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A multimodal interactive system for audio-haptic integration is presented in this paper. Preliminary subjective tests with a virtual reality setup were conducted with the goal of interpreting cognitive mechanisms and improving performances in orientation & mobility protocols for visually impaired subjects, where spatial representations need to be developed using residual sensory channels. An object recognition experiment was performed in order to investigate the contribution of dynamic spatial audio cues when integrated with haptic feedback. Audio cues took the form of anchor sound delivered through headphones using customized Head-related Transfer Functions (HRTFs). This setup was employed in the exploration of simplified virtual audio-tactile environments. Overall results on recognition time reveal a relationship between anchor position and object shape. Moreover, a qualitative analysis of the exploration paths highlights behavioral changes between unimodal and multimodal conditions.
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Multichannel recordings are usually performed by means of microphone arrays. In many cases "sparse" and discrete microphone arrays are used, where each microphone is employed for capturing one of the channels, which in turn is routed to one loudspeaker. However, also the usage of "dense" microphone arrays has a long history, dating back to the first MS-matrixed microphones setups and passing through the whole Ambisonics saga. A dense microphone array is employed differently from a sparse array: each channel is obtained by a combination of the signals coming from all the capsules, by means of different matrixing and filtering approaches. And similarly, each loudspeaker feed results from a re-matrixing of all the transmitted channels. This paper is the third of a series: in the previous two [1,2] a numerical method for computing a matrix of FIR filter was employed for processing the microphone signals (encoding, [1]) and for computing the speaker feeds (decoding, [2]). In this third paper, the same numerical approach is extended to intermediate processing (rotation, zooming, stretching, spatial equalization, etc.): hence we have now a general meta-theory, providing a unique framework capable of processing the signals for any kind of dense microphone array, providing any kind of intermediate manipulation, and finally projecting the signal to every kind of loudspeaker arrays. The same framework can operate according to different standards and formats, including A-format (raw signals), B-format (High Order Ambisonics signals), G-format (speaker feeds) and P-format (Spatial PCM Sampling signals), and can be used for converting freely among them. Experimental results are presented, including "traditional" tetrahedral probes, a commercial spherical microphone array, and two newly-developed massive microphone arrays developed by the authors, a cylindrical and a planar array, both incorporating 32 high-quality condenser microphones and a panoramic video camera.
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This work proposes a scheme for improving binaural reproduction of 5.1 channel surround sound using dynamic rendering with low cost head-tracking module. The head-tracking module consists of a magnetic sensor and an accelerometer. A microcontroller unit is used to calculate the horizontal orientation of the listener's head. The orientation data is transmitted to the PC via the serial port for dynamic binaural synthesis. Psychoacoustic experiment indicates that the scheme eliminates the front-back confusion and in-head-localization and therefore improves the localization performance in the binaural 5.1 channel reproduction.
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This study investigates the audio-visual spatial congruence in distance dimension in the case of virtual environments. A perceptual study is conducted in which visual targets are presented using a passive 3D projector, and accompanying virtual sounds are simulated by Wave Field Synthesis. Audio-visual stimuli are spatially congruent in azimuth and elevation, but a discrepancy is introduced in the distance dimension. For each tested visual distance (2, 3, 4 and 5 m), three audio stimuli are presented at 10 distances (1, 1.5, 2, 2.5, 3, 3.5, 4, 5, 7, and 10 m). After each presentation, participants have to estimate whether or not the sound is at the same place as the visual object. Results show that there is a significant range of audio conditions which are perceived as spatially congruent for each tested visual distance. Moreover, these integration windows tend to increase as the distance of visual stimuli increases. Finally, no significant effect of the stimulus is found in this study even though slight differences are locally observed.
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Subjective comparisons of room acoustics require high-fidelity auralizations. Earlier research has shown that room impulse response measurements and directional analysis with the Spatial decomposition method provides accurate reproductions of concert hall acoustics in multi-channel listening. Moreover, timbral aspects have been found important for the overall audio quality, even more than spatial fidelity. This paper explores the effect by the number of true and virtual loudspeakers, and application of amplitude panning, to brightness of the sound. Results from a listening test with concert hall auralizations suggest that amplitude panning reduces the perceived brightness in both loudspeaker and headphone listening.
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In this paper, the reproduction of industrial or vehicle sound environments is investigated. The method relies on the combination of microphone array recording, beamforming, Wave Field Synthesis (WFS), and bass equalization. The approach is based on fixed looking-direction beamformers to separate signals depending on their incoming direction. The beamformer signals are then processed by WFS in order to recreate the sound environment as multiple plane waves. A theoretical parametric study based on the number of beamformer steering directions is provided. Comparisons of the target and reproduced sound environments are reported in order to identify appropriate system parameters for the reproduction of sound environments as found in vehicle or industrial contexts.
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This paper studies the analysis of spatial room impulse responses recorded with a microphone array, at a single frequency, and assuming more than one plane wave per analysis window. A parametric method, weighted subspace fitting, developed in array processing some 20 years ago, is applied to the analysis of the spatial room impulse responses. The previous approaches for spatial room impulse response analysis are known to have limitations since they assume only plane wave per time window. This paper shows by room acoustic simulations that assuming more than one plane wave improves the analysis, which will benefit the synthesis of spatial room impulse responses in sound reproduction techniques.
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MPEG Surround provides an efficient representation of multi-channel audio signals by a set of down-mixed signals and spatial parameters. These parameters describe the inter-channel spatial properties, which are utilized by the decoder to reconstruct the multi-channel signals from the down-mixed signals. The encoding process includes time/frequency analysis-synthesis filtering and signal down-mixing; hence the computational requirement is highly dependent on the number of surround audio playback channels. Meanwhile, coincident microphone techniques offer compact microphone array geometry and low number of microphones to produce surround sound recording. Usually, the recorded signals are post-processed into multi-channel surround sound signals prior to encoding. In this paper, an MPEG Surround encoding scheme for coincident microphone recording is presented. The main feature is the direct derivation of the spatial parameters and down-mixed signals from the microphone array signals. It minimizes the dependency of the computational demand on the number of surround sound audio channels. Implemented on ARM-based embedded platform, the direct encoding scheme requires approximately 50% less memory and 40% less processing power than the reference encoding scheme.
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Numerical simulations offer a viable alternative to measurements for generating personalized head-related transfer functions (HRTFs). The fast multipole boundary element method (FM-BEM) is a popular method for simulating the HRTFs since it requires a surface mesh of the head (and torso) only. The FM-BEM simulation of the HRTF at a single frequency can be computed in a few minutes. Utilizing cloud computing, the entire audible frequency range can be simulated in less than an hour. A bottleneck in the fast acquisition of the personalized HRTFs has been the complexity of generating good quality head models for the simulation. We compare three photography based geometry acquisition methods, ranging from a system of 52 cameras to a method using a single mobile phone camera only.
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Reverberation is considered as one of the fundamental perceived properties of an acoustical space. Literature is available on the topic and currently a range of sciences have contributed in understanding the properties of reverberant sound fields and the relevant auditory processes. This paper summarises the current literature following a top-down approach. It identifies the perceptual aspects of reverberation and attempts to establish links to physical measures, focussing on small rooms. Results indicate that the current acoustical metrics often have limited correlation to the perceptual attributes of reverberation and conclusive measurement data is restricted, especially for small spaces. A proposal for perceptual-based experiments is presented, aiming to further understand the links between physical properties of rooms and their effects on perception.
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