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I'm studying the effectiveness of ultrasound as a wolf deterrent. A first step in this, is measuring how loud the speakers that we use actually are. Decibel meters which operate in the ultrasound range are very expensive, and DIY solutions don't always seem very accurate. The solutions we are left with are either a) build our own decibelmeter and calibrator or b) find a lab (in Germany) where these types of tests are being done, so we can bring our own equipment there to test it.
We would only need to test the speakers a limited number of times, so we were thinking that building our own device might require too much time and resources.
Does anyone have any suggestions or possible solutions?
Thank you very much!
Melissa
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We bought a large diaphragm microphone, sound card and a noise source for calibration from a bioacoustics (bat) company. The setup can use a PC program for analysis And you can get out dBA values or spectrograms. However, the system costed about 4000 euro. Not far from a normal pro sound level meter.
An other route could be to use a 200 kHz soundcard and a mic working up to 60 kHz. Calibrate it at 1kHz as usual, (you need that calibrator at 1000 euro, or borrow it) and use that with a good PC sound processing software. Some sound recording enthusiasts work up to 200 kHz and have mics for that. You need a 1/2 inch (i think) for the calibrator or an adaptor. I have 1/4 inch mics from Earthworks that goes to 30 kHz with an adapter, maybe they have something going higher in frequency now. But they will be insensitive with those small diaphragms, possibly not a problem when the measurement is done close to the source.
Air damp high frequencies dependant on the relative humidity, effects that are much higher for ultrasound than audible for us. Dry air damp more than humid. The effect should be measureable by varying the distance and testing the effect against theoretical -6dB per doubling. You will get more. :-)
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Can anyone recommend a usable resource/tool for species detection from acoustic data please? A middle ground between a phone app and something like Arbimon would be about the level. Briefly, a phone app lacks flexibility in data collection, i.e. it cant be left out all night or left running for long periods. However, Arbimon is not that useful to anyone below ecologist level, as it only tells the user what species is present if the user completes their own validation, i.e. the user has to identify all species themselves. I'm looking for something that can analyse data from an AudioMoth, uploaded by citizen scientist participants, and actually identify species.
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Very interesting project.
There also are some other interesting findings when doing a google search for "training data animal sounds"
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Is there any relationship between stl and absorption coefficient?
Suppose i am getting a peak at 400Hz (90% absorption) then at the same frequency what will be the sound transmission loss(it will be minimum or maximum or we can not say).
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There is more than a relation between room acoustic and the physical composition: it is a real correspondence. – One example: A room with surrounding walls of stone with plaster (maybe with wallpaper on it, doesn't really matter). You will have a greater problem with deep frequences. Another room with walls of thin plaster plates (20 mm oder 25 mm), 50 – 80 mm mineral fleece, than again 20 - 25 mm plaster plate, and the next room. This to the right and left side + the wall to the floor. Deep frequences pass through the wall (the mineral fleece does only matter very less) and in result you have less problems or maybe no problems with deep frequences inside your room. Room acoustic is directly corresponding with physical aspects of the building. And you will have (nearly) NO chance to correct the problem with disturbance in the neighbour rooms by noise in the area of deep frequences. – Nevertheless you CAN create a good speech intelligibility and clarity of sound INSIDE the room. So or so (with thin walls of plaster plates or heavy stone walls) there is a possibility to create a good room acoustic parallel to a bad constitution of the building (causing disturbance from outside). This is (so I think) what you mean, when saying that there is a "relation but not strict correspondence" between room acoustic and the pysical behavior of a building.
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Please see attachment, here two terms appeared in the inhomogeneous Helmholtz equation. What is the use of these two terms for a propagating wave in a particular direction.
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Paul Kinsler Thank you sir for your last reply.
This form will be union.
I have one more question . I want to calculate the transmission loss through a structure having a membrane and cavity and supported by an acrylic plate.
I am simulating using comsol multiphysics.
Over membrane I have chosen solid mechanics. The air cavity is assigned pressure acoustics but what about that backing acrylic plate.
I have to assign it as pressure acoustics or solid mechanics.
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Hello everyone,
I am looking for links of audio datasets that can be used in classification tasks in machine learning. Preferably the datasets have been exposed in scientific journals.
Thank you for your attention and valuable support.
Regards,
Cecilia-Irene Loeza-Mejía
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In our work we used UrbanSound8K .
Here is our work with code provided. Do not forget to cite our work
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Hi everyone, I have a query regarding the elbow and its insertion loss captured in ASHRAE.
The elbow captured in the ASHRAE is referring to what kind dimension? I have attached three common scenarios which I have quite often encounter during FCU noise control design stage.
Looking for advise.
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You will have to find it yourself, if it is there.
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I am searching for good recordings of distress calls emitted by American crocodiles, in particular by hatchlings. The longer the better. If not, also distress call recordings of other crocodilian species are fine.
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Differences in distress: Variance and production of American Crocodile ( Crocodylus acutus ) distress calls in Belize
Visit to that article
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i am working on in which i has to trace band limited noise using Bellhop matlab but i am new in matlab and also not much familiar with acoustics can anyone please help me it's urgent..
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Hi how to work with bellhop acoustic simulator using matlab for underwater acoustic communication
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I was reading papers on muffler design and optimization and a question pupped up to me. Why are the frequency range for these researchs between 1-3000Hz?
e.g.
or
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Because the human can hear it.
mechanic parts has an eigenvalue lover then 200Hz (12.000 rpm) and this is normally enough for all design.
Over 3000Hz is only beep sound and it is not easy to excite.
More then 8000Hz is only rush, and for most of the People not perceptible.
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Dears,
I'm a spatial planning master student, I'm studying the soundscape in Jordan- Amman, the methodology I will use is varied from a questionnaire (Soundwalk) to a binaural recording, and I'm wondering if you have any recommendation for questions to be asked in the Soundwalk?
Thank You
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Hello,
Will you conduct noise level measurements at the soundwalk stops? If so, you could ask about the perceived loudness using a scoring method and then correlate the objective and the subjective results.
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I am an undergraduate working on modelling generator noise through an aperture and writing my literature review has been a little extra difficult.
I have been trying to understand the methods of noise analysis in acoustic engineering. There are four of them which are mostly used. And I need to understand each of them to be able to choose the method I would be using in my modelling.
If you have any material, book or journal that could assist me with my research work please do send in, would be very much appreciated.
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see
1-Fundamentals of noise and vibration analysis for engineers
Cambridge University Press
M P Norton, D G Karczub
Year:
2003
2-Noise and Vibration Analysis: Signal Analysis and Experimental Procedures
Wiley
Anders Brandt
Year:
2011
3-Fundamentals of Noise and Vibration Analysis for Engineers
Cambridge University Press
Norton M.P., Karczub D.G.
Year:
2003
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I want to simulate this using the Acoustic structure interaction module.
For ease this can be done by taking a single unit.
How can I do this in comsol?
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Dear Nitish Katiyar
I suggest "scatterer on substrate" mph file availbe both in COMSOL website and applican builder within the software.
