Equalization filter design for downfiring flat television speakers

Many loudspeakers in newer flat television systems are mounted on the bottom of the television, with their diaphragms facing downward, so as to be hidden inside the TV frame. This kind of loudspeaker installation, called downfiring, induces relatively large reflections from the walls of a room. The increased reflections inevitably change the impulse response of the loudspeakers, which leads to a perceptible distortion in sound quality. In this study, an equalization procedure to resolve the distortion due to downfiring loudspeakers is presented. The change in the early-arriving sound from the room reflections was analyzed, and the specific experimental environment was set up to measure the early reflections for designing a single-channel equalization filter. The inverse technique for the single-input multiple-output (SIMO) system was applied such that both the magnitude and phase responses from downfiring loudspeakers can be equalized at multiple listener positions. We also investigated the possible equalization range in space and time, for which the single-channel filter can work effectively. The performance of the filter designed for the equalization range was then demonstrated by experiments. The experiments were performed using a flat television mock-up installed in a reflective environment. For the performance evaluation of the proposed method, we employed two different measures that can represent the magnitude and phase distortions: the mean squared error of the magnitude and an energy decay curve. The experimental results demonstrate that the single-channel filter can reduce the early reflections from a downfiring loudspeaker by a considerable amount.     

Robustness of spatial average equalization: a statistical reverberation model approach

Traditionally, multiple listener room equalization is performed to improve sound quality at all listeners, during audio playback, in a multiple listener environment (e.g., movie theaters, automobiles, etc.). A typical way of doing multiple listener equalization is through spatial averaging, where the room responses are averaged spatially between positions and an inverse equalization filter is found from the spatially averaged result. However, the equalization performance, will be affected if there is a mismatch between the position of the microphones (which are used for measuring the room responses for designing the equalization filter) and the actual center of listener head position (during playback). In this paper, we will present results on the effects of microphone-listener mismatch on spatial average equalization performance. The results indicate that, for the analyzed rectangular configuration, the region of effective equalization depends on (i) the distance of a listener from the source, (ii) the amount of mismatch between the responses, and (iii) the frequency of the audio signal. We also present some convergence analysis to interpret the results.     

Acoustic model for robustness analysis of optimal multipoint room equalization

In this paper, an acoustic model for the robustness analysis of optimal multipoint room equalization is proposed. The optimal multipoint equalization aims to have the optimal performance in a least-squares sense for all measured points. The model can be used for theoretical robustness estimation depending on the critical design parameters such as the number of measurement points, the distance between measurements, or the frequency before applying real equalization system. The analysis results show that it is important to set the appropriate number of measurement points and the distances between measurement points to ensure the enlarged equalization region at a specific frequency.     

Power-output regularization in global sound equalization

The purpose of equalization in room acoustics is to compensate for the undesired modification that an enclosure introduces to signals such as audio or speech. In this work, equalization in a large part of the volume of a room is addressed. The multiple point method is employed with an acoustic power-output penalty term instead of the traditional quadratic source effort penalty term. Simulation results demonstrate that this technique gives a smoother decline of the reproduction performance away from the control points.     

Spatially extended sound equalization in rectangular rooms.

The results of a theoretical study on global sound equalization in rectangular rooms at low frequencies are presented. The zone where sound equalization can be obtained is a continuous three-dimensional region that occupies almost the complete volume of the room. It is proved that the equalization of broadband signals can be achieved by the simulation of a traveling plane wave using FIR filters. The optimal solution has been calculated following the traditional least-squares approximation, where a modeling delay has been applied to minimize reverberation. An advantage of the method is that the sound field can be estimated with sensors placed in the limits of the equalization zone. As a consequence, a free space for the listeners can be obtained.     

Toward broadband electroacoustic resonators through optimized feedback control strategies

This paper presents a methodology for the design of broadband electroacoustic resonators for low-frequency room equalization. An electroacoustic resonator denotes a loudspeaker used as a membrane resonator, the acoustic impedance of which can be modified through proportional feedback control, to match a target impedance. However, such impedance matching only occurs over a limited bandwidth around resonance, which can limit its use for the low-frequency equalization of rooms, requiring an effective control at least up to the Schroeder frequency. Previous experiments have shown that impedance matching can be achieved over a range of a few octaves using a simple proportional control law. But there is still a limit to the feedback gain, beyond which the feedback-controlled loudspeaker becomes non-dissipative. This paper evaluates the benefits of using PID control and phase compensation techniques to improve the overall performance of the electroacoustic resonator. More specifically, it is shown that some adverse effects due to high-order dynamics in the moving-coil transducer can be mitigated. The corresponding control settings are also identified with equivalent electroacoustic resonator parameters, allowing a straightforward design of the controller. Experimental results using PID control and phase compensation are finally compared in terms of sound absorption performances. As a conclusion the overall performances of electroacoustic resonators for damping the modal resonances inside a duct are presented, along with general discussions on practical implementation and the extension to actual room modes damping.     

