scholarly journals ACOUSTIC CHARACTERISTICS OF PULSED SOUND SOURCES OF VARIOUS TYPES/ĮVAIRIŲ TIPŲ IMPULSINIŲ GARSO ŠALTINIŲ AKUSTINĖS CHARAKTERISTIKOS

1998 ◽  
Vol 4 (4) ◽  
pp. 311-315 ◽  
Author(s):  
Vytautas Stauskis ◽  
Vytautas Kunigėlis
Keyword(s):  
Pulse Duration ◽  
Sound Source ◽  
Short Pulse ◽  
Sound Field ◽  
Correct Selection ◽  
Sound Sources ◽  
Major Drawback ◽  
Maximum Width ◽  
Acoustic Characteristics ◽  
Sound Energy

The paper examines the acoustic characteristics of explosion-type pulsed sound sources of four types. These include a Calibre 8 sound gun, a start gun, a Calibre 16 hunting gun, and a toy gun. The latter was included both because of its short pulse duration and for comparison purposes. Correct selection of a source is very important because it largely determines the results of acoustic measurements. Certain requirements are set for a sound source. In order to concentrate as much energy as possible at the given moment, the signal bandwidth-duration product must be as large as possible. The range frequencies to be excited depend on the pulse duration. The latter also determines whether interference phenomena will occur in the room and whether individual reflections will merge. The experiments were conducted in a room of 12 m2. The distance between the microphone and the pulsed sound source was 1 m. The structure of reflections depends on the pulse by means of which the sound field is excited. The smallest number of reflections is generated by a sound source. During a 20 ms experiment, the amplitudes of these reflections almost coincided with the direct sound amplitude. A sound gun emits more sound energy than other pulses. When the sound field is excited by means of a start gun and a hunting gun, the reflection structure, by amplitude, is very different from that produced by a sound gun. A dense reflection structure is formed by a toy gun but it emits less energy. The structure of reflections generated by a hunting gun is acceptable but its shots are very unstable, which is a major drawback in an experiment. The shots from a sound gun differ only by about 0.1% among themselves by amplitude, ie they are sufficiently stable. Among the four sound sources, the best reflection structure is produced by a sound gun. A sound gun is characterised both by the longest pulse duration (about 0.55 ms) and the highest levels of energy emitted. The pulse duration of the rest three guns is almost equal and is about 0.15 ms, ie is 3.6 times shorter than that of a sound gun. The forms of signals emitted by these sound sources are also very different. The spectrum of a sound source was established on Fourier transformation basis. The spectrum is largely dependent on the type of a gun by means of which the sound field is excited. The maximum width of the spectrum generated by a sound gun occupies almost two octaves, from 500 to 2000 Hz, and the radiation in this range is quite uniform. The spectra of a start gun and a hunting gun are similar but these guns emit less sound energy than a sound gun. The structure of reflections generated by them is also quite different. A toy gun radiates energy in a less narrower band, the width of which occupies about a half of octave, with a maximum at 2000 Hz. This is not very good because too small quantities of low- and medium-frequency sound energy are radiated.

Optimization of the Sound Source Position for Shaping Acoustically-Structurally Coupled Cavities

2001 ◽  
Author(s):  
Arzu Gonenc Sorguc ◽  
Ichiro Hagiwara ◽  
Qinzhong Shi ◽  
Haldun Akagunduz
Keyword(s):  
Sound Source ◽  
Sequential Quadratic Programming ◽  
Normal Modes ◽  
Sound Field ◽  
Source Strength ◽  
Source Position ◽  
Sound Sources ◽  
Coupled Cavities ◽  
Structural Loading ◽  
Initial Starting

Abstract In this study, sound field inside acoustically-structurally coupled rectangular cavity excited by structural loading and sound sources is shaped by optimizing the position of the sound source. In the optimization, Most Probable Optimal Design (MPOD) based on Holographic Neural Network is employed and the results are compared with Sequential Quadratic Programming (SQP). It is shown that source position, rather than source strength, is more effective in acoustically controlled modes. The nodal positions for in-vacuo acoustical normal modes are good candidates for initial starting points.

