Spatial sound for computer games and virtual reality

In this chapter, we discuss spatial sound within the context of Virtual Reality and other synthetic environments such as computer games. We review current audio technologies, sound constraints within immersive multi-modal spaces, and future trends. The review process takes into consideration the wide-varying levels of audio sophistication in the gaming and VR industries, ranging from standard stereo output to Head Related Transfer Function implementation. The level of sophistication is determined mostly by hardware/system constraints (such as mobile devices or network limitations), however audio practitioners are developing novel and diverse methods to overcome many of these challenges. No matter what approach is employed, the primary objectives are very similar—the enhancement of the virtual scene and the enrichment of the user experience. We discuss how successful various audio technologies are in achieving these objectives, how they fall short, and how they are aligned to overcome these shortfalls in future implementations

Object-based reverberation for spatial audio

Object-based audio is gaining momentum as a means for future audio content to be more immersive, interactive, and accessible. Recent standardization developments make recommendations for object formats; however, the capture, production, and reproduction of reverberation is an open issue. In this paper parametric approaches for capturing, representing, editing, and rendering reverberation over a 3D spatial audio system are reviewed. A framework is proposed for a Reverberant Spatial Audio Object (RSAO), which synthesizes reverberation inside an audio object renderer. An implementation example of an object scheme utilizing the RSAO framework is provided, and supported with listening test results, showing that: the approach correctly retains the sense of room size compared to a convolved reference; editing RSAO parameters can alter the perceived room size and source distance; and, format-agnostic rendering can be exploited to alter listener envelopment

Backward Compatible Spatialized Teleconferencing based on Squeezed Recordings

Commercial teleconferencing systems currently available, although offering sophisticated video stimulus of the remote participants, commonly employ only mono or stereo audio playback for the user. However, in teleconferencing applications where there are multiple participants at multiple sites, spatializing the audio reproduced at each site (using headphones or loudspeakers) to assist listeners to distinguish between participating speakers can significantly improve the meeting experience (Baldis, 2001; Evans et al., 2000; Ward & Elko 1999; Kilgore et al., 2003; Wrigley et al., 2009; James & Hawksford, 2008). An example is Vocal Village (Kilgore et al., 2003), which uses online avatars to co-locate remote participants over the Internet in virtual space with audio spatialized over headphones (Kilgore, et al., 2003). This system adds speaker location cues to monaural speech to create a user manipulable soundfield that matches the avatar’s position in the virtual space. Giving participants the freedom to manipulate the acoustic location of other participants in the rendered sound scene that they experience has been shown to provide for improved multitasking performance (Wrigley et al., 2009). A system for multiparty teleconferencing requires firstly a stage for recording speech from multiple participants at each site. These signals then need to be compressed to allow for efficient transmission of the spatial speech. One approach is to utilise close-talking microphones to record each participant (e.g. lapel microphones), and then encode each speech signal separately prior to transmission (James & Hawksford, 2008). Alternatively, for increased flexibility, a microphone array located at a central point on, say, a meeting table can be used to generate a multichannel recording of the meeting speech A microphone array approach is adopted in this work and allows for processing of the recordings to identify relative spatial locations of the sources as well as multichannel speech enhancement techniques to improve the quality of recordings in noisy environments. For efficient transmission of the recorded signals, the approach also requires a multichannel compression technique suitable to spatially recorded speech signals

The creation of a binaural spatialization tool

The main focus of the research presented within this thesis is, as the title suggests, binaural spatialization. Binaural technology and, especially, the binaural recording technique are not particu-larly recent. Nevertheless, the interest in this technology has lately become substantial due to the increase in the calculation power of personal computers, which started to allow the complete and accurate real-time simulation of three-dimensional sound-fields over headphones. The goals of this body of research have been determined in order to provide elements of novelty and of contribution to the state of the art in the field of binaural spatialization. A brief summary of these is found in the following list: • The development and implementation of a binaural spatialization technique with Distance Simulation, based on the individual simulation of the distance cues and Binaural Reverb, in turn based on the weighted mix between the signals convolved with the different HRIR and BRIR sets; • The development and implementation of a characterization process for modifying a BRIR set in order to simulate different environments with different characteristics in terms of frequency response and reverb time; • The creation of a real-time and offline binaural spatialization application, imple-menting the techniques cited in the previous points, and including a set of multichannel(and Ambisonics)-to-binaural conversion tools. • The performance of a perceptual evaluation stage to verify the effectiveness, realism, and quality of the techniques developed, and • The application and use of the developed tools within both scientific and artistic “case studies”. In the following chapters, sections, and subsections, the research performed between January 2006 and March 2010 will be described, outlining the different stages before, during, and after the development of the software platform, analysing the results of the perceptual evaluations and drawing conclusions that could, in the future, be considered the starting point for new and innovative research projects

