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1.
Since the classical room acoustics can not be used for long enclosures due to the inhomogeneous sound field, much work has been carried out recently to investigate the sound propagation in long enclosures, which are helpful to the acoustic design of practical long enclosures, such as the high-speed railway tunnels. However, most of these works focuses on the straight long enclosures without branches or with one branch. In this paper, the effects of the multiple branches on sound propagation of long enclosures are studied. The sound pressure level (SPL) attenuation, early decay time (EDT), and reverberation time (T30) of long enclosures with multiple branches have been investigated by physical scale models based acoustic experiments. Several interesting results have been obtained concerning the sound propagation of long enclosures with multiple branches. It shows that the sound field of long enclosures with multiple branches is more complex and inhomogeneous than that of the long enclosures without branches or with one branch.  相似文献   

2.
Detailed acoustical measurements were made in 41 working elementary school classrooms near Ottawa, Canada to obtain more representative and more accurate indications of the acoustical quality of conditions for speech communication during actual teaching activities. This paper describes the room acoustics characteristics and noise environment of 27 traditional rectangular classrooms from the 41 measured rooms. The purpose of the work was to better understand how to improve speech communication between teachers and students. The study found, that on average, the students experienced: teacher speech levels of 60.4 dB A, noise levels of 49.1 dB A, and a mean speech-to-noise ratio of 11 dB A during teaching activities. The mean reverberation time in the occupied classrooms was 0.41 s, which was 10% less than in the unoccupied rooms. The reverberation time measurements were used to determine the average absorption added by each student. Detailed analyses of early and late-arriving speech sounds showed these sound levels could be predicted quite accurately and suggest improved approaches to room acoustics design.  相似文献   

3.
The study of mosque acoustics, with regard to acoustical characteristics, sound quality for speech intelligibility, and other applicable acoustic criteria, has been largely neglected. In this study a background as to why mosques are designed as they are and how mosque design is influenced by worship considerations is given. In the study the acoustical characteristics of typically constructed contemporary mosques in Saudi Arabia have been investigated, employing a well-known impulse response. Extensive field measurements were taken in 21 representative mosques of different sizes and architectural features in order to characterize their acoustical quality and to identify the impact of air conditioning, ceiling fans, and sound reinforcement systems on their acoustics. Objective room-acoustic indicators such as reverberation time (RT) and clarity (C50) were measured. Background noise (BN) was assessed with and without the operation of air conditioning and fans. The speech transmission index (STI) was also evaluated with and without the operation of existing sound reinforcement systems. The existence of acoustical deficiencies was confirmed and quantified. The study, in addition to describing mosque acoustics, compares design goals to results obtained in practice and suggests acoustical target values for mosque design. The results show that acoustical quality in the investigated mosques deviates from optimum conditions when unoccupied, but is much better in the occupied condition.  相似文献   

4.
This paper presents the results of new studies based on speech intelligibility tests in simulated sound fields and analyses of impulse response measurements in rooms used for speech communication. The speech intelligibility test results confirm the importance of early reflections for achieving good conditions for speech in rooms. The addition of early reflections increased the effective signal-to-noise ratio and related speech intelligibility scores for both impaired and nonimpaired listeners. The new results also show that for common conditions where the direct sound is reduced, it is only possible to understand speech because of the presence of early reflections. Analyses of measured impulse responses in rooms intended for speech show that early reflections can increase the effective signal-to-noise ratio by up to 9 dB. A room acoustics computer model is used to demonstrate that the relative importance of early reflections can be influenced by the room acoustics design.  相似文献   

5.
The sound power of a number of test objects was determined from spatially averaged intensity measurements. The results show that the influence of room acoustics is insignificant even for rooms of widely different room constants, if the measuring surfaces are exactly defined and if a good space-averaging technique is used. The intensity integrated over a closed surface defining a source-free space compared to the sound pressure integrated over the same surface gives a measure of the capability of a specific intensity measuring system to suppress external noise. For the test arrangements measured with broad band noise, this suppression was found to be 14–18 dB(A). A similar value of 15 dB was found from sound power measurements on a source with high external sound and an analysis of the results in one-third octave bands. From these measurements an analytical function was derived which describes the average error of the spatially averaged intensity as a function of the difference between the external sound level and the source sound level. For practical measurement situations a further analytical function was derived which gives this intensity error as a function of the difference between the measured (spatially averaged) pressure and intensity levels. Thus it is possible to estimate the error of intensity measurements directly from measured intensity and pressure data.  相似文献   

