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This thesis is concerned specifically with the implementation of large-scale professional DMCs. The design of such multi-DSP audio products is extremely challenging: one cannot simply lash together n DSPs and obtain /7-times the performance of a sole device. M-P models developed here show that topology and IPC mechanisms have critical design implications. Alternative processor technologies are investigated with respect to the requirements of DMC architectures. An extensive analysis of M-P topologies is undertaken using the metrics provided by the TPG tool. Novel methods supporting DSP message-passing connectivity lead to the development of a hybrid audio M-P (HYMIPS) employing these techniques. A DMC model demonstrates the impact of task allocation on ASP M-P architectures. Five application-specific heuristics and four static-labelling schemes are developed for scheduling console taskforces on M-Ps. An integrated research framework and DCS engine enable scheduling strategies to be analysed with regard to the DMC problem domain. Three scheduling algorithms — CPM, DYN and AST — and three IPC mechanisms — FWE, NSL and NML — are investigated. Dynamic-labelling strategies and mix-bus granularity issues are further studied in detail. To summarise, this thesis elucidates those topologies, construction techniques and scheduling algorithms appropriate to professional DMC systems.

Cette thèse s'inscrit dans le contexte de l'analyse/transformation/synthèse des signaux audio utilisant des représentations temps-fréquence, de type transformation de Gabor. Dans ce contexte, la complexité des transformations permettant de relier des sons peut être modélisée au moyen de multiplicateurs de Gabor, opérateurs de signaux linéaires caractérisés par une fonction de transfert temps-fréquence, à valeurs complexes, que l'on appelle masque de Gabor. Les multiplicateurs de Gabor permettent deformaliser le concept de filtrage dans le plan temps-fréquence. En agissant de façon multiplicative dans le plan temps-fréquence, ils sont a priori bien adaptés pour réaliser des transformations sonores telles que des modifications de timbre des sons. Dans un premier temps, ce travail de thèses intéresse à la modélisation du problème d'estimation d'un masque de Gabor entre deux signaux donnés et la mise en place de méthodes de calculs efficaces permettant de résoudre le problème. Le multiplicateur de Gabor entre deux signaux n'est pas défini de manière unique et les techniques d'estimation proposées de construire des multiplicateurs produisant des signaux sonores de qualité satisfaisante. Dans un second temps, nous montrons que les masques de Gabor contiennent une information pertinente capable d'établir une classification des signaux,et proposons des stratégies permettant de localiser automatiquement les régions temps-fréquence impliquées dans la différentiation de deux classes de signaux. Enfin, nous montrons que les multiplicateurs de Gabor constituent tout un panel de transformations sonores entre deux sons, qui, dans certaines situations, peuvent être guidées par des descripteurs de timbre / Analysis/Transformation/Synthesis is a generalparadigm in signal processing, that aims at manipulating or generating signalsfor practical applications. This thesis deals with time-frequencyrepresentations obtained with Gabor atoms. In this context, the complexity of a soundtransformation can be modeled by a Gabor multiplier. Gabormultipliers are linear diagonal operators acting on signals, andare characterized by a time-frequency transfer function of complex values, called theGabor mask. Gabor multipliers allows to formalize the conceptof filtering in the time-frequency domain. As they act by multiplying in the time-frequencydomain, they are "a priori'' well adapted to producesound transformations like timbre transformations. In a first part, this work proposes to model theproblem of Gabor mask estimation between two given signals,and provides algorithms to solve it. The Gabor multiplier between two signals is not uniquely defined and the proposed estimationstrategies are able to generate Gabor multipliers that produce signalswith a satisfied sound quality. In a second part, we show that a Gabor maskcontain a relevant information, as it can be viewed asa time-frequency representation of the difference oftimbre between two given sounds. By averaging the energy contained in a Gabor mask, we obtain a measure of this difference that allows to discriminate different musical instrumentsounds. We also propose strategies to automaticallylocalize the time-frequency regions responsible for such a timbre dissimilarity between musicalinstrument classes. Finally, we show that the Gabor multipliers can beused to construct a lot of sounds morphing trajectories,and propose an extension

The complex relationship between array gain patterns and microphone distributions limits the application of traditional optimization algorithms on irregular arrays, which show enhanced beamforming performance for human speech capture in immersive environments. This work analyzes the relationship between irregular microphone geometries and spatial filtering performance with statistical methods. Novel geometry descriptors are developed to capture the properties of irregular microphone distributions showing their impact on array performance. General guidelines and optimization methods for regular and irregular array design are proposed in immersive (near-field) environments to obtain superior beamforming ability for speech applications. Optimization times are greatly reduced through the objective function rules using performance-based geometric descriptions of microphone distributions that circumvent direct array gain computations over the space of interest. In addition, probabilistic descriptions of acoustic scenes are introduced to incorporate various levels of prior knowledge for the source distribution. To verify the effectiveness of the proposed optimization methods, simulated gain patterns and real SNR results of the optimized arrays are compared to corresponding traditional regular arrays and arrays obtained from direct exhaustive searching methods. Results show large SNR enhancements for the optimized arrays over arbitrary randomly generated arrays and regular arrays, especially at low microphone densities. The rapid convergence and acceptable processing times observed during the experiments establish the feasibility of proposed optimization methods for array geometry design in immersive environments where rapid deployment is required with limited knowledge of the acoustic scene, such as in mobile platforms and audio surveillance applications.

