Loudspeaker Positions Research Articles

A raytracing method to create inhomogeneous wave fields for collaborative virtual reality systems

A new rapid, raytracing method was developed and implemented for collaborative virtual reality systems. The goal of the method was to auralize historic acoustic venues in Rensselaer’s Collaborative-Research Augmented Immersive Virtual Environment Laboratory (CRAIVE-Lab). The CRAIVE-Lab provides a physical/digital environment for collaborative tasks using seamless video from multiple projectors and a 128-channel wave-field system with 6 additional ceiling loudspeakers. For the raytracing method, scanned floor plans are marked up by the user to create a two-dimensional geometric room model to calculate the temporal and spatial early-reflection pattern for the horizontal array of 128 loudspeakers resulting in an individual impulse response for each of the 128 loudspeakers. For this purpose, the loudspeaker array is virtually placed within the floor plan to calculate the impulse responses at each loudspeaker position considering the angles of incidence. The user can define wall materials and a directivity pattern for each sound source. The late reverberation tail is generated using a stochastic model based on the volume and surface characteristics of the space. An individual exponentially decaying reverberation tail is computed for each of the 134 loudspeakers with frequency-specific decay times. [Work supported by NSF Grant Nos. #1229391 and CISL.]

Sound Field Reproduction via the Alternating Direction Method of Multipliers Based Lasso Plus Regularized Least-Square

This paper proposes a 3-D sound field reproduction (SFR) approach through the combination of alternating direction method of multipliers (ADMM)-based least-absolute shrinkage and selection operator (Lasso) and regularized least square (LS). The proposed SFR method is split into two parts through the pressure matching optimization of loudspeaker positions and the computation of driving signals. At the first part, a plurality of candidate positions of loudspeakers in planar array is given and, then, the active speaker selection method is proposed based on ADMM complex Lasso algorithm for selecting the optimal loudspeaker positions. Afterwards, regularized LS is adopted to calculate the selected loudspeaker weights and control the total power. The numerical simulation experiments demonstrate that the proposed SFR scheme outperforms the existing sparse loudspeakers’ placement and weight optimization algorithms especially in under-sampled sound fields. Meanwhile, the evaluations also confirmed that the proposed method could significantly reduce the computational complexity of the active loudspeaker selection compared to the state-of-the-art Lasso-based SFR. Effectively, the proposed method uses a relatively small number of loudspeakers for a satisfying reproduction quality.

A reverberation-time-aware DNN approach leveraging spatial information for microphone array dereverberation

A reverberation-time-aware deep-neural-network (DNN)-based multi-channel speech dereverberation framework is proposed to handle a wide range of reverberation times (RT60s). There are three key steps in designing a robust system. First, to accomplish simultaneous speech dereverberation and beamforming, we propose a framework, namely DNNSpatial, by selectively concatenating log-power spectral (LPS) input features of reverberant speech from multiple microphones in an array and map them into the expected output LPS features of anechoic reference speech based on a single deep neural network (DNN). Next, the temporal auto-correlation function of received signals at different RT60s is investigated to show that RT60-dependent temporal-spatial contexts in feature selection are needed in the DNNSpatial training stage in order to optimize the system performance in diverse reverberant environments. Finally, the RT60 is estimated to select the proper temporal and spatial contexts before feeding the log-power spectrum features to the trained DNNs for speech dereverberation. The experimental evidence gathered in this study indicates that the proposed framework outperforms the state-of-the-art signal processing dereverberation algorithm weighted prediction error (WPE) and conventional DNNSpatial systems without taking the reverberation time into account, even for extremely weak and severe reverberant conditions. The proposed technique generalizes well to unseen room size, array geometry and loudspeaker position, and is robust to reverberation time estimation error.

In situ Probe Microphone Measurement for Testing the Direct Acoustical Cochlear Stimulator.

