text:
Acoustics/Filter Design and Implementation

Introduction
Acoustic filters, or mufflers, are used in a number of applications requiring the suppression or attenuation of sound. Although the idea might not be familiar to many people, acoustic mufflers make everyday life much more pleasant. Many common appliances, such as refrigerators and air conditioners, use acoustic mufflers to produce a minimal working noise. The application of acoustic mufflers is mostly directed to machine components or areas where there is a large amount of radiated sound such as high pressure exhaust pipes, gas turbines, and rotary pumps.

Although there are a number of applications for acoustic mufflers, there are really only two main types which are used. These are absorptive and reactive mufflers. Absorptive mufflers incorporate sound absorbing materials to attenuate the radiated energy in gas flow. Reactive mufflers use a series of complex passages to maximize sound attenuation while meeting set specifications, such as pressure drop, volume flow, etc. Many of the more complex mufflers today incorporate both methods to optimize sound attenuation and provide realistic specifications.

In order to fully understand how acoustic filters attenuate radiated sound, it is first necessary to briefly cover some basic background topics. For more information on wave theory and other material necessary to study acoustic filters please refer to the references below.

Basic wave theory
Although not fundamentally difficult to understand, there are a number of alternate techniques used to analyze wave motion which could seem overwhelming to a novice at first. Therefore, only 1-D wave motion will be analyzed to keep most of the mathematics as simple as possible. This analysis is valid, with not much error, for the majority of pipes and enclosures encountered in practice.

Plane-wave pressure distribution in pipes
The most important equation used is the wave equation in 1-D form.

Therefore, it is reasonable to suggest, if plane waves are propagating, that the pressure distribution in a pipe is given by:

$$\mathbf{p}=\mathbf{Pi}e^{j[\omega t-kx]}+\mathbf{Pr}e^{j[\omega t+kx]}$$

where Pi and Pr are incident and reflected wave amplitudes respectively. Also note that bold notation is used to indicate the possibility of complex terms. The first term represents a wave travelling in the +x direction and the second term, -x direction.

Since acoustic filters or mufflers typically attenuate the radiated sound power as much as possible, it is logical to assume that if we can find a way to maximize the ratio between reflected and incident wave amplitude then we will effectively attenuated the radiated noise at certain frequencies. This ratio is called the reflection coefficient and is given by:

$$\mathbf{R}=\left( \frac{\mathbf{Pr}}{\mathbf{Pi}} \right)$$

It is important to point out that wave reflection only occurs when the impedance of a pipe changes. It is possible to match the end impedance of a pipe with the characteristic impedance of a pipe to get no wave reflection. For more information see [1] or [2].

Although the reflection coefficient isn't very useful in its current form since we want a relation describing sound power, a more useful form can be derived by recognizing that the power intensity coefficient is simply the magnitude of reflection coefficient square [1]:

$$R_{\pi}=\left|\mathbf{R}\right|^2$$

As one would expect, the power reflection coefficient must be less than or equal to one. Therefore, it is useful to define the transmission coefficient as:

$$T_{\pi}=\left(1-R_{\pi}\right)$$

which is the amount of power transmitted. This relation comes directly from conservation of energy. When talking about the performance of mufflers, typically the power transmission coefficient is specified.

Basic filter design
For simple filters, a long wavelength approximation can be made to make the analysis of the system easier. When this assumption is valid (e.g. low frequencies) the components of the system behave as lumped acoustical elements. Equations relating the various properties are easily derived under these circumstances.

The following derivations assume long wavelength. Practical applications for most conditions are given later.

Low-pass filter

These are devices that attenuate the radiated sound power at higher frequencies. This means the power transmission coefficient is approximately 1 across the band pass at low frequencies(see figure to right).

This is equivalent to an expansion in a pipe, with the volume of gas located in the expansion having an acoustic compliance (see figure to right). Continuity of acoustic impedance (see Java Applet at: Acoustic Impedance Visualization) at the junction, see [1], gives a power transmission coefficient of:

$$T_{\pi} \approx \frac{1}{1+\left(\displaystyle\frac{S_{1}-S}{2S}kL\right)^2}$$

where k is the wavenumber (see Wave Properties), L & $$S_{1}$$ are length and area of expansion respectively, and S is the area of the pipe.

The cut-off frequency is given by:

$$f_{c}=\left(\frac{c S}{\pi L(S_{1}-S)}\right)$$

High-pass filter

These are devices that attenuate the radiated sound power at lower frequencies. Like before, this means the power transmission coefficient is approximately 1 across the band pass at high frequencies (see figure to right).

This is equivalent to a short side branch (see figure to right) with a radius and length much smaller than the wavelength (lumped element assumption). This side branch acts like an acoustic mass and applies a different acoustic impedance to the system than the low-pass filter. Again using continuity of acoustic impedance at the junction yields a power transmission coefficient of the form [1]:

$$T_{\pi}=\left(\frac{1}{1+\left(\frac{\pi a^2}{2SLk}\right)^2}\right)$$

where a and L are the area and effective length of the small tube, and S is the area of the pipe.

The cut-off frequency is given by:

$$f_{c}=\left(\frac{ca^2}{2SL}\right)$$

Band-stop filter

These are devices that attenuate the radiated sound power over a certain frequency range (see figure to right). Like before, the power transmission coefficient is approximately 1 in the band pass region.

