<p>The advancements of computational power in low cost FPGAs are giving the opportunityto implement real-time compensation of loudspeakers and audio systems. The need for expensive commercial audio systems is reduced when the fidelity ofmuch cheaper audio systems easily can be improved by real-time compensation. The topic of this thesis is to investigate and evaluate methods for digital compensationof distortion in audio systems. More specifically, a VHDL module isimplemented to, when necessary, alleviate the problem of drastically deterioratingfidelity of the bass appearing when the input power is too high.</p>

The advancements of computational power in low cost FPGAs are giving the opportunityto implement real-time compensation of loudspeakers and audio systems. The need for expensive commercial audio systems is reduced when the fidelity ofmuch cheaper audio systems easily can be improved by real-time compensation. The topic of this thesis is to investigate and evaluate methods for digital compensationof distortion in audio systems. More specifically, a VHDL module isimplemented to, when necessary, alleviate the problem of drastically deterioratingfidelity of the bass appearing when the input power is too high.

Les systèmes audios incluent les instruments de musique traditionnels (percussions, cordes, vents, voix) et les systèmes électro-acoustiques (amplificateurs de guitares, pédales d’effets, synthétiseurs analogiques). Ces systèmes multi-physiques possèdent une propriété commune : hors des sources d’excitation (les générateurs), ils sont tous passifs. Nous présentons dans cette thèse un ensemble de méthodes automatiques dédiées à leur modélisation, leur simulation et leur contrôle, qui garantissent explicitement et exploitent la passivité du système original. Nous utilisons dans ce travail le formalisme des systèmes hamiltoniens à ports (SHP), introduits en automatique et théorie des systèmes au début des années 1990. Pour la modélisation, on exploite le fait que la connexion de systèmes décrits dans ce formalisme préserve explicitement la dynamique de la puissance dissipée de l'ensemble, pour développer une méthode automatique de modélisation d'instruments complets à partir de modèles élémentaires rassemblés dans un dictionnaire. Pour la simulation, une méthode numérique qui préserve la structure passive des SHP à temps discret a été développée, garantissant ainsi la stabilité des simulations (pour lesquelles le code C++ est généré automatiquement). Concernant le contrôle, on exploite la structure d'interconnexion afin de déterminer automatiquement une forme découplée (sous-systèmes hiérarchisés) pour une certaine classe de SHP. Les systèmes de cette classe sont dits systèmes hamiltonien à ports plats, au sens de la propriété de platitude différentielle, à partir de laquelle une loi de commande en boucle ouverte exacte sur le modèle est générée. / The class of audio systems includes traditional musical instruments (percussion, string, wind, brass, voice) and electro-acoustic systems (guitar amplifiers, analog audio processing, synthesizers). These multi-physical systems have a common property: out of excitation sources (generators), they are all passive. We present a set of automatic methods dedicated to their modeling, simulation and control, which explicitly guarantee and exploit the passivity of the original system. This class of systems is that of port-Hamiltonian systems (PHS), introduced in system theory in the early 1990s. Regarding the models, we exploit the fact that the interconnection of systems described in this formalism explicitly preserves the dynamics of total dissipated power. This enabled the development of an automated method that builds models of complete instruments based on a dictionary of elementary models. Regarding the simulations, we developed a numerical method that preserves the passive structure of PHS in discrete-time domain. This ensures the stability of simulations (for which the C++ code is automatically generated). Regarding the control, we exploit the interconnection structure to automatically build an input-to-output decoupled form for a class of PHS. Systems of this class are flat, within the meaning of the differential flatness approach. A formula that yields the (open loop) control law for these systems is provided.

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/10251/40651 / TESIS / Premios Extraordinarios de tesis doctorales

In spreads pectrum (SS)watermarking of audio signals, since the watermark acts as an additive noise to the host audio signal, the most important challenge is to maintain perceptual transparency. Human perception is a very sensitive apparatus, yet can be exploited to hide some information, reliably. SS watermark embedding has been proposed, in which psycho-acoustically shaped pseudo-random sequences are embedded directly into the time domain audio signal. However, these watermarking schemes use informed detection, in which the original signal is assumed available to the watermark detector. Blind detection of psycho-acoustically shaped SS watermarking is not well addressed in the literature. The problem is still interesting, because, blind detection is more practical for audio signals and, psycho-acoustically shaped watermarks embedding offers the maximum possible watermark energy under requirements of perceptual transparency. In this thesis we study the blind detection of psycho-acoustically shaped SS watermarks in time domain audio signals. We focus on a class of watermark sequences known as random phase watermarks, where the watermark magnitude spectrum is defined by the perceptual criteria and the randomness of the sequence lies in their phase spectrum. Blind watermark detectors, which do not have access to the original host signal, may seem handicapped, because an approximate watermark has to be re-derived from the watermarked signal. Since the comparison of blind detection with fully informed detection is unfair, a hypothetical detection scheme, denoted as semi-blind detection, is used as a reference benchmark. In semi-blind detection, the host signal as such is not available for detection, but it is assumed that sufficient information is available for deriving the exact watermark, which could be embedded in the given signal. Some reduction in performance is anticipated in blind detection over the semi-blind detection. Our experiments revealed that the statistical performance of the blind detector is better than that of the semi-blind detector. We analyze the watermark-to-host correlation (WHC) of random phase watermarks, and the results indicate that WHC is higher when a legitimate watermark is present in the audio signal, which leads to better detection performance. Based on these findings, we attempt to harness this increased correlation in order to further improve the performance. The analysis shows that uniformly distributed phase difference (between the host signal and the watermark) provides maximum advantage. This property is verified through experimentation over a variety of audio signals. In the second part, the correlated nature of audio signals is identified as a potential threat to reliable blind watermark detection, and audio pre-whitening methods are suggested as a possible remedy. A direct deterministic whitening (DDW) scheme is derived, from the frequency domain analysis of the time domain correlation process. Our experimental studies reveal that, the Savitzky-Golay Whitening (SGW), which is otherwise inferior to DDW technique, performs better when the audio signal is predominantly low pass. The novelty of this work lies in exploiting the complementary nature of the two whitening techniques and combining them to obtain a hybrid whitening (HbW) scheme. In the hybrid scheme the DDW and SGW techniques are selectively applied, based on short time spectral characteristics of the audio signal. The hybrid scheme extends the reliability of watermark detection to a wider range of audio signals. We also discuss enhancements to the HbW technique for robustness to temporal offsets and filtering. Robustness of SS watermark blind detection, with hybrid whitening, is determined through a set of experiments and the results are presented. It is seen that the watermarking scheme is robust to common signal processing operations such as additive noise, filtering, lossy compression, etc.