Showing papers in "Acta Acustica United With Acustica in 2009"
TL;DR: In this paper, the authors proposed a method to measure and characterize the acoustical conditions of the whole office space, including both short and long distances from the speaker, using an omni-directional loudspeaker.
Abstract: Noise is the most detrimental factor of the indoor environment in open-plan offices. Speech is the most distracting source of noise. Distraction and speech privacy can be estimated objectively by determining speech intelligibility between workstations. Previously used measurement methods have focused on two neighbouring workstations. However, noise complaints are not restricted to the nearest workstation. The aim of this paper is to suggest a new method to measure and characterize the acoustical conditions of the whole office space, including both short and long distances from the speaker. The method should result in compact and expedient single-number quantities to improve the utilization of the method in building design. The measurement is carried out between workstations along a line consisting of at least 4 workstations. Measurements are made for background noise level, spatial reduction of Speech Transmission Index, STI, and spatial reduction of the SPL of normal effort speech. An omni-directional loudspeaker is used. The acoustic performance of offices could be logically described by three single number quantities: distraction distance, r D , spatial attenuation rate of A-weighted SPL of speech, DL 2 , and SPL of speech at a distance of 4 metres, L p.S,4m . The method has been validated in 16 offices which varied significantly in room geometry, furniture and absorption. The differences between offices were unexpectedly large. Recommendations are presented for new target values which are already adopted in two Finnish guidelines.
86 citations
TL;DR: In this paper, the third octave band spectra of water falling onto water, gravel, bricks and small boulders and various combinations were captured under controlled conditions and these were used later to enable the subjective assessment of the tranquillity of the sounds.
Abstract: There is increasing concern with protecting quiet and tranquil areas from intrusive noise. Noise reduction at source and barriers to transmission are mitigation measures often considered. An alternative is to attempt to mask or distract attention away from the noise source. The masking or distracting sound source should be pleasant so that it does not add to any irritation caused by the noise source alone. The laboratory measurements described in this paper consisted of capturing under controlled conditions the third octave band spectra of water falling onto water, gravel, bricks and small boulders and various combinations. These spectra were then matched with typical traffic noise spectra to assess the degree of masking that could be expected for each option. Recordings were also taken during each measurement and these were used later to enable the subjective assessment of the tranquillity of the sounds. It was found that there were differences between water sounds both in terms of masking and their subjective impact on tranquillity.
70 citations
TL;DR: This work offers a detailed analysis of several suitable sounds for human echolocation, comparing their merits from a physical point of view, and has important applications to design systematic and optimized training protocols for accurate eCholocation awareness.
Abstract: Active human echolocation can be an extremely useful aid for blind people. Active echolocation can be trained with both artificial and organic signals. Organic signals offer some advantages over artificial ones. Very detailed studies of organic signals in animals have been done. However, in the case of humans, the scientific literature is very scarce and not systematic. This is the first paper of a series on the properties of several suitable sounds for human echolocation. In this work, we offer a detailed analysis of these sounds, comparing their merits from a physical point of view. The results of this study have important applications to design systematic and optimized training protocols for accurate echolocation awareness.
61 citations
TL;DR: In this paper, three common methods, namely the equivalent radiated power, the lumped parameter model and an approximation based on the volume velocity, are investigated for estimating the radiated sound power functions of a car and the radiation of a diesel engine under realistic load cases.
Abstract: The radiated sound power value is often used to evaluate the sound radiation of a machine or a product. Since its estimation requires the sound pressure on a surrounding surface of the radiating object, the sound power value is mostly computed under high numerical costs due to the acoustic field that has to be modeled. Therefore, approximations of the sound pressure are widely popular. In this article three common methods namely the equivalent radiated power, the lumped parameter model and an approximation based on the volume velocity are investigated. It is the goal of this paper to test these methods on realistic examples. The radiated sound power functions of the floor panel of a car and the radiation of a diesel engine under realistic load cases are estimated.