Hypothesis: Acoustical measurements can be used for functional control of a direct acoustic cochlear stimulator (DACS).Background: The DACS is a recently released active hearing implant that works on the principle of a conventional piston prosthesis driven by the rod of an electromagnetic actuator. An inherent part of the DACS actuator is a thin titanium diaphragm that allows for movement of the stimulation rod while hermetically sealing the housing. In addition to mechanical stimulation, the actuator emits sound into the mastoid cavity because of the motion of the diaphragm.Methods: We investigated the use of the sound emission of a DACS for intra-operative testing. We measured sound emission in the external auditory canal (PEAC) and velocity of the actuators stimulation rod (Vact) in five implanted ears of whole-head specimens. We tested the influence various positions of the loudspeaker and a probe microphone on PEAC and simulated implant malfunction in one example.Results: Sound emission of the DACS with a signal-to-noise ratio >10 dB was observed between 0.5 and 5 kHz. Simulated implant misplacement or malfunction could be detected by the absence or shift in the characteristic resonance frequency of the actuator. PEAC changed by <6 dB for variations of the microphone and loudspeaker position.Conclusion: Our data support the feasibility of acoustical measurements for in situ testing of the DACS implant in the mastoid cavity as well as for post-operative monitoring of actuator function.

A Reverberation-Time-Aware Approach to Speech Dereverberation Based on Deep Neural Networks

A reverberation-time-aware deep-neural-network DNN-based speech dereverberation framework is proposed to handle a wide range of reverberation times. There are three key steps in designing a robust system. First, in contrast to sigmoid activation and min-max normalization in state-of-the-art algorithms, a linear activation function at the output layer and global mean-variance normalization of target features are adopted to learn the complicated nonlinear mapping function from reverberant to anechoic speech and to improve the restoration of the low-frequency and intermediate-frequency contents. Next, two key design parameters, namely, frame shift size in speech framing and acoustic context window size at the DNN input, are investigated to show that RT60-dependent parameters are needed in the DNN training stage in order to optimize the system performance in diverse reverberant environments. Finally, the reverberation time is estimated to select the proper frame shift and context window sizes for feature extraction before feeding the log-power spectrum features to the trained DNNs for speech dereverberation. Our experimental results indicate that the proposed framework outperforms the conventional DNNs without taking the reverberation time into account, while achieving a performance only slightly worse than the oracle cases with known reverberation times even for extremely weak and severe reverberant conditions. It also generalizes well to unseen room sizes, loudspeaker and microphone positions, and recorded room impulse responses.

Evaluating acoustical features of 3D sound rendering systems by using a visualizing tool of sound intensities

The authors have developed a simple measurement/analyzing tool which visualizes 3D acoustical properties of sound by using sound intensity information. The tool, VSV (Virtual Source Visualizer), has already being used for practical acoustic design works, and the authors have also begun to use it for evaluating acoustical features of 3D sound rendering systems. On the audio industries, various 3D sound reproduction systems have already been commercialized. Therefore, further skill is now required to design acoustic specifications of their production studios. From this point of view, development of a simple method to obtain relationship between loudspeaker positions and source locations rendered by a 3D reproduction system would be helpful. This research introduces analyzing flows of the measurement tool briefly, and shows measured results of physical and/or phantom sound locations detected from sound intensity information in some different 3D reproduction environments including commercial audio strategies such as Dolby Atmos and 22.2ch, and a wave theory based Boundary Surface Control system, BoSC. Verifying differences between target source positions and measured ones, the authors suggest the feasibility of the evaluation method of a rendered 3D sound field by a simple measurement method.

Methods of field measurements of facade sound insulation

The purpose of this article is to describe generally usable methods to measure the sound insulation of a facade in cases where a setup according to standards such as ISO 140/5 or ISO 16283/3 cannot be achieved. The methods discussed use a loudspeaker as the source of a test signal since background noise on the interior receiving side in many cases prevents the use of traffic as a noise source. The measurements presented in this article have been made using two loudspeaker positions at different angles. Three different ways of placing the microphone are considered. The microphone can be placed in fixed points directly on the facade, in fixed points at the same distance in front of the facade, or swept in front of the facade. The data presented are all taken from actual field measurements. The results are shown as a comparison between the levels acquired and the uncertainty associated with the different methods. (C) 2015 Institute of Noise Control Engineering. (Less)

Coded excitation in space and time for the identification of geometric parameters using audio transducer arrays

Impulse responses of a room can provide a lot of geometric information of acoustic systems. Loudspeaker and microphone array positions, locations of walls and scattering bodies are representative examples. To extract geometric information from the room impulse responses, the measurement time should be short enough such that the degradation of time delay estimation performance due to non-stationary environmental noises can be minimized. In this work, we attempt to enhance the SNR of the short-term measurement, by using a coded excitation technique in space and time. Both the spatially and temporally coded excitation signals are applied to a set of loudspeakers combined with microphones, and it is shown that the measurement time can be shortened with minimal loss of SNR and time delay estimation performance.