6.
A novel panel-form loudspeaker in which the panel of the speaker is excited by the forces generated through the flat voice coil of a rectangular electro-magnetic type exciter for sound radiation is presented. The exciter when properly designed has the advantage of exerting appropriate loads to the panel so that the major sound pressure level (SPL) dips of the speaker can be suppressed or even eliminated. For designing such panel-form speaker, a method formulated on the basis of the classical plate theory (CPT), Ritz method, and first Rayleigh integral is proposed for predicting the SPL curve of the speaker. An experimental investigation was performed to verify the feasibility of the proposed method. The effects of some system parameters on the major SPL dips of the proposed panel-form speakers are investigated by means of several numerical examples. The optimal locations of flat voice coils for exciting several panel-form speakers are determined to illustrate the important role of excitation location for enhancing sound quality of such speakers via the removal or suppression of the major SPL dips.  相似文献   

7.
M.K. Lim 《Applied Acoustics》1981,14(4):245-252
This paper describes a corona-type point source suitable for acoustic measurement purposes, especially model studies in acoustics. This source operates within a wide frequency range from 1 kHz to 100 kHz. The characteristics of this sound source were investigated and were found to be very close to those of a theoretical point source. It also gives a satisfactory sound power level.  相似文献   

8.
刘力  蔡野锋  吴鸣  杨军 《应用声学》2015,34(1):7-16
针对目前室外及超大型室内空间的扩声应用中声场分布不均匀的问题,本文提出了一种基于最小二乘法声场重建方法的扩声技术。该技术通过对目标声场的逼近来计算线性扬声器阵列各通道的输入参数,实现扩声区域内声压级的均匀分布,同时约束非扩声区域的声能量以获得较好指向性。本文通过仿真研究算法各参数的改变对控制结果的影响,探讨不同扩声区域和目标所对应参数的选取方法。仿真和实验比较了该算法以及未经控制的声场、相移法波束控制的效果,证明该方法可以获得更好的声场均匀度。  相似文献   

9.
High speed switching of current in gradient coils within high magnetic field strength magnetic resonance imaging (MRI) scanners results in high acoustic sound pressure levels (SPL) in and around these machines. Many studies have already been conducted to characterize the sound field in and around MRIs and various methods have been investigated to attenuate the noise generated. In the work presented here a computational vibro-acoustic model was developed based on an iteratively modified and validated finite element (FE) model to characterize the acoustic noise properties of the gradient coil. The simulation results from the computational model were verified through experimental noise measurement for the gradient coil insert in a 4 T MRI scanner by using swept sinusoidal time waveform inputs. Comparisons show that the computational model predicts the noise characteristic properties extremely accurately. There are three dominant frequency bands where the SPL is much higher than those at other frequencies. The SPL in the horizontal direction is much higher than that in the vertical direction due to the excitation to the horizontally placed X coil. The SPL to the inner surface of the coil is higher than far from the inner surface, which proves that the acoustic noise is radiated from the inner surface and primarily caused by the normal vibration of the inner surface. Further verification was conducted by using two types of trapezoidal sequence inputs usually used, which is to simulate real scanning sequences for small animals. Again the accuracy of the developed model is verified. The validated acoustic computational model could be used as an effective method to predict the noise that would be produced by a coil in the design stage. Modification of the structural design or the excitation pulse could be performed to reduce the acoustic noise when the gradient coil is in scanning.  相似文献   

10.
Vocal intensity is studied as a function of fundamental frequency and lung pressure. A combination of analytical and empirical models is used to predict sound pressure levels from glottal waveforms of five professional tenors and twenty five normal control subjects. The glottal waveforms were obtained by inverse filtering the mouth flow. Empirical models describe features of the glottal flow waveform (peak flow, peak flow derivative, open quotient, and speed quotient) in terms of lung pressure and phonation threshold pressure, a key variable that incorporates the Fo dependence of many of the features of the glottal flow. The analytical model describes the contributions to sound pressure levels SPL by the vocal tract. Results show that SPL increases with Fo at a rate of 8-9 dB/octave provided that lung pressure is raised proportional to phonation threshold pressure. The SPL also increases at a rate of 8-9 dB per doubling of excess pressure over threshold, a new quantity that assumes considerable importance in vocal intensity calculations. For the same excess pressure over threshold, the professional tenors produced 10-12 dB greater intensity than the male nonsingers, primarily because their peak airflow was much higher for the same pressure. A simple set of rules is devised for predicting SPL from source waveforms.  相似文献   

