Loudspeaker Positions Research Articles

Energy-Preserving Ambisonic Decoding

Ambisonics with height is a three-dimensional sound field reproduction technique for spherical loudspeaker arrangements surrounding the reproduction area. It employs spherical harmonics up to a given order to expand incident sound fields with a limited angular resolution. The expansion coefficients describe the spatial sound scene. For reproduction, these coefficients are decoded to a set of surrounding loudspeakers. Common decoding approaches either sample the spherical harmonic excitation at the loudspeaker positions or match the excitation modes of a continuous sound field to those of the loudspeakers. For well-designed spherical loudspeaker arrays, both decoding approaches achieve good perceptual localization of virtual sound sources. However, both approaches perform unsatisfactorily with non-uniformly arranged arrays. Sounds from directions with only sparse loudspeaker coverage appear with altered loudness levels. This distracting effect results from variations in the decoded energy. The present article demonstrates an improved decoding technique, which preserves the decoded energy. Using available objective estimators, the localization qualities of these energy-preserving decoders are shown to lie between both common decoding approaches.

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.

Conversion of Multichannel Sound Signal Maintaining Physical Properties of Sound in Reproduced Sound Field

In this paper, we describe a new method for converting the signal of the original multichannel sound system into that of an alternative system with a different number of channels while maintaining the physical properties of sound at the listening point in the reproduced sound field. Such a conversion problem can be described by the underdetermined linear equation. To obtain an analytical solution to the equation, the method partitions the sound field of the alternative system on the basis of the positions of three loudspeakers and solves the “local solution” in each subfield. As a result, the alternative system localizes each channel signal of the original sound system at the corresponding loudspeaker position as a phantom source. The composition of the local solutions introduces the “global solution,” that is, the analytical solution to the conversion problem. 22-channel signals of a 22.2 multichannel sound system without the two low-frequency effect channels were converted into 10-, 8-, and 6-channel signals by the method. Subjective evaluations showed that the proposed method could reproduce the spatial impression of the original 22-channel sound with eight loudspeakers.

Robust active noise control in the loadmaster area of a military transport aircraft

The active noise control (ANC) method is based on the superposition of a disturbance noise field with a second anti-noise field using loudspeakers and error microphones. This method can be used to reduce the noise level inside the cabin of a propeller aircraft. However, during the design process of the ANC system, extensive measurements of transfer functions are necessary to optimize the loudspeaker and microphone positions. Sometimes, the transducer positions have to be tailored according to the optimization results to achieve a sufficient noise reduction. The purpose of this paper is to introduce a controller design method for such narrow band ANC systems. The method can be seen as an extension of common transducer placement optimization procedures. In the presented method, individual weighting parameters for the loudspeakers and microphones are used. With this procedure, the tailoring of the transducer positions is replaced by adjustment of controller parameters. Moreover, the ANC system will be robust because of the fact that the uncertainties are considered during the optimization of the controller parameters. The paper describes the necessary theoretic background for the method and demonstrates the efficiency in an acoustical mock-up of a military transport aircraft.

Letter to the Editor: Critical Distance in Churches: Comments on Carvalho & Lencastre (2000)

The goal of this letter is to comment on the article “Catholic Churches, Sound Reinforcement Systems and RASTI” by Carvalho & Lencastre (2000). Carvalho & Lencastre could have calculated or measured critical distance in each church to know the reason of decreasing RASTI in the direct field. It is concluded that in spaces with long reverberation time, the number and position of loudspeakers with regard to critical distance must be taken into consideration.

A binaural model to evaluate surround sound reproduction systems using head movements.

Recently, a binaural model was proposed, which utilized head movements to resolve front-back confusions [Braasch et al., J. Acoust. Soc. Am. 127, 1886]. Meanwhile, the binaural model has been extended to evaluate the benefit of surround loudspeakers for the reproduction of diffuse sound. For this purpose, two to four loudspeaker locations were simulated using head-related transfer functions (HRTFs). To simulate head movements, the HRTFs were adjusted in steps of 1 deg to maintain the loudspeaker positions in a room-related coordinate system. Uncorrelated white-noise bursts (1-s duration) were fed to each loudspeaker channel. The superposed spatialized signals were processed through a gammatone filter bank to measure coherence and interaural-level (ILDs) differences in individual frequency bands. Various loudspeaker configurations were tested. The study shows that coherence and ILDs for diffuse sound vary less with head movements if surround speakers (e.g., ±120 deg) are used in addition to a pair of front speakers (e.g., ±30 deg). For example, if the head is turned between ±40 deg, the maximum coherence drops from 0.58 to 0.39 if surround loudspeakers are added (600-Hz frequency band). At 5000 Hz, the maximum ILD magnitude is reduced from 10 to 4 dB with surround loudspeakers.

