Robust Personal Audio Geometry Optimization in the SVD-Based Modal Domain

© 2014 IEEE. Personal audio generates sound zones in a shared space to provide private and personalized listening experiences with minimized interference between consumers. Regularization has been commonly used to increase the robustness of such systems against potential perturbations in the sound reproduction. However, the performance is limited by the system geometry such as the number and location of the loudspeakers and controlled zones. This paper proposes a geometry optimization method to find the most geometrically robust approach for personal audio amongst all available candidate system placements. The proposed method aims to approach the most 'natural' sound reproduction so that the solo control of the listening zone coincidently accompanies the preferred quiet zone. Being formulated in the SVD-based modal domain, the method is demonstrated by applications in three typical personal audio optimizations, i.e., the acoustic contrast control, the pressure matching, and the planarity control. Simulation results show that the proposed method can obtain the system geometry with better avoidance of 'occlusion,' improved robustness to regularization, and improved broadband equalization

Optimizing Source and Sensor Placement for Sound Field Control: An Overview

International audienceIn order to control an acoustic field inside a target region, it is important to choose suitable positions of secondary sources (loudspeakers) and sensors (control points/microphones). This paper provides an overview of state-of-the-art source and sensor placement methods in sound field control. Although the placement of both sources and sensors greatly affects control accuracy and filter stability, their joint optimization has not been thoroughly investigated in the acoustics literature. In this context, we reformulate five general source and/or sensor placement methods that can be applied for sound field control. We compare the performance of these methods through extensive numerical simulations in both narrowband and broadband scenarios. Index Terms-source and sensor placement, sound field control , sound field reproduction, subset selection, interpolation

2D to 3D ambience upmixing based on perceptual band allocation

3D multichannel audio systems employ additional elevated loudspeakers in order to provide listeners with a vertical dimension to their auditory experience. Listening tests were conducted to evaluate the feasibility of a novel vertical upmixing technique called “perceptual band allocation (PBA),” which is based on a psychoacoustic principle of vertical sound localization, the “pitch height” effect. The practical feasibility of the method was investigated using 4-channel ambience signals recorded in a reverberant concert hall using the Hamasaki-Square microphone technique. Results showed that the PBA-upmixed 3D stimuli were significantly stronger than or similar to 9-channel 3D stimuli in 3D listener-envelopment (LEV), depending on the sound source and the crossover frequency of PBA. They also significantly produced greater 3D LEV than the 7-channel 3D stimuli. For the preference tests, the PBA stimuli were significantly preferred over the original 9-channel stimuli

Perceptual Optimization of Room-In-Room Reproduction with Spatially Distributed Loudspeakers

It is often desirable to reproduce a specific room-acoustic scene, e.g. a concert hall in a playback room, in such a way that the listener has a plausible and authentic spatial impression of the original sound source including the room acoustical properties. In this study a perceptually motivated approach for spatial audio reproduction is developed. This approach optimizes the spatial and monaural cues of the direct and reverberant sound separately. More specifically, the (monaural) spectral cues responsible for the timbre and the (binaural) interaural cross correlation (IACC) cues, responsible for the listener envelopment, were optimized in the playback room to restore the auditory impression of the recording room. The direct sound recorded close to the source is processed with an auditory motivated gammatone filterbank such that the spectral cues, ITD’s and ILD’s are comparable to the direct sound in the recording room. Additionally, the reverberant sound, which was recorded at two distant locations from the source, is played back via dipole loudspeakers. Due to the arrangement of the two dipole loudspeakers, only the diffuse sound field in the playback room is excited, therefore the spectral cues and the IACC of the reverberant sound field can be controlled independently to match the cues that were present in the recording room. As indicated by a preliminary listening test the applied optimization is perceptually similar to the reference signal and is generally preferred when compared to a conventional room-in-room reproduction.DFG, FOR 1732, Individualisierte Hörakustik: Modelle, Algorithmen und Systeme für die Sicherstellung der akustischen Wahrnehmung für alle in allen Situatione

Microphone and Loudspeaker Array Signal Processing Steps towards a “Radiation Keyboard” for Authentic Samplers

To date electric pianos and samplers tend to concentrate on authenticity in terms of temporal and spectral aspects of sound. However, they barely recreate the original sound radiation characteristics, which contribute to the perception of width and depth, vividness and voice separation, especially for instrumentalists, who are located near the instrument. To achieve this, a number of sound field measurement and synthesis techniques need to be applied and adequately combined. In this paper we present the theoretic foundation to combine so far isolated and fragmented sound field analysis and synthesis methods to realize a radiation keyboard, an electric harpsichord that approximates the sound of a real harpsichord precisely in time, frequency, and space domain. Potential applications for such a radiation keyboard are conservation of historic musical instruments, music performance, and psychoacoustic measurements for instrument and synthesizer building and for studies of music perception, cognition, and embodiment

Synthesis of Soundfields through Irregular Loudspeaker Arrays Based on Convolutional Neural Networks

Most soundfield synthesis approaches deal with extensive and regular loudspeaker arrays, which are often not suitable for home audio systems, due to physical space constraints. In this article we propose a technique for soundfield synthesis through more easily deployable irregular loudspeaker arrays, i.e. where the spacing between loudspeakers is not constant, based on deep learning. The input are the driving signals obtained through a plane wave decomposition-based technique. While the considered driving signals are able to correctly reproduce the soundfield with a regular array, they show degraded performances when using irregular setups. Through a Convolutional Neural Network (CNN) we modify the driving signals in order to compensate the errors in the reproduction of the desired soundfield. Since no ground-truth driving signals are available for the compensated ones, we train the model by calculating the loss between the desired soundfield at a number of control points and the one obtained through the driving signals estimated by the network. Numerical results show better reproduction accuracy both with respect to the plane wave decomposition-based technique and the pressure-matching approach