Acoustic contrast, planarity and robustness of sound zone methods using a circular loudspeaker array

Since the mid 1990s, acoustics research has been undertaken relating to the sound zone problem—using loudspeakers to deliver a region of high sound pressure while simultaneously creating an area where the sound is suppressed—in order to facilitate independent listening within the same acoustic enclosure. The published solutions to the sound zone problem are derived from areas such as wave field synthesis and beamforming. However, the properties of such methods differ and performance tends to be compared against similar approaches. In this study, the suitability of energy focusing, energy cancelation, and synthesis approaches for sound zone reproduction is investigated. Anechoic simulations based on two zones surrounded by a circular array show each of the methods to have a characteristic performance, quantified in terms of acoustic contrast, array control effort and target sound field planarity. Regularization is shown to have a significant effect on the array effort and achieved acoustic contrast, particularly when mismatched conditions are considered between calculation of the source weights and their application to the system

Application of sound source separation methods to advanced spatial audio systems

This thesis is related to the field of Sound Source Separation (SSS). It addresses the development and evaluation of these techniques for their application in the resynthesis of high-realism sound scenes by means of Wave Field Synthesis (WFS). Because the vast majority of audio recordings are preserved in twochannel stereo format, special up-converters are required to use advanced spatial audio reproduction formats, such as WFS. This is due to the fact that WFS needs the original source signals to be available, in order to accurately synthesize the acoustic field inside an extended listening area. Thus, an object-based mixing is required. Source separation problems in digital signal processing are those in which several signals have been mixed together and the objective is to find out what the original signals were. Therefore, SSS algorithms can be applied to existing two-channel mixtures to extract the different objects that compose the stereo scene. Unfortunately, most stereo mixtures are underdetermined, i.e., there are more sound sources than audio channels. This condition makes the SSS problem especially difficult and stronger assumptions have to be taken, often related to the sparsity of the sources under some signal transformation. This thesis is focused on the application of SSS techniques to the spatial sound reproduction field. As a result, its contributions can be categorized within these two areas. First, two underdetermined SSS methods are proposed to deal efficiently with the separation of stereo sound mixtures. These techniques are based on a multi-level thresholding segmentation approach, which enables to perform a fast and unsupervised separation of sound sources in the time-frequency domain. Although both techniques rely on the same clustering type, the features considered by each of them are related to different localization cues that enable to perform separation of either instantaneous or real mixtures.Additionally, two post-processing techniques aimed at improving the isolation of the separated sources are proposed. The performance achieved by several SSS methods in the resynthesis of WFS sound scenes is afterwards evaluated by means of listening tests, paying special attention to the change observed in the perceived spatial attributes. Although the estimated sources are distorted versions of the original ones, the masking effects involved in their spatial remixing make artifacts less perceptible, which improves the overall assessed quality. Finally, some novel developments related to the application of time-frequency processing to source localization and enhanced sound reproduction are presented.Cobos Serrano, M. (2009). Application of sound source separation methods to advanced spatial audio systems [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/8969Palanci

Subjective evaluation and electroacoustic theoretical validation of a new approach to audio upmixing