A study on sound source apparent shape and wideness

Proceedings of the 9th International Conference on Auditory Display (ICAD), Boston, MA, July 7-9, 2003.This work is intended as an initial investigation into the perception of wideness and shape of sound sources. A method that employs multiple uncorrelated point sources is used in order to form ``sound shapes''. Several experiments were carried out in which, after some initial training, subjects were asked to indentify the shapes that were being played. Results indicate that differences in vertical and horizontal source wideness are easily perceived and scenes that use broad sound sources to represent normally large sound objects are selected 70% of the time over point source versions. However, shape identification was found to be more ambiguous except for certain types of signals where results were above statistical probability. The work indicates that shape and wideness of sound sources could be effectively used as extra cues in virtual auditory displays and generally improve the realism of virtual 3D sound scenes. This work was performed as a Core Experiment within the MPEG Audio Subgroup with the intention of possible integration of source wideness into MPEG-4 AudioBIFS

On binaural spatialization and the use of GPGPU for audio processing

3D recordings and audio, namely techniques that aim to create the perception of sound sources placed anywhere in 3 dimensional space, are becoming an interesting resource for composers, live performances and augmented reality. This thesis focuses on binaural spatialization techniques. We will tackle the problem from three different perspectives. The first one is related to the implementation of an engine for audio convolution, this is a real implementation problem where we will confront with a number of already available systems trying to achieve better results in terms of performances. General Purpose computing on Graphic Processing Units (GPGPU) is a promising approach to problems where a high parallelization of tasks is desirable. In this thesis the GPGPU approach is applied to both offline and real-time convolution having in mind the spatialization of multiple sound sources which is one of the critical problems in the field. Comparisons between this approach and typical CPU implementations are presented as well as between FFT and time domain approaches. The second aspect is related to the implementation of an augmented reality system having in mind an “off the shelf” system available to most home computers without the need of specialized hardware. A system capable of detecting the position of the listener through a head-tracking system and rendering a 3D audio environment by binaural spatialization is presented. Head tracking is performed through face tracking algorithms that use a standard webcam, and the result is presented over headphones, like in other typical binaural applications. With this system users can choose audio files to play, provide virtual positions for sources in an Euclidean space, and then listen as if they are coming from that position. If users move their head, the signals provided by the system change accordingly in real-time, thus providing the realistic effect of a coherent scene. The last aspect covered by this work is within the field of psychoacoustic, long term research where we are interested in understanding how binaural audio and recordings are perceived and how then auralization systems can be efficiently designed. Considerations with regard to the quality and the realism of such sounds in the context of ASA (Auditory Scene Analysis) are propose

Automatic musical instrument recognition for multimedia indexing

Trabalho apresentado no âmbito do Mestrado em Engenharia Informática, como requisito parcial para obtenção do grau de Mestre em Engenharia InformáticaThe subject of automatic indexing of multimedia has been a target of numerous discussion and study. This interest is due to the exponential growth of multimedia content and the subsequent need to create methods that automatically catalogue this data. To fulfil this idea, several projects and areas of study have emerged. The most relevant of these are the MPEG-7 standard, which defines a standardized system for the representation and automatic extraction of information present in the content, and Music Information Retrieval (MIR), which gathers several paradigms and areas of study relating to music. The main approach to this indexing problem relies on analysing data to obtain and identify descriptors that can help define what we intend to recognize (as, for instance,musical instruments, voice, facial expressions, and so on), this then provides us with information we can use to index the data. This dissertation will focus on audio indexing in music, specifically regarding the recognition of musical instruments from recorded musical notes. Moreover, the developed system and techniques will also be tested for the recognition of ambient sounds (such as the sound of running water, cars driving by, and so on). Our approach will use non-negative matrix factorization to extract features from various types of sounds, these will then be used to train a classification algorithm that will be then capable of identifying new sounds

Audio for Virtual, Augmented and Mixed Realities: Proceedings of ICSA 2019 ; 5th International Conference on Spatial Audio ; September 26th to 28th, 2019, Ilmenau, Germany

The ICSA 2019 focuses on a multidisciplinary bringing together of developers, scientists, users, and content creators of and for spatial audio systems and services. A special focus is on audio for so-called virtual, augmented, and mixed realities. The fields of ICSA 2019 are: - Development and scientific investigation of technical systems and services for spatial audio recording, processing and reproduction / - Creation of content for reproduction via spatial audio systems and services / - Use and application of spatial audio systems and content presentation services / - Media impact of content and spatial audio systems and services from the point of view of media science. The ICSA 2019 is organized by VDT and TU Ilmenau with support of Fraunhofer Institute for Digital Media Technology IDMT