11.
吕亚东  田静 《声学学报》2010,35(2):101-106
首先回顾了全国声学标准化技术委员会自1980年成立以来,在国家声学标准制、修订方面所取得的一批重要的标准研究成果。经过全国声学领域历届委员和专家30年来的共同努力,截止2009年共制、修订声学方面国家标准166项次,形成了目前归口声标委管理的131项现行有效的国家标准,由此也建立了涵盖声学基础、噪声、建筑声学、超水声四大类标准的国家声学基础测量方法标准体系。这些声学基础测量方法标准体系与国家声环境质量、噪声排放等环保标准和工程建设标准、电声和音视频等标准共同构成了较为完整的国家声学标准体系。最后结合声标委今后要开展的标准工作,提出了进一步完善国家声学基础测量方法标准体系的细部构想,可以相信,将来随着这些标准的陆续出台,我国声学标准体系会更趋完善。   相似文献   

12.
《Journal of voice》2020,34(3):485.e33-485.e43
PurposeThe present study aimed at measuring the smoothed and non-smoothed cepstral peak prominence (CPPS and CPP) in teachers who considered themselves to have normal voice but some of them had laryngeal pathology. The changes of CPP, CPPS, sound pressure level (SPL) and perceptual ratings with different voice tasks were investigated and the influence of vocal pathology on these measures was studied.MethodEighty-four Finnish female primary school teachers volunteered as participants. Laryngoscopically, 52.4% of these had laryngeal changes (39.3% mild, 13.1% disordered). Sound recordings were made for phonations of comfortable sustained vowel, comfortable speech, and speech produced at increased loudness level as used during teaching. CPP, CPPS and SPL values were extracted using Praat software for all three voice samples. Sound samples were also perceptually evaluated by five voice experts for overall voice quality (10 point scale from poor to excellent) and vocal firmness (10 point scale from breathy to pressed, with normal in the middle).ResultsThe CPP, CPPS and SPL values were significantly higher for vowels than for comfortable speech and for loud speech compared to comfortable speech (P < 0.001). Significant correlations were found between SPL and cepstral measures. The loud speech was perceived to be firmer and have a better voice quality than comfortable speech. No significant relationships of the laryngeal pathology status with cepstral values, perceptual ratings, or voice SPLs were found (P > 0.05).ConclusionNeither the acoustic measures (CPP, CPPS, and SPL) nor the perceptual evaluations could clearly distinguish teachers with laryngeal changes from laryngeally healthy teachers. Considering no vocal complaints of the subjects, the data could be considered representative of teachers with functionally healthy voice.  相似文献   

13.
Moving-coil loudspeakers typify the interdisciplinary nature of acoustics: in order to reproduce sound, these devices employ principles of electricity, magnetism, mechanics, and acoustics. The widespread use of loudspeakers has made them a familiar and valuable opportunity to introduce students to acoustics. A low-cost loudspeaker project/demonstration is presented here that is built from scratch using common household items and craft supplies. A variety of educational topics may be illustrated with this device, making it appropriate for a wide range of academic levels.  相似文献   

14.
Sound field reproduction has applications in music reproduction, spatial audio, sound environment reproduction, and experimental acoustics. Sound field reproduction can be used to artificially reproduce the spatial character of natural hearing. The objective is then to reproduce a sound field in a real reproduction environment. Wave field synthesis (WFS) is a known open-loop technology which assumes that the reproduction environment is anechoic. The room response thus reduces the quality of the physical sound field reproduction by WFS. In recent research papers, adaptive wave field synthesis (AWFS) was defined as a potential solution to compensate for these quality reductions from which WFS objective performance suffers. In this paper, AWFS is experimentally investigated as an active sound field reproduction system with a limited number of reproduction error sensors to compensate for the response of the listening environment. Two digital signal processing algorithms for AWFS are used for comparison purposes, one of which is based on independent radiation mode control. AWFS performed propagating sound field reproduction better than WFS in three tested reproduction spaces (hemianechoic chamber, standard laboratory space, and reverberation chamber).  相似文献   

15.
It is known that the sound field in a long space is not diffuse, and that the classic theory of room acoustics is not applicable. A theoretical model is developed for the prediction of reverberation time and speech transmission index in rectangular long enclosures, such as corridors and train stations, where the acoustic quality is important for speech. The model is based on an image-source method, and both acoustically hard and impedance boundaries are investigated. An approximate analytical solution is used to predict the frequency response of the sound field. The reverberation time is determined from the decay curve which is computed by a reverse-time integration of the squared impulse response. The angle-dependence of reflection coefficients of the boundaries and the change of phase upon reflection are incorporated in this model. Due to the relatively long distance of sound propagation, the effect of atmospheric absorption is also considered. Measurements of reverberation time and speech transmission index taken from a real tunnel, a corridor, and a model tunnel are presented. The theoretical predictions are found to agree well with the experimental data. An application of the proposed model has been suggested.  相似文献   