Since the band-stop filter is essentially a cross between a low and high pass filter, one might expect to create one by using a combination of both techniques. This is true in that the combination of a lumped acoustic mass and compliance gives a band-stop filter. This can be realized as a helmholtz resonator (see figure to right). Again, since the impedance of the helmholtz resonator can be easily determined, continuity of acoustic impedance at the junction can give the power transmission coefficient as [1]:

$$T_{\pi}=\left(\frac{1}{1+\left(\frac{c/2S}{\omega L/S_{b}-c^2/\omega V}\right)^2}\right)$$

where $$S_{b}$$ is the area of the neck, L is the effective length of the neck, V is the volume of the helmholtz resonator, and S is the area of the pipe. It is interesting to note that the power transmission coefficient is zero when the frequency is that of the resonance frequency of the helmholtz. This can be explained by the fact that at resonance the volume velocity in the neck is large with a phase such that all the incident wave is reflected back to the source [1].

The zero power transmission coefficient location is given by:

$$f_{c}=\left(\frac{c}{2\pi}\right)\sqrt{\left(\frac{S_{b}}{LV}\right)}$$

This frequency value has powerful implications. If a system has the majority of noise at one frequency component, the system can be "tuned" using the above equation, with a helmholtz resonator, to perfectly attenuate any transmitted power (see examples below).

Design
If the long wavelength assumption is valid, typically a combination of methods described above are used to design a filter. A specific design procedure is outlined for a helmholtz resonator, and other basic filters follow a similar procedure (see 1).

Two main metrics need to be identified when designing a helmholtz resonator [3]:

 * 1) Resonance frequency desired: $$f_{c}=\frac{c}{2\pi}\frac{\sqrt{C_{o}}}{V}$$ where $$C_{o}=\frac{S}{L}$$.
 * 2) - Transmission loss: $$\frac{\sqrt{C_{o}V}}{2S}=const$$ based on TL level.  This constant is found from a TL graph (see HR pp. 6).

This will result in two equations with two unknowns which can be solved for the unknown dimensions of the helmholtz resonator. It is important to note that flow velocities degrade the amount of transmission loss at resonance and tend to move the resonance location upwards [3].

In many situations, the long wavelength approximation is not valid and alternative methods must be examined. These are much more mathematically rigorous and require a complete understanding acoustics involved. Although the mathematics involved are not shown, common filters used are given in the section that follows.

Actual filter design
As explained previously, there are two main types of filters used in practice: absorptive and reactive. The benefits and drawback of each will be briefly explained, along with their relative applications (see Absorptive Mufflers.

Absorptive
These are mufflers which incorporate sound absorbing materials to transform acoustic energy into heat. Unlike reactive mufflers which use destructive interference to minimize radiated sound power, absorptive mufflers are typically straight through pipes lined with multiple layers of absorptive materials to reduce radiated sound power. The most important property of absorptive mufflers is the attenuation constant. Higher attenuation constants lead to more energy dissipation and lower radiated sound power.

Examples

There are a number of applications for absorptive mufflers. The most well known application is in race cars, where engine performance is desired. Absorptive mufflers don't create a large amount of back pressure (as in reactive mufflers) to attenuate the sound, which leads to higher muffler performance. It should be noted however, that the radiated sound is much higher. Other applications include plenum chambers (large chambers lined with absorptive materials, see picture below), lined ducts, and ventilation systems.

Reactive
Reactive mufflers use a number of complex passages (or lumped elements) to reduce the amount of acoustic energy transmitted. This is accomplished by a change in impedance at the intersections, which gives rise to reflected waves (and effectively reduces the amount of transmitted acoustic energy). Since the amount of energy transmitted is minimized, the reflected energy back to the source is quite high. This can actually degrade the performance of engines and other sources. Opposite to absorptive mufflers, which dissipate the acoustic energy, reactive mufflers keep the energy contained within the system. See #The_reflector_muffler Reactive Mufflers for more information.

Examples

Reactive mufflers are the most widely used mufflers in combustion engines 1. Reactive mufflers are very efficient in low frequency applications (especially since simple lumped element analysis can be applied). Other application areas include: harsh environments (high temperature/velocity engines, turbines, etc.), specific frequency attenuation (using a helmholtz like device, a specific frequency can be toned to give total attenuation of radiated sound power), and a need for low radiated sound power (car mufflers, air conditioners, etc.).

Performance
There are 3 main metrics used to describe the performance of mufflers; Noise Reduction, Insertion Loss, and Transmission Loss. Typically when designing a muffler, 1 or 2 of these metrics is given as a desired value.

Noise Reduction (NR)
Defined as the difference between sound pressure levels on the source and receiver side. It is essentially the amount of sound power reduced between the location of the source and termination of the muffler system (it doesn't have to be the termination, but it is the most common location) [3].

$$NR = \left(L_{p1}-L_{p2}\right)$$

where $$L_{p1}$$ and $$L_{p2}$$ is sound pressure levels at source and receiver respectively. Although NR is easy to measure, pressure typically varies at source side due to standing waves [3].

Insertion Loss (IL)
Defined as difference of sound pressure level at the receiver with and without sound attenuating barriers. This can be realized, in a car muffler, as the difference in radiated sound power with just a straight pipe to that with an expansion chamber located in the pipe. Since the expansion chamber will attenuate some of the radiate sound power, the pressure at the receiver with sound attenuating barriers will be less. Therefore, a higher insertion loss is desired [3].

$$IL = \left(L_{p,without}-L_{p,with}\right)$$

where $$L_{p,without}$$ and $$L_{p,with}$$ are pressure levels at receiver without and with a muffler system respectively. Main problem with measuring IL is that the barrier or sound attenuating system needs to be removed without changing the source [3].