57 citations
Abstract: The facsimile or true-to-original reproduction of sound events is of great interest in acoustics and related areas and has been researched for many years. One form of achieving this is binaural technology. Many consider binaural technology a very modern technology and some even consider that it is strictly related to and was invented for sound quality research. However, binaural technology, especially recording technology, has been established for some time and, in fact, the first steps were made in around 1880. Over the decades this technology has made enormous advances, due to the dedication of many people, but some challenges related to achieving a true facsimile are still to be resolved. The most important milestones and also the remaining challenges are presented herein and the prospects for the near future are discussed.
48 citations
TL;DR: In this article, a method for modeling room acoustics and auralizing the results in real time with a moving listener is presented, based on the acoustic radiance transfer technique which is capable of modeling arbitrary reflection properties of different materials.
Abstract: A method for modeling room acoustics and auralizing the results in real time with a moving listener is presented. The acoustics modeling is based on the acoustic radiance transfer technique which is capable of modeling arbitrary reflection properties of different materials. The novel idea of implementing this technique in frequency domain allows modeling of all frequencies at once as the time domain technique requires separate runs for each frequency band. Since the auralization of the results requires scaling, adding, and delaying responses as well as convolving them with head-related impulse responses, the massive parallel computation capacity of modern graphics hardware is utilized. Thus, realistic interactive walkthroughs are possible in typical room models with a stationary source.
43 citations
TL;DR: In this paper, the authors used a compressible URANS approach in combination with appropriate acoustic boundary conditions for the generation and propagation of entropy noise in a non-reactive tube flow.
Abstract: In the present work, the generation and propagation of entropy noise is computed using a compressible URANS approach in combination with appropriate acoustic boundary conditions. The Entropy Wave Generator (EWG) experiment is best suited for validating the proposed approach and for investigating the acoustic sources of entropy noise. The EWG involves a non-reactive tube flow, where entropy modes are imposed to an incoming air flow by using a heating module. The generated temperature non-uniformities are accelerated with the mean flow downstream in a convergent divergent nozzle, thus producing entropy noise. Simulation results of pressure fluctuations and their spectra for a defined standard configuration as well as for diff erent operating points of the EWG agree very well with the respective experimental data. Additionally, an analysis of the acoustic sources was performed. For this purpose the acoustic sources caused by the acceleration of density inhomogeneities were calculated. For the first time, a numerical method is introduced for the localization of the acoustic sources of entropy noise in acceleration/deceleration regions.
39 citations
TL;DR: In this article, an auditory experiment was carried out to investigate the relationship between various level indices and subjective responses to floor impact sounds and it was found that sound quality ratings (LL z and N max ) as well as instrumental metrics (L Aeq, L Amax and L m, 1/1/1(63−500) ) showed good correlation with impact sound annoyance.
Abstract: Measurements of impact sounds generated by an impact ball (heavy/soft impact source) were conducted in apartment buildings with box-frame type reinforced concrete structures. Variations in frequency characteristics were found in the impact ball sounds and these were classified into three frequency groups. An auditory experiment was carried out to investigate the relationship between various level indices and subjective responses to floor impact sounds. It was found that sound quality ratings (LL z and N max ) as well as instrumental metrics (L Aeq , L Amax and L m,1/1(63―500) ) showed good correlation with impact sound annoyance. Among them, L Amax was suggested as a practical descriptor of the auditory sensation of impact ball sounds on the basis of measurement and calculation procedures. Additional auditory experiments were conducted to characterize the classified impact sound groups and to evaluate the annoyance of impact ball sounds through paired comparison and semantic differential tests. The results show that the impact ball sound with a dominant sound pressure level at 250 Hz was most annoying. Sound Quality (SQ) metrics were used to explain the annoyance level of heavy-weight floor impact sounds. The annoyance obtained from the paired comparison test correlated well with loudness and fluctuation strength. In the semantic differential test, it was found that the adjectives describing loudness had a dominant effect on the subjects' annoyance levels.
38 citations
TL;DR: In this paper, the frequency-dependent excitation signals for each transducer were calculated using a discretely measured target radiation pattern to calculate the frequency dependent excitation signal for each source.