Measurement of confined acoustic sources using near-field acoustic holography

Due to excessive reverberation or to the presence of secondary noise sources, characterization of sound sources in enclosed space is rather difficult to perform. In this paper a process layer is used to recover the pressure field that the studied source would have radiated in free space. This technique requires the knowledge of both acoustic pressure and velocity fields on a closed surface surrounding the source. The calculation makes use of boundary element method and is performed in two steps. First, the outgoing pressure field is extracted from the measured data using a separation technique. Second, the incoming field then scattered by the tested source body is subtracted from the outgoing field to recover free field conditions. The studied source is a rectangular parallelepiped with seven mid-range loudspeakers mounted on it. It stands at 40 cm from the rigid ground of a semi-anechoic chamber which strongly modifies the radiated pressure field, especially on the underside. After the measured data have been processed, the loudspeaker positions are recovered with a fairly good accuracy. The acoustic inverse problem is also solved to calculate the velocity field on the source surface.

Theory and Design of Soundfield Reproduction Using Continuous Loudspeaker Concept

Reproduction of a soundfield is a fundamental problem in acoustic signal processing. A common approach is to use an array of loudspeakers to reproduce the desired field where the least-squares method is used to calculate the loudspeaker weights. However, the least-squares method involves matrix inversion which may lead to errors if the matrix is poorly conditioned. In this paper, we use the concept of theoretical continuous loudspeaker on a circle to derive the discrete loudspeaker aperture functions by avoiding matrix inversion. In addition, the aperture function obtained through continuous loudspeaker method reveals the underlying structure of the solution as a function of the desired soundfield, the loudspeaker positions, and the frequency. This concept can also be applied for the 3-D soundfield reproduction using spherical harmonics analysis with a spherical array. Results are verified through computer simulations.

System for configuring audio system

A system is provided for configuring an audio system for a given space. The system may statistically analyze potential configurations of the audio system to configure the audio system. The potential configurations may include positions of the loudspeakers, numbers of loudspeakers, types of loudspeakers, listening positions, correction factors, or any combination thereof. The statistical analysis may indicate at least one metric of the potential configuration including indicating consistency of predicted transfer functions, flatness of the predicted transfer functions, differences in overall sound pressure level from seat to seat for the predicted transfer functions, efficiency of the predicted transfer functions, or the output of predicted transfer functions. The system also provides a methodology for selecting loudspeaker locations, the number of loudspeakers, the types of loudspeakers, correction factors, listening positions, or a combination of these schemes in an audio system that has a single listening position or multiple listening positions.

An etude in hearing‐aid directivity measurement and benchmarking

In this study, a three‐dimensional, automated scanning system is used to measure the directional performance of hearing aids mounted on a test manikin. A diffuse field is simulated in an anechoic environment by sequentially positioning a loudspeaker along the surface of a 1‐m‐radius sphere circumscribing the manikin and device under test. Particular attention is given to the type of stimulus, the spatial resolution of loudspeaker positions, and postprocessing of the impulse responses; each affects the measurement precision of the Directivity Index and the 3D polar responses. The purpose of this research is to describe the recipe that yielded minimal measurement bias and variance for ITEs, BTEs, and OTEs of varying directivity.

Acoustic rendering of a virtual environment based on virtual microphone control and binaural room scanning

Binaural room scanning (BRS) is a convolution technique that utilizes a set of spatially indexed impulse responses that are selected in response to listener head movements during virtual acoustic rendering, such that the direction of sonic elements is automatically updated according to the angle determined by a head tracker. Since the room impulse responses have to be measured for only a few loudspeaker positions, a very successful application of BRS is the simulation of a control room. In order to create a flexible headphone‐based virtual environment, it is proposed to simulate the input signals for the BRS system using virtual microphone control (ViMiC). ViMiC renders a virtual environment by simulating the output signals of virtual microphones that can be used to address a surround loudspeaker system. The advantages of this approach are twofold. First, the measured impulse responses of the BRS system ensure high spatial density in overall reflection patterns, avoiding the typical gaps in between the early reflections that occur when using solely mirror images or ray tracing. Second, the resulting imaging is closer to common audio engineering practice, where a pleasant but plausible environment is favored over a more purely realistic rendering.

