Digital filter design using root moments for sum-of-all-pass structures from complete and partial specifications

Published versio

DSP compensation for distortion in RF filters

There is a growing demand for the high quality TV programs such as High Definition TV (HDTV). The CATV network is often a suitable solution to address this demand using a CATV modem delivering high data rate digital signals in a cost effective manner, thereby, utilizing a complex digital modulation scheme is inevitable. Exploiting complex modulation schemes, entails a more sophisticated modulator and distribution system with much tighter tolerances. However, there are always distortions introduced to the modulated signal in the modulator degrading signal quality. In this research, the effect of distortions introduced by the RF band pass filter in the modulator will be considered which cause degradations on the quality of the output Quadrature Amplitude Modulated (QAM) signal. Since the RF filter's amplitude/group delay distortions are not symmetrical in the frequency domain, once translated into the base band they have a complex effect on the QAM signal. Using Matlab, the degradation effects of these distortions on the QAM signal such as Bit Error Rate (BER) is investigated. In order to compensate for the effects of the RF filter distortions, two different methods are proposed. In the first method, a complex base band compensation filter is placed after the pulse shaping filter (SRRC). The coefficients of this complex filter are determined using an optimization algorithm developed during this research. The second approach, uses a pre-equalizer in the form of a Feed Forward FIR structure placed before the pulse shaping filter (SRRC). The coefficients of this pre-equalizer are determined using the equalization algorithm employed in a test receiver, with its tap weights generating the inverse response of the RF filter. The compensation of RF filter distortions in base band, in turn, improves the QAM signal parameters such as Modulation Error Ratio (MER). Finally, the MER of the modulated QAM signal before and after the base band compensation is compared between the two methods, showing a significant enhancement in the RF modulator performance

A Variational Approach for Designing Infinite Impulse Response Filters With Time-Varying Parameters

Filter design with short transient state is a problem encountered in many fields of circuits, systems, and signal processing. In this paper, a novel low-pass filter design technique with time-varying parameters is introduced in order to minimize the rise-time parameter. Through the use of calculus of variations a method is developed to obtain the optimal closed-form expression for adjusting the parameters. In this context, two cases are addressed: the ideal case in which infinite bandwidth is required and a solution of finite bandwidth. The latter is obtained by means of a proper constraint formulation in the frequency domain. The proposed filter achieves the shortest rise time and allows better preservation of the edge shape in comparison with other existing filtering methods. The analysis, synthesis, and performance of the proposed system are discussed and illustrated with the aid of simulations