Abstract: Sound sources for measurements in room acoustics are of omni-directional type, in general. With respect to auralization applications, an omni-directionally measured room impulse response may not be the ideal choice since it does not represent the real life situation playing an instrument in the room. To achieve the directivity of a real source (like a musical instrument or human voice) with a technical sound source (like a loudspeaker) requires either to copy the entire body and the sound radiation (i.e. the surface velocity) of that particular source or to reproduce the directional pattern of the radiation using a multiple source configuration like a dodecahedron or icosahedron loudspeaker array with independent excitation of each transducer. The advantage of the latter method is obvious since one single source is able to provide a large variety of different directivities by simply changing the excitation profile. To maintain the appropriate excitation of each individual transducer, different approaches can be made. In this paper a method is described using a discretely measured target radiation pattern to calculate the frequency dependent excitation signals for each transducer. Hereby, the directivity pattern of the source transducers and a phase optimization of the energy averaged radiation of musical instruments are used. The advantage of this method is a very flexible computation that is able to match the radiation pattern at the points of measurement very well. The resulting input filters for the platonic sound source can be used either for a real time convolution or offline processing of measured signals.
37 citations
TL;DR: In this paper, a procedure to obtain more accurate parameters to be used in vibration and sound insulation models is presented, and the results show that they must be obtained from dynamic models and/or experiments.
Abstract: Steel studs are used in double walls to provide structural stability. This creates a vibration transmission path between leaves that can often be more critical than the airborne path through the cavity. Some of the existing models for sound transmission consider the studs as elastic springs. The spring stiff ness may be taken as the cross-section elastic stiff ness of the stud, but this leads to an underestimation of the vibration transmission. A procedure to obtain more accurate parameters to be used in vibration and sound insulation models is presented. The results show that they must be obtained from dynamic models and/or experiments.
35 citations
TL;DR: In this article, a low-voltage high-energy spark discharges in water and associated bubble oscillations are studied experimentally to determine the radiated acoustic energy and potential energy of the bubbles.
Abstract: In this paper low-voltage high-energy spark discharges in water and associated bubble oscillations are studied experimentally. Pressure waves emitted during spark discharges and during following bubble oscillations have been recorded and analyzed to determine the radiated acoustic energy and potential energy of the bubbles. These energies are then used to determine the efficiency of energy conversion. For example, it has been found out that only 2% to 8% of electrical energy stored in the capacitor bank is converted into the potential energy of the bubble at its first maximum volume and only about 30% of this potential energy is radiated as acoustic energy in the first bubble pulse. Due to specific features of the low-voltage apparatus used, bubbles of different sizes have been generated and these bubbles were found to oscillate with different intensities. To explain this great diversity in the generated bubbles, the electrical circuit of the apparatus and the early stages of the spark discharge are analyzed. While it is confirmed that the bubble size grows with the energy used for discharge, the relation between the intensity of bubble oscillations and the circuit parameters is more complex. To explain this relation, two hypotheses, partly supported by the data available, are presented. The first hypothesis concerns the rate of electrical energy delivery into the discharge channel, and the second hypothesis concerns the efficiency of electrical energy conversion into heat in the discharge channel.
TL;DR: In this paper, a new technique based on the time reversal sink concept is used to detect active sound sources with a limited number of measurement points, which allows high-resolution imaging and provides a new method of characterization and detection of sound sources.
Abstract: Numerous practical applications ― such as non destructive evaluation of industrial structures, acoustic characterization of musical instruments, and acoustic mapping of sound sources in a known propagation medium ― involve source detection and characterization. In the past, this problem has been investigated using different beamforming and backpropagation methods. In this work, a new technique, based on the time reversal sink concept, is used to detect active sound sources with a limited number of measurement points. The theory and application of super-resolution focusing of sound and vibration using a time-reversal sink (TRS) have been studied, both in ultrasonic regime and in audible range. A high-resolution imaging technique based on a numerical time reversal sink has recently been developed by the authors for vibrational imaging of active sources in a dispersive medium. In this paper, the numerical time reversal sink imaging technique is adapted to the case of high-resolution acoustic imaging of active sound sources in a three-dimensional free field. This technique allows high-resolution imaging and provides a new method of characterization and detection of sound sources. All results show the high resolution imaging capabilities of this new technique when compared with classical time-reversal (TR) backpropagation. More than simply detecting the position of the acoustic source, this technique allows to detect the size of the active sources. This technique provides an alternative to other imaging and source detection techniques, such as three-dimensional acoustic holography and beamforming.