Transmission Loss (TL)
Defined as the difference between the sound power level of the incident wave to the muffler system and the transmitted sound power. For further information see Transmission Loss [3].

$$TL = 10log\left(\frac{1}{\tau}\right)$$ with $$\tau =\left(\frac{I_{t}}{I_{i}} \right)$$

where $$I_{t}$$ and $$I_{i}$$ are the transmitted and incident wave power respectively. From this expression, it is obvious the problem with measure TL is decomposing the sound field into incident and transmitted waves which can be difficult to do for complex systems (analytically).

Examples
(1) - For a plenum chamber (see figure below):

$$TL = -10 \log\left[S\left(\frac{\cos\theta}{2\pi d^2}+\frac{1-\alpha}{\alpha S_{w}}\right)\right]$$ in dB

where $$\alpha$$ is average absorption coefficient.

(2) - For an expansion (see figure below):

$$NR = 10 \log\left[ \frac{1}{2}\left| e^{-ikx_{s}}+\left( \frac{1-S}{1+S} \right)e^{ikx_{s}} \right|^2\left( 1+S \right)^2 \right]$$

$$IL = 10 \log\left[ \frac{\left( 1+S \right)^2}{4} \right]$$

$$TL = 10 \log\left[ \frac{\left( 1+S \right)^2}{4S} \right]$$

where $$S=\left( \frac{A_{2}}{A_{1}} \right)$$

(3) - For a helmholtz resonator (see figure below):

$$TL = 10 \log\left[ 1+\left( \frac{\left( \frac{c}{2S} \right)}{\omega LS_{b} - \left( \frac{c^2}{\omega V} \right)} \right)^2 \right]$$ in dB

Links

 * 1) Muffler/silencer applications and descriptions of performance criteria Exhaust Silencers
 * 2) Engineering Acoustics, Purdue University - ME 513.
 * 3) Sound Propagation Animations
 * 4) Exhaust Muffler Design
 * 5) Project Proposal & Outline


text:
False Friends of the Slavist/Map olovo

olovo

license: Creative Commons - Attribution Share-Alike - https://creativecommons.org/licenses/by-sa/4.0/

title: False Friends of the Slavist/Map olovo


text:
Acoustics/Flow-induced Oscillations of a Helmholtz Resonator

Introduction
The importance of flow excited acoustic resonance lies in the large number of applications in which it occurs. Sound production in organ pipes, compressors, transonic wind tunnels, and open sunroofs are only a few examples of the many applications in which flow excited resonance of Helmholtz resonators can be found.[4] An instability of the fluid motion coupled with an acoustic resonance of the cavity produce large pressure fluctuations that are felt as increased sound pressure levels.

Passengers of road vehicles with open sunroofs often experience discomfort, fatigue, and dizziness from self-sustained oscillations inside the car cabin. This phenomenon is caused by the coupling of acoustic and hydrodynamic flow inside a cavity which creates strong pressure oscillations in the passenger compartment in the 10 to 50 Hz frequency range. Some effects experienced by vehicles with open sunroofs when buffeting include: dizziness, temporary hearing reduction, discomfort, driver fatigue, and in extreme cases nausea. The importance of reducing interior noise levels inside the car cabin relies primarily in reducing driver fatigue and improving sound transmission from entertainment and communication devices.

This Wikibook page aims to theoretically and graphically explain the mechanisms involved in the flow-excited acoustic resonance of Helmholtz resonators. The interaction between fluid motion and acoustic resonance will be explained to provide a thorough explanation of the behavior of self-oscillatory Helmholtz resonator systems. As an application example, a description of the mechanisms involved in sunroof buffeting phenomena will be developed at the end of the page.

Feedback loop analysis
As mentioned before, the self-sustained oscillations of a Helmholtz resonator in many cases is a continuous interaction of hydrodynamic and acoustic mechanisms. In the frequency domain, the flow excitation and the acoustic behavior can be represented as transfer functions. The flow can be decomposed into two volume velocities.

 * qr: flow associated with acoustic response of cavity
 * qo: flow associated with excitation

Lumped parameter model
The lumped parameter model of a Helmholtz resonator consists of a rigid-walled volume open to the environment through a small opening at one end. The dimensions of the resonator in this model are much less than the acoustic wavelength, in this way allowing us to model the system as a lumped system.

Figure 2 shows a sketch of a Helmholtz resonator on the left, the mechanical analog on the middle section, and the electric-circuit analog on the right hand side. As shown in the Helmholtz resonator drawing, the air mass flowing through an inflow of volume velocity includes the mass inside the neck (Mo) and an end-correction mass (Mend). Viscous losses at the edges of the neck length are included as well as the radiation resistance of the tube. The electric-circuit analog shows the resonator modeled as a forced harmonic oscillator. [1] [2][3]

Figure 2

V: cavity volume

$$\rho$$: ambient density

c: speed of sound

S: cross-section area of orifice

K: stiffness

$$M_a$$: acoustic mass

$$C_a$$: acoustic compliance

The equivalent stiffness K is related to the potential energy of the flow compressed inside the cavity. For a rigid wall cavity it is approximately:

$$ K = \left(\frac{\rho c^2}{V}\right)S^2 $$

The equation that describes the Helmholtz resonator is the following:

$$ S \hat{P}_e =\frac{\hat{q}_e}{j\omega S}(-\omega ^2 M + j\omega R + K) $$

$$\hat{P}_e$$: excitation pressure

M: total mass (mass inside neck Mo plus end correction, Mend)

