accuracy and speed. It is shown that passive ladder prototypes of a minimum-node form can be efficiently simulated by SC networks without additional op-amps. A special technique is introduced for canonic realisation of SC ladder networks from transfer functions with finite transmission at high frequency, solving instability and synthesis difficulties. SC ladder structures are further simplified by synthesising the zeros at +/-2fs which are introduced into the transfer function by bilinear transformation. They cause cancellation of feedthrough branches and yield simplified LDI-type SC filter structures, although based solely on the bilinear transform. Matrix methods are used to design the SC filter simulations. They are shown to be a very convenient and flexible vehicle for computer processing of the linear equations involved in analogue filter design. A wide variety of filter structures can be expressed in a unified form. Scaling and analysis can readily be performed on the system matrices with great efficiency. Finally, the techniques are assembled in a filter compiler for SC filters called PANDDA. The application of the above techniques to practical design problems is then examined. Exact correction of sinc(x), LDI termination error, pre-filter and local loop telephone line weightings are illustrated. An optimisation algorithm is described, which uses the arbitrary passband weighting to predistort the transfer function for response distortions. Compensation of finite amplifier gain-bandwidth and switch resistance effects in SC filters is demonstrated. Two commercial filter specifications which pose major difficulties for traditional design methods are chosen as examples to illustrate PANDDA's full capabilities. Significant reductions in order and total area are achieved. Finally, test results of several SC filters designed using PANDDA for a dual-channel speech-processing ASIC are presented. The speed with which high-quality, standard SC filters can be produced is thus proven

MICROWAVE FILTERS FOR NEXT GENERATION RADIO FREQUENCY TRANSCEIVERS

Increased data rates in wireless communications enforce unprecedented performance metrics on the front-end filters to operate in crowded spectral bands. These requirements include strong selectivity, low insertion loss, and good out-of-band (OOB) rejection in addition to the applicability in complementary metal oxide semiconductor (CMOS) integrated circuit layouts. The acoustic wave (AW) resonator based filter design technology has gained a very important role in the on-chip filter design techniques due to chip-scale physical resonator sizes and the ability of achieving high quality factor values at microwave frequencies. However, conventional synthesis methods used in the design of AW resonator based microwave filters suffer from limited achievable fractional bandwidth (FBW) and weak OOB rejection. The origin of these issues is the limitations on increasing the electromechanical coupling coefficient (kt2) of the resonators, which is an intrinsic property of the piezoelectric material in its design. This dissertation proposes a new class of hybrid acoustic-electromagnetic (Hybrid-ACEM) filters to overcome both of the aforementioned limitations of AW resonator-based filters. In other words, the main goal of this new topology is to maximize the ratio between the achievable FBW and the required kt2. This is achieved by employing one or two electromagnetic (EM) resonators that are placed at purposefully selected stages within the design. In addition, cross-coupling mechanisms are systematically used to reduce the required electromechanical coupling coefficient in certain filter orders. Altogether, the proposed method can achieve much larger FBW values and stronger OOB rejection compared to the conventionally synthesized ladder acoustic wave filters. The effect of finite quality factor of the EM resonators is analyzed. A new algorithm to convert extracted-pole sections to Butterworth-Van-Dyke (BVD) model for large FBW values is also presented. It has been shown in the simulations that FBW-to-kt2 ratios of four or above is achievable with this method. As a proof-of-concept, a sixth-order hybrid canonical prototype with a center frequency of 2.67 GHz and 11.2% FBW is designed and fabricated. The acoustic wave resonators used in the fabrication have kt2 values of 3.5%. The fabricated prototype proves the validity of the proposed method for achieving FBW values of 30% with required kt2 values of 7.5%, which is available with the common aluminum nitride (AlN) based bulk acoustic wave resonator technologies of today. The developed technique opens a new pathway to reduce the limitations of integrating microwave filters for future fully on-chip microwave transceivers