TL;DR: In this paper, the authors investigated the perception of speech quality over telephone channels with time-varying transmission characteristics for simulated conversational structures and established a relationship between subjective quality associated with short speech samples (5−6 seconds) and quality associated to overall conversations (1−2 minutes).
Abstract: This study investigates the perception of speech quality over telephone channels with time-varying transmission characteristics for simulated conversational structures. The aim is to establish a relationship between subjective quality associated with short speech samples (5―6 seconds) and quality associated with overall conversations (1― 2 minutes). Two two-part experiments were conducted. In the first part of each experiment, dialog-final ratings within the temporal structure of a telephone conversation were assessed. Varying transmission characteristics were realized with ten different degradation profiles of preprocessed speech samples obtained mainly from real mobile channels to ensure authentic types of degradation. The second part was carried out to obtain separate short-term ratings of the speech samples used in the first part. Experiments 1 and 2 tested different conversation durations (1 and 2 minutes). The results demonstrate that dialog-final ratings vary with respect to the degradation profile, revealing a recency effect and a strong impact of individual bad samples. Two related models which implement these findings are presented. With these models, dialog-final quality ratings can be estimated significantly better than by plain averaging of short sample ratings (about 10% absolute improvement). They also perform better than two algorithms taken from literature. Both models can be applied to the instrumental method described in ITU-T Rec. P.862 [1], resulting in about 13% absolute improvement. They were evaluated with the results of two different experiments, which were performed independently but on the basis of our test procedure. In these experiments similar profiles but a different type of quality degradation, different sample durations, and different speech material were used. The models proved to be valid and reliable for the time span investigated (1―2 minutes) and for the profiles used. One of them is now being recommended by the ETSI STQ mobile group.
TL;DR: In this paper, a two-part study of the quality of car horn sounds is presented, focusing on the agreement among listeners in categorizing sounds as being members of the car horn category.
Abstract: This paper is the second of a two-part study of the quality of car horn sounds. It aims to provide insights into the design of new sounds. It is based on the assumption that hearing a car horn sound warns road users because they recognize the sound of a car horn, i.e. they know what this sound means, and what they have to do as a consequence. The three experiments reported in this paper are grounded in a psychoacoustical framework. They seek to provide car horn builders with recommendations allowing them to create new sounds. In the first part [1], we studied the perception of the timbre of existing car horn sounds.We found that, from their perception of the sounds, listeners were able to make inferences concerning the different mechanisms causing the sound, and that the perceived differences between the sounds were based on the integration of three elementary sensations, correlated with three acoustical descriptors. In this second part, we focus on the agreement among listeners in categorizing sounds as being members of the car horn category. Membership agreement is operationally defined as the result of a two-alternative forced-choice task. We first study recordings of existing sounds. The results allow us to define relationships that predict membership agreement from a set of acoustical descriptors. To extend these results, we create a new set of sounds in a second step, which we submit to a timbre study similar to the one reported in [1]. We finally study membership agreement
TL;DR: In this paper, a high resolution method, called ESPRIT, is used to separate the spectral components which are very close one to another, and the eigenfrequencies and mode shapes of these modes are investigated using a suitable model of the instrument.
Abstract: The concert harp is composed of a soundboard, a cavity with sound holes and 47 strings. When one string is plucked, other strings are excited and induce a characteristic 'halo of sound'. This phenomenon, called sympathetic vibrations is due to a coupling between strings via the instrument's body. These sympathetic modes generate the presence of multiple spectral components in each partial of the tone. Resolution of Fourier analysis does not permit their identification. A high resolution Method, called ESPRIT, is used to separate the spectral components which are very close one to another. Some of the measured spectral components in the analysed partials correspond to the response of sympathetic modes. The eigenfrequencies and mode shapes of these modes are investigated using a suitable model of the instrument : this model is based on a waveguide approach in which bending and longitudinal motions of 35 strings connected to an equivalent beam representing the soundboard are described. Identified experimental sympathetic modes are very well captured by the model.
