33 research outputs found

    PERFORMANCE IMPROVEMENT OF MULTICHANNEL AUDIO BY GRAPHICS PROCESSING UNITS

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    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

    Student Teaching and Research Laboratory Focusing on Brain-computer Interface Paradigms - A Creative Environment for Computer Science Students -

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    This paper presents an applied concept of a brain-computer interface (BCI) student research laboratory (BCI-LAB) at the Life Science Center of TARA, University of Tsukuba, Japan. Several successful case studies of the student projects are reviewed together with the BCI Research Award 2014 winner case. The BCI-LAB design and project-based teaching philosophy is also explained. Future teaching and research directions summarize the review.Comment: 4 pages, 4 figures, accepted for EMBC 2015, IEEE copyrigh

    Accelerating multi-channel filtering of audio signal on ARM processors

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    The researchers from Universitat Jaume I are supported by the CICYT projects TIN2014-53495-R and TIN2011-23283 of the Ministerio de Economía y Competitividad and FEDER. The authors from the Universitat Politècnica de València are supported by projects TEC2015-67387-C4-1-R and PROMETEOII/2014/003. This work was also supported from the European Union FEDER (CAPAP-H5 network TIN2014-53522-REDT)

    A Parallel Approach to HRTF Approximation and Interpolation Based on a Parametric Filter Model

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    "© 2017 IEEE. Personal use of this material is permitted. Permissíon from IEEE must be obtained for all other uses, in any current or future media, including reprinting/republishing this material for advertisíng or promotional purposes, creating new collective works, for resale or redistribution to servers or lists, or reuse of any copyrighted component of this work in other works."[EN] Spatial audio-rendering techniques using head-related transfer functions (HRTFs) are currently used in many different contexts such as immersive teleconferencing systems, gaming, or 3-D audio reproduction. Since all these applications usually involve real-time constraints, efficient processing structures for HRTF modeling and interpolation are necessary for providing real-time binaural audio solutions. This letter presents a parametric parallel model that allows us to perform HRTF filtering and interpolation efficiently from an input HRTF dataset. The resulting model, which is an adaptation from a recently proposed modeling technique, not only reduces the size of HRTF datasets significantly, but also allows for simplified interpolation and real-time computation over parallel processors. In order to discuss the suitability of this new model, an implementation over a graphic processing unit is presented.This work was supported by the Spanish Ministry of Economy and Competitiveness under Grant TEC2012-37945-C02-02 and FEDER funds and by the UNKP-16-4-III New National Excellence Program of the Hungarian Ministry of Human Capacities. The work of J. A. Belloch was supported by GVA Postdoctoral Contract APOSTD/2016/069.Ramos Peinado, G.; Cobos Serrano, M.; Bank, B.; Belloch Rodríguez, JA. (2017). A Parallel Approach to HRTF Approximation and Interpolation Based on a Parametric Filter Model. IEEE Signal Processing Letters. 24(10):1507-1511. https://doi.org/10.1109/LSP.2017.2741724S15071511241

    Multichannel massive audio processing for a generalized crosstalk cancellation and equalization application using GPUs

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    [EN] Multichannel acoustic signal processing has undergone major development in recent years due to the increased com- plexity of current audio processing applications, which involves the processing of multiple sources, channels, or filters. A gen- eral scenario that appears in this context is the immersive reproduction of binaural audio without the use of headphones, which requires the use of a crosstalk canceler. However, generalized crosstalk cancellation and equalization (GCCE) requires high com- puting capacity, which is a considerable limitation for real-time applications. This paper discusses the design and implementation of all the processing blocks of a multichannel convolution on a GPU for real-time applications. To this end, a very efficient fil- tering method using specific data structures is proposed, which takes advantage of overlap-save filtering and filter fragmentation. It has been shown that, for a real-time application with 22 inputs and 64 outputs, the system is capable of managing 1408 filters of 2048 coefficients with a latency time less than 6 ms. The proposed GPU implementation can be easily adapted to any acoustic environment, demonstrating the validity of these co-processors for managing intensive multichannel audio applications.This work has been partially funded by Spanish Ministerio de Ciencia e Innovacion TEC2009-13741, Generalitat Valenciana PROMETEO 2009/2013 and GV/2010/027, and Universitat Politecnica de Valencia through Programa de Apoyo a la Investigacion y Desarrollo (PAID-05-11).Belloch Rodríguez, JA.; Gonzalez, A.; Martínez Zaldívar, FJ.; Vidal Maciá, AM. (2013). Multichannel massive audio processing for a generalized crosstalk cancellation and equalization application using GPUs. Integrated Computer-Aided Engineering. 20(2):169-182. https://doi.org/10.3233/ICA-130422S16918220

    SoundFields: A Virtual Reality Game Designed to Address Auditory Hypersensitivity in Individuals with Autism Spectrum Disorder

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    Individuals with autism spectrum disorder (ASD) are characterised as having impairments in social-emotional interaction and communication, alongside displaying repetitive behaviours and interests. Additionally, they can frequently experience difficulties in processing sensory information with particular prevalence in the auditory domain. Often triggered by everyday environmental sounds, auditory hypersensitivity can provoke self-regulatory fear responses such as crying and isolation from sounds. This paper presents SoundFields, an interactive virtual reality game designed to address this area by integrating exposure based therapy techniques into game mechanics and delivering target auditory stimuli to the player rendered via binaural based spatial audio. A pilot study was conducted with six participants diagnosed with ASD who displayed hypersensitivity to specific sounds to evaluate the use of SoundFields as a tool to reduce levels of anxiety associated with identified problematic sounds. During the course of the investigation participants played the game weekly over four weeks and all participants actively engaged with the virtual reality (VR) environment and enjoyed playing the game. Following this period, a comparison of pre- and post-study measurements showed a significant decrease in anxiety linked to target auditory stimuli. The study results therefore suggest that SoundFields could be an effective tool for helping individuals with autism manage auditory hypersensitivity