Front-back confusions when sources and listeners rotate

Tonal stimuli presented from directly in-front or directly in-back of a stationary listener are often miss-located in that sounds presented from in-front are occasionally perceived as being in-back of the listener; and often sounds presented in-back of a listener are perceived as being in-front. These types of front-back/back-front confusions in the azimuth plane were investigated as a function of the type of stimuli (e.g., tones, speech, and wideband noise) used and as a function of the sound or listener rotating around an azimuthal 24-loudspeaker array. Front-back confusions/back-front confusions depended on the type of stimulus; and on the number of loudspeaker positions the sound rotated through and the rate of sound or listener rotation. For example, if the sound rotated through two loudspeakers from in-front to in-back, there were a large number of confusions; if the sound rotated through each of the 24 loudspeakers on the circular array there were almost no confusions. [Research supported by an AFOSR grant.]

Parametric uncertainty quantification of sound insulation values.

A probabilistic framework is developed for quantifying the combined effect of uncertain parameters in sound insulation measurements, such as test sample dimensions, room properties, and loudspeaker positions, on the sound insulation values. The joint probability distribution of the uncertain parameters is constructed from the available information by means of a maximum entropy approach. The resulting sound insulation predictions are fully compatible with the available information but otherwise maximally conservative, so that the robustness of the predictions is guaranteed. Fundamental insight in the inherent uncertainty of the measurement procedure for airborne sound insulation is obtained by combining the method with detailed numerical simulations of the measurement procedure for single and double walls. The resulting uncertainty levels are very large, especially in the lowest frequency bands, and agree with experimental results. Furthermore, the probability distribution of the band-averaged sound reduction index of modally sparse walls can be of bimodal form.

Significance of height loudspeaker positioning for perceived immersive sound field reproduction

Recently, new multichannel audio formats incorporating height loudspeakers have caught researchers' attention due to their ability to reproduce an immersive sound field. This study investigated the influence of the height loudspeaker positions and their signals on individually perceived sound quality. The authors generated nine-channel sound sources by convolving two anechoic musical pieces with nine selected room impulse responses measured in different distances and heights. In the listening test two layers of loudspeakers were used: the horizontal layer, following the standard ITU-R BS 775 five-channel loudspeaker configuration and the height layer (with elevation of 30°) with a total of twelve loudspeakers, located at ±30, ±50, ±70, ±90, ±110, and ±130 degrees. Four height signals were reproduced through eight different configurations of four height loudspeakers. Twelve listeners participated in the experiment, wherein they were asked to compare the randomly presented eight configurations and rank them based on their individually perceived sound quality. The experimental results indicate that despite the perceptual differences related to the room impulse responses, the perceived overall quality is significantly influenced by the positioning of the four height loudspeakers.

Active field control using sound field generation technique—Case study of a Live concert at a virtual Renaissance church

In the Renaissance area, musical culture was centered at and spread across churches. In order to appreciate Renaissance music, therefore, it is important to account for the influence of acoustics of churches at that era so that audiences today can experience homogeneous musical appreciation. We used an Active Field Control system to create the acoustics using measured the impulse responses (IRs) of a church. The system consists of directional microphones, head amps, a convolution engine, a matrix processor, amplifiers, and loudspeakers. And it picks up the direct response of performance, convolves it with the measured IRs, and reproduces the resulting sound using loudspeakers around the room. Loudspeaker positions in the performance hall are equivalent to the positions where the IRs had been measured at the church. This technique allows us to convincingly recreate not only the reverberation time but also the spatial impressions of the church at the performance hall. In addition, we modified the IRs so that the inherent acoustical characters of the performance space would less influence to recreation of the target church acoustics. We used a 6-channel recording/reproduction system to evaluate the modification to the IRs and the impressions of the recreated acoustics in the room. This paper presents the results of the experiment.