16.
Male bullfrogs will vocalize in response to playbacks of the mating (advertisement) calls of conspecifics. This behavior was studied in response to playbacks of bullfrog mating calls presented at six different sound intensity levels. The lowest sound intensity level tested (50 dB SPL) was insufficient to evoke calling from any of the animals. Calling was evoked by playback levels of 60 dB SPL and higher. The data suggest that behavioral evoked calling thresholds lie between 50-60 dB SPL for these animals. Playback intensity levels of 80 dB SPL were more effective in evoking responses than were intensity levels up to 20 dB higher or lower. This was true both in terms of the total number of evoked responses and the trial number at which responding ceased. Moreover, significantly less habituation of evoked calling occurred at levels of 80 dB SPL than at higher or lower levels. The data suggest that a sound pressure level of 80 dB represents a behaviorally preferred intensity level for evoked calling in the bullfrog. Field recordings of bullfrog choruses show that the intensity produced by an individual calling male reaches a level of 80 dB SPL at a distance of 1 m. This intensity level is identical to that producing maximal evoked calling in the laboratory.  相似文献   

17.
A theoretical framework is established, for the robustness of multichannel sound equalization in reverberant environments. Using results from statistical room acoustics, a closed-form expression is derived that predicts the degradation in performance of an equalization system as the sound source moves from its nominal position inside the enclosure. The presented analysis also provides means of identifying the performance bounds that can be expected when using such a system in an actual room. Using extensive computer simulations, the effect of physical parameters such as the relative positions of the source and the receivers, as well as effects of different design parameters are investigated. Based on the conditions imposed by these parameters, it is shown that, depending on the array geometry and the exact form of the equalizers, slight performance gains can be expected as the number of receivers is increased.  相似文献   

18.
Computer algorithms are described for constructing virtual acoustic models of various rooms that should satisfy some specific sound quality criteria. The algorithms are based on the ray tracing method, which, in the general case, allows calculation of the amplitude of an acoustic ray that survived multiple reflections from arbitrary curved surfaces. As a result, calculations of room acoustics are reduced to tracing the trajectories of all the acoustic rays in the course of their propagation with multiple reflections from reflecting surfaces to the point of their complete decay. For this approach to be used, the following physical properties of a room should be known: the geometry of the reflecting surfaces, the absorption and diffusion coefficients on each of these surfaces, and the decay law for rays propagating in air. The proposed models allow for the solution of the important problem of architectural acoustics called the auralization problem, i.e., to predict how any given audio segment will sound in any given hall on the basis of computer simulation alone, without any full-scale testing in specific halls.  相似文献   

19.
A hypophonic voice, characterized perceptually as weak and breathy, is associated with voice disorders such as vocal fold atrophy and unilateral vocal fold paralysis. Although voice therapy programs for hypophonia typically address the vocal folds or the sound source, twang voice quality was examined in this study as an alternative technique for increasing vocal power by altering the epilarynx or the sound filter. OBJECTIVE: This study investigated the effect of twang production on physiologic, acoustic, and perceived voice handicap measures in speakers with hypophonia. DESIGN/METHODS: This prospective pilot study compared the vocal outcomes of six participants with hypophonia at pre- and posttreatment time points. Outcome measures included mean airflow rate, intensity in dB sound pressure level (SPL), maximum phonation time, and self-report of voice handicap. RESULTS: All subjects improved in at least three of the four vocal outcome measures. Wilcoxon signed-rank test of paired differences revealed significant differences between pre- and posttherapy group means for airflow rate, SPL, and Voice Handicap Index scores. CONCLUSION: The twang voice quality as a manipulation of the sound filter offers a clinical complement to traditional voice therapies that primarily address the sound source.  相似文献   

20.
Numerical modeling of urban sound fields by a diffusion process   总被引:1,自引:0,他引:1  
In this paper, we present the numerical implementation of a sound field model used in urban acoustics. The mathematical model being based on a classic diffusion equation for the sound energy, a simple finite difference scheme is applied. We give also some finite difference equations for simple boundary conditions, like absorption by a wall and at building edges. The two-dimensional numerical scheme is then compared to analytical solutions of the sound field propagation in a rectangular street with a good agreement, both in the steady state and in the time varying state. Finally it is suggested that the adjustment of usual softwares for heat transfer could be an interesting and low cost way to develop powerful acoustic softwares for the prediction of noise in urban areas.  相似文献   

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