Comparison of Wideband Earpiece Integrations in Mobile Phone

Perinteisesti puhelinverkoissa välitettävä puhe on ollut kapeakaistaista, kaistan ollessa 300 - 3400 Hz. Voidaan kuitenkin olettaa, että laajakaistaiset puhepalvelut tulevat saamaan markkinoilla enemmän jalansijaa tulevina vuosina. Tässä lopputyössä esitellään puheenkoodauksen perusteet laajakaistaisen adaptiivisen moninopeuspuhekoodekin (AMR-WB) kanssa. Laajakaistainen puhekoodekki laajentaa puhekaistan 50-7000 Hz käyttäen 16 kHz näytetaajuutta. Käytännössä laajempi kaista tarkoittaa parannuksia puheen ymmärrettävyyteen ja tekee siitä luonnollisemman ja mukavamman kuuloista. Tämän lopputyön päätavoite on vertailla kahden eri laajakaistaisen matkapuhelinkuulokkeen integrointia. Kysymys kuuluu, kuinka paljon käyttäjä hyötyy isommasta kuulokkeesta matkapuhelimessa? Kuulokkeiden suorituskyvyn selvittämiseksi niille tehtiin objektiivisia mittauksia vapaakentässä. Mittauksia tehtiin myös puhelimelle pää- ja torsosimulaattorissa (HATS) johdottamalla kuuloke suoraan vahvistimelle, sekä lisäksi puhelun ollessa aktiivisena GSM ja WCDMA verkoissa. Objektiiviset mittaukset osoittivat kahden eri integroinnin väliset erot kuulokkeiden taajuusvasteessa ja särössä erityisesti matalilla taajuuksilla. Lopuksi tehtiin kuuntelukoe tarkoituksena selvittää erottaako loppukäyttäjä pienemmän ja isomman kuulokkeen välistä eroa käyttäen kapeakaistaisia ja laajakaistaisia puhelinääninäytteitä. Kuuntelukokeen tuloksien pohjalta voidaan sanoa, että käyttäjä erottaa kahden eri integroinnin erot ja miespuhuja hyötyy naispuhujaa enemmän isommasta kuulokkeesta laajakaistaisella puhekoodekilla.The speech in telecommunication networks has been traditionally narrowband ranging from 300 Hz to 3400 Hz. It can be expected that wideband speech call services will increase their foothold in the markets during the coming years. In this thesis speech coding basics with adaptive multirate wideband (AMR-WB) are introduced. The wideband codec widens the speech band to new range from 50 Hz to 7000 Hz using 16 kHz sampling frequency. In practice the wider band means improvements to speech intelligibility and makes it more natural and comfortable to listen to. The main focus of this thesis work is to compare two different wideband earpiece integrations. The question is how much the end-user will benefit from using a larger earpiece in a mobile phone? To find out speaker performance, objective measurements in free field were done for the earpiece modules. Measurements were performed also for the phone on head and torso simulator (HATS) by wiring the earpieces directly to a power amplifier and with over the air on GSM and WCDMA networks. The results of objective measurements showed differences between the earpiece integrations especially on low frequencies in frequency response and distortion. Finally the subjective listening test is done for comparison to see if the end-user notices the difference between smaller and larger earpiece integrations using narrowband and wideband speech samples. Based on these subjective test results it can be said that the user can differentiate between two different integrations and that a male speaker benefits more from a larger earpiece than a female speaker

Synthesis of Filters for Digital Wireless Communications

Firstly, a new synthesis method for the generation of the generalized Chebyshev characteristic polynomials has been presented. The general characteristic function is generated by a linear combination of Chebyshev basis characteristic functions. The basis functions for different filtering functions may easily be determined based on the number and position of reflection and transmission zeros. These basis functions enable direct synthesis of both lumped and distributed filter networks. Different filter functions including but not limited to low-pass, bandpass and dual bandpass filters, have been synthesised to demonstrate the general application of the synthesis method. Secondly, a new method for the design of a new class of distributed low-pass filter has been presented that enables exact realisation of the series short circuited transmission lines which are normally approximated via unit elements in other filter realisations. The filters are based on parallel coupled high impedance transmission lines which are terminated at one end in open-circuited stubs. The approach enables realisation of both finite and quarter-wave frequency transmission zeros hence giving improved stopband performance. A complete design is presented and the fabricated low-pass filter demonstrates excellent performance in good agreement with theory. Finally, design techniques for microwave bandpass filters using re-entrant resonators are presented. The key feature is that each re-entrant resonator in the filter generates a passband resonance and a finite frequency transmission zero, above the passband. Thus an Nth degree filter can have N finite frequency transmission zeros with a simple physical realization. A new synthesis technique for pseudo-elliptic low-pass filters suitable for designing re-entrant bandpass filter has also been show-cased. A physically symmetrical 5 pole re-entrant bandpass prototype filter with 5 transmission zeros above the passband was designed and fabricated. Measured results showed good correspondence with theories

Digital signal processing and digital-to-analog converters for wide-band transmitters