Experimental implementation of a low-frequency global sound equalization method based on free field propagation

An experimental implementation of a global sound equalization method in a rectangular room using active control is described in this paper. The main purpose of the work has been to provide experimental evidence that sound can be equalized in a continuous three-dimensional region, the listening zone, which occupies a considerable part of the complete volume of the room. The equalization method, based on the simulation of a progressive plane wave, was implemented in a room with inner dimensions of 2.70 m × 2.74 m × 2.40 m. With this method, the sound was reproduced by a matrix of 4 × 5 loudspeakers in one of the walls. After traveling through the room, the sound wave was absorbed on the opposite wall, which had a similar arrangement of loudspeakers, by means of active control. A set of 40 digital FIR filters was used to modify the original input signal before it was fed to the loudspeakers, one filter for each transducer. The optimal arrangement of the loudspeakers and the maximum frequency that can be equalized is analyzed theoretically in this paper. The presented experimental results show that sound equalization was possible from 10 Hz to approximately 425 Hz in the listening zone. A flat frequency response with deviations within ±5 decibels from the desired value was achieved. A higher demanding performance with deviations within ±1.5 decibels from a flat frequency response was attained in the interval between 20 Hz and 280 Hz. At the same time, the impulse response was quite well approximated to a delayed delta function in the listening zone. Examples of the spatial distribution of the sound field are also shown.     

Self-adaptive grids for noise mapping refinement

Noise maps are usually represented as contour or isolines maps describing the sound levels in a region. Using this kind of representation the user can easily find the noise level assigned to every location in the map.But the acoustic calculations behind the map are not performed for every single location on it; they are only performed in a grid of receivers. The results in this calculation grid are interpolated to draw the isolines or contours. Therefore, the resolution of the calculation grid and the way it was created (rectangular, triangulated, random…) have an effect on the resulting map.In this paper we describe a smart iterative procedure to optimize the quality of the map at a really low additional computational cost, using self-adaptive grids for the acoustic calculations. These self-adaptive grids add new receivers to the sampling grid in those locations where they are expected to be more useful, so that the performance at the output of the interpolator is enhanced.Self-adaptive sampling grids can be used for minimizing the overall error of the map (improving its quality), or for reducing calculation times, and can be also applied selectively to target areas or contour lines. This can be done by the user customizing the maximum number of iterations, the number of new receivers for each iteration, the target isolines, the target quality…     

A method to predict the transmission of sound through corridors

This paper describes research carried out to provide a method by means of which sound levels in rooms adjacent to a corridor, produced by the transmission of sound through the corridor from another room, can be predicted. This has been done by using an acoustic scale model to show how the physical parameters of the corridor affect the transmission of sound into, out of and along the corridor. From analysis of the experimental results, empirical formulae are obtained to predict: (1) sound levels along a corridor produced by a sound source in the corridor or an adjacent room, (2) sound levels along a side corridor produced by a sound source in the main corridor and (3) sound levels in rooms adjacent to the corridor produced by a sound source in another room.     

Perceptual validation of virtual room acoustics: Sound localisation and speech understanding

The reliability of algorithms for room acoustic simulations has often been confirmed on the basis of the verification of predicted room acoustical parameters. This paper presents a complementary perceptual validation procedure consisting of two experiments, respectively dealing with speech intelligibility, and with sound source front–back localisation.The evaluated simulation algorithm, implemented in software ODEON®, is a hybrid method that is based on an image source algorithm for the prediction of early sound reflection and on ray-tracing for the later part, using a stochastic scattering process with secondary sources. The binaural room impulse response (BRIR) is calculated from a simulated room impulse response where information about the arriving time, intensity and spatial direction of each sound reflection is collected and convolved with a measured Head Related Transfer Function (HRTF). The listening stimuli for the speech intelligibility and localisation tests are auralised convolutions of anechoic sound samples with measured and simulated BRIRs.Perception tests were performed with human subjects in two acoustical environments, i.e. an anechoic and reverberant room, by presenting the stimuli to subjects in a natural way, and via headphones by using two non-individualized HRTFs (artificial head and hearing aids placed on the ears of the artificial head) of both a simulated and a real room.Very good correspondence is found between the results obtained with simulated and measured BRIRs, both for speech intelligibility in the presence of noise and for sound source localisation tests. In the anechoic room an increase in speech intelligibility is observed when noise and signal are presented from sources located at different angles. This improvement is not so evident in the reverberant room, with the sound sources at 1-m distance from the listener. Interestingly, the performance of people for front–back localisation is better in the reverberant room than in the anechoic room.The correlation between people’s ability for sound source localisation on one hand, and their ability for recognition of binaurally received speech in reverberation on the other hand, is found to be weak.     