Efficient Algorithms for Immersive Audio Rendering Enhancement

Il rendering audio immersivo è il processo di creazione di un’esperienza sonora coinvolgente e realistica nello spazio 3D. Nei sistemi audio immersivi, le funzioni di trasferimento relative alla testa (head-related transfer functions, HRTFs) vengono utilizzate per la sintesi binaurale in cuffia poiché esprimono il modo in cui gli esseri umani localizzano una sorgente sonora. Possono essere introdotti algoritmi di interpolazione delle HRTF per ridurre il numero di punti di misura e per creare un movimento del suono affidabile. La riproduzione binaurale può essere eseguita anche dagli altoparlanti. Tuttavia, il coinvolgimento di due o più gli altoparlanti causa il problema del crosstalk. In questo caso, algoritmi di cancellazione del crosstalk (CTC) sono necessari per eliminare i segnali di interferenza indesiderati. In questa tesi, partendo da un'analisi comparativa di metodi di misura delle HRTF, viene proposto un sistema di rendering binaurale basato sull'interpolazione delle HRTF per applicazioni in tempo reale. Il metodo proposto mostra buone prestazioni rispetto a una tecnica di riferimento. L'algoritmo di interpolazione è anche applicato al rendering audio immersivo tramite altoparlanti, aggiungendo un algoritmo di cancellazione del crosstalk fisso, che considera l'ascoltatore in una posizione fissa. Inoltre, un sistema di cancellazione crosstalk adattivo, che include il tracciamento della testa dell'ascoltatore, è analizzato e implementato in tempo reale. Il CTC adattivo implementa una struttura in sottobande e risultati sperimentali dimostrano che un maggiore numero di bande migliora le prestazioni in termini di errore totale e tasso di convergenza. Il sistema di riproduzione e le caratteristiche dell'ambiente di ascolto possono influenzare le prestazioni a causa della loro risposta in frequenza non ideale. L'equalizzazione viene utilizzata per livellare le varie parti dello spettro di frequenze che compongono un segnale audio al fine di ottenere le caratteristiche sonore desiderate. L'equalizzazione può essere manuale, come nel caso dell'equalizzazione grafica, dove il guadagno di ogni banda di frequenza può essere modificato dall'utente, o automatica, la curva di equalizzazione è calcolata automaticamente dopo la misurazione della risposta impulsiva della stanza. L'equalizzazione della risposta ambientale può essere applicata anche ai sistemi multicanale, che utilizzano due o più altoparlanti e la zona di equalizzazione può essere ampliata misurando le risposte impulsive in diversi punti della zona di ascolto. In questa tesi, GEQ efficienti e un sistema adattativo di equalizzazione d'ambiente. In particolare, sono proposti e approfonditi tre equalizzatori grafici a basso costo computazionale e a fase lineare e quasi lineare. Gli esperimenti confermano l'efficacia degli equalizzatori proposti in termini di accuratezza, complessità computazionale e latenza. Successivamente, una struttura adattativa in sottobande è introdotta per lo sviluppo di un sistema di equalizzazione d'ambiente multicanale. I risultati sperimentali verificano l'efficienza dell'approccio in sottobande rispetto al caso a banda singola. Infine, viene presentata una rete crossover a fase lineare per sistemi multicanale, mostrando ottimi risultati in termini di risposta in ampiezza, bande di transizione, risposta polare e risposta in fase. I sistemi di controllo attivo del rumore (ANC) possono essere progettati per ridurre gli effetti dell'inquinamento acustico e possono essere utilizzati contemporaneamente a un sistema audio immersivo. L'ANC funziona creando un'onda sonora in opposizione di fase rispetto all'onda sonora in arrivo. Il livello sonoro complessivo viene così ridotto grazie all'interferenza distruttiva. Infine, questa tesi presenta un sistema ANC utilizzato per la riduzione del rumore. L’approccio proposto implementa una stima online del percorso secondario e si basa su filtri adattativi in sottobande applicati alla stima del percorso primario che mirano a migliorare le prestazioni dell’intero sistema. La struttura proposta garantisce un tasso di convergenza migliore rispetto all'algoritmo di riferimento.Immersive audio rendering is the process of creating an engaging and realistic sound experience in 3D space. In immersive audio systems, the head-related transfer functions (HRTFs) are used for binaural synthesis over headphones since they express how humans localize a sound source. HRTF interpolation algorithms can be introduced for reducing the number of measurement points and creating a reliable sound movement. Binaural reproduction can be also performed by loudspeakers. However, the involvement of two or more loudspeakers causes the problem of crosstalk. In this case, crosstalk cancellation (CTC) algorithms are needed to delete unwanted interference signals. In this thesis, starting from a comparative analysis of HRTF measurement techniques, a binaural rendering system based on HRTF interpolation is proposed and evaluated for real-time applications. The proposed method shows good performance in comparison with a reference technique. The interpolation algorithm is also applied for immersive audio rendering over loudspeakers, by adding a fixed crosstalk cancellation algorithm, which assumes that the listener is in a fixed position. In addition, an adaptive crosstalk cancellation system, which includes the tracking of the listener's head, is analyzed and a real-time implementation is presented. The adaptive CTC implements a subband structure and experimental results prove that a higher number of bands improves the performance in terms of total error and convergence rate. The reproduction system and the characteristics of the listening room may affect the performance due to their non-ideal frequency response. Audio equalization is used to adjust the balance of different audio frequencies in order to achieve desired sound characteristics. The equalization can be manual, such as in the case of graphic equalization, where the gain of each frequency band can be modified by the user, or automatic, where the equalization curve is automatically calculated after the room impulse response measurement. The room response equalization can be also applied to multichannel systems, which employ two or more loudspeakers, and the equalization zone can be enlarged by measuring the impulse responses in different points of the listening zone. In this thesis, efficient graphic equalizers (GEQs), and an adaptive room response equalization system are presented. In particular, three low-complexity linear- and quasi-linear-phase graphic equalizers are proposed and deeply examined. Experiments confirm the effectiveness of the proposed GEQs in terms of accuracy, computational complexity, and latency. Successively, a subband adaptive structure is introduced for the development of a multichannel and multiple positions room response equalizer. Experimental results verify the effectiveness of the subband approach in comparison with the single-band case. Finally, a linear-phase crossover network is presented for multichannel systems, showing great results in terms of magnitude flatness, cutoff rates, polar diagram, and phase response. Active noise control (ANC) systems can be designed to reduce the effects of noise pollution and can be used simultaneously with an immersive audio system. The ANC works by creating a sound wave that has an opposite phase with respect to the sound wave of the unwanted noise. The additional sound wave creates destructive interference, which reduces the overall sound level. Finally, this thesis presents an ANC system used for noise reduction. The proposed approach implements an online secondary path estimation and is based on cross-update adaptive filters applied to the primary path estimation that aim at improving the performance of the whole system. The proposed structure allows for a better convergence rate in comparison with a reference algorithm