R: total resistance (radiation loss plus viscous loss)

From the electrical-circuit we know the following:

$$ M_a = \frac{L \rho}{S} $$

$$ C_a = \frac{V}{\rho c^2} $$

$$ L ' = \ L + \ 1.7 \ re $$

The main cavity resonance parameters are resonance frequency and quality factor which can be estimated using the parameters explained above (assuming free field radiation, no viscous losses and leaks, and negligible wall compliance effects)

$$ \omega_r^2 = \frac{1}{M_a C_a} $$

$$ f_r = \frac{c}{2 \pi} \sqrt{\frac{S}{L' V}} $$

The sharpness of the resonance peak is measured by the quality factor Q of the Helmholtz resonator as follows:

$$ Q = 2 \pi \sqrt{V (\frac{L'} {S})^3} $$

$$ f_r$$: resonance frequency in Hz

$$\omega_r$$: resonance frequency in radians

 * L: length of neck
 * L': corrected length of neck

From the equations above, the following can be deduced:

 * The greater the volume of the resonator, the lower the resonance frequencies.
 * If the length of the neck is increased, the resonance frequency decreases.

Production of self-sustained oscillations
The acoustic field interacts with the unstable hydrodynamic flow above the open section of the cavity, where the grazing flow is continuous. The flow in this section separates from the wall at a point where the acoustic and hydrodynamic flows are strongly coupled. [5]

The separation of the boundary layer at the leading edge of the cavity (front part of opening from incoming flow) produces strong vortices in the main stream. As observed in Figure 3, a shear layer crosses the cavity orifice and vortices start to form due to instabilities in the layer at the leading edge.

Figure 3

From Figure 3, L is the length of the inner cavity region, d denotes the diameter or length of the cavity length, D represents the height of the cavity, and $$\delta$$ describes the gradient length in the grazing velocity profile (boundary layer thickness).

The velocity in this region is characterized to be unsteady and the perturbations in this region will lead to self-sustained oscillations inside the cavity. Vortices will continually form in the opening region due to the instability of the shear layer at the leading edge of the opening.

How are vortices formed during buffeting?
In order to understand the generation and convection of vortices from the shear layer along the sunroof opening, the animation below has been developed. At a certain range of flow velocities, self-sustained oscillations inside the open cavity (sunroof) will be predominant. During this period of time, vortices are shed at the trailing edge of the opening and continue to be convected along the length of the cavity opening as pressure inside the cabin decreases and increases. Flow visualization experimentation is one method that helps obtain a qualitative understanding of vortex formation and conduction.

The animation below shows, in the middle, a side view of a car cabin with the sunroof open. As the air starts to flow at a certain mean velocity Uo, air mass will enter and leave the cabin as the pressure decreases and increases again. At the right hand side of the animation, a legend shows a range of colors to determine the pressure magnitude inside the car cabin. At the top of the animation, a plot of circulation and acoustic cavity pressure versus time for one period of oscillation is shown. The symbol x moving along the acoustic cavity pressure plot is synchronized with pressure fluctuations inside the car cabin and with the legend on the right. For example, whenever the x symbol is located at the point where t=0 (when the acoustic cavity pressure is minimum) the color of the car cabin will match that of the minimum pressure in the legend (blue).

The perturbations in the shear layer propagate with a velocity of the order of 1/2Uo which is half the mean inflow velocity. [5] After the pressure inside the cavity reaches a minimum (blue color) the air mass position in the neck of the cavity reaches its maximum outward position. At this point, a vortex is shed at the leading edge of the sunroof opening (front part of sunroof in the direction of inflow velocity). As the pressure inside the cavity increases (progressively to red color) and the air mass at the cavity entrance is moved inwards, the vortex is displaced into the neck of the cavity. The maximum downward displacement of the vortex is achieved when the pressure inside the cabin is also maximum and the air mass in the neck of the Helmholtz resonator (sunroof opening) reaches its maximum downward displacement. For the rest of the remaining half cycle, the pressure cavity falls and the air below the neck of the resonator is moved upwards. The vortex continues displacing towards the downstream edge of the sunroof where it is convected upwards and outside the neck of the resonator. At this point the air below the neck reaches its maximum upwards displacement.[4] And the process starts once again.

How to identify buffeting
Flow induced tests performed over a range of flow velocities are helpful to determine the change in sound pressure levels (SPL) inside the car cabin as inflow velocity is increased. The following animation shows typical auto spectra results from a car cabin with the sunroof open at various inflow velocities. At the top right hand corner of the animation, it is possible to see the inflow velocity and resonance frequency corresponding to the plot shown at that instant of time.

It is observed in the animation that the SPL increases gradually with increasing inflow velocity. Initially, the levels are below 80 dB and no major peaks are observed. As velocity is increased, the SPL increases throughout the frequency range until a definite peak is observed around a 100 Hz and 120 dB of amplitude. This is the resonance frequency of the cavity at which buffeting occurs. As it is observed in the animation, as velocity is further increased, the peak decreases and disappears.

In this way, sound pressure level plots versus frequency are helpful in determining increased sound pressure levels inside the car cabin to find ways to minimize them. Some of the methods used to minimize the increased SPL levels achieved by buffeting include: notched deflectors, mass injection, and spoilers.