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Hello,
I am trying to account for amplitude damping of acoustic vibration of a polymer disc, using Ansys software. I am currently using Harmonic Acoustics module in Ansys, which seems to be the only one allowing FSI boundary between the surrounding acoustic medium (water) and the structure (polymer disc).
I recently asked Ansys support, and turns out the effect of damping on amplitude of vibration cannot be modeled with it: https://forum.ansys.com/discussion/comment/125750#Comment_125750
Now my question is, is there any way to work around this issue by changing the mechanical properties of the polymer in my mode, to account for viscous damping of water? Like probably to calculate some kind of equivalent damping to assign to polymer material in my model instead?
Thanks in advance for your help.
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Parinaz Sadri-Moshkenani You would need to know the correct damping in order to make a 'hack' like that, so that you can fit your structural mechanics model. Or switch to COMSOL Multiphysics.
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Hello, may I know how to compute the overall sound absorption coefficient of 2 or more items together (coefficient for overall system)?
For example,
1. item 34 on the surface of item 28?
2. item 12 on the surface of item 38?
Kindly refer to the attachment to get the item.
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is this a book?
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such as Fourier , Hilbert or Laplace transforms in order to analysis and studies in the field of music acoustics.
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You my see
1-Transform methods in applied mathematics: An introduction
Wiley-Interscience
Peter Lancaster, Kestutis Šalkauskas
Year:
1996
Language:
english
2-Integral transforms in applied mathematics
Cambridge University Press
John Miles
Year:
1971
Language:
english
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Hello,
I am trying to find acoustic round-trip latency of Android smartphones. I have using the technique of where I play a beep and listen to it through the phone and then perform convolution and get time index where we get the peak value.
Issues:
Here the issue is that when we play our beep, the AEC and NS of Android smartphones cancel and attenuate the beep and it cant be seen in recorded data.
Background & Relevant Information:
Most of the applications that measure acoustic latency such as the OboeTester app etc. use VOICE_RECOGNITION (https://developer.android.com/reference/android/media/MediaRecorder.AudioSource#VOICE_RECOGNITION) mode. In this mode, no DSP is performed and we get raw data. But this excludes the latency of DSP algorithms performed by the built-in components.
What I want:
I want to find the round trip latency using VOICE_COMMUNICATION mode. In this mode, all components of AEC, AGC, and NS are activated. But this cancels our beep and we can't get accurate results.
Is there any want to find latency while AEC and NS are working. Looking forward for the solution.
Regards,
Khubaib
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Let's say I play a voice signal that will from a phone speaker and then capture it with the same phone's mic. If AEC is there, a played signal will be set as a reference signal and it will remove the captured signal. That's what I think.
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Whether Acoustic Vibrations can be used to damage / fracture the CORONA VIRUS (COVID-19) structure ?
Acoustics with resonance frequency of the COVID-19 can be used to prevent this Pandemic ?
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Please see the attachment.
1- I have to use Perfectly matched layer while using port boundary condition or not.
2- My port shape is hexagonal.
So from available options, I am choosing user defined port. But while computation it is showing error.
Plz suggest.
Thanks.
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Thank you very much sir. René Christensen
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I am pondering whether or not the vibrato traditionally present in Western classical singing (Bel Canto) bears any significant influence on perturbation or noise measures – vibrato has been defined as variance in frequency and amplitude, as well as having aerodynamic interaction with vocal-fold vibration/oscillation. I am unsure if clinical measures such as VTI, HNR, RAP, etc will be confounded by its presence in sung voice samples. The majority of literature I've looked at supports CPP(S) as one of the most robust acoustic voice measurements across a wide variety of tasks, but I am wanting to include other measures in my analysis of singing students too. Any insight or ideas? Thank you.
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Probably not. Loudness and a parameter describing the efficiency of the singers formant may be relevant. Intonation as well.
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I have modelled a unit-cell of an anechoic structure using ANSYS. The layer is silicone and has been assigned as an acoustics physics region, with an air cavity inside (the air has been modelled and also assigned as an acoustics physics region). There is a steel backplate as well that is structural. In order to simulate the water closure on the silicone layer face I have applied an impedance boundary condition, assigned a port to this front surface and used the body of the same layer to assign the inside surface bodies. A planar wave has been applied using a 'port in duct' excitation condition (with 10000 Pa) and the acoustic absorbance is calculated.
I have used the same material parameters that I have seen in many papers, however I am getting very different results and wondered if anybody could please highlight where I have modelled it incorrectly or explain the observed behaviour? I have included an image of the absorption coefficient with regards to frequency in a 0-6000 kHz range.
Thank you for any help.
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Hi
A hand sketch or similar of the setup would help.
Impedance BC only apply for normal angle of incidence. They work on boundaries at a distance from the source, say one wavelength out or so and then, still are reflecting.
As per usual - the advice when tweaking models - start with simulating a Kundt's tube experiment first, as you there can use impedance BCs. Move to 3d from there.
/C
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I am a final year university student and for my honours project I am studying the effects of the size of air-filled cavities in an anechoic layer. The layer is made of silicone and has a steel plate backing. It is surrounded by a water enclosure, the surfaces in the x and y planes of the anechoic layer have a frictionless support applied to them to stop displacement in the normal directions. I am however having trouble with the remainder of the boundary conditions, namely the mesh sizing and whether I have to use commands to apply specific nodal values (using FLUID 30?).
Any help would be greatly appreciated. Thank you.
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Hi
The rule of thumb is seven nodes per wavelength for free wave propagation and nine nodes per wavelength for reactive fields.
By the sound of it, try to use nine close to the body and less dense away from it.
/C
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Good afternoon everyone,
I'm planning on studying and comparing the amount of underwater "noise" in different areas, and later, potentially, linked it with marine mammals occupancy. However I was wondering, what is the most suitable way to do it? Would it be by doing transects and dragging a hydrophone behind a sailing boat, or by submerging a few at different locations? What would be the advantages or disadvantages of both methods?
We would like to sample these oceanic sounds at three different times as well.
Thank you in advance!
G.M.
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Towing is not advisable if you can avoid it due to costly ship/boat time, bad weather, and especially flow noise on hydrophones.
There are many commercial battery-powered buoys in wide use that can log underwater acoustic data for months. Check out the AMAR buoys at JASCO Canada for instance, which I think may include marine mammal call recognition software (https://www.jasco.com/amar-g4-specifications).
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I am working on acoustic characterization of microbubbles using two transducers arranged at 90 degrees. During calibration, I am receiving response in dBm or mV. How I can know the PNP of my acoustic pulse through it??
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Ijaz Durrani
Thanks but for all that, I need needle hydrophone as it can only give sensitivity in V/kPas.
Is there any way to calibrate transducer without needle hydrophone?
And are they available in Pakistan? As import takes time and I need it urgent.
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I did an experiment on a glass composite thin plate and generated Lamb wave. I increased the stiffness of the plate by forming a sandwich structure. I could find Ao mode with very attenuated amplitude but So is very diminishing. Confused, as per my understanding, I should get both modes frequencies even if the amplitude is small. But only Ao I could obtain its frequency but So not.