A theoretical model for the evaluation of measurement uncertainty of a sound level meter calibration by comparison method in an anechoic room

A theoretical model for the evaluation of measurement uncertainty of a sound level meter (hereafter as `SLM') calibration by comparison method in an anechoic room was developed. Through this model, the uncertainties in the semi-automatic calibration and that in the full-automatic calibration were estimated for the recently developed SLM calibration system. In order to estimate the standard uncertainty against the SLM positioning, which is a significant uncertainty component, the sound field curve-fitting formulae were adopted. The validity of the curve-fitting method was proven by the similarity of the spatial distributions of radiation sound field produced by the plane circular piston source and that by the cone shape source. A linear equation was used to fit the measurements of the sound field distribution along the radiation axis. A quadratic equation was used to fit the measurements along the radial axis normal to the radiation axis. The fitting parameters gave us the sensitivity coefficients of the propagation of the uncertainty. In addition, one of the quadratic fitting parameters was found to be a positional uncertainty itself. Using this model, the expanded uncertainties were evaluated for the semi-automatic and full-automatic calibration of SLM.     

Directional properties of an array of sound receivers positioned in an impedance screen recess

Expressions for calculating the directional characteristics of an array of sound receivers positioned in a waveguide with impedance walls are obtained from the solution to the problem on the diffraction of a plane sound wave by the waveguide open end with impedance flanges. The waveguide can be of a finite length, and, in this case, it can be considered as an open cavity in an impedance screen. The solution of the integral equation for the sound pressure distribution over the opening area is reduced to the solution of an infinite system of algebraic equations for the coefficients of the field expansion in normal waveguide waves. Examples of calculated directional characteristics are presented for arrays with receivers positioned at different distances from the opening and for different values of the impedances of the waveguide walls and flanges.     

Experimental studies of sound field suppression at discrete frequencies

Practical implementation of an active sound control system ensuring sound suppression in outer space is described as applied to sound insulation problems for equipment whose total noise level is mainly due to low-frequency discrete spectral components. The operational principle of the proposed system is based on inverse field generation with respect to the field of the initial source of quasi-monochromatic signals. The inverse field is formed by a set of radiators, which are controlled by the signals of pressure receivers positioned in the near field of the source. Experimental studies carried out with the proposed sound control system demonstrate its efficiency and testify to the stability of its operation.     

基于时间反转的复杂声场拾声传声器阵列性能研究

探讨时间反转技术在复杂声场传声器阵列拾声中应用的可行性及其机理,给出其一般规律和性能。研究表明:在自由空间中,其拾声性能与频率,阵列形状和半径有关,频率越高,半径越大,拾声效果越好。在普通房间中,在语音频段内,圆弧阵列在预定目标点处的阵列增益性能要比离预定目标点约25 cm远处的位置处大5 dB以上。在普通房间和混响室中的实验验证了上述结论。      

Sound propagation and speech transmission in a branching underground tunnel

The characteristics of sound propagation and speech transmission along a tunnel with a "T" intersection were investigated. At receivers within sight of the sound source, low frequencies were mainly attenuated around the intersection than high frequencies. At receivers out of sight of the source, high frequencies were extensively attenuated. The overall pattern of sound attenuation along the different sections of tunnel, which was calculated by the conical beam method, agreed well with the measurements in this study. Numerical calculations of reflected and diffracted waves with minimum transmission paths in a two-dimensional plane showed that reflected waves were the primary contributors to sound fields out of sight of the source. The articulation scores measured at receivers within sight of the source were high, and most of the confusion concerned syllables that could easily be misheard, even if there were a high signal-to-noise ratio. The types of syllable confusions observed at the receivers out of sight of the source appeared to have been caused by the greater deterioration in speech signals along this part of the tunnel, especially at high frequencies. The evaluation by rapid speech transmission indices (RASTI) appeared to be overestimated at the receivers out of sight of the source. Taking into account the early decay times of impulsive sound and the calculation procedures used in RASTI, it is concluded that speech intelligibility may not have been evaluated correctly by RASTI.     