In this thesis, the implementation methods of digital signal processing and digital-to-analog converters for wide-band transmitters are researched. With digital signal processing, the problems of analog signal processing, such as sensitivity to interference and nonidealities of the semiconductor processes, can be avoided. Also, the programmability can be implemented digitally more easily than by means of analog signal processing. During the past few years, wireless communications has evolved from analog to digital, and signal bandwidths have increased, enabling faster and faster data transmission. The evolution of semiconductor processes, decreasing linewidth and supply voltages, has decreased the size of the electronics and power dissipation, enabling the integration of larger and larger systems on single silicon chips. There is little overall benefit in decreasing linewidths to meet the needs of analog design, since it makes the design process more difficult as the device sizes cannot be scaled according to minimum linewidth and because of the decreasing supply voltage. On the other hand, the challenges of digital signal processing are related to the efficient realization of signal processing algorithms in such a way that the required area and power dissipation does not increase extensively. In this book, the problems related to digital filters, upconversion algorithms and digital-to-analog converters used in digital transmitters are researched. Research results are applied to the implementation of a transmitter for a third-generation WCDMA base-station. In addition, the theory of factors affecting the linearity and performance of digital-to-analog converters is researched, and a digital calibration algorithm for enhancement of the static linearity has been presented. The algorithm has been implemented together with a 16-bit converter; its functionality has been demonstrated with measurements.Tässä väitöskirjassa on tutkittu digitaalisen signaalinkäsittelyn toteuttamista ja digitaalisesta analogiseksi -muuntimia laajakaistaisiin lähettimiin. Digitaalisella signaalinkäsittelyllä voidaan välttää monia analogiseen signaalinkäsittelyyn liittyviä ongelmia, kuten häiriöherkkyyttä ja puolijohdeprosessien epäideaalisuuksien vaikutuksia. Myös ohjelmoitavuus on helpommin toteutettavissa digitaalisesti kuin analogisen signaalinkäsittelyn keinoin. Viime vuosina on langattomien tietoliikennejärjestelmien kehitys kulkenut analogisesta digitaaliseen, ja käytettävät signaalikaistanleveydet ovat kasvaneet mahdollistaen yhä nopeamman tiedonsiirron. Puolijohdeprosessien kehitys, kapeneva minimiviivanleveys ja pienemmät käyttöjännitteet, on pienentänyt elektroniikan kokoa ja tehonkulutusta mahdollistaen yhä suurempien kokonaisuuksien integroimisen yhdelle piisirulle. Viivanleveyksien pieneneminen ei kuitenkaan suoraan hyödytä analogiasuunnittelua, jossa piirielementtien kokoa ei välttämättä voida pienentää viivanleveyden pienentyessä, ja jossa madaltuva käyttöjännite ennemminkin hankaloittaa kuin helpottaa suunnittelua. Siksi yhä suurempi osa signaalinkäsittelystä pyritään tekemään digitaalisesti. Digitaalisen signaalinkäsittelyn ongelmat puolestaan liittyvät algoritmien tehokkaaseen toteuttamiseen siten, että piirien pinta-ala ja tehonkulutus eivät kasva liian suuriksi. Tässä kirjassa on tutkittu digitaalisessa lähettimessä tarvittavien digitaalisten suodattimien, ylössekoitusalgoritmien ja digitaalisesta analogiseksi -muuntimien toteuttamiseen liittyviä ongelmia. Tutkimustuloksia on sovellettu kolmannen sukupolven WCDMA-tukiasemalähettimen toteutuksessa. Lisäksi on tutkittu digitaalisesta analogiseksi -muuntimien lineaarisuuteen ja suorituskykyyn vaikuttavien seikkojen teoriaa, ja esitetty digitaalinen kalibrointialgoritmi muuntimen staattisen suorituskyvyn parantamiseksi. Algoritmi on toteutettu 16-bittisen muuntimen yhteydessä ja se on osoitettu toimivaksi mittauksin.reviewe

Band-pass waveguide filters and multiplexers design by the structure segmentation technique

This work presents a technique for the efficient design of bandpass waveguide microwave filters using a segmentation technique. The segmentation technique was originally developed for in-line filters, and the main contribution of this work is in the combination of this technique with the coupling matrix formalism. In this way we have used this useful design technique for complex coupling topologies, beyond the in-line configuration. As an example some dual mode filters are designed using the new coupling matrix formalism, validating the theory presented. In addition, this technique has been used in the design of diplexers and triplexers with the previously designed filters. Furthermore, a novel dual-mode filter topology that can be implemented through this technique is proposed. Finally, in order to validate these contributions, a dual-mode filter has been designed with the introduced topology for a real application in the space sector in collaboration with the European Space Agency.Universidad Politécnica de Cartagen

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