Universal Silicon Microwave Photonic Spectral Shaper

Optical modulation plays arguably the utmost important role in microwave photonic (MWP) systems. Precise synthesis of modulated optical spectra dictates virtually all aspects of MWP system quality including loss, noise figure, linearity, and the types of functionality that can be executed. But for such a critical function, the versatility to generate and transform analog optical modulation is severely lacking, blocking the pathways to truly unique MWP functions including ultra-linear links and low-loss high rejection filters. Here we demonstrate versatile RF photonic spectrum synthesis in an all-integrated silicon photonic circuit, enabling electrically-tailorable universal analog modulation transformation. We show a series of unprecedented RF filtering experiments through monolithic integration of the spectrum-synthesis circuit with a network of reconfigurable ring resonators

CARIOCA: a fast binary front-end implemented in 0.25μ0.25\mu CMOS using a Novel current-mode technique for the LHCb muon detector

The CARIOCA front-end is an amplifier discriminator chip, using 0.25mm CMOS technology, developed with a very fast and low noise preamplifier. This prototype was designed to have input impedance below 10W. Measurements showed a peaking time of 14ns and noise of 450e- at zero input capacitance, with a noise slope of 37.4 e-/pF. The sensitivity of 8mV/fC remains almost unchanged up to a detector capacitance of 120pF

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

Novel Front-end Electronics for Time Projection Chamber Detectors

Este trabajo ha sido realizado en la Organización Europea para la Investigación Nuclear (CERN) y forma parte del proyecto de investigación Europeo para futuros aceleradores lineales (EUDET). En física de partículas existen diferentes categorías de detectores de partículas. El diseño presentado esta centrado en un tipo particular de detector de trayectoria de partículas denominado TPC (Time Projection Chamber) que proporciona una imagen en tres dimensiones de las partículas eléctricamente cargadas que atraviesan su volumen gaseoso. La tesis incluye un estudio de los objetivos para futuros detectores, resumiendo los parámetros que un sistema de adquisición de datos debe cumplir en esos casos. Además, estos requisitos son comparados con los actuales sistemas de lectura utilizados en diferentes detectores TPC. Se concluye que ninguno de los sistemas cumple las restrictivas condiciones. Algunos de los principales objetivos para futuros detectores TPC son un altísimo nivel de integración, incremento del número de canales, electrónica más rápida y muy baja potencia. El principal inconveniente del estado del arte de los sistemas anteriores es la utilización de varios circuitos integrados en la cadena de adquisición. Este hecho hace imposible alcanzar el altísimo nivel de integración requerido para futuros detectores. Además, un aumento del número de canales y frecuencia de muestreo haría incrementar hasta valores no permitidos la potencia utilizada. Y en consecuencia, incrementar la refrigeración necesaria (en caso de ser posible). Una de las novedades presentadas es la integración de toda la cadena de adquisición (filtros analógicos de entrada, conversor analógico-digital (ADC) y procesado de señal digital) en un único circuito integrado en tecnología de 130nm. Este chip es el primero que realiza esta altísima integración para detectores TPC. Por otro lado, se presenta un análisis detallado de los filtros de procesado de señal. Los objetivos más importantes es la reduccióGarcía García, EJ. (2012). Novel Front-end Electronics for Time Projection Chamber Detectors [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/16980Palanci

An investigation into the real-time manipulation and control of three-dimensional sound fields

This thesis describes a system that can be used for the decoding of a three dimensional audio recording over headphones or two, or more, speakers. A literature review of psychoacoustics and a review (both historical and current) of surround sound systems is carried out. The need for a system which is platform independent is discussed, and the proposal for a system based on an amalgamation of Ambisonics, binaural and transaural reproduction schemes is given. In order for this system to function optimally, each of the three systems rely on providing the listener with the relevant psychoacoustic cues. The conversion from a five speaker ITU array to binaural decode is well documented but pair-wise panning algorithms will not produce the correct lateralisation parameters at the ears of a centrally seated listener. Although Ambisonics has been well researched, no one has, as yet, produced a psychoacoustically optimised decoder for the standard irregular five speaker array as specified by the ITU as the original theory, as proposed by Gerzon and Barton (1992) was produced (known as a Vienna decoder), and example solutions given, before the standard had been decided on. In this work, the original work by Gerzon and Barton (1992) is analysed, and shown to be suboptimal, showing a high/low frequency decoder mismatch due to the method of solving the set of non-linear simultaneous equations. A method, based on the Tabu search algorithm, is applied to the Vienna decoder problem and is shown to provide superior results to those shown by Gerzon and Barton (1992) and is capable of producing multiple solutions to the Vienna decoder problem. During the write up of this report Craven (2003) has shown how 4th order circular harmonics (as used in Ambisonics) can be used to create a frequency independent panning law for the five speaker ITU array, and this report also shows how the Tabu search algorithm can be used to optimise these decoders further. A new method is then demonstrated using the Tabu search algorithm coupled with lateralisation parameters extracted from a binaural simulation of the Ambisonic system to be optimised (as these are the parameters that the Vienna system is approximating). This method can then be altered to take into account head rotations directly which have been shown as an important psychoacoustic parameter in the localisation of a sound source (Spikofski et al., 2001) and is also shown to be useful in differentiating between decoders optimised using the Tabu search form of the Vienna optimisations as no objective measure had been suggested. Optimisations for both Binaural and Transaural reproductions are then discussed so as to maximise the performance of generic HRTF data (i.e. not individualised) using inverse filtering methods, and a technique is shown that minimises the amount of frequency dependant regularisation needed when calculating cross-talk cancellation filters.EPRS