Adaptiiviset läpikuuluvuuskuulokkeet

Hear-through equalization can be used to make a headset acoustically transparent, i.e.~to produce sound perception that is similar to perception without the headset. The headset must have microphones outside the earpieces to capture the ambient sounds, which is then reproduced with the headset transducers after the equalization. The reproduced signal is called the hear-through signal. Equalization is needed, since the headset affects the acoustics of the outer ear. \\ In addition to the external microphones, the headset used in this study has additional internal microphones. Together these microphones can be used to estimate the attenuation of the headset online and to detect poor fit. Since the poor fit causes leaks and decreased attenuation, the combined effect of the leaked sound and the hear-through signal changes, when compared to proper fit situation. Therefore, the isolation estimate is used to control the hear-through equalization in order to produce better acoustical transparency. Furthermore, the proposed adaptive hear-through algorithm includes manual controls for the equalizers and the volume of the hear-through signal. \\ The proposed algorithm is found to transform the used headset acoustically transparent. The equalization controls improve the performance of the headset, when the fit is poor or when the volume of the hear-through signal is adjusted, by reducing the comb-filtering effect due to the summation of the leaked sound and the hear-through signal inside the ear canal. The behavior of the proposed algorithm can be demonstrated with an implemented Matlab simulator.Läpikuuluvuusekvalisoinnilla voidaan saavuttaa akustinen läpinäkyvyys kuulokkeita käytettäessä, eli tuottaa samankaltainen ääniaistimus kuin mikä havaittaisiin ilman kuulokkeita. Käytetyissä kuulokkeissa tulee olla mikrofonit kuulokkeen ulkopinnalla, joiden avulla voidaan tallentaa ympäröiviä ääniä. Mikrofonisignaalit ekvalisoidaan, jolloin niistä tulee läpikuuluvuussignaalit, ja toistetaan kuulokkeista. Ekvalisointi on tarpeellista, sillä kuulokkeet muuttavat ulkokorvan akustiikka ja siten myös äänihavaintoa. \\ Tässä diplomityössä käytetyssä prototyyppikuulokeparissa on edellä mainittujen mikrofonien lisäksi myös toiset, korvakäytävän sisälle asettuvat mikrofonit. Yhdessä näiden kahden mikrofonin avulla voidaan määrittää reaaliaikainen estimaatti kuulokkeen vaimennukselle ja tunnistaa huono istuvuus. Koska huonosti asetettu kuuloke vuotaa enemmän ääntä korvakäytävän sisään kuin kunnolla asetettu, kuulokkeen äänen ja vuotavan äänen yhteisvaikutus muuttuu. Tästä syystä vaimennusestimaattia käytetään läpikuuluvuusekvalisoinnin säätöön, jotta akustinen läpinäkyvyys ei kärsisi. Lisäksi esitellyssä algoritmissa on manuaaliset säädöt ekvalisaattoreille ja läpikuuluvuussignaalin voimakkuudelle.\\ Esitetyn algoritmin havaitaan tuottavan akustinen läpinäkyvyys, kun sitä käytetään prototyyppikuulokkeiden kanssa. Ekvalisointisäädöt parantavat kuulokkeiden toimintaa istuvuuden ollessa huono tai säädettäessä läpikuuluvuussignaalin voimakkuutta, koska ne vähentävät kampasuodatusefektiä, joka voi aiheutua vuotavan äänen ja läpikuuluvuussignaalin summautuessa. Esitellyn algoritmin toimintaa voidaan havainnollistaa toteutetulla Matlab-simulaattorilla