Study of Spherical Near-Field Acoustic Holography with Rigid Spherical Microphone Array

2013 ◽  
Vol 546 ◽  
pp. 156-163
Author(s):  
Xin Guo Qiu ◽  
Ming Zong Li ◽  
Huan Cai Lu ◽  
Wei Jiang
Keyword(s):  
Sound Source ◽  
Microphone Array ◽  
Near Field ◽  
Sound Field ◽  
Acoustic Holography ◽  
Real Situation ◽  
Helmholtz Equations ◽  
Sound Sources ◽  
Array Configuration ◽  
Sphere Radius

The aim of this paper is to investigate the impacts of various parameters of rigid spherical microphone array in detecting and locating interior sound source. Helmholtz Equations are adopted to express the sound field produced by the incident field and scattered field. The gradient of the pressure is zero at the surface for the sphere is rigid. Both the incident and scattered coefficient could be obtained by solving the Helmholtz Equation using the boundary condition. Then the interior sound field could be detected and located on with the methodology of spherical near-field acoustic holography (SNAH). This study is developed in two aspects，one is configuring the microphone in various distribution in the same sphere radius, and the other one is changing the radius of sphere array. Numerical simulations are carried out to determine the optimum microphone array configuration and structure parameters. One, two, and three sound sources are arranged respectively in different displacement to the sphere center and in different angle direction to simulate the real situation. During the experiments, Omni-directional speakers and beeps are adopted as sound sources. The result shows that the method to detect and locate sound source in interior sound field is valid.

The Influence of the Shape Space on the Scattering of the Sound Energy

2013 ◽  
Vol 855 ◽  
pp. 237-240
Author(s):  
Alena Pernišová ◽  
Dušan Dlhý
Keyword(s):  
Sound Source ◽  
Sound Propagation ◽  
Shape Space ◽  
Limited Range ◽  
Sound Level ◽  
Sound Sources ◽  
Sound Energy ◽  
Empty Area ◽  
The Way

The sound level adjacent to the sound sources is mainly characterized by the straight sound. The dispersion sound ratio is increasing with distance increasing and within the limited range round the sound source, the sound level is higher, in the area with dispersion bodies even higher than in an empty area. The laws of sound propagation in empty areas are derived on classical geometric base. The laws of sound propagation in large areas with dispersion objects are also derived from these laws complemented with the Kuttruff ́s equation of reverberation process in media with dispersion bodies. Simultaneously the sound energy is according to the purpose divided into the straight sound and the reverberation. The straight sound is the energetic ratio of sound, that is during the way to destination not dispersed and the propagation laws are equal to empty areas propagation laws are equal to empty areas propagation laws. The dispersed sound is the ratio of sound energy that reaches the destination after one or more reflections. The energy result is then the sum of densities of dispersion and straight sound.

Reproduction of Phantom Sources Improves with Separation of Direct and Reflected Sounds

2015 ◽  
Vol 40 (4) ◽  
pp. 575-584
Author(s):  
Piotr Kleczkowski ◽  
Aleksandra Król ◽  
Paweł Małecki
Keyword(s):  
Sound Source ◽  
Sound Field ◽  
Angular Separation ◽  
Impulse Responses ◽  
Sound Sources ◽  
Perceptual Effect ◽  
Multichannel Systems ◽  
Virtual Acoustics

AbstractIn virtual acoustics or artificial reverberation, impulse responses can be split so that direct and reflected components of the sound field are reproduced via separate loudspeakers. The authors had investigated the perceptual effect of angular separation of those components in commonly used 5.0 and 7.0 multichannel systems, with one and three sound sources respectively (Kleczkowski et al., 2015, J. Audio Eng. Soc. 63, 428-443). In that work, each of the front channels of the 7.0 system was fed with only one sound source. In this work a similar experiment is reported, but with phantom sound sources between the front loud- speakers. The perceptual advantage of separation was found to be more consistent than in the condition of discrete sound sources. The results were analysed both for pooled listeners and in three groups, according to experience. The advantage of separation was the highest in the group of experienced listeners.

The acoustic characteristics of the “Dives in Misericordia” Church in Rome

2020 ◽  
pp. 1351010X2094865
Author(s):  
Giuseppe Ciaburro ◽  
Gino Iannace ◽  
Amelia Trematerra ◽  
Ilaria Lombardi ◽  
Maurizio Abeti
Keyword(s):  
Side Wall ◽  
Sound Field ◽  
Reverberation Time ◽  
Acoustic Measurements ◽  
Internal Length ◽  
Maximum Width ◽  
Acoustic Characteristics ◽  
Current Configuration ◽  
Double Curve ◽  
The Church

This paper discusses the acoustic characteristics of the “Dives in Misericordia” Church in Rome. The church was designed by architect Richard Meier and opened in 2003. It was made entirely of white concrete and consists of three septa with a double curve shaped like a sail. The nave roof is glass. The volume is approximately 14.000 cubic meters. The highest measuring is approximately 26 m. the width of the nave is 19.5 m, while the maximum width is 29.5 m, while the internal length is 32.0 m, while the total length is 45.6 m. It can seat approximately 240 people. The acoustic measurements were taken by placing a microphone at different points of the nave (the area occupied by the audience), with the sound source being placed on the altar. It was therefore possible to obtain a spatial distribution of the average acoustic characteristics inside the church. At a frequency of 1000 Hz, the average values of the reverberation time is about 10 s. In its current configuration, the church is neither suitable for understanding speech nor listening to music. A 3D virtual model was created and with the help of the building acoustics software it was possible to study the sound field inside the church. The possibility to carry out an appropriate acoustic correction was analyzed, in order to reduce the values of the reverberation time, by pacing on a side wall of the church an adequate number of sound-absorbing polyester panels.