=Useful websites=

This link: takes you to the website of EXA Corporation, a developer of PowerFlow for Computational Fluid Dynamics (CFD) analysis.

This link: is a small news article about the current use of(CFD) software to model sunroof buffeting.

This link: is a small industry brochure that shows the current use of CFD for sunroof buffeting.

=References=
 * 1) Acoustics: An introduction to its Physical Principles and Applications ; Pierce, Allan D., Acoustical Society of America, 1989.
 * 2) Prediction and Control of the Interior Pressure Fluctuations in a Flow-excited Helmholtz resonator ; Mongeau, Luc, and Hyungseok Kook., Ray W. Herrick Laboratories, Purdue University, 1997.
 * 3) Influence of leakage on the flow-induced response of vehicles with open sunroofs ; Mongeau, Luc, and Jin-Seok Hong., Ray W. Herrick Laboratories, Purdue University.
 * 4) Fluid dynamics of a flow excited resonance, part I: Experiment ; P.A. Nelson, Halliwell and Doak.; 1991.
 * 5) An Introduction to Acoustics ; Rienstra, S.W., A. Hirschberg., Report IWDE 99–02, Eindhoven University of Technology, 1999.


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Acoustics/Acoustics in Violins

For a detailed anatomy of the violin, please refer to Atelierla Bussiere.

General concept
When a violinist bows a string, which can produce vibrations with abundant harmonics, the vibrations of the strings are structurally transmitted to the bridge and the body of the instrument through the bridge. The bridge transmits the vibrational energy produced by the strings to the body through its feet, further triggering the vibration of body. The vibration of the body determines sound radiation and sound quality, along with the resonance of the cavity.

String
The vibration pattern of the strings can be easily be observed. To the naked eye, the string appears to move back and forth in a parabolic shape (see figure), which resembles the first mode of free vibration of a stretched string. The vibration of strings was first investigated by Hermann Von Helmholtz, the famous mathematician and physicist in 19th century. A surprising scenario was discovered that the string actually moves in an inverse “V” shape rather than parabolas (see figure). What we see is just an envelope of the motion of the string. To honor his findings, the motion of bowed strings had been called “Helmholtz motion.”

Bridge
The primary role of the bridge is to transform the motion of vibrating strings into periodic driving forces by its feet to the top plate of the violin body. The configuration of the bridge can be referred to the figure. The bridge stands on the belly between f holes, which have two primary functions. One is to connect the air inside the body with outside air, and the other one is to make the belly between f holes move more easily than other parts of the body. The fundamental frequency of a violin bridge was found to be around 3000 Hz when it is on a rigid support, and it is an effective energy-transmitting medium to transmit the energy from the string to body at frequencies from 1 kHz to 4 kHz, which is in the range of keen sensitivity of human hearing. In order to darken the sound of violin, the player attaches a mute on the bridge. The mute is actually an additional mass which reduces the fundamental frequency of the bridge. As a result, the sound at higher frequencies is diminished since the force transferred to the body has been decreased. On the other hand, the fundamental frequency of the bridge can be raised by attaching an additional stiffness in the form of tiny wedges, and the sound at higher frequencies will be amplified accordingly.

The sound post connects the flexible belly to the much stiffer back plate. The sound post can prevent the collapse of the belly due to high tension force in the string, and, at the same time, couples the vibration of the plate. The bass bar under the belly extends beyond the f holes and transmits the force of the bridge to a larger area of the belly. As can be seen in the figure, the motion of the treble foot is restricted by the sound post, while, conversely, the foot over bass bar can move up and down more easily. As a result, the bridge tends to move up and down, pivoting about the treble foot. The forces appearing at the two feet remain equal and opposite up to 1 kHz. At higher frequencies, the forces become uneven. The force on the soundpost foot predominates at some frequencies, while it is the bass bar foot at some.

Body
The body includes top plate, back plate, the sides, and the air inside, all of which serve to transmit the vibration of the bridge into the vibration of air surrounding the violin. For this reason, the violin needs a relatively large surface area to push enough amount of air back and forth. Thus, the top and back plates play important roles in the mechanism. Violin makers have traditionally paid much attention to the vibration of the top and back plates of the violin by listening to the tap tones, or, recently, by observing the vibration mode shapes of the body plates. The vibration modes of an assembled violin are, however, much more complicated.

The vibration modes of top and back plates can be easily observed in a similar technique first performed by Ernest Florens Friedrich Chaldni (1756–1827), who is often respectfully referred “the father of acoustics.” First, the fine sand is uniformly sprinkled on the plate. Then, the plate can be resonated, either by a powerful sound wave tuned to the desired frequencies, by being bowed by a violin bow, or by being excited mechanically or electromechanically at desired frequencies. Consequently, the sand disperses randomly due to the vibration of plate. Some of the sand falls outside the region of plate, while some of the sand is collected by the nodal regions, which have relatively small movement, of the plate. Hence, the mode shapes of the plate can be visualized in this manner, which can be referred to the figures in the reference site, Violin Acoustics. The first seven modes of the top and back plates of violin are presented, with nodal lines depicted by using black sands.

The air inside the body is also important, especially in the range of lower frequencies. It is like the air inside a bottle when you blow into the neck, or, as known as Helmholtz resonance, which has its own modes of vibration. The air inside the body can communicate with air outside through the f holes, and the outside air serves as medium carrying waves from the violin.

See www.violinbridges.co.uk for more articles on bridges and acoustics.