Evaluation of different spatial windows for a multi-channel audio interpolation system

The adoption rate of multi-channel audio systems has dramatically increased in recent years. It is common to find 5.1- or 7.1-channel systems in typical home theaters. However, most users do not setup the satellite loudspeakers at the prescribed positions for aesthetic reasons or due to space constraints. Recently, we introduced a technique to optimize multi-channel contents for reproduction over non-ideal loudspeaker setups. Our proposal improves localization accuracy for sound sources reproduced by horizontal loudspeaker arrays. It can also be extended to handle full 3D contents, like those of the upcoming 22.2-channel standard. The proposed method works by applying a set of spatial windows centered at the loudspeaker positions and interpolating along the angles using the spherical harmonics. We now extend our previous results by evaluating the performance of six different spatial window functions. We consider the 5-channel distribution of ITU recommendation BS.775-2, as well as four variations that end-users are likely to deploy. Apparent sound source locations are estimated from the energy and velocity vectors at the sweet spot. Our study found that using the Slepian window with our proposal and a non-ideal loudspeaker layout leads to a reproduced sound field that is closer to that of the ideal configuration.

Large-scale sound field rendering with graphics processing unit cluster for three-dimensional audio with loudspeaker array

The sound field rendering is a technique to simulate the sound field from the three-dimensional numerical models constructed in the computer, and it is the same concept as the graphics rendering in the computer graphics. In this paper, a graphics processing unit (GPU) cluster system is applied to the sound field rendering for a large room simulation. The compact explicit finite difference time domain (CE-FDTD) method is implemented on the GPU cluster system. The CE-FDTD method is a kind of the finite difference method in which the wave equation is directly discretized based on the central differences. The developed GPU cluster system consists of eight PC nodes in which four GPUs are mounted respectively. The rendering results are reproduced by a speaker array system in which 157 speakers surround a small room. The sound field renderings are performed for a large room with a volume capacity of about 5000 cubic meters, in which the impulse responses of one-second length with a sampling rate of 40 kHz are calculated at 157 points corresponding to the loudspeaker positions. The impulse responses are then convoluted with dry music sources. The sound field rendering with the 157ch loudspeaker array system provides the realistic sound field reproduction with natural reverberation.

Active Acoustic Systems for the Control of Room Acoustics

The acoustic design of auditoria involves the specification of the room geometry and boundary properties, and any additional acoustic elements such as reflectors or diffusers, to usefully direct sound to produce a desired subjective experience, quantified by measurable acoustic parameters. This design must take into account the reflection of sound within the stage area, the early reflections from the stage to the audience and the reverberant response of the room. The sound produced by the audience can also be an important consideration. Active acoustic systems provide an alternative approach to controlling subjective experience. They use microphones, electronic processors and loudspeakers to create reflections and reverberation in addition to those produced by the naturally-occurring sound field. The acoustic properties can be changed instantly, and the enhanced acoustic properties of the auditorium can typically be varied over a wider range than can be produced by variable passive techniques. The design of active acoustics follows that of passive approaches, but rather than the physical arrangement of the room surfaces, it commences with an existing passive space with some minimum acoustic condition, and requires the arrangement of microphones to detect relevant sound and the choice of processors and loudspeaker positions to direct it usefully back into the room to produce a desired set of acoustic parameters. While active systems have historically been developed with the goal of enhancing either the stage or audience sound, they must generally provide the same control of sound as passive acoustic design. This paper discusses the principles of active acoustic systems and how they are used to achieve the required range of control. A survey of current commercial systems is given and some implications for the future of live performance are explored.

APPARATUS AND METHOD FOR GENERATING A NUMBER OF LOUDSPEAKER SIGNALS FOR A LOUDSPEAKER ARRAY WHICH DEFINES A REPRODUCTION SPACE

An apparatus for generating a number of loudspeaker signals for a loudspeaker array defining a reproduction space includes a prestage configured to generate a plurality of output audio signals while using one or more audio signals associated with one or more virtual positions, each output audio signal being associated to a loudspeaker position such that the plurality of output audio signals together replicate a reproduction of the input audio signal(s) at the virtual position(s), and a number of output audio signals being smaller than a number of loudspeaker signals. The apparatus further includes a main stage configured to obtain the plurality of output audio signals and further to obtain, as a virtual position for each output audio signal, the loudspeaker positions, and to generate the number of loudspeaker signals for the loudspeaker array such that the loudspeaker positions are replicated as a virtual sources by the loudspeaker array.

Method for determining the loudspeaker positions for the implementation of crosstalk cancellers robust to head rotation

Method for determining the loudspeaker positions for the implementation of crosstalk cancellers robust to head rotation

Development of Auditory Localization Accuracy and Auditory Spatial Discrimination in Children and Adolescents

The present study investigated the development of two parameters of spatial acoustic perception in children and adolescents with normal hearing, aged 6–18 years. Auditory localization accuracy was quantified by means of a sound source identification task and auditory spatial discrimination acuity by measuring minimum audible angles (MAA). Both low- and high-frequency noise bursts were employed in the tests, thereby separately addressing auditory processing based on interaural time and intensity differences. Setup consisted of 47 loudspeakers mounted in the frontal azimuthal hemifield, ranging from 90° left to 90° right (–90°, +90°). Target signals were presented from 8 loudspeaker positions in the left and right hemifields (±4°, ±30°, ±60° and ±90°). Localization accuracy and spatial discrimination acuity showed different developmental courses. Localization accuracy remained stable from the age of 6 onwards. In contrast, MAA thresholds and interindividual variability of spatial discrimination decreased significantly with increasing age. Across all age groups, localization was most accurate and MAA thresholds were lower for frontal than for lateral sound sources, and for low-frequency compared to high-frequency noise bursts. The study also shows better performance in spatial hearing based on interaural time differences rather than on intensity differences throughout development. These findings confirm that specific aspects of central auditory processing show continuous development during childhood up to adolescence.

Procedure for verification of sound source coverage over façades according to the International Standard ISO 140–5

This paper presents a new verification procedure for sound source coverage according to ISO 140–5 requirements. The ISO 140–5 standard applies to the measurement of façade insulation and requires a sound source able to achieve a sufficiently uniform sound field in free field conditions on the façade under study. The proposed method involves the electroacoustic characterisation of the sound source in laboratory free field conditions (anechoic room) and the subsequent prediction by computer simulation of the sound free field radiated on a rectangular surface equal in size to the façade being measured. The loudspeaker is characterised in an anechoic room under laboratory controlled conditions, carefully measuring directivity, and then a computer model is designed to calculate the acoustic free field coverage for different loudspeaker positions and façade sizes. For each sound source position, the method provides the maximum direct acoustic level differences on a façade specimen and therefore determines whether the loudspeaker verifies the maximum allowed level difference of 5dB (or 10dB for façade dimensions greater than 5m) required by the ISO standard. Additionally, the maximum horizontal dimension of the façade meeting the standard is calculated and provided for each sound source position, both with the 5dB and 10dB criteria. In the last section of the paper, the proposed procedure is compared with another method used by the authors in the past to achieve the same purpose: in situ outdoor measurements attempting to recreate free field conditions. From this comparison, it is concluded that the proposed method is able to reproduce the actual measurements with high accuracy, for example, the ground reflection effect, at least at low frequencies, which is difficult to avoid in the outdoor measurement method, and it is fully eliminated with the proposed method to achieve the free field requisite.

Performance of optimized sound field control techniques in simulated and real acoustic environments

It is of interest to create regions of increased and reduced sound pressure ('sound zones') in an enclosure such that different audio programs can be simultaneously delivered over loudspeakers, thus allowing listeners sharing a space to receive independent audio without physical barriers or headphones. Where previous comparisons of sound zoning techniques exist, they have been conducted under favorable acoustic conditions, utilizing simulations based on theoretical transfer functions or anechoic measurements. Outside of these highly specified and controlled environments, real-world factors including reflections, measurement errors, matrix conditioning and practical filter design degrade the realizable performance. This study compares the performance of sound zoning techniques when applied to create two sound zones in simulated and real acoustic environments. In order to compare multiple methods in a common framework without unduly hindering performance, an optimization procedure for each method is first used to select the best loudspeaker positions in terms of robustness, efficiency and the acoustic contrast deliverable to both zones. The characteristics of each control technique are then studied, noting the contrast and the impact of acoustic conditions on performance.