PERFORMANCE IMPROVEMENT OF MULTICHANNEL AUDIO BY GRAPHICS PROCESSING UNITS

Multichannel acoustic signal processing has undergone major development in recent years due to the increased complexity of current audio processing applications. People want to collaborate through communication with the feeling of being together and sharing the same environment, what is considered as Immersive Audio Schemes. In this phenomenon, several acoustic e ects are involved: 3D spatial sound, room compensation, crosstalk cancelation, sound source localization, among others. However, high computing capacity is required to achieve any of these e ects in a real large-scale system, what represents a considerable limitation for real-time applications. The increase of the computational capacity has been historically linked to the number of transistors in a chip. However, nowadays the improvements in the computational capacity are mainly given by increasing the number of processing units, i.e expanding parallelism in computing. This is the case of the Graphics Processing Units (GPUs), that own now thousands of computing cores. GPUs were traditionally related to graphic or image applications, but new releases in the GPU programming environments, CUDA or OpenCL, allowed that most applications were computationally accelerated in elds beyond graphics. This thesis aims to demonstrate that GPUs are totally valid tools to carry out audio applications that require high computational resources. To this end, di erent applications in the eld of audio processing are studied and performed using GPUs. This manuscript also analyzes and solves possible limitations in each GPU-based implementation both from the acoustic point of view as from the computational point of view. In this document, we have addressed the following problems: Most of audio applications are based on massive ltering. Thus, the rst implementation to undertake is a fundamental operation in the audio processing: the convolution. It has been rst developed as a computational kernel and afterwards used for an application that combines multiples convolutions concurrently: generalized crosstalk cancellation and equalization. The proposed implementation can successfully manage two di erent and common situations: size of bu ers that are much larger than the size of the lters and size of bu ers that are much smaller than the size of the lters. Two spatial audio applications that use the GPU as a co-processor have been developed from the massive multichannel ltering. First application deals with binaural audio. Its main feature is that this application is able to synthesize sound sources in spatial positions that are not included in the database of HRTF and to generate smoothly movements of sound sources. Both features were designed after di erent tests (objective and subjective). The performance regarding number of sound source that could be rendered in real time was assessed on GPUs with di erent GPU architectures. A similar performance is measured in a Wave Field Synthesis system (second spatial audio application) that is composed of 96 loudspeakers. The proposed GPU-based implementation is able to reduce the room e ects during the sound source rendering. A well-known approach for sound source localization in noisy and reverberant environments is also addressed on a multi-GPU system. This is the case of the Steered Response Power with Phase Transform (SRPPHAT) algorithm. Since localization accuracy can be improved by using high-resolution spatial grids and a high number of microphones, accurate acoustic localization systems require high computational power. The solutions implemented in this thesis are evaluated both from localization and from computational performance points of view, taking into account different acoustic environments, and always from a real-time implementation perspective. Finally, This manuscript addresses also massive multichannel ltering when the lters present an In nite Impulse Response (IIR). Two cases are analyzed in this manuscript: 1) IIR lters composed of multiple secondorder sections, and 2) IIR lters that presents an allpass response. Both cases are used to develop and accelerate two di erent applications: 1) to execute multiple Equalizations in a WFS system, and 2) to reduce the dynamic range in an audio signal.Belloch Rodríguez, JA. (2014). PERFORMANCE IMPROVEMENT OF MULTICHANNEL AUDIO BY GRAPHICS PROCESSING UNITS [Tesis doctoral]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/40651TESISPremios Extraordinarios de tesis doctorale

Surround by Sound: A Review of Spatial Audio Recording and Reproduction

In this article, a systematic overview of various recording and reproduction techniques for spatial audio is presented. While binaural recording and rendering is designed to resemble the human two-ear auditory system and reproduce sounds specifically for a listener’s two ears, soundfield recording and reproduction using a large number of microphones and loudspeakers replicate an acoustic scene within a region. These two fundamentally different types of techniques are discussed in the paper. A recent popular area, multi-zone reproduction, is also briefly reviewed in the paper. The paper is concluded with a discussion of the current state of the field and open problemsThe authors acknowledge National Natural Science Foundation of China (NSFC) No. 61671380 and Australian Research Council Discovery Scheme DE 150100363

An in-band full-duplex radio receiver with a passive vector modulator downmixer for self-interference cancellation

In-band full-duplex (FD) wireless, i.e., simultaneous transmission and reception at the same frequency, introduces strong self-interference (SI) that masks the signal to be received. This paper proposes a receiver in which a copy of the transmit signal is fed through a switched-resistor vector modulator (VM)that provides simultaneous downmixing, phase shift, and amplitude scaling and subtracts it in the analog baseband for up to 27 dB SI-cancellation. Cancelling before active baseband amplification avoids self-blocking, and highly linear mixers keep SIinduced distortion low, for a receiver SI-to-noise-and-distortionratio (SINDR) of up to 71.5 dB in 16.25 MHz BW. When combined with a two-port antenna with only 20 dB isolation, the low RX distortion theoretically allows sufficient digital cancellation for over 90 dB link budget, sufficient for short-range, low-power FD links

Methods for reducing visual discomfort in stereoscopic 3D: A review

This work was supported by the EPSRC Grant EP/M01469X/1, “Geometric Evaluation of Stereoscopic Video”