3-D audio using loudspeakers

Thesis (Ph. D.)--Massachusetts Institute of Technology, Program in Media Arts & Sciences, 1997.Includes bibliographical references (p. 145-153).by William G. Gardner.Ph.D

Filter Bank Multicarrier Modulation for Spectrally Agile Waveform Design

In recent years the demand for spectrum has been steadily growing. With the limited amount of spectrum available, Spectrum Pooling has gained immense popularity. As a result of various studies, it has been established that most of the licensed spectrum remains underutilized. Spectrum Pooling or spectrum sharing concentrates on making the most of these whitespaces in the licensed spectrum. These unused parts of the spectrum are usually available in chunks. A secondary user looking to utilize these chunks needs a device capable of transmitting over distributed frequencies, while not interfering with the primary user. Such a process is known as Dynamic Spectrum Access (DSA) and a device capable of it is known as Cognitive Radio. In such a scenario, multicarrier communication that transmits data across the channel in several frequency subcarriers at a lower data rate has gained prominence. Its appeal lies in the fact that it combats frequency selective fading. Two methods for implementing multicarrier modulation are non-contiguous orthogonal frequency division multiplexing (NCOFDM)and filter bank multicarrier modulation (FBMC). This thesis aims to implement a novel FBMC transmitter using software defined radio (SDR) with modulated filters based on a lowpass prototype. FBMCs employ two sets of bandpass filters called analysis and synthesis filters, one at the transmitter and the other at the receiver, in order to filter the collection of subcarriers being transmitted simultaneously in parallel frequencies. The novel aspect of this research is that a wireless transmitter based on non-contiguous FBMC is being used to design spectrally agile waveforms for dynamic spectrum access as opposed to the more popular NC-OFDM. Better spectral containment and bandwidth efficiency, combined with lack of cyclic prefix processing, makes it a viable alternative for NC-OFDM. The main aim of this thesis is to prove that FBMC can be practically implemented for wireless communications. The practicality of the method is tested by transmitting the FBMC signals real time by using the Simulink environment and USRP2 hardware modules

Deep Learning for Audio Effects Modeling

PhD Thesis.Audio effects modeling is the process of emulating an audio effect unit and seeks to recreate the sound, behaviour and main perceptual features of an analog reference device. Audio effect units are analog or digital signal processing systems that transform certain characteristics of the sound source. These transformations can be linear or nonlinear, time-invariant or time-varying and with short-term and long-term memory. Most typical audio effect transformations are based on dynamics, such as compression; tone such as distortion; frequency such as equalization; and time such as artificial reverberation or modulation based audio effects. The digital simulation of these audio processors is normally done by designing mathematical models of these systems. This is often difficult because it seeks to accurately model all components within the effect unit, which usually contains mechanical elements together with nonlinear and time-varying analog electronics. Most existing methods for audio effects modeling are either simplified or optimized to a very specific circuit or type of audio effect and cannot be efficiently translated to other types of audio effects. This thesis aims to explore deep learning architectures for music signal processing in the context of audio effects modeling. We investigate deep neural networks as black-box modeling strategies to solve this task, i.e. by using only input-output measurements. We propose different DSP-informed deep learning models to emulate each type of audio effect transformations. Through objective perceptual-based metrics and subjective listening tests we explore the performance of these models when modeling various analog audio effects. Also, we analyze how the given tasks are accomplished and what the models are actually learning. We show virtual analog models of nonlinear effects, such as a tube preamplifier; nonlinear effects with memory, such as a transistor-based limiter; and electromechanical nonlinear time-varying effects, such as a Leslie speaker cabinet and plate and spring reverberators. We report that the proposed deep learning architectures represent an improvement of the state-of-the-art in black-box modeling of audio effects and the respective directions of future work are given