A study of bone conduction and its applications for audio signal transmission

Human can perceive a sound in two different ways: one is sound transmission through the ear canal, another is that through the skull. The bone conduction technology has been used mainly for hearing impaired listeners and as communication tools in the military organization. Recently, because of a rapid improvement of the bone conduction device, its technology has spread in the field of audio. However, the relationship between the physical characteristic and the subjective responses of bone conduction are not clearly understood. For example, the effect of attachment position and excitation force of bone conduction loudspeakers has been still under discussion. In addition, since the bone conduction units also generate the airborne sound, listeners can hear the sound composed of both the bone conduction sound and the airborne sound. In our previous work, by using of a loudness curve and a threshold of hearing, a fundamental performance of the bone conduction was discussed. In this paper, further investigation will be carried out in order to establish a transmission system of the bone conduction, and the airborne sound on the eardrum that is caused by the vibration of the bone conduction loudspeaker will be measured.

Accuracy of an acoustic location system for monitoring the position of duetting songbirds in tropical forest

A field test was conducted on the accuracy of an eight-microphone acoustic location system designed to triangulate the position of duetting rufous-and-white wrens (Thryothorus rufalbus) in Costa Rica's humid evergreen forest. Eight microphones were set up in the breeding territories of 20 pairs of wrens, with an average intermicrophone distance of 75.2+/-2.6 m. The array of microphones was used to record antiphonal duets broadcast through stereo loudspeakers. The positions of the loudspeakers were then estimated by evaluating the delay with which the eight microphones recorded the broadcast sounds. Position estimates were compared to coordinates surveyed with a global-positioning system (GPS). The acoustic location system estimated the position of loudspeakers with an error of 2.82+/-0.26 m and calculated the distance between the "male" and "female" loudspeakers with an error of 2.12+/-0.42 m. Given the large range of distances between duetting birds, this relatively low level of error demonstrates that the acoustic location system is a useful tool for studying avian duets. Location error was influenced partly by the difficulties inherent in collecting high accuracy GPS coordinates of microphone positions underneath a lush tropical canopy and partly by the complicating influence of irregular topography and thick vegetation on sound transmission.

Qualification and performance of a reverberation chamber equipped with lightweight diffusers

A study was conducted in the reverberation chamber of the Integrated Acoustics Laboratory at Georgia Tech to evaluate the efficacy of lightweight (1.3 kg/m 2 ), fiberglass diffusers in increasing room diffusion while affording easy chamber reconfiguration. To this end, performance and qualification tests were conducted in accordance with the American Society of Testing and Materials (ASTM) standard C423-02a, the American standard for sound absorption measurements in reverberation chambers. Also, comparative absorption tests were conducted per the standard for comparison with other laboratories. Preliminary performance testing per C423-X1 included determining the optimum diffuser surface area and quantifying variation of sound decay rate with loudspeaker position. During performance testing, the sound absorption coefficient of a specimen increased with increasing diffuser area until the coefficient reached a maximum. The configuration that yielded this maximum, identified in Appendix X1 of the standard as optimum, had a diffuser-to-chamber surface area ratio of 19%; the diffusers were stationary for these tests. With respect to loudspeaker position, the decay rate was relatively independent of position. Qualification testing per C423-A3 required demonstration that decay rate was sufficiently independent of measurement location and specimen position. With the optimum stationary diffuser configuration the chamber did not qualify per ASTM C423-A3. In five of eighteen frequency bands, variation of decay rate with respect to measurement location exceeded the qualification limitations; likewise, variation with respect to specimen position exceeded the limits in five bands. However, a combination of rotating and stationary lightweight diffusers produced a sound environment that was diffuse enough to qualify per ASTM C423 in all frequency bands. To compare performance to other laboratories, absorption testing of round robin specimens was replicated in the IAL chamber. It should be noted that all specimens were Amounted according to ASTM Practices E795-00 (to chamber floor); the results of this paper apply to this mounting only. With both diffuser configurations, the measured absorption coefficients and their repeatabilities were within the confidence interval developed by the round robin testing. Interestingly, the absorption coefficients measured with the diffuser configuration that did not qualify the chamber were statistically no different from those from qualified laboratories. These results call into question the assumption that dense diffusers and strict diffusion requirements are necessary for sound absorption measurements in reverberation chambers.

Territorial responses of male blue tits to simulated dynamic intrusions: effects of song overlap and intruder location

Territorial songbirds frequently challenge each other with intrusions and close-range singing interactions. The timing of songs in such interactions may signal aggressive intentions and experiments have shown that song overlapping may indicate a higher willingness to escalate an interaction than song alternating. A territory owner should therefore perceive an intruder overlapping its songs and staying inside its territory as a greater threat than one alternating and exiting the territory quickly. To test this hypothesis we used playback to interact with territorial male blue tits, Parus caeruleus, by varying the amount of overlap of a subject's song; we also used multiple loudspeakers to simulate an intruder changing song posts. The territory owners approached and followed simulated intruders that changed song post inside their territories more slowly when more of their songs had been overlapped at the first loudspeaker, and they did not follow an intruder outside their territory. Nevertheless their responses remained high throughout playback and interaction terms between proportion of songs overlapped and loudspeaker position inside or outside the territory indicated that male blue tits combined and used information from both singing and intrusion patterns.