TL;DR: In this paper, a sensor has been developed which allows measurement of the force exerted by the bow on the string (bow force) during violin performance, based on the transversal sensors.
Abstract: A sensor has been developed which allows measurement of the force exerted by the bow on the string ( bow force) during violin performance. The bow force is deduced from measurement of the transvers ...
TL;DR: In this article, a statistical analysis of head, torso, and pinna dimensions of subjects including children and adults of all ages has been conducted for the first time to create a detailed database.
Abstract: This study deals with the question of how much influence the head, torso, and pinna dimensions have on spatial hearing, particularly as far as children and their growth are concerned. A statistical analysis of head, torso, and pinna dimensions of subjects including children and adults of all ages has been conducted for the first time to create a detailed database. Spatial cues are derived from CAD 1 models of the head using the Boundary Element Method (BEM) with a basic simplified model. Anthropometric parameters for the head and torso (height, breadth, length, and the distance from the ear to the shoulder) and for the pinna (height, breadth, and rotation angle, and cavum concha height, depth, and breadth) are analyzed and their effects are discussed. Varying each parameter individually makes it possible to isolate the effect of this parameter on the HRTF and on the spatial cues. The same approach is taken for various incidence angles. As far as the future construction of artificial heads for specific groups of population, including various ages, is concerned, the results presented here show where larger tolerances are acceptable regarding the different dimensions and where the dimensions should be chosen more accurately.
TL;DR: In this article, the authors investigate the temporal distribution of arrivals and the estimation of mixing time, which is defined as the time it takes for initially adjacent sound rays to spread uniformly across the room.
Abstract: In Room Acoustics, the quantity that fully describes the hall is a set of room impulse responses (RIRs), which are composed of the succession of arrivals (i.e., some sound rays which have undergone one or more reflections on their way from the source to the receiver). The mixing time is defined as the time it takes for initially adjacent sound rays to spread uniformly across the room. This paper proposes to investigate the temporal distribution of arrivals and the estimation of mixing time. A method based on maxima of correlations (Matching Pursuit) between the source impulse and the RIR allows to estimate in practice arrivals. This paper compares the cumulative distribution function of arrivals of experimental and synthesized RIRs (using a stochastic model). The mixing time is estimated when the arrival density becomes constant. The dependence of mixing time upon the distance source/receiver is investigated with measured and synthesized RIRs.
TL;DR: The string player controls variations in spectral content mainly via bow velocity, bow-bridge distance and bow force, and many combinations of the bowing parameters influence the pitch noticeably as we...
Abstract: The string player controls variations in spectral content mainly via bow velocity, bow-bridge distance and bow force. Many combinations of the bowing parameters influence the pitch noticeably as we ...
TL;DR: It is shown that the proposed MSO and LSO models are sensitive to interaural differences in time and level in a way that accounts for some known psychoacoustical phenomena.
Abstract: Some recent neurophysical studies suggest that mammalian binaural decoding is based on count comparison. When a signal is presented earlier or with higher level to one ear, the neural signals are stronger in the auditory pathways leading to the contralateral hemisphere in such mechanisms. This paper describes functional count-comparison models of two brainstem nuclei, medial superior olive (MSO) and lateral superior olive (LSO), both of which exist in both hemispheres. The topology of the organs and the connections between them as presented in the current neuroanatomical studies are imitated in the functional model. The parameters of the functional models are selected to fit existing neurophysiological and psychoacoustical data. It is shown that the proposed MSO and LSO models are sensitive to interaural differences in time and level in a way that accounts for some known psychoacoustical phenomena.
TL;DR: In this paper, the authors compare the acoustics of the Herodes Odeion in its current open-air form to a detailed acoustic model reconstruction of its original roofed version and illustrate the significant differences in acoustic properties between the two spaces.