scholarly journals THE INFLUENCE OF ADDITIONAL ABSORPTION ON THE ACOUSTICAL INDICATORS OF A HALL/PAPILDOMOS ABSORBCIJOS ĮTAKA SALĖS MODELIO AKUSTINIAMS RODIKLIAMS

1998 ◽  
Vol 4 (1) ◽  
pp. 86-90
Author(s):  
Vytautas Stauskis
Keyword(s):  
Sound Source ◽  
Sound Absorption ◽  
Sound Field ◽  
Reverberation Time ◽  
Frequency Range ◽  
Time Intervals ◽  
Scaled Model ◽  
Additional Absorption ◽  
Low Frequencies ◽  
Sound Energy

The influence of the slits between the walls and the floor of the model upon the objective acoustical indicators was examined in a scaled model of a hall. The Small Hall of the Lithuanian National Philharmonic Society was selected for the investigations. The hall is of rectangular form, 13.6 m in length, 10.7 m in width and 7 m in height. The hall model was scaled 1:25. The floor and the ceiling of the model were made of cloth-based laminate, while the walls of plywood 8 mm thick, with three layers of varnish. Thus, all materials employed in the model were similar to those of the real hall by their sound-absorption properties. There were 1 to 3 mm slits between the floor and the walls of the model. Their overall length was about 10–12 m (converted to real values). A spark sound source was used for the radiation of signals within the required spectrum. The sound source was put through a hole in the floor in order to improve the directivity diagram of the radiation. The positions of both the source and the ¼ microphone coincided in all cases. The frequencies examined fell in the range between 1250 Hz and 50000 Hz. The frequency of quantization of the signal was 166.6 kHz and the quantization time was 6 mcs. All frequencies were converted into real ones in the diagrams. A 2000 Hz upper limit was established to ensure that the Nyquist frequency exceeds 3. The experiments showed that the slits in the model influenced the muffling of the sound energy starting from 200 ms. With the slits present, the muffling occurs faster and the greatest difference of 2–3 dB is observed in the interval of 1000—2500 ms. Given small slit dimensions and overall slit length, the change of 2-3 dB is quite significant. The muffling of the sound field of the model is not exponent in character. The muffling varies on differently in different time intervals. Then the reverberation times of a non-filtered signal must be different when the muffling is approximated every 10 dB. The investigation showed that, with the slits present, the reverberation time values were reduced by 0.4–0.8 s throughout the interval when the muffling was approximated every 10 dB, starting from 0 to—30 dB and from—5 to—35 dB. This means that the slits absorb the sound energy on all intervals of the muffling of the sound field. The largest sound absorption is reached when the muffling of the sound field is approximated every 10 dB from 0 to—30 dB and amounts to as much as 3-6 m2. The influence of the slits is weaker when the muffling is approximated on other intervals. The slits also produce effect upon subjective acoustical indicators of a non-filtered signal, which vary between 1 to 2 dB. This shows that the intensity of reflections is changed in various time intervals by the slits. The influence exerted by the slits over the early reverberation time manifests itself both at the low and high frequencies. The greatest difference of about 0.8 s is observed at 100 Hz and 160 Hz. Within the frequency range from 500 Hz to 1000 Hz, the difference is not so marked and amounts to about 0.5 s. Within the range from 200 Hz to 400 Hz, the early reverberation time is only slightly influenced by the slits. The effect produced by the slits on the standard reverberation time, as compared with the early reverberation time, is not significant up to 160 Hz, while in the frequency range of 200—2,000 Hz the standard reverberation time is cut by about 0.4–0.6 s. The smallest sound absorption brought about by the slits is observed at low frequencies (around 1 m2). In the frequency range of 200—500 Hz, the sound absorption amounts to 3–4 m2, and at the frequencies exceeding 630 Hz to 2–7 m2. At low frequencies, the music sound clarity index is increased by the slits by about 0.5 dB. From 200 Hz and on, the clarity index is increased by 2 to 4 dB. These results show that the slits in the model alter the intensity of the early sound reflections. Beginning with 250 Hz, the sound absorption amounts to 3.2–9.0 m2. Such absorption is already significant, therefore the slit factor must be taken into consideration while conducting investigations in the hall model.

scholarly journals Influence estimation of the inclination angle of the top of the noise protection barrier on its efficiency

2021 ◽  
Vol 1 (1(57)) ◽  
pp. 12-16
Author(s):  
Vitaly Zaets
Keyword(s):  
Inclination Angle ◽  
Sound Source ◽  
Sound Field ◽  
Wide Frequency Range ◽  
Sound Insulation ◽  
Frequency Range ◽  
Noise Barriers ◽  
Sound Sources ◽  
Sound Levels ◽  
Number Of Publications