    GPU Implementation of multichannel adaptive algorithms for local active noise control

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    © 2014 IEEE. Personal use of this material is permitted. Permission from IEEE must be obtained for all other uses, in any current or future media, including reprinting/republishing this material for advertising or promotional purposes, creating new collective works, for resale or redistribution to servers or lists, or reuse of any copyrighted component of this work in other worksMultichannel active noise control (ANC) systems are commonly based on adaptive signal processing algorithms that require high computational capacity, which constrains their practical implementation. Graphics Processing Units (GPUs) are well known for their potential for highly parallel data processing. Therefore, GPUs seem to be a suitable platform for multichannel scenarios. However, efficient use of parallel computation in the adaptive filtering context is not straightforward due to the feedback loops. This paper compares two GPU implementations of a multichannel feedforward local ANC system working as a real-time prototype. Both GPU implementations are based on the filtered-x Least Mean Square algorithms; one is based on the conventional filtered-x scheme and the other is based on the modified filtered-x scheme. Details regarding the parallelization of the algorithms are given. Finally, experimental results are presented to compare the performance of both multichannel ANC GPU implementations. The results show the usefulness of many-core devices for developing versatile, scalable, and low-cost multichannel ANC systems.This work was supported by the European Union ERDF and Spanish Government under Project TEC2012-38142-C04, and Generalitat Valenciana under Project PROMETEO/2009/013. The associate editor coordinating the review of this manuscript and approving it for publication was Prof. Thushara D. Abhayapala.Lorente Giner, J.; Ferrer Contreras, M.; Diego Antón, MD.; Gonzalez, A. (2014). GPU Implementation of multichannel adaptive algorithms for local active noise control. IEEE Transactions on Audio, Speech and Language Processing. 22(11):1624-1635. https://doi.org/10.1109/TASLP.2014.2344852S16241635221

    De la prise de son à la diffusion: mystères et mécanique de la perspective sonore.

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    Il s'agit ici de proposer un aperçu de quelques techniques mises en œuvre dans l'objectif de proposer à l'auditeur, ou l'audio-spectateur une représentation convaincante de la notion d'espace sonore.Après une description des sensations éprouvées qui concourent à une représentation de l'espace, nous évoquerons quelques procédés techniques et leurs perspectives.After a concise statement of the auditory perception, an overview of some audio production processes implemented in order to cogently figure sound space in filmmaking

    A survey on hardware and software solutions for multimodal wearable assistive devices targeting the visually impaired

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    The market penetration of user-centric assistive devices has rapidly increased in the past decades. Growth in computational power, accessibility, and cognitive device capabilities have been accompanied by significant reductions in weight, size, and price, as a result of which mobile and wearable equipment are becoming part of our everyday life. In this context, a key focus of development has been on rehabilitation engineering and on developing assistive technologies targeting people with various disabilities, including hearing loss, visual impairments and others. Applications range from simple health monitoring such as sport activity trackers, through medical applications including sensory (e.g. hearing) aids and real-time monitoring of life functions, to task-oriented tools such as navigational devices for the blind. This paper provides an overview of recent trends in software and hardware-based signal processing relevant to the development of wearable assistive solutions

    Optimized Fundamental Signal Processing Operations for Energy Minimization on Heterogeneous Mobile Devices

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    [EN] Numerous signal processing applications are emerging on both mobile and high-performance computing systems. These applications are subject to responsiveness constraints for user interactivity and, at the same time, must be optimized for energy efficiency. The increasingly heterogeneous power-versus-performance profile of modern hardware introduces new opportunities for energy savings as well as challenges. In this line, recent systems-on-chip (SoC) composed of low-power multicore processors, combined with a small graphics accelerator (or GPU), yield a notable increment of the computational capacity while partially retaining the appealing low power consumption of embedded systems. This paper analyzes the potential of these new hardware systems to accelerate applications that involve a large number of floating-point arithmetic operations mainly in the form of convolutions. To assess the performance, a headphone-based spatial audio application for mobile devices based on a Samsung Exynos 5422 SoC has been developed. We discuss different implementations and analyze the tradeoffs between performance and energy efficiency for different scenarios and configurations. Our experimental results reveal that we can extend the battery lifetime of a device featuring such an architecture by a 238% by properly configuring and leveraging the computational resources.This work was supported by the Spanish Ministerio de Economia y Competitividad projects under Grant TIN2014-53495-R and Grant TEC2015-67387-C4-1-R, in part by the University Project UJI-B2016-20, in part by the Project PROMETEOII/2014/003. The work of J. A. Belloch was supported by the GVA Post-Doctoral Contract under Grant APOSTD/2016/069. This paper was recommended by Associate Editor Y. Ha.Belloch Rodríguez, JA.; Badia Contelles, JM.; Igual Peña, FD.; Gonzalez, A.; Quintana Ortí, ES. (2017). Optimized Fundamental Signal Processing Operations for Energy Minimization on Heterogeneous Mobile Devices. IEEE Transactions on Circuits and Systems I Regular Papers. 65(5):1614-1627. https://doi.org/10.1109/TCSI.2017.2761909S1614162765
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