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The dimensions of the thin plate, the configuration of shaker or input device, support and/or constraint of the specimen during testing, the location or distribution of area being measured for signal to test for Ao & So, as well as what transducer(s) or optical technique(s) is/are being used for those measurements, and what circuits and software are being used to condition and process the signal after acquisition.
The are many reasons that may be causing attenuation or lack of effective detection or isolation of the desired or expected signals, including that they actually may not be there at sufficient energy level for effective detection. Are you sure your estimated values for So of the increased-thickness sandwich plate are reasonable and don't actually predict the lower So values? Are you measuring 'lamb' waves on one skin of the sandwich only or of the entire plate (i.e. do you have transducers on both surfaces of the plate with careful and accurate reproduction of phase in both cases, and relative to one and other? Do you isolate or effectively filter frequency or wavelength anywhere in the process of your data collection, conditioning and/or post-acquisition processing? Are the 'lamb' waves on the skin or in the overall plate, what is the ratio of characteristic wavelength to plate thickness and how did this change by your structural alteration?).
Without knowing more of your setup, you experiment and your instrumentation, it is difficult or impossible to offer meaningful guidance.
In terms of a general vibration measurement problem, what signal are you expecting and why? Does the measured value represent something that seems impossible or not, it the result within or nearly within the expected range, and if not, is there something in the method or instrumentation or conditioning/processing that might explain any observed discrepancy? Hope this helps.... -TH
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I'm in the process of buying new equipment to record the vocalizations of wild mammals, and have used Marantz's solid state recorders with great success in the past. However, it seems that their PMD range of recorders are not easily available anymore. I've looked around and the Tascam DR-1 portable solid state recorder seems like a good option, but I have no experience with that product.
I'd appreciate any advice on the best, portable recorders to be used in the field. Nothing too fancy (I don't expect ultrasonic vocalizations, for example), but I'd love to hear from you about the pro's and cons of different recording tools that you've used. I would need the recordings to be of good enough quality to analyze properly in programs like AviSoft.
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Hi Alina,
I have used many recorders throughout my career and it wasn't until I discovered the Sound Devices Model 722 that I stopped buying other field recorders. It is a two-channel digital recorder (they also make a 4-channel model), made in the US, with a wide, flat frequency response that enables you to record up to the high-frequency limit of your microphone. So if we want to record ultrasound, we use an ultrasonic microphone. If we don't need this capability, we use a standard directional mike that drops off at 18 kHz or so. The microphone signal is digitized at a sampling rate selected by the user via a menu. The digitized signal is stored as a .wav file on the internal compact flash card. When this is filled, you can easily transfer the contents of the CF card to the internal 40 GByte hard drive. This is a small, lightweight recorder that can run on rechargeabale batteries that give a recording time of 6-12 h. Take a look at their website- I think you will be pleased. By the way, I have no connection to this company!
Hope this helps-
Peter
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When some of the call properties show negative correlation with temperature . Is it necessary to perform temperature correction of such properties ? What are the steps involved in the correction.
Platz and forester, 1988 gives the formula D14 = Damb - (Tamb. -14.0) (-0.0974) . Can we apply our desired temperature here (Eg. 20 °C ? Or is there any better way to perform this correction
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Hi Vineeth,
In the paper
Narins PM (1995) Temperature dependence of auditory function in the frog. In: Advances in Hearing Research (GA Manley, GM Klump, C Köppl, H Fastl, H Oeckinghaus eds.) World Scientific Publishers, Singapore 198-206, the authors present examples of how temperature afects frog call parameters. The following is one example from this paper of how to calculate the effect of a 10 degree C temperature change (the thermal Q10) for a frequency shift of s octaves:
In all frog species tested to date, increasing the temperature results in an upward shift in the CF and a concomitant reduction in CF-threshold of the tuning curves for fibers innervating the amphibian papilla (low- and mid-frequency fibers). In contrast, basilar papilla (high-frequency) fibers appear to have temperature-independent CFs and CF-thresholds.
Temperature induced shifts of CFs varied from 0.08 octaves/degree C for low-frequency fibers to no CF shift for high-frequency BP fibers. The frequency shift/degree C may be expressed as a thermal Q10 dB value to simplify comparisons with other temperature- dependent processes (Eatock and Manley,1981). The thermal Q10 for a frequency shift of s octaves may be calculated by:
Q1O = e**s (10/delta T)ln2 where ** means "raised to the power of."
Hope this is useful,
Peter
PS- If you want just want to compare frog calls recorded at different temperatures, then it is useful to shift all the temporal parameters to a common temperature (say 20 degrees C) for a valid comparison.
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I'm curious to know if one can find the force of the collision by analyzing the audio or extracting features from the audio signal that estimates the force.
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I'm looking for any kind of legislation, zoning, best practices, etc. either required or recommended by the state, the EU/other federation or local government.
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Dear Yael,
Wind turbines emit a relatively weak but characteristic noise. The noise is mainly generated by the movement of the blades through the air. This produces a swishing sound in rate with the rotation of the blades, as well as noise from the turbine machinery.
Limits should only be placed on noise over a range of wind speeds up to 12 m/s when measured at 10 m height on the wind farm site, as faster wind speeds will typically mask the noise produced by a wind farm. Separate noise limits should apply for day-time and night-time.
Turbines are audible from distances as great as 2 km only in ideal conditions; when there is very little wind where the listener is, but sufficient wind where the turbines are on a ridge-top to power them.
Peer-reviewed studies have found that living near wind turbines does not pose a risk on human health. The studies looked at a range of health effects from hearing loss, nausea, and sleep disorders to dizziness, blood pressure, tinnitus, and more.
Noise guidelines
Unlike typical noise sources at licensed sites, wind turbine noise is directly related to wind speed. Wind turbine noise guidelines in Ireland are set out by the DoEHLG guidance and are based on the principle that turbine noise should be controlled with reference to fixed limits when background noise is low, or relative to background noise itself as it increases with wind speed, whichever is the greater. A common interpretation of these limits is that turbine attributable noise should be limited to:
1. 43 dB LA90 or 5 dB above background noise, whichever is the greater at NSL for night-time· hours
2. 45 dB LA90 or 5 dB above background noise , whichever is the greater at NSL for daytime hours 35 to 40 dB LA90 or 5 dB above background, whichever is the greater, at NSL for daytime· hours where background noise is less than 30 dB LA90
In this context, background noise is defined as a function of wind speed, over the relevant period (day or night) which is quantified by measurements prior to the site being built. It is important that such background noise measurements are referenced with wind speed at the hub height of the turbine(s) proposed, as it is this which sets the level of noise generated by the turbine.
WHO Guidelines
World Health Organisation Community Noise Guidelines, 1999 This guidance document recommends an external day-time limit of 55 dB LAeq to prevent serious annoyance during the daytime and evening, or 50 dB LAeq to prevent moderate annoyance, together with a night-time external noise limit of 45 dB LAeq to protect against sleep disturbance.
WHO Night Noise Guidelines for Europe
The World Health Organisation has recently published updated guidance on night-time noise levels designed to protect the public, based on external noise levels as averaged over a whole year xii. This recommends a target value of Lnight,outside of 40 dB where Lnight,outside is the external LAeq over a 1 year period xiii. It should be noted that this guideline limit is intended to cover noise from all sources at a specific location.