The object of research is the sound field from linear sound sources around a rounded noise barrier of the same height and different angles of inclination of the top part of the barrier. It is known that the effectiveness of noise protection barriers depends primarily on the geometric dimensions of the barrier and the relative position of the sound source, barrier and area of noise protection. A large number of publications have been devoted to the study of the influence of these factors and some others, such as the influence of the earth's surface, sound absorption, sound insulation of the barrier. However, these works did not study the effect of the angle of the top part of the barrier on the change in the barrier efficiency. In this paper, the reduction of sound levels from linear sound sources around noise barriers with different inclination angle of the top part of the barrier is investigated. Rounded barriers of the same height with different radii are considered, which made it possible to simulate barriers in which the top part of the barrier has a different inclination angle. An effectiveness of such barriers for various locations of the sound source, which could also affect the establishment of a pattern of changes in the effectiveness of barriers, is also considered. In addition, the results were analyzed over a wide frequency range. The calculation of the field around such a barrier was carried out using computer simulation using the finite element method. This method allows to easily change the geometric parameters of the barrier and the position of the sound source. The barriers were considered acoustically hard. Thus, an influence of the inclination angle of the top part of the barrier on the sound field around the barrier from various locations of sound sources in a wide frequency range is analysed. The results must be taken into account when designing noise barriers to reduce noise levels from traffic flows

scholarly journals Binaural (pre)processing for contralateral sound field attenuation and improved speech-in-noise recognition

2021 ◽  
Author(s):  
Enrique A. Lopez-Poveda ◽  
Almudena Eustaquio-Martín ◽  
Fernando Martín San Victoriano
Keyword(s):  
Sound Source ◽  
Transfer Functions ◽  
Signal To Noise Ratio ◽  
Free Field ◽  
Sound Field ◽  
Directivity Pattern ◽  
Sound Sources ◽  
Speech In Noise ◽  
Speech Reception ◽  
Speech Reception Thresholds

ABSTRACTUnderstanding speech presented in competition with other sound sources can be challenging. Here, we reason that this task can be facilitated by improving the signal-to-noise ratio (SNR) in either of the two ears and that in free-field listening scenarios, this can be achieved by attenuating contralateral sounds. We present a binaural (pre)processing algorithm that improves the SNR in the ear ipsilateral to the target sound source by linear subtraction of the weighted contralateral stimulus. Although the weight is regarded as a free parameter, we justify setting it equal to the ratio of ipsilateral to contralateral head-related transfer functions averaged over an appropriate azimuth range. The algorithm is implemented in the frequency domain and evaluated technically and experimentally for normal-hearing listeners in simulated free-field conditions. Results show that (1) it can substantially improve the SNR (up to 20 dB) and the short-term intelligibility metric in the ear ipsilateral to the target source, particularly for speech-like maskers; (2) it can improve speech reception thresholds for sentences in competition with speech-shaped noise by up to 8.5 dB in bilateral listening and 10.0 dB in unilateral listening; (3) it hardly affects sound-source localization; and (4) the improvements, and the algorithm’s directivity pattern depend on the weights. The algorithm accounts qualitatively for binaural unmasking for speech in competition with multiple maskers and for multiple target-masker spatial arrangements, an unexpected property that can inspire binaural intelligibility models.

scholarly journals Research of Space Positioning Method Based on Sound Field HBT Interference

2018 ◽  
Vol 232 ◽  
pp. 04028
Author(s):  
Jing Zou ◽  
Lei Nie ◽  
Mengran Liu ◽  
Chuankai Jiang
Keyword(s):  
Source Localization ◽  
Sound Source ◽  
Target Location ◽  
Sound Propagation ◽  
Sound Field ◽  
Long Distance ◽  
Sound Sources ◽  
Positioning Method ◽  
New Ideas ◽  
Propagation Media

Based on Hanbury Brown-Twiss (HBT) interference in the sound field, a space positioning method is presented to realize the long-distance and high-precision positioning of sound sources in media. Firstly, theoretical model of HBT interference positioning is established. Location of the sound source can be acquired by analyzing the correlation function of the output signals. Then, sound source localization under different signal-to-noise ratios (SNR) shows that by this method, the sound source can be accurately found with six sensors (two arrays) even the SNR is low to 0.04. Positioning experiment in air is carried out, and the experimental results show that the sound source can be accurately located at 42 meters, and the positioning error is low to 0.1 meters. Thus the validity and accuracy of the HBT interference space location principle is demonstrated. It provides new ideas for the research of long-range target location in sound propagation media (air, water, etc.).

Sign in / Sign up

Export Citation Format

Share Document