Examination of multi-dimensional vibration isolation measures and their correlation to sound radiation over a broad frequency range

This article examines alternate vibration isolation measures for a multi-dimensional system. The isolator and receiver are modelled by the continuous system theory. The source is assumed to be rigid and both force and moment excitations are considered. Our analysis is limited to a linear time-invariant system, and the mobility synthesis method is adopted to describe the overall system behavior. Inverted ‘L’ beam and plate receivers are employed here to incorporate the contribution of their in-plane motions to vibration powers and radiated sound. Multi-dimensional transmissibilities and effectivenesses are comparatively evaluated along with power-based measures for the inverted ‘L’ beam receiver and selected source configurations. Further, sound pressures radiated from the inverted ‘L’ beam receiver are calculated and correlated with power transmitted to the receiver. Interactions within the ‘L’ beam receiver are also analyzed and measures that could identify dominant transfer paths within a system are examined. Sound measurements and predictions for the inverted ‘L’ plate receiver demonstrate that a rank order based on free field sound pressures, at one or more locations, may be regarded as a measure of isolation performance. Measured insertion losses for sound pressure match well with those based on computed results although further study is needed in relation to some discrepancies shown in the results. Finally, several emerging research topics are identified.     

Blind estimation of reverberation time

The reverberation time (RT) is an important parameter for characterizing the quality of an auditory space. Sounds in reverberant environments are subject to coloration. This affects speech intelligibility and sound localization. Many state-of-the-art audio signal processing algorithms, for example in hearing-aids and telephony, are expected to have the ability to characterize the listening environment, and turn on an appropriate processing strategy accordingly. Thus, a method for characterization of room RT based on passively received microphone signals represents an important enabling technology. Current RT estimators, such as Schroeder's method, depend on a controlled sound source, and thus cannot produce an online, blind RT estimate. Here, a method for estimating RT without prior knowledge of sound sources or room geometry is presented. The diffusive tail of reverberation was modeled as an exponentially damped Gaussian white noise process. The time-constant of the decay, which provided a measure of the RT, was estimated using a maximum-likelihood procedure. The estimates were obtained continuously, and an order-statistics filter was used to extract the most likely RT from the accumulated estimates. The procedure was illustrated for connected speech. Results obtained for simulated and real room data are in good agreement with the real RT values.     

Source excitation strategies for obtaining impulse responses in finite difference time domain room acoustics simulation

This paper considers source excitation strategies in finite difference time domain room acoustics simulations for auralization purposes. We demonstrate that FDTD simulations can be conducted to obtain impulse responses based on unit impulse excitation, this being the shortest, simplest and most efficiently implemented signal that might be applied. Single, rather than double, precision accuracy simulations might be implemented where memory use is critical but the consequence is a remarkably increased noise floor. Hard source excitation introduces a discontinuity in the simulated acoustic field resulting in a shift of resonant modes from expected values. Additive sources do not introduce such discontinuities, but instead result in a broadband offset across the frequency spectrum. Transparent sources address both of these issues and with unit impulse excitation the calculation of the compensation filters required to implement transparency is also simplified. However, both transparent and additive source excitation demonstrate solution growth problems for a bounded space. Any of these approaches might be used if the consequences are understood and compensated for, however, for room acoustics simulation the hard source is the least favorable due to the fundamental changes it imparts on the underlying geometry. These methods are further tested through the implementation of a directional sound source based on multiple omnidirectional point sources.     

Determination of the sound energy level of a gunshot and its applications in room acoustics

In some cases an impulsive noise source such as a gunshot can be a preferred alternative when investigating building acoustics, including sound insulation measurements, when compared to conventional steady state noise sources. A gun equipped with blank cartridges is an impulsive noise source that is lightweight and small enough to be easily transported. The differences in the noise characteristics between individual cartridges for the same gun are usually small, so the impulsive source can be replicated to a high degree. This paper is focused on the practical application of the sound exposure levels produced by a gunshot with a known sound energy level in the rooms under investigation. In this way, the equipment and methods required by the conventional method are simplified significantly. Furthermore, reverberation times need not be measured, since the equivalent absorption area can be directly obtained from the measured sound exposure levels. Using Green’s theorem, the roles of the sound source and measuring microphone were exchanged, which simplified the determination of sound insulation as it was easier to change the position of the gun than the microphone. The results obtained using the impulsive noise source were in good agreement with those obtained using the conventional method. Above 100 Hz, their difference in any frequency band of interest was less than 1 dB.