WHO guidelines for Holland
Holland Wind turbine noise is restricted to an Lden value of 47 dB and an Lnight value of 41 dB.
WHO for germany
Germany No specific wind turbine noise guidance is available so the following generic limits apply. iN genral it is 35 to 70 dB(night) and Day 50-70 dB.
WHO for Itly
A guidance document on wind turbine noise is currently being drafted but, until such time as this is available, the following generic limits apply: 40-70dB(night) and 50-70 in day for school, hospital, mixed activity, commercial and industrial etc.
Hope content is helpful for you.
Ashish
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While working on squirrel calls, I found that the frequency of the sound produced by individuals from mean sea level is quite more than the same from an elevation of about 2000 metres. Does the frequency depends upon the elevation of habitat and changes with change in the height or their is something else behind the aforesaid phenomenon.
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Hi Anish,
It is interesting that vocalization frequency tends to get lower as one proceeds up an altitudinal gradient. Moreover, dominant frequency also decreases with increasing body size. For many animals, body size also increases as one proceeds up an altitudinal gradient, due to harsher conditions at higher altitudes. Harsher conditions favor larger animals with smaller surface area-to-volume ratios.
In a study (refernce below) we did on hearing in tropical treefrogs found along an altitudinal gradient in Puerto Rico, we concluded:
"We suggest that the animal’s body size determines the frequency particulars of the call apparatus and the inner ear."
Meenderink SWF, Kits M and Narins PM (2010) Frequency matching of vocalizations to inner-ear sensitivity along an altitudinal gradient in the coqui frog. Biol Letters 6: 278-281. doi:101098/rsbl20090763.
I hope this helps,
Peter
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Anyone have any idea on how to harvest the acoustic energy from a line sound source? The line sound source is in small scale, maybe in a centimeter range, and the sound pressure is very small, around uPa I guess. 
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Dear Sheng,
welcome,
I think you had already solved the problem. However i would like to introduce a solution. There are two types of electrical vibration harvesters. The electrostaic MEMES converter and the piezoelectric transducer. You can use four flat converters to collect the sound from all sides. You can also put the sound line in in the focus of parabolic reflector and receiver the reflected sound by one flat converters. The second solution may be better. as it uses only one flat converter.
Best wishes
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How many interfacial SH-waves can propagate in the PEMs?
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My pleasure Dear Prof Aleksey. Regards
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Hi, everyone. If there is any public sound data with the marks, such as bird songs, children crying, traffic congestions, vehicle noise, vehicle horn, or other labels that marks the sound events.
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Analyzing a sound's spectral makeup (fundamental frequency plus harmonics) is easy with commercial or free software (I often use SpectraPLUS, which has a 30-day trial). What isn't so easy to access is a sound's intensity or sound pressure level (SPL) unless the recording comes with a calibration file. Inexpensive sound level meters (SLMs) may help when making live measurements, but these SLMs often present sounds as A-weighted only, meaning the measurements filter out very low and high frequencies. Very low level sounds (i.e. very soft sounds) are not easy to measure, as microphone noise alone can obscure the measurement. So, depending on what attributes of a noise or sound you're investigating, publicly available sound snippets may not be sufficient. Eric C. (former Principle Engineer, Doppler Labs; former Chief Engineer, Countryman Associates).
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some metal surfaces, rubbed against each other squeaks in a shrilling high frequency. Why? What is the role of surface roughness, friction factor, geometric shape, stress distribution,solid and medium sound velocity, strength, resilience and microstructure to frequency spectrum of the sound? Are there any set of differential equations that describe this phenomena? How to construct a model concerning the phenomena?
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This phenomenon is called "stick-slip" and it is caused by the friction and surface roughness. however, the resultant noise is governed by the geometry of the components. For example: a violin bow rubs against a tuned string, and you hear a combination of a saw-toothed waveform of the frequency of the string, and not the bow, or the geometry of the horse hairs in the bow, nor the rosin that causes the high friction. Theerfore, concentrate on modelling the components, and don't concentrate on the surface roughness as this only causes a broad-band exitation of the ctuned omponent resonance. you hear. I hope you understand my explanation? Best regards, Mike.
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Suppose avg. grain size, crystal structure, young's Modulus, fracture strength, velocity of sound and surface roughness of both the crushed crystals (before and after crushing ) and the crushing surfaces, as well as load on grinding surface, static/dynamic friction coefficients are known. Then, is it possible to estimate the crushing sound of the crystals?
Conversely, if crushing sound of the crystals are analyzed, is it possible to find any mathematical relation between the variables outlined?
I am asking the questions since crushing minerals and recording-analyzing the sound require no sophisticated instrument at all.
Please provide relevant research links.
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Dear Sumit,
Sound of cracking crystals between two hard surface depends on couple of important metallurgical as well as mechanics factors such as increasing strains, grain distribution reaches log normal packing, coarsening of crystal grains, mis orientation angles, level of temperature distribution due to fraction between two surface with grain misalignment effect , critical grain size micro-cracking and thermal enhancement, cohesion crack nucleation, defects, cavities etc.
In general, If the size of the crystal grains is closed to the wavelength of the supersonic ray, the part will not be transparent to ultrasound, in this case the there will be no ultrasound bottom of signal. it is also seen that size of the crystal grain increases the intensity of supersonic wave decreases.
Sound products can be listen at high extrusion speeds.
Atomistic modeling is highly applicable in easy understanding of crack sound between hard metal.
Ashish
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I am using FFT(Matlab) to convert time domain curve in frequency domain but it is showing me the number of samples in X-axis instead of frequency. How to obtain the frequency in X axis in frequency domain curve obtained from time domain using FFT?
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For example, if you have N data in a range of T seconds, then the steplength in frequency domain is 1/T Hz. Keep in mind, FFT will also return N data in frequency domain.
Thus, what you get is exactly [0:(N-1)] * 1/T Hz in frequency domain.
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Hi, I built a 2D plain strain model, put a concentrated time-harmonic force below the free surface, and detected Rayleigh wave on the free surface, I found that the amplitude of Rayleigh wave decay with the distances increasing. The material is aluminum, I just set density, Young modulus, no damping. I have no idea why the amplitude of Rayleigh wave would decay with distance.
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A point force can generate other waves in addition to a Rayleigh wave (which is an interface mode owing to the discontinuity of elastic properties at the aluminum boundary). I think that a simple point force can also generate P-SV waves that radiate down into the aluminum. So in your graph you may be seeing the superposition of Rayleigh waves and P-SV body waves.
Additional waves are suggested by the evanescent waves in Bernard Garnier's answer above, where he is no doubt thinking more generally of many different modes of vibration in stratified media, of which the Rayleigh wave is only one type of mode. But in a homogeneous solid half space, P-SV body waves can be attributed to the contribution of two branch line integrals -- one for P-waves, and one for S-waves -- in the complex wave-number plane for cylindrical wave analysis.
Unlike Rayleigh waves, P-SV body waves should exhibit the usual 2D decay (geometric spreading loss) with travel distance. But in your model Rayleigh waves and P-SV body waves will be superposed on each other.