Abstract: Ancient Greek / Roman odeia were semi-enclosed theaters that currently survive without their original roof sections. The work compares the acoustics of the well-preserved Herodes Odeion in its current open-air form to a detailed acoustic model reconstruction of its original roofed version and illustrates the significant differences in acoustics between the two spaces. It is shown that in their original state, the odeia had acoustics appropriate for music performances in contrast to their current open-air form that have acoustic properties appropriate for speech reproduction, similar to the larger open-air theaters of the time which were used specifically for ancient drama performances.
TL;DR: In this paper, an alternative finite element formulation for poroelastic materials modelled with the Biot-Allard theory is presented, based on the solid and total displacement fields of the porous medium.
Abstract: The use of finite element modeling for porous sound absorbing materials is often limited by the numerical cost of the resolution scheme. To overcome this limitation, an alternative finite element formulation for poroelastic materials modelled with the Biot-Allard theory is first presented. This formulation is based on the solid and total displacement fields of the porous medium. Three resolution methods (one semi-analytical and two numerical) based on normal modes are proposed secondly. These methods take benefit from the decoupling properties of normal modes. The semi-analytical method is associated with problems in which the shear wave can be neglected. The numerical methods are a direct and an iterative scheme. The direct method allows a reduction by 2 of the number of degrees without making any approximation. The iterative method provides an approximation corresponding to a controlled tolerance. The finite element formulation is validated by comparison with an analytical model in two mono-dimensional configurations corresponding to a single and a multilayered problem. The efficiency of the two numerical resolution methods is also illustrated in term of computation time in comparison with classical formulations, such as the mixed displacement-pressure formulation.
TL;DR: In this article, a modification to the room-acoustic diffusion model is proposed to take different amounts of wall scattering into account, and an empirical law can be proposed between the diffusion constant and the scattering coefficient.
Abstract: In this paper, a modification to the room-acoustic diffusion model is proposed to take different amounts of wall scattering into account. An extensive set of numerical simulations using a cone-tracing software has first been carried out, in order to highlight the impact of the scattering coefficient on the diffusion process in rooms, in terms of sound pressure levels. An iterative method is then proposed to identify, for a given value of the wall's scattering coefficient, the diffusion constant that allows the stationary sound field to be governed by a diffusion process, regardless of the room's geometry. Using this method, an empirical law can be proposed between the diffusion constant and the scattering coefficient. The empirical diffusion model is then compared to scale model experiments, as well as to other models from the literature, with a satisfactory agreement for the sound pressure level. However, the empirical diffusion model fails to predict the sound decay for rooms with perfectly specularly reflecting surfaces, due to the inherent concept of a diffusion process.
TL;DR: Pheasant et al. as discussed by the authors reported the results of a study that set out to validate and further develop a Tranquillity Rating Predication Tool, which utilized the percentage of natural features contained within a scene and either of the noise indices LAmax or LAeq.
Abstract: Summary In both urban and rural environments, ‘tranquil space’ is predominantly constructed through the sensory information received by the auditory and visual modalities. This paper reports the results of a study that set out to validate and further develop a ‘Tranquillity Rating Predication Tool’, previously proposed by the authors [Pheasant et al. JASA 123(3), 1446‐1457 (2008)], which utilises the percentage of natural features contained within a scene and either of the noise indices LAmax or LAeq, as the key components. The results of the validation study show that when responding to combined audio-visual stimuli the A-weighted sound pressure level LAeq is most significantly related to the Tranquillity Rating. It is suggested that a modified measure of the visual composition of a scene incorporating contextual aspects is required. The findings of this work should be of particular interest to those charged with landscape management, such as the National Park Authorities, Regional Councils, and other agencies concerned with providing and maintaining public amenity.
TL;DR: In this paper, the European Directive 2002/49/EC requires an estimation of the number of people living in dwellings that are exposed to different bands of Lden values in dB 4 m above the ground on the most exposed facade.