Synthesis and monolithic integration of analogue signal processing networks

Data traffic of future 5G telecommunication systems is projected to increase 10 000-fold compared to current rates. 5G fronthaul links are therefore expected to operate in the mm-wave spectrum with some preliminary International Telecommunication Union specifications set for the 71-76 and 81-86 GHz bands. Processing 5 GHz as a single contiguous band in real-time, using existing digital signal processing (DSP) systems, is exceedingly challenging. A similar challenge exists in radio astronomy, with the Square Kilometer Array project expecting data throughput rates of 15 Tbits/s at its completion. Speed improvements on existing state-of-the-art DSPs of 2-3 orders of magnitude are therefore required to meet future demands. One possible mitigating approach to processing wideband data in real-time is to replace some DSP blocks with analog signal processing (ASP) equivalents, since analogue devices outperform their digital counterparts in terms of cost, power consumption and the maximum attainable bandwidth. The fundamental building block of any ASP is an all-pass network of prescribed response, which can always be synthesized by cascaded first- and second-order all-pass sections (with two cascaded first-order sections being a special case of the latter). The monolithic integration of all-pass networks in commercial CMOS and BiCMOS technology nodes is a key consideration for commercial adaptation of ASPs, since it supports mass production at reduced costs and operating power requirements, making the ASP approach feasible. However, this integration has presented a number of yet unsolved challenges. Firstly, the state-of-the-art methods for synthesizing quasi-arbitrary group delay functions using all-pass elements lack a theoretical synthesis procedure that guarantees minimum-order networks. In this work an analytically-based solution to the synthesis problem is presented that produces an all-pass network with a response approximating the required group delay to within an arbitrary minimax error. This method is shown to work for any physical realization of second-order all-pass elements, is guaranteed to converge to a global optimum solution without any choice of seed values as an input, and allows synthesis of pre-defined networks described either analytically or numerically. Secondly, second-order all-pass networks are currently primarily implemented in off-chip planar media, which is unsuited for high volume production. Component sensitivity, process tolerances and on-chip parasitics often make proposed on-chip designs impractical. Consequently, to date, no measured results of a dispersive on-chip second-order all-pass network suitable for ASP applications (delay Q-value (QD) larger than 1) have been presented in either CMOS or BiCMOS technology nodes. In this work, the first ever on-chip CMOS second-order all-pass network is proposed with a measured QD-value larger than 1. Measurements indicate a post-tuning bandwidth of 280 MHz, peak-to-nominal delay variation of 10 ns, QD-value of 1.15 and magnitude variation of 3.1 dB. An active on-chip mm-wave second-order all-pass network is further demonstrated in a 130 nm SiGe BiCMOS technology node with a bandwidth of 40 GHz, peak-to-nominal delay of 62 ps, QD-value of 3.6 and a magnitude ripple of 1.4 dB. This is the first time that measurement results of a mm-wave bandwidth second-order all-pass network have been reported. This work therefore presents the first step to monolithically integrating ASP solutions to conventional DSP problems, thereby enabling ultra-wideband signal processing on-chip in commercial technology nodes.Thesis (PhD)--University of Pretoria, 2018.Square Kilometer Array (SKA) project - postgraduate scholarshipElectrical, Electronic and Computer EngineeringPhDUnrestricte

A room acoustics measurement system using non-invasive microphone arrays

This thesis summarises research into adaptive room correction for small rooms and pre-recorded material, for example music of films. A measurement system to predict the sound at a remote location within a room, without a microphone at that location was investigated. This would allow the sound within a room to be adaptively manipulated to ensure that all listeners received optimum sound, therefore increasing their enjoyment. The solution presented used small microphone arrays, mounted on the room's walls. A unique geometry and processing system was designed, incorporating three processing stages, temporal, spatial and spectral. The temporal processing identifies individual reflection arrival times from the recorded data. Spatial processing estimates the angles of arrival of the reflections so that the three-dimensional coordinates of the reflections' origin can be calculated. The spectral processing then estimates the frequency response of the reflection. These estimates allow a mathematical model of the room to be calculated, based on the acoustic measurements made in the actual room. The model can then be used to predict the sound at different locations within the room. A simulated model of a room was produced to allow fast development of algorithms. Measurements in real rooms were then conducted and analysed to verify the theoretical models developed and to aid further development of the system. Results from these measurements and simulations, for each processing stage are presented