The Rayleigh wave presumably dominates without decay at long distances and near the surface of the aluminum. Your graph seems to be showing this leveling off of amplitude at greater distance. If your model allows you to go out further in distance, then you should check for leveling off in amplitude at much greater distances. That is where the Rayleigh wave dominates.
But P-SV body waves may dominate at greater depth, where the Rayleigh wave motions do not penetrate owing to the exponential decay with depth. So you might check for body waves by plotting the wave amplitude at depth.
In any case, there may be more wave excitation going on than the excitation of Raleigh waves alone when the forcing is a point force. In order to generate a Rayleigh wave alone, and no other waves, you will require a rather complex distribution of forces and not just a single point force (use Huygens' principle for wavefront reconstruction, applied to elastic waves). To isolate the Rayleigh wave, it is much easier to simply sample the waves far from their source, as distances where the Rayleigh wave dominates all of the other waves generated by the source.
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Hi I need to model PA system for a train station with EASE.
I got this doubt, while a speaker makes announcement, sound from it have all frequencies or it will be at one particular frequency.
Because in EASE, it divides electric power equally among all frequencies.
If there are 21 frequencies, and electric power provided is 0.75W, then for each frequency the power is (0.75/21). We use pink noise.
Since the power gets distrbuted we are not getting the desired SPL.
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As Fabrizio above has correctly mentioned, you need to consider just the frequency range for human voice in your application, which is much more limited than the hearing frequency range. In addition to the acoustical response of the room you are interested in to model (a train station), an important detail is to consider the background noise as accurate as possible. This will also affect the total required sound power of your PA system in order to get an appropriate intelligibility level.
Best
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Sound Power Level of a speaker is computed from the sound power of the speaker.
Sound power from the speaker is the total sound energy emitted by a speaker per unit time.
How this sound power is related to the electric power given to the speaker. For example if I give 6W electric power to the speaker then sound power from the speaker will be only 6W.
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Unqualified comments :-
Time weighted negotiating of a mass, collection of matter considering field effects ( say gravity, possibly friction and so on) with a directional effort is work, forced displacment sort of effects.
Rate of doign work is power (watts) , sort of time rated expense of energy. Describing power has nothing to do with underlying phenomenon.
Most important thing to understand or find or reconcile is
proxy of or displacments/field
proxy of or efforts,
and of course ratio of expended against recoved power offers clues about nature of losses , mostly thermal, and proverbial efficiency etc.
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I have difference frequency iterations and pressure values for each iteration. I want a single final frequency-pressure curve. How to convert all the frequency-pressure curve to single resultant frequency-pressure curve?
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Hi Kapil
Not excatly what you are asking but perhaps anway useful ?
The below link contains links to excellent material for self study on signal analysis. The writings are old but the material is among the best available. Start from the top and continue downward.
Sincerely
Claes
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Dear All
What is the minimum computer specification required for the following simulations?
Suppose the following simulations should be apply for Lundell Alternator (with around 80 parts assembled on each other) and Starter Motor (With around 100 parts)
- Air flow simulation
- Dynamic Analysis
- Acoustic Analysis
- Static Analysis
- Fatigue Analysis
I should mention that the geometry of the parts in some cases are too complicated.
Thank you for giving me your time in advance.
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i7 core with 16GB RAM minimum.
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Dear colleagues,
I need to analyse the mean fundamental frequency of disordered voices and I found errors using the standard advanced settings of Praat autocorrelation procedure.
I found different articles which suggest to modify pitch floor and pitch ceiling but I cannot use a narrow pitch range because I am measuring the pitch range of dysphonic men and women.
Are you aware of guidelines on specific advanced settings (silence and voicing threshold, octave cost, octave-jump cost, voiced/unvoiced cost) for the alaysis of dysphonic voices?
Thank you for your help!
Valentina Di Natale
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Please see the link below: https://pdfs.semanticscholar.org/2c7f/30a9d9a516399e07f4210a32eed8595884c6.pdf. Acoustic Analysis of Voice: A Tutorial.
Best regards
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I would like to measure STI (speech transmission index) in a 3D environment. I have been thinking about use Unity and reproduce the sound from a speaker and record the sound from the character ears but I would like to find something more profesional. All has to be in 3D because the building is not real any more.
Thank you for your time.
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I completely agree with the answers from Tim Ziemer and Xenia Kontogianni . We did a similar investigation on the STI in a virtual room. Check out the poster and let me know if you have any questions.
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Hi,
I'm currently using Random Forests to classify acoustic data to corresponding bat species, however I was wondering if there is a way to also incorporate unsupervised cluster analysis into my classifier (such as K-Means or Hierarchical modelling)?
Whilst I have training data for ~40% of my target species, there is likely to be a large proportion of my field dataset which corresponds to species that haven't yet been described in the supervised model and therefore I would like for the classifier to be able to assign new clusters/call types to new classes (that hopefully can be matched to a species later).
I have been looking at consensus maximisation algorithms but I'm not sure if this would be the best method.
Any help is greatly appreciated!
(I'm not a computer scientist so please forgive me if this is a very straightforward question)
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What do you mean for semi-supervised learning?
Reinforcement learning?
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I am currently working on anthropogenic effects on the vocalisation of howler monkeys in the urban environment.
I record the vocalisations with a TASCAM DR-07MKii and I want to extract acoustic measurement such as frequency, pitch, rates and lengths of vocalisations.
Which program is easiest to use for a beginner to extract these variables?
I currently have: Sound Analysis Pro, Praat and Audacity.
All tips and additional information are very welcome
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I agree that Audacity is of no use-- it's primarily an editing program, not for measurement. Praat and SAP would both probably work for the simple measurements you described. I believe Praat (which was developed for work on speech) has some tools for measuring speech-related features such as formants, which might be useful for nonhuman primate vocalizations. Raven Pro (http://ravensoundsoftware.com/) has (I'm told) an easier learning curve than Praat and includes basic time, frequency, and relative power measurements (though no formant measurements). There's also a set of freely available tutorial videos available (see "Training" on that website).
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Dear All,
Please give me your recommendations (or some papers) how I can examine acoustic properties of the titanium printed flute.
How in general acoustic properties in metals are examined and evaluated?
Sincerely yours,
Anastasia
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Can you please clarify by "acoustic properties"?
A metal alloy as every solid propagates compressional waves that are "sound waves" with a sound speed of SQR(Young Modulus/Density) with an hysteretic damping generally close to 1E-3; measuring the Young Modulus is a simpler lab test than measuring the hysteretic damping, both can be made on a simple parallelepipedic sample; sound speed and sound absorption ((1-hysteretic damping)*number of wavelengthes) can be also assessed explicitly with an ultrasound probe on a rod (send a short pulse and compare with the echo).
If you manufacture a musical instrument and want to characterize the influence of the alloy on the musical quality of this instrument, this is an absolutely different issue calling for decades of research (the most popular is trying to understand the role of the varnish versus wood selection in Stradivarius violins): I would just ask a good musician to assess it by playing and ranking various instruments manufactured conventionally and by additive printing...