Sound radiation
A complete description of sound radiation of a violin should include the information about radiation intensity as functions both of frequency and location. The sound radiation can be measured by a microphone connected to a pressure level meter which is rotatably supported on a stand arm around the violin, while the violin is fastened at the neck by a clip. The force is introduced into the violin by using a miniature impact hammer at the upper edge of the bridge in the direction of bowing. The detail can be referred to Martin Schleske, master studio for violinmaking. The radiation intensity of different frequencies at different locations can be represented by directional characteristics, or acoustic maps. The directional characteristics of a violin can be shown in the figure in the website of Martin Schleske, where the radial distance from the center point represents the absolute value of the sound level (re 1Pa/N) in dB, and the angular coordinate of the full circle indicates the measurement point around the instrument. According to the directional characteristics of violins, the principal radiation directions for the violin in the horizontal plane can be established. For more detail about the principal radiation direction for violins at different frequencies, please refer to reference (Meyer 1972).

References and other links

 * Violin Acoustics
 * Paul Galluzzo's Homepage
 * Martin Schleske, master studio for violinmaking
 * Atelierla Bussiere
 * Fletcher, N. H., and Rossing, T. D., The physics of musical instrument, Springer-Verlag, 1991
 * Meyer, J., "Directivity of bowed stringed instruments and its effect on orchestral sound in concert halls", J. Acoustic. Soc. Am., 51, 1972, pp. 1994–2009

title: Acoustics/Acoustics in Violins


text:
Acoustics/Noise from Cooling Fans

Proposal
As electric/electronic devices get smaller and functional, the noise of cooling device becomes important. This page will explain the origins of noise generation from small axial cooling fans used in electronic goods like desktop/laptop computers. The source of fan noises includes aerodynamic noise as well as operating sound of the fan itself. This page will be focused on the aerodynamic noise generation mechanisms.

Introduction
If one opens a desktop computer, they may find three (or more) fans. For example, a fan is typically found on the heat sink of the CPU, in the back panel of the power supply unit, on the case ventilation hole, on the graphics card, and even on the motherboard chipset if it is a recent one. Computer noise which annoys many people is mostly due to cooling fans, if the hard drive(s) is fairly quiet. When Intel Pentium processors were first introduced, there was no need to have a fan on the CPU, however, contemporary CPUs cannot function even for several seconds without a cooling fan. As CPU densities increase, the heat transfer for nominal operation requires increased airflow, which causes more and more noise. The type of fans commonly used in desktop computers are axial fans, and centrifugal blowers in laptop computers. Several fan types are shown here (pdf format). Different fan types have different characteristics of noise generation and performance. The axial flow fan is mainly considered in this page.

Noise Generation Mechanisms
The figure below shows a typical noise spectrum of a 120 mm diameter electronic device cooling fan. One microphone was used at a point 1 m from the upstream side of the fan. The fan has 7 blades, 4 struts for motor mounting and operates at 13V. Certain amount of load is applied. The blue plot is background noise of anechoic chamber, and the green one is sound loudness spectrum when the fan is running. (*BPF = Blade Passing Frequency) Each noise elements shown in this figure is caused by one or more of following generation mechanisms.

Blade Thickness Noise - Monopole (But very weak)
Blade thickness noise is generated by volume displacement of fluid. A fan blade has its thickness and volume. As the rotor rotates, the volume of each blade displaces fluid volume, then they consequently fluctuate pressure of near field, and noise is generated. This noise is tonal at the running frequency and generally very weak for cooling fans, because their RPM is relatively low. Therefore, thickness of fan blades hardly affects to electronic cooling fan noise. (This kind of noise can become severe for high speed turbomachines like helicopter rotors.)

Uniform Inlet Flow (Negligible)
The sound generation due to uniform and steady aerodynamic force has a characteristic very similar to the blade thickness noise. It is very weak for low speed fans, and depends on fan RPM. Since some steady blade forces are necessary for a fan to do its duty even in an ideal condition, this kind of noise is impossible to avoid. It is known that this noise can be reduced by increasing the number of blades.

Non-uniform Inlet Flow
Non-uniform (still steady) inlet flow causes non-uniform aerodynamic forces on blades as their angular positions change. This generates noise at blade passing frequency and its harmonics. It is one of the major noise sources of electronic cooling fans.

Rotor-Casing interaction
If the fan blades are very close to a structure which is not symmetric, unsteady interaction forces to blades are generated. Then the fan experiences a similar running condition as lying in non-uniform flow field. See Acoustics/Rotor Stator Interactions for details.

Impulsive Noise (Negligible)
This noise is caused by the interaction between a blade and blade-tip-vortex of the preceding blade, and not severe for cooling fans.

Rotating Stall

 * Click here to read the definition and an aerodynamic description of stall.

The noise due to stall is a complex phenomenon that occurs at low flow rates. For some reason, if flow is locally disturbed, it can cause stall on one of the blades. As a result, the upstream passage on this blade is partially blocked. Therefore, the mean flow is diverted away from this passage. This causes increasing of the angle of attack on the closest blade at the upstream side of the originally stalled blade, the flow is again stalled there. On the other hand, the other side of the first blade is un-stalled because of reduction of flow angle. repeatedly, the stall cell turns around the blades at about 30~50% of the running frequency, and the direction is opposite to the blades. This series of phenomenon causes unsteady blade forces, and consequently generates noise and vibrations.