Abstract: In noise mapping, the European Directive 2002/49/EC requires an estimation, to the nearest hundred, of the number of people living in dwellings that are exposed to different bands of Lden values in dB 4 m above the ground on the most exposed facade. Even though this methodology always overestimates the effects caused by noise, this estimation is accurate enough for detached or semidetached houses in urban areas. However, for large and tall buildings in urban configuration, this estimation could be confusing and even very vague. In fact, differences regarding the percentage of people affected by noise taken from an exact calculation and from European Directive criterion could be unacceptable. When facade noise map calculation is not possible, either due to being too time consuming or unavailability of software resources, estimating people affected by noise based on the nearest grid point approximation criterion is clearly preferable to the European Directive approach.
TL;DR: In this article, a wind field model was coupled with a new method for predicting sound from vegetation in order to examine the sound generated when the wind interacts with vegetation and examine the effect of the wind on the vegetation.
Abstract: This article examines the sound generated when the wind interacts with vegetation. A wind field model has been coupled to a new method for predicting sound from vegetation. This includes prediction ...
TL;DR: In this article, a case study of Saga urban areas, the analysis of urban soundscape was performed by means of soundscapegraphy, which was composed of objective information, subjective information, environmental information and analysis information.
Abstract: The establishment of the concept of soundscape has brought about a new aspect for sound environment, and many conceptual and practical researches have been performed around the world. However, the notation of soundscape, in other words, the way how we can record, describe or express the soundscape, still need to be studied furthermore. Soundscape is the concept of an environment of sound (sonic environment) with emphasis on the way how it is perceived and understood by individual, or by a society, and thus it depends on the relationship between individual and any such environment. Therefore, the notation of soundscape should put emphasis on not only the physical characteristics but also the perception of the people, as well as the relationship between them. In this paper, firstly the notation of soundscape was reviewed with the concept of soundscapegraphy, which was composed of objective information, subjective information, environmental information and analysis information. Then, the methodology of making soundscapegraphy was discussed from the stage of data collection, data processing, mapping to analysis, and in the data processing GIS technique was applied. Next, with the case study of Saga urban areas, the analysis of urban soundscape was performed by means of soundscapegraphy. Finally, the applications of soundscapegraphy and the prospect for future research were discussed. Soundscapegraphy can be a proper and applicable method for the notation and analysis of soundscape.
TL;DR: In this article, three different mechanisms accentuating temporal onsets of sounds were tested: adaptive compression, adaptive auditory filters and bandwidth-dependent integration, and a comparison between simulations and experimental data indicated that, in principle, all mechanisms lead to increased spectral loudness summation for short noise bursts.
Abstract: Recent studies have shown that spectral loudness summation depends on duration. Modifications of a current loudness model were investigated with respect to their ability to predict this effect. The original version of the model could not simulate the duration dependence of spectral loudness summation. To reconcile the model with the loudness data, three different mechanisms accentuating temporal onsets of sounds were tested: (i) adaptive compression, (ii) adaptive auditory filters and (iii) bandwidth-dependent integration. A comparison between simulations and experimental data indicated that, in principle, all mechanisms lead to increased spectral loudness summation for short noise bursts, but bandwidth-dependent integration may be the most realistic approach. Such a modified model also predicts the spectral loudness summation of repeated noise bursts as a function of repetition rate.
TL;DR: In this paper, the use of acoustic diffusors to reduce noise in urban areas is investigated by computer simulation with a sound particles tracing based code, which allows to model the sound reflection, diffusion and absorption by building facades with any arrangement of acoustic Diffusors.
Abstract: The use of acoustic diffusors to reduce noise in urban areas is investigated by computer simulation. The numerical simulations have been performed with a sound particles tracing based code, which allows to model the sound reflection, diffusion and absorption by building facades with any arrangement of acoustic diffusors. Several parameters have been studied, among which, the diffusion pattern, the relative surface on the facades, the arrangement and the size of diffusors. In addition, the combined effect of absorption and diffusion by diffusors has also been investigated. The numerical results show that a significant noise reduction can be observed on the sound attenuation as well as on the sound decay in a street, by considering diffusors creating uniform diffusion, with a relative surface larger than 20 %, and if possible locating on the lower part of the facades. Lastly, and as expected, by adding sound absorption on diffusors, it increases the noise reduction.