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We have multiple arrays of microphones, each array well seperated from others on the ground. Our task is to track an aircraft that happens to pass by. But the terrain is such that there is a lot of reverberation. Now:
1) What prefiltering technique is best to remove reverberation effects.
2) Which weightings during GCC(Generalised Cross Correlation) are best against reverberation, while at the same time without increasing error by much.
Thank you very much.
Lay Jain,
Massachusetts Institute of Technology
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Maybe "reverberation" might be considered simply as "multipath" for aircraft noise in outdoor environments, and in that case you might solve it explicitly e.g using adaptative beamforming algorithms. Acoustic tracking of low flying helos is a conventional problem solved satisfactorily from many years in open field, I know some systems based on likelihood maximisation, it take a fraction of a second to get iteratively the most likely position, then your tracking algorithm will have to reject aberrant dots. A key issue is now far your aircraft noise and course is stationary or not, as rightly commented by Ronald! Of course if you deal with supersonic low flying jets, you will find the Mach wave direction, not directly the plane heading... [An issue solved in acoustic sniper detection systems by tabulating the speed (hence the Mach cone angle) for a library of ammunitions and associated signatures]
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Plz, suggest reference also.
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I want to classify the reason for baby cry - unconformable, hungry, feeling Hot or cold etc by listing acoustical analysis of the audio signal.
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couldn't find any standard database but yes i guess this would be helpful https://opendata.stackexchange.com/questions/6771/required-a-audio-format-baby-crying-data-set
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Many applications in numerical acoustics occur in exterior domains. The advantage of benchmark problems is a closed form solution (e.g. circumferentially harmonic radiation from an infinite cylinder or radiation from a portion of the infinite cylinder). Different models such as perfectly matched layers (PML) and absorbing boundary conditions ABCs (like Enquist Majda, Bayliss Turkel) are used in the finite element method to approximate Sommerfelds radiation condition. I am interested in a publication, where these approaches are compared.
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WHO ARE WE
Master Acoustics International Corp.was founded in Taiwan in 2017 by our Chief Director Mr. Jack Chen, who has a brilliant sense of both sound and frequency. Jack has used sound techniques to invent two of our innovative products: the sound optimized resonator and a sound optimized coating which can be painted on the surface of instruments. Both of these innovations dramatically improve the timbre of the instrument to which they are applied. The sound optimized resonator has already been patented in China with patent applications in process in Taiwan and Europe.
WHAT WE WANT TO DO – A draft blueprint for the project.
We conceived of this project and would propose three goals:
1. To examine our inventions to assess their effectiveness
2. To use collaboration to create new ideas/innovations/inventions
3. To formally present the project findings for everyone to see
LOOKING FOR PARTNERS
If you are interested in knowing more about Master Acoustics International and this project, or you can recommend any institutes or researchers. Please let us know, thanks!
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Hi, i'm an acoustics professor in Colombia and i'm really interested in the project and in general in Master Acoustics International , could you give me more information.
Best regards
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I want to compare “neutral” baseline data with data recorded in a test session to finally be able to evaluate arousal/affect of the infant.
Which software would you recommend? Do you have any literature advice?
Any advice would be appreciated!
All the best
Sam
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Do you know this paper?
Shigeaki Amano, Tadahisa Kondo, and Sachiyo Kajikawa 2001. Analysis on infant speech with longitudinal recordings The Journal of the Acoustical Society of America 110, 2703 (2001);
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Dear all,
I need to detect the values represented by the red rectangle, in the attached picture, automatically.
This means I need to detect the part where the curve change significantly
Thank you
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Let me add that after applying any program finding just peaks probably some elimination procedure would be desired. Indeed, the example in the red box supplied by Hadjer shows that any automatic program returning all local maxima and minima will display too many points - in the same manner those which significant as the non-significant. Of course, the expected correct answer should eliminate some of them. This depends on the assumed DEFINITION of a significant point of changes. Perhaps it would be enough to implement in the code a suitable criterion like the following:
For a given fixed half-width t>0 do the following:
1. for every point of maximum (x,y) apply:
If [yn <= y whenever x-t <= xn <= x+t], then decide: [ (x,y) is a significant point of local maximum]
2. for every point of minimum (x,y) apply:
If [yn >= y whenever x-t <= xn <= x+t], then decide: [ (x,y) is a significant point of local minimum]
3. Eliminate a point if it is not unique among other points of the same type within distance not exceeding t.
The third step is just to choose one of points in cases when e.g. at two neiboring maximal points say x1 and x2 the values y1 and y2 are almost equal:) Perhaps some average of the similar x-s and y-s would be better? Or, before running the program finding max and min points, to perform some standard smoothing procedure?
Regards, Joachim
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When obtaining sound absorption coefficient of Kenaf (Hibiscus Cannabinus- natural fiber) mix with latex rubber, results show that a drop performance at mid range frequency. Refer to the graph, Black line is sample with pure fiber contain. Blue dotted are Kenaf mix with latex rubber. Any strong evidence or reason behind the slop drop after 1000Hz for latex mixed sample and 2k Hz - 3k Hz for pure fiber sample? Reference journal / paper could be great if attach together.
Sound absorption coefficient obtain by using Impedance Tube Model SCS9020B Kundt/T60.
Thank you.
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Hi, Mathan,
As additional resources for Trevor J Cox book, you may find an interesting detailed explanation in J.F. Allard and N. Atalla’s book Propagation of Sound in Porous Media (Willey, 2009). You may also find another brief explanation in Hoda S. Seddeq’s paper published in Australian Journal of Basic and Applied Sciences, 3(4): 4610-4617, 2009: Factors Influencing Acoustic Performance of Sound Absorptive Materials.  Almost similar topic provided by Tiuc et al. in their paper entitled The analysis of factors that influence the sound absorption coefficient of porous materials which are available at RG.
According to my own experience in working with felt, it is possible to increase the sound absorption of the fibrous material based sound absorber in the mid-frequency range. The simplest way is by using thin elastic membrane backed fibrous layer. When the fibrous layer coated with a thin elastic membrane, it changes the reactance of the structure and brings the possibility for absorbing the sound waves energy through combined resonant and viscous damping mechanism simultaneously. Should you need further discussion, please feel free to send an email privately.
Best regards,
Iwan.
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Hi everyone,
I am writing my MSc thesis about vocal communication in woolly monkeys and I want to make a general description of their different types of calls. I want to obtain various acoustic parameters such as duration, frequency range, low frequency, high frequency, maximum amplitude, average frequency, initial frequency, and final frequency. Hence, I have to analyse my recordings using SoundRuler, but I've never used this software before. I've read the instructions but I have some questions anyway.
- I recorded in stereo, so when I introduce the recording in the software, it asks me if I want to analyse left or right channel. Can I analyse both separately and then calculate the mean of both channels?
- Also, when I introduce the recording, I mark the section that I want to analyse using green bars. Once this section is marked, I proceed to do the analysis. Is it as easy as clicking the "manual" button? When I do it, it appears a table with the different values of the parameters, but I don't know if it is as "simple" as that.
That's all at the moment. Thank you for your answers!
Laura.