Non-uniform Rotor Geometry
Asymmetry of rotor causes noise at the rotating frequency and its harmonics (not blade passing frequency obviously), even when the inlet flow is uniform and steady.

Unsteady Flow Field
Unsteady flow causes random forces on the blades. It spreads the discrete spectrum noises and makes them continuous. In case of low-frequency variation, the spread continuous spectral noise is around rotating frequency, and narrowband noise is generated. The stochastic velocity fluctuations of inlet flow generates broadband noise spectrum. The generation of random noise components is covered by the following sections.

Turbulent Boundary Layer
Even in the steady and uniform inlet flow, there exist random force fluctuations on the blades. That is from turbulent blade boundary layer. Some noise is generated for this reason, but dominant noise is produced by the boundary layer passing the blade trailing edge. The blade trailing edges scatter the non-propagating near-field pressure into a propagatable sound field.

Incident Turbulent
Velocity fluctuations of the intake flow with a stochastic time history generate random forces on blades, and a broadband spectrum noise.

Vortex Shedding
For some reason, a vortex can separate from a blade. Then the circulating flow around the blade starts to be changed. This causes non-uniform forces on blades, and noises. A classical example for this phenomenon is 'Karman vortex street'. (some images and animations.) Vortex shedding mechanism can occur in a laminar boundary layer of low speed fan and also in a turbulent boundary layer of high frequency fan.

Flow Separation
Flow separation causes stall explained above. This phenomenon can cause random noise, which spreads all the discrete spectrum noises, and turns the noise into broadband.

Tip Vortex
Since cooling fans are ducted axial flow machines, the annular gap between the blade tips and the casing is important parameter for noise generation. While rotating, there is another flow through the annular gap due to pressure difference between upstream and downstream of fan. Because of this flow, tip vortex is generated through the gap, and broadband noise increases as the annular gap gets bigger.

Installation Effects
Once a fan is installed, even though the fan is well designed acoustically, unexpected noise problem can come up. It is called as installation effects, and two types are applicable to cooling fans.

Effect of Inlet Flow Conditions
A structure that affects the inlet flow of a fan causes installation effects. For example Hoppe & Neise [3] showed that with and without a bellmouth nozzle at the inlet flange of 500mm fan can change the noise power by 50dB (This application is for much larger and noisier fan though).

Acoustic Loading Effect
This effect is shown on duct system applications. Some high performance graphic cards apply duct system for direct exhaustion. The sound power generated by a fan is not only a function of its impeller speed and operating condition, but also depends on the acoustic impedances of the duct systems connected to its inlet and outlet. Therefore, fan and duct system should be matched not only for aerodynamic noise reasons but also because of acoustic considerations.

Closing Comment
Noise reduction of cooling fans has some restrictions: 1. Active noise control is not economically effective. 80mm cooling fans are only 5~10 US dollars. It is only applicable for high-end electronic products. 2. Restricting certain aerodynamic phenomenon for noise reduction can cause serious performance reduction of the fan. Increasing RPM of the fan is of course much more dominant factor for noise. Different stories of fan noise are introduced at some of the linked sites below like active RPM control or noise comparison of various bearings used in fans. If blade passing noise is dominant, a muffler would be beneficial.

Links to Interesting Sites about Fan Noise
Some practical issue of PC noise are presented at the following sites. Cooling Fan Noise Comparison - Sleeve Bearing vs. Ball Bearing (pdf format) Brief explanation of fan noise origins and noise reduction suggestions Effect of sweep angle comparison Comparisons of noise from various 80mm fans Noise reduction of a specific desktop case Noise reduction of another specific desktop case Informal study for noise from CPU cooling fan Informal study for noise from PC case fans Active fan speed optimizators for minimum noise from desktop computers
 * Some general fan noise reduction techniques
 * Brüel & Kjær - acoustic testing device company. Various applications are presented in "applications" tap
 * "Fan Noise Solutions" by cpemma
 * How To Build A Computer/Quiet PC

title: Acoustics/Noise from Cooling Fans


text:
Talk:Find Employment

pages
current format of this book is relatively monolithic. Of course, there isn't enough material yet to justify expanding to multiple pages, yet. --Whiteknight (talk) (current) 22:07, 7 March 2006 (UTC)
 * ✅. Good job, everyone, on expanding this book to multiple pages. --DavidCary (talk) 22:24, 1 December 2009 (UTC)

Move/Merge
"Preparing for an Employment Interview": I think that this book should be merged with the Writing a Résumé book, and simultaneously moved to a better namespace, such as "Getting A Job", or something like that. --Whiteknight (talk) (current) 23:51, 13 April 2006 (UTC)
 * Books have been merged and moved to Find Employment. --Whiteknight (talk) (projects) 13:43, 23 May 2006 (UTC)

I think it was right to merge in the resume part. I am not so sure about the title though. The current title "Find Employment" is very descriptive, I like that.

Offer
I am the webmaster of [WorkOne.com] a career and employment portal, we need a book like this for our educational purposes. We would be willing to sponsor someone to extend the scope of this book to a full length more in depth ebook. We would be willing to pay in the houndreds of USD for this job to be done. Please contact me at workone@workone.com for details. Thank you. May 23. 2006.
 * I sent you an email. Feel free to contact me by email, or leave messages here. --Whiteknight (talk) (projects) 13:43, 23 May 2006 (UTC)
 * PS Never mind, the email address you supplied came back as invalid. provide a better email address, and I'll send you some information. --Whiteknight (talk) (projects) 13:45, 23 May 2006 (UTC)

Hi there Whiteknight, that was fast! :) Nice. I will send you an email.