This research focuses on biologically inspired audio signal processing using programmable analog circuitry. This research is inspired by the biology of the human cochlea since biology far outperforms any engineered system at converting audio signals into meaningful electrical signals. The human cochlea efficiently decomposes any sound into the respective frequency components by harnessing the resonance nature of the basilar membrane, essentially forming a bank of bandpass filters. In a similar fashion, this work revolves around developing a filter bank composed of continuous-time, low-power, analog bandpass filters that serve as the core front end to this silicon audio-processing system. Like biology, the individual bandpass filters are tuned to have narrow bandwidths, moderate amounts of resonance, and exponentially spaced center frequencies. This audio front end serves to efficiently convert incoming sounds into information useful to subsequent signal-processing elements, and it does so by performing a frequency decomposition of the waveform with extremely low-power consumption and real-time operation. To overcome mismatch and offsets inherent in CMOS processes, floating-gate transistors are used to precisely tune the time constants in the filters and to allow programmability of analog components.

The Generalized Sidelobe Canceller is an adaptive algorithm for optimally estimating the parameters for beamforming, the signal processing technique of combining data from an array of sensors to improve SNR at a point in space. This work focuses on the algorithm’s application to widely-separated microphone arrays with irregular distributions used for human voice capture. Methods are presented for improving the performance of the algorithm’s blocking matrix, a stage that creates a noise reference for elimination, by proposing a stochastic model for amplitude correction and enhanced use of cross correlation for phase correction and time-difference of arrival estimation via a correlation coefficient threshold. This correlation technique is also applied to a multilateration algorithm for an efficient method of explicit target tracking. In addition, the underlying microphone array geometry is studied with parameters and guidelines for evaluation proposed. Finally, an analysis of the stability of the system is performed with respect to its adaptation parameters.

Over the years, a variety of array processing techniques have been applied to the problem of enhancing degraded speech to improve automatic speech recognition. In this context, linear beamforming has long been the approach of choice, for reasons including good performance, robustness and analytical simplicity. While various non-linear techniques - typically based to some extent on the study of auditory scene analysis - have also been of interest, they tend to lag behind their linear counterparts in terms of simplicity, scalability and exibility. Nonlinear techniques are also more difficult to analyze and lack the systematic descriptions available in the study of linear beamformers. This work focuses on a class of nonlinear processing, known as time-frequency (T-F) masking - a.k.a. spectro-temporal masking { whose variants comprise a significant portion of the existing techniques. T-F masking is based on accepting or rejecting individual time-frequency cells based on some estimate of local signal quality. Analyses are developed that attempt to mirror the beam patterns used to describe linear processing, leading to a view of T-F masking as "nonlinear beamforming". Two distinct formulations of these "nonlinear beam patterns" are developed, based on different metrics of the algorithms behavior; these formulations are modeled in a variety of scenarios to demonstrate the flexibility of the idea. While these patterns are not quite as simple or all-encompassing as traditional beam patterns in microphone-array processing, they do accurately represent the behavior of masking algorithms in analogous and intuitive ways. In addition to analyzing this class of nonlinear masking algorithm, we also attempt to improve its performance in a variety of ways. Improvements are proposed to the baseline two-channel version of masking, by addressing both the mask estimation and the signal reconstruction stages; the latter more successfully than the former. Furthermore, while these approaches have been shown to outperform linear beamforming in two-sensor arrays, extensions to larger arrays have been few and unsuccessful. We find that combining beamforming and masking is a viable method of bringing the benefits of masking to larger arrays. As a result, a hybrid beamforming-masking approach, called "post-masking", is developed that improves upon the performance of MMSE beamforming (and can be used with any beamforming technique), with the potential for even greater improvement in the future.

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In this dissertation, we discuss a reading profiling system, a biological data visualization system and a sports visualization system. Self-tracking is getting increasingly popular in the field of personal informatics. Reading profiling can be used as a personal data collection method. We present UUAT, an unintrusive user attention tracking system. In UUAT, we used user interaction data to develop technologies that help to pinpoint a users reading region (RR). Based on computed RR and user interaction data, UUAT can identify a readers reading struggle or interest. A biomarker is a measurable substance that may be used as an indicator of a particular disease. We developed CancerVis for visual and interactive analysis of cancer data and demonstrate how to apply this platform in cancer biomarker research. CancerVis provides interactive multiple views from different perspectives of a dataset. The views are synchronized so that users can easily link them to a same data entry. Furthermore, CancerVis supports data mining practice in cancer biomarker, such as visualization of optimal cutpoints and cutthrough exploration. Tennis match summarization helps after-live sports consumers assimilate an interested match. We developed TennisVis, a comprehensive match summarization and visualization platform. TennisVis offers chart- graph for a client to quickly get match facts. Meanwhile, TennisVis offers various queries of tennis points to satisfy diversified client preferences (such as volley shot, many-shot rally) of tennis fans. Furthermore, TennisVis offers video clips for every single tennis point and a recommendation rating is computed for each tennis play. A case study shows that TennisVis identifies more than 75% tennis points in full time match.

The main goal of this thesis is design of simple digital audio synthesizer. The synthesis of piano tones are descrribed. Final part is devoted to the basic principle of sound sample processing.