Intelligent sound field tuning system for home theater systems

In this paper, we propose an intelligent sound field tuning system for home theater systems that includes room acoustic compensation and additive artificial sound field processing. Room acoustic compensation is performed based on the information about room transfer characteristics measured by a three-channel microphone array. With a three-channel microphone array, we estimate the exact angular position, distance, level, and frequency response of loudspeakers. Using this information, the proposed system can compensate the degradation induced by non-ideal loudspeaker arrangement with virtual loudspeaker positioning and binaural energy level alignment algorithm. After room acoustic compensation, additive artificial sound field processing is performed for more realistic spacious impression. For artificial sound field processing, directional impulse responses are measured in real acoustic space by three-channel microphone array.

Loudspeaker and listener positions for optimal low-frequency spatial reproduction in listening rooms

This paper will briefly describe the physical and physiological mechanisms that enable low-frequency externalization and spatial reproduction in listening rooms. These mechanisms depend on reproducing a time-varying interaural time difference at the listening position through interference between symmetric and asymmetric room modes. The effect works successfully when the symmetric and asymmetric room modes are driven by independent portions of a multi-channel signal, typically the in-phase and the anti-phase component of a two-channel recording. The paper will describe how this overlap can be optimized by adjusting the loudspeaker and listener positions for a variety of playback rooms. The results show that certain common room shapes yield low spatiality regardless of loudspeaker and listener positions.

개구부를 가지는 실내의 능동소음제어시스템의 최적 트랜스듀서 위치

Optimal transducer positions are important as much as the control algorithms and hardware performance in the active noise control system. This study is similar to the past researches on the optimal transducer locations but with a far field noise source having a plane wave characteristic and the noise coming through an opening such as a window in an enclosure. Optimization techniques are used to find sets of optimal loudspeaker positions from a larger possible loudspeaker positions. Loudspeakers are placed on the surface of opening at the wall and inside of the enclosure. Using the measured acoustic transfer impedances and numerical simulations with the optimization technique, optimal positions are identified and compared. When a small number of loudspeakers are used. loudspeaker positions on the opening near the center seems to be the best place, but when a larger number of loudspeakers are used it was difficult to find simple patterns in the optimal positions. With the optimally positioned loudspeakers, optimal microphone positions are also studied.

Estimated variability of real-ear insertion response (REIR) due to loudspeaker type and placement

The real-ear insertion response (REIR) of a hearing aid is estimated as the difference between the aided response and the unaided response in the ear canal. Changes in the position of the loudspeaker relative to the head, between the two measurements, may reduce the accuracy of the estimate. The spatial variability of the sound field at distances close to the loudspeaker is less for a ‘flat-panel’ loudspeaker than for a conventional cone loudspeaker; the panel might thus lead to a reduced influence of loudspeaker position. To assess this, we measured the real-ear unaided response (REUR) as each of three loudspeakers (two cone type and one panel) was moved in a threedimensional space centred at either 0° or 45° azimuth, at a distance of 50 cm from a KEMAR manikin. Contrary to our expectation, the variability of the REUR was larger for the panel than for the cone loudspeakers. The REUR varied less with position for the 0° than for the 45° azimuth. The variability of the REUR decreased with increasing distance of the loudspeaker from KEMAR. We tentatively suggest that loudspeaker-to-client distances of 40–50 cm should be used and that a 0° azimuth is preferable.

Controlling loudspeakers in rooms

How to control the perceived timbre of a loudspeaker in different listening positions, using different loudspeaker positions in different listening rooms is a central problem, when designing and using loudspeakers is discussed. Loudspeaker directivity has been found to be one of the critical parameters in the solution to this problem. This paper presents three hypotheses for the optimal loudspeaker directivity and a novel realization of that directivity, i.e., the Acoustic Lense (ALT). The dominant problem at low frequencies has been found to relate to the fact, that the acoustic power output of loudspeakers is highly dependent on the loudspeaker position and on the acoustic properties of the listening room. This paper also presents a novel system for adapting a loudspeaker to its position and to the acoustic properties of the listening room, i.e., Adaptive Bass Control (ABC).