Audio signal processing systems for converting two-channel (stereo) recordings to four or five channels are increasingly relevant. These audio upmixers can be used with conventional stereo sound recordings and reproduced with multichannel home theatre or automotive loudspeaker audio systems to create a more engaging and natural-sounding listening experience. This dissertation discusses existing approaches to audio upmixing for recordings of musical performances and presents specific design criteria for a system to enhance spatial sound quality. A new upmixing system is proposed and evaluated according to these criteria and a theoretical model for its behavior is validated using empirical measurements.The new system removes short-term correlated components from two electronic audio signals using a pair of adaptive filters, updated according to a frequency domain implementation of the normalized-least-means-square algorithm. The major difference of the new system with all extant audio upmixers is that unsupervised time-alignment of the input signals (typically, by up to +/-10 ms) as a function of frequency (typically, using a 1024-band equalizer) is accomplished due to the non-minimum phase adaptive filter. Two new signals are created from the weighted difference of the inputs, and are then radiated with two loudspeakers behind the listener. According to the consensus in the literature on the effect of interaural correlation on auditory image formation, the self-orthogonalizing properties of the algorithm ensure minimal distortion of the frontal source imagery and natural-sounding, enveloping reverberance (ambiance) imagery.Performance evaluation of the new upmix system was accomplished in two ways: Firstly, using empirical electroacoustic measurements which validate a theoretical model of the system; and secondly, with formal listening tests which investigated auditory spatial imagery with a graphical mapping tool and a preference experiment. Both electroacoustic and subjective methods investigated system performance with a variety of test stimuli for solo musical performances reproduced using a loudspeaker in an orchestral concert-hall and recorded using different microphone techniques.The objective and subjective evaluations combined with a comparative study with two commercial systems demonstrate that the proposed system provides a new, computationally practical, high sound quality solution to upmixing

Design and Implementation of Switching Voltage Integrated Circuits Based on Sliding Mode Control

The need for high performance circuits in systems with low-voltage and low-power requirements has exponentially increased during the few last years due to the sophistication and miniaturization of electronic components. Most of these circuits are required to have a very good efficiency behavior in order to extend the battery life of the device. This dissertation addresses two important topics concerning very high efficiency circuits with very high performance specifications. The first topic is the design and implementation of class D audio power amplifiers, keeping their inherent high efficiency characteristic while improving their linearity performance, reducing their quiescent power consumption, and minimizing the silicon area. The second topic is the design and implementation of switching voltage regulators and their controllers, to provide a low-cost, compact, high efficient and reliable power conversion for integrated circuits. The first part of this dissertation includes a short, although deep, analysis on class D amplifiers, their history, principles of operation, architectures, performance metrics, practical design considerations, and their present and future market distribution. Moreover, the harmonic distortion of open-loop class D amplifiers based on pulse-width modulation (PWM) is analyzed by applying the duty cycle variation technique for the most popular carrier waveforms giving an easy and practical analytic method to evaluate the class D amplifier distortion and determine its specifications for a given linearity requirement. Additionally, three class D amplifiers, with an architecture based on sliding mode control, are proposed, designed, fabricated and tested. The amplifiers make use of a hysteretic controller to avoid the need of complex overhead circuitry typically needed in other architectures to compensate non-idealities of practical implementations. The design of the amplifiers based on this technique is compact, small, reliable, and provides a performance comparable to the state-of-the-art class D amplifiers, but consumes only one tenth of quiescent power. This characteristic gives to the proposed amplifiers an advantage for applications with minimal power consumption and very high performance requirements. The second part of this dissertation presents the design, implementation, and testing of switching voltage regulators. It starts with a description and brief analysis on the power converters architectures. It outlines the advantages and drawbacks of the main topologies, discusses practical design considerations, and compares their current and future market distribution. Then, two different buck converters are proposed to overcome the most critical issue in switching voltage regulators: to provide a stable voltage supply for electronic devices, with good regulation voltage, high efficiency performance, and, most important, a minimum number of components. The first buck converter, which has been designed, fabricated and tested, is an integrated dual-output voltage regulator based on sliding mode control that provides a power efficiency comparable to the conventional solutions, but potentially saves silicon area and input filter components. The design is based on the idea of stacking traditional buck converters to provide multiple output voltages with the minimum number of switches. Finally, a fully integrated buck converter based on sliding mode control is proposed. The architecture integrates the external passive components to deliver a complete monolithic solution with minimal silicon area. The buck converter employs a poly-phase structure to minimize the output current ripple and a hysteretic controller to avoid the generation of an additional high frequency carrier waveform needed in conventional solutions. The simulated results are comparable to the state-of-the-art works even with no additional post-fabrication process to improve the converter performance