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Hi Laura,
I agree with Pavel regarding the channels and amplitude. I also don't use SoundRuler (sorry!) but I thought it might be useful to add that you need to be sure there is no background noise overlapping your calls of interest. If for example these recordings were made at a zoo, there may be visitors chatting, or in the field there could be other animals calling etc. on your recordings. If there is, then you can either need to filter it (if it does not overlap in frequency with the monkey calls), or if that's not possible, you could manually extract the frequency measures, or simply eliminate those calls from your analyses. It might be that you can use the read-out on all your "clean" calls (i.e. no background noise, one individual calling at a time) and in that case it could be just as simple as you say!
Good luck with your interesting project!
All the best,
Esther
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Given that the data points are approx. 100,000, Fast Fourier Transform in Excel or other software does not support more than 4096 data points. Specific software or technique will be appreciated.
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You must try Long Term Average Spectrum
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For the forensic evidence analysis of speech samples, the speaker specific phonetic and other linguistic parameters of acoustic signals are to be measued to compute the likelyhood ratio.  
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A group of parameters are important for that purpose, for example:
Pitch, Harmonics, First five formants at list, Duration, Melody, Intonation, Sonority, Transitions of phonemes, etc.
Best regards,
D. Escobedo
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In what way SNR of an acoustic source can be measured in real time? Theoretically, we know that SNR=Psignal/Pnoise, but how we can put this into practice, i.e. how to implement it on an embedded controller?  
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Dear Mad Helmi Bin Ab. Majid,
It matters which type of signal do you measure (i.e. stationary, nonstationary) which is implicated by the source of the signal. Is it a biological/medical, artificially generated periodical signal or other?
I guess, the question refers to acoustics, so the easiest and the most accurate way to measure the SNR is to do it directly by device performing the acquisition.  However, this method is frequently not available or results not transfered to data sheet.
You can use some easy approximation: divide mean value of the signal by a standard deviation of the signal measured within certain time window. If the recording device does not provide any additional pink noise or the only noise present is the white noise - the task is easy. According to theory, the expected value of white noise refers to mean value of normal distribution, and autocorrelation function should obtain value 0 (or close to it in practice). Then, the approximation above is sufficient for SNR.
You can also perform  a similar action in frequency domain, which requires firstly to calculate an amplitude spectrum (or power spectrum) of the signal in certain small/big time window (or for the whole measurement at once). Though, you need to be aware of two effects:
A) the wider time window - the better resolution of the spectrum, but the longer time of acquisition and worse time resolution of the results;
B) You need to know, which part of the spectrum your noise accupies, especially if the noise is not white.
The easiest way - you can divide the amplitude (or sum of the amplitudes present in the certain part of the spectrum) referring to your 'usable' signal, by the same referring to part occupied by noise.
For power spectrum SNR = (average signal power)/ average noise power), which in dB refers to (SNRdB=10 log10(SNR).
Please, search some information about the type of the signal you use and the possible noise influencing your results. It might be, that in the spectrum you will see a strong influence of a low freaquency or only high frequency noise (if there is no additional source of low frequency noise added).
This is the fast, simple answer. If you need more help, please don't hesitate to write.
PS> I hope the publication attached would be helpful:
Regards,
Richard Gomolka
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I have decided to do research on noise pollution modeling. Kindly tell me open source software for noise pollution modeling around airport.
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Hi
Here is another opensource software, Code Tympan, that uses raytracing. It might not yet be exactly what you want but may still be worth your taking a look at as anew solver is in the works.
"A new solver called ANIME3D will be soon available and includes a transformation based method to take into account refraction by sound speed celerity gradients. "
It seems that Code Tympan currently is slower than SoundPlan. 
Sincerely
Claes
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Hi all,
I used intependent component analysis to sepereate speech from noise.i used 2 microphones and two sources (speech and noise) . I got improvement in SNR when I used ICA but when I used ICA in speaker verification . I got bad performance compared with noisy test signal. Does any one used ICA to remove noise and used the enhanced speech after ICA in speaker verification?
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I did some experiments on noisy speaker recognition using speech enhancement with non-negative matrix factorization (check FASST toolkit)
The approach works well, but only if you apply it for both noisy train and test data. To my knowledge, any speech separation algorithm leaves some residual noise, and consequently, there is still a mismatch between clean speech and the enhanced one. If your training data are clen, I'd suggest to apply artificial noising techniques.
It might also depend on your classifier. For me the best was obtained with i-vector extractor and plda classifier trained on noisy data enhanced with NMF approach
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I mean if we would want to modify a muffler based on parametric design, which parameters should be evaluated in the study, for example the cross section of the tube, the number of mufflers (inline or parallel), etc.
What kind of engine specifications should be evaluated in the study?
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Hi
For muffler design, all dimensions matter, though some decide the base performance more than other. These tend to decide the system fundamental resonances. 
I believe it is worth your while taking a look here
and to take a look at what Prof. Åbom & Prof Boden have been doing for quite some time. 
For design, perhaps some of my ramblings can be of interest though they only adress Test, Simulation, QA and Optimization in general rather than muffler design in particular?
For what it is worth. It is worthwhile to simulate muffler not only regarding acoustics as they should be able to withstand also fatigue from pressure pulsation loading, and if the muffler is extended (large with respect to system dimensions), you must identify also good suspension positions, mechanical resonance should not lie at bad motor RPMs or at typical cruise speed, fit within a limited space envelope, etc. 
Sincerely
Claes 
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I have two acoustic signals (amplitude as a function of time). I do an FFT or Welch analysis to see, for which frequencies the two spectra are similar and where they differ. I am thinking of a correlation coefficient as a function of frequency. How do I do this (e.g. in matlab)?
PS the spectra are quite noisy.
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Best way I know of is what Sina said with the CPSD or the coherence. To do this with MATLAB's built in functions, use the "cpsd" for the cross-power spectral density or mscohere function for the coherence.
If they're noisy signals as you say, you'll may want to use a Hanning window with some overlap for the window edge effects. We usually use 75% overlap for aeroacoustic broadband signals.
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Hi,
In the past few years I been involved in sound analysis and condition monitoring of the car engine where the misfiring engine operating in workshop environment is successfully identified. I was also involved in research on Music Information Retrieval (MIR) where the system is provided with a sample record or part of a record and the system looks for similar records or records with similar queried part within them, based on their similarity and not their genre or style. 
I am looking for a data set to apply my previous experiences in sound analysis on animal identification and even in the field of animal psychology.
This will be a inter/multi disciplinary work and scientific inputs are welcomed.
Many thanks
Peyman
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The Bird Audio Detection project may be of interest, although the aim is detection rather than identification. The dataset has 17000 recordings tagged as either containing or not containing birds. The survey paper reviews the most common methodologies.
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I want to make a speaker segmentation of the train base in TIMIT database.
I need the speaker labels to make evaluation with test label which i have done.
so Howcan do the speaker labeling on train data on TIMIT database?
Note: Speaker labels found in test are in TEXT Grid format.
Waitting your response...!! Thank you in advance!!!
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Thank you so much!!!
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Is there a "sound density limit" beyond which sound energy fails to be recorded and/or played back?
Example: one of the largest known choirs consisted of 121,440 people - if I wanted to record such an event (or as many overdubs as those or many more), would there be a density limit I would reach and if yes, how can it be calculated ?
What about natural events ? Imagine hailstroms for example. What would happen if I recorded many and created a sound file with dozens, even hundreds of those and played them back ? Would I be reaching any playback (or hearing) limits ? Would such density create some sort of coloured noise ? 
All your ideas, suggestions and explorations will be very welcome  
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If :played back "means sounding all at once in the same place, your question devolves mathematically into asking how many numbers can sum to, say, 3.  Mathematicians are inclined to invoke such ideas as infinity in such cases, though the answer is clearly larger than we can name.
If you at what point does adding yet another sound to a large mix become irrelevant, the answer is also clearly large, but far from infinite.  Having sung in large choruses, I opine that after about a hundred or two voice, the addition (or subtraction) one more (or less) hardly matters in any musical sense.  (The gospel choir at UCSD once numbered over 500, but their major problem was finding a space in which they could rehearse).
The interesting part of your question lies in between 0 and infinity.  Modern psycholofy and cognitive science both suggest that  7 plus or minus 2 is about the limit of our "attention".  But they also suggest we don't multitask, but scan, i.e., shift our focused attention rapidly among competing messages.  Any instrument-rated pilot can tell you that simultaneously scanning (meaning observing and understanding) 7 or so instruments (as well as looking out the window) stretches human capabilities.  Not impossible, but difficult, especially if your life depends on it.
So even though I can't truly answer your question, my best guess would be the well-established psychological number associated with short-term memory: seven plus or minus two.  Seems our brain - evolved over million of years - have figured that before that number, we stand a chance to figure things out.  After that number, run!
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I want to give some reasonable explanation of my calculated results on the zeroth-order acoustical Bessel beam scattering from the forward half-space for rigid spheroids and finite cylinders. There is an obvious difference between the ordinary plane waves and Bessel beams. Could you suggest some appropriate physical models which may help to describe the results? Thank you very much.
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Dear Zhixiong Gong, please, you will be able to learn some answers to your questions if lament my geometrical works on interpretation paraxial (Gaussian bunches), it is in RG.
It is also necessary to understand that the main geometry of Euclid is a geometry on the plane. That is at the heart of the exact sciences, there is a two-dimensional space. And only about 5% new can be found in three-dimensional space, in comparison with two-dimensional.
My rabbit on the country site grew up and eats bread, potato and carrot which I leaving it. He at distance of three meters from me gets up on hinder legs, its height about one meter.
Other my friends consider that it is after all a hare, but not a rabbit.
Sincerely, Alexander
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I am trying to use ABAQUS for FEM analysis and LMS VirtualLab for BEM for an acoustic analysis. But there is not much documentation available on how to use them together for. Where do can I find the necessary tutorials?
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Hi Anvesh, you can contact the GTAC service to take your questions. It is an excellent tool (http://www.plm.automation.siemens.com/en_in/support/gtac/index.shtml)
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Does anyone knows the Acoustic impedance or sheer modulus of pentacene ??
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Please start  GOOGLE SCHOLAR    and not GOOGLE .
give your question in the mask (e.g.What is the Acoustic Impedance of pentacene ?)
and you will obtain  991 hits. Read them and extract the infos related to your research.
Good luck
JRG
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on acoustic emission reaearch ,many authors used the formation on amplitude distibution model .n(a)=(a/a0)-b.and i want to know how to obtain b value?,i need you can list the example .who can help me ?
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I use the method described by Claused et al. for fitting.
[1]    Clauset, A., Shalizi, C., Newman, M., 2009. Power-law distributions in empirical data. SIAM Rev. 51 (4), 661–703.
There are practical Matlab functions and  Python packages to do the job: 
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Dear Physicists, please I would like to know if there is a relation between the intensity of a signal acoustic wave and the ability of that wave signal to shatter a body. For instance, if a tuning fork is emitting an acoustic wave into body X (such as a glass cup) at a resonant frequency of body X (for instance the fundamental frequency) is there a threshold intensity (e.g. wave amplitude, wave power) of the wave emitted by the tuning fork which needs to be exceeded before body X will shatter?  It seems well known that the shattering effect will occur at specific frequencies (the fundamental frequency or its harmonics) however, does the wave amplitude, power or other material properties of the body play a part? If yes, please what relation or formula governs these? Answers appreciated. Thanks
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Yes, if you need to.
BB-C 
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We want to study noise in  scaled down room to utilize better noise control. We can use dimensionless analysis however it's better to stick to standards.
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The principles of acoustic scale modelling are simple.  For auditorium acoustics scales of 1:8 to 1:50 have been used, larger scales are simpler to operate but take up more space.  Scale modelling offers 'perfect scaling' EXCEPT in the case of air absorption.  Dealing with this and other issues involves a certain investment in time etc.  For simple acoustic situations, computer simulation modelling may offer sufficiently accurate predictions.  The attached paper is old but considers most of the issues relevant to scale modelling.
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In doing acoustic analysis of a certain segmental or supra-segmental feature of a language, a researcher usually decides on the age range of speakers to be recorded and analyzed. This age range should represent a "generation". What is the most appropriate age range that better represents the current status of the intended feature in the language under analysis? 
Some studies record males or females only, while some studies record both and report certain variations based on gender difference. The question is, is it essential to have both sexes recorded even if the researcher is not interested in gender variation? 
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Ammar: For methodological tips on informant choice, as well as sampling and analysis of previously unrecorded languages, the Handbook of the IPA may be a good place to start:
The Journal of the IPA regularly publishes phonetic illustrations of previously unrecorded languages (vowels, consonants, prosody), and is dedicated to research on the phonetics of any language. Browsing a couple of journal issues will give you a few more tips:
You may even consider submitting to this journal an article on your findings about the vowel system of the language you’re working on, and/or a full illustration of its phonemic inventory? Instructions for contributions are here:
All best for your research
Madalena
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Ecoacoustics has become a tool to assess biodiversity of calling animals using Passive Acoustic Monitoring (PAM) of soundscapes (for examples, search for the respective keywords https://www.researchgate.net/search.Search.html?type=publication&query=ecoacoustics&tabViewId=57865c2140485491311aff27&previous=researcher). Several projects host accessible recordings, allowing re-analysis and data-mining for vocalisations of certain target species (my personal interest: Orthoptera). However, it seems there is no federated database allowing one-stop access to distinct servers - acoustic data servers are fragmented, or even unstable, and  I did not even find a comprehensive list of available sites.
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I would like to add that there is a difference between sound (species-specific recordings) and soundscape files. The information associated to these two categories of files are little bit different and sound libraries are not all adapted to receive soundscape recordings.
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The mean processed signal to noise ratio was calculated to be 30 dB for the Raytheon sonar, and 13 dB for the Klein sonar. Using the Receiver Operating Characteristic (ROC) curve displayed (figure with calculations from Urick,1983 is attached: file name is "ROC curves calculations.bmp"), and given the desired false alarm probability of 0.5%, the probability of detection corresponding to the mean processed signal to noise ratio for each sonar
was calculated at the false alarm level. The probability of detection was calculated to be 0.998 for the Raytheon sonar (green lines on the plot attach