 * If you send me an email, make sure it has "Wikibooks" or something similar in the subject, or it is liable to be deleted as spam. --Whiteknight (talk) (projects) 14:28, 9 August 2006 (UTC)

Please note that I have stated adding information to this book project from: Occupational Outlook Handbook (OOH), 2006-07 Edition http://www.bls.gov/oco/ This book is provided by the Department of Labor and everything in it is in the public domain, as seen here (bottom of page): http://www.bls.gov/oco/oohinfo_faq.htm?/oco/home.htm#pub3 "# May I reproduce material from the Occupational Outlook Handbook or Career Guide to Industries sites? Material on these sites is in the public domain and, with appropriate credit, may be reproduced without permission."

Information that might be added
We should look into good tips and advice around the web and other places to find material and best practices that we can write into the book. Here are some sources that deserves investigating.

Get invited to the interview

Books: What colour is you parashute? This is the industry classic, it has a lot of good advice in it. The advice as such is not proprietary and some of it can we rewritten.

Ideas for other topics in the book: - How to use the various online services like Monster, Job.com etc. Tricks and what to expect. - How to use the different career companies. - Tips and tricks and advice from more professionals. - How to dress for sucess. - Specifics for different industries, The Handbook has a lot of material on this. I have not so far started adding this, but it might be a good idea. - Statistics on salaries. - How to negotiate wages. - How to stay motivated during the job search. - Advice on moving.

In the extension a book like this can easily border into self improvement. Where does the line go between "Dress well, dont swear..." and pure self improvement advice. Is self improvement something this book should do, or is it that a topic for a different book?

Rearanging Content
I have taken the initiative (being bold) to rearrange content into chapters that seem more appropriate. Also I would add related resources: tips on how to improve my resume, career coach, etc. For example, in the "your career" chapter, there is/was a lot of info about finding prospects, which seems to make more sense to be in the finding prospects chapter... Feel free to give feedback if I am on the wrong track. But it seems that there is a lot of good content in this wikibook, it just isn't well organized yet. --Jacob Walker 00:00, 5 September 2007 (UTC)

Disagreement about Audience
I think the note that this textbook is geared for the highschool level is too limiting. I teach adults, and I find many of them really need this knowledge. Unless someone is independently wealthy from an outside source, or an entrepreneur, then they will need to find employment. --Jacob Walker (talk) 04:58, 2 February 2008 (UTC)

Sources for Book
The following might be useful as sources to cite for the wikibook, although probably should not be used directly as to not "advertise" them:

 * Susan Ireland Resumes

title: Talk:Find Employment


text:
False Friends of the Slavist/Map otьcь

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title: False Friends of the Slavist/Map otьcь


text:
Acoustics/Human Vocal Fold

Physiology of vocal fold
The human vocal fold is a set of lip-like tissues located inside the larynx, and is the source of sound for humans and many animals.

The larynx is located at the top of the trachea. It is mainly composed of cartilages and muscles, and the largest cartilage, thyroid, is well known as the "Adam's Apple."

The organ has two main functions; to act as the last protector of the airway, and to act as a sound source for voice. This page focuses on the latter function.

Links on Physiology:
 * Discover The Larynx

Voice production
Although the science behind sound production for a vocal fold is complex, it can be thought of as similar to a brass player's lips, or a whistle made out of grass. Basically, vocal folds (or lips or a pair of grass) make a constriction to the airflow, and as the air is forced through the narrow opening, the vocal folds oscillate. This causes a periodical change in the air pressure, which is perceived as sound.

Vocal Folds Video

When the airflow is introduced to the vocal folds, it forces open the two vocal folds which are nearly closed initially. Due to the stiffness of the folds, they will then try to close the opening again. And now the airflow will try to force the folds open etc... This creates an oscillation of the vocal folds, which in turn, as I stated above, creates sound. However, this is a damped oscillation, meaning it will eventually achieve an equilibrium position and stop oscillating. So how are we able to "sustain" sound?

As it will be shown later, the answer seems to be in the changing shape of vocal folds. In the opening and the closing stages of the oscillation, the vocal folds have different shapes. This affects the pressure in the opening, and creates the extra pressure needed to push the vocal folds open and sustain oscillation. This part is explained in more detail in the "Model" section.

This flow-induced oscillation, as with many fluid mechanics problems, is not an easy problem to model. Numerous attempts to model the oscillation of vocal folds have been made, ranging from a single mass-spring-damper system to finite element models. In this page I would like to use my single-mass model to explain the basic physics behind the oscillation of a vocal fold.

Information on vocal fold models: National Center for Voice and Speech

Model

The most simple way of simulating the motion of vocal folds is to use a single mass-spring-damper system as shown above. The mass represents one vocal fold, and the second vocal fold is assumed to be symmetry about the axis of symmetry. Position 3 represents a location immediately past the exit (end of the mass), and position 2 represents the glottis (the region between the two vocal folds).

The pressure force
The major driving force behind the oscillation of vocal folds is the pressure in the glottis. The Bernoulli's equation from fluid mechanics states that:

$$P_1 + \frac{1}{2}\rho U^2 + \rho gh = Constant$$ -EQN 1

Neglecting potential difference and applying EQN 1 to positions 2 and 3 of Figure 1,

$$P_2 + \frac{1}{2}\rho U_2^2 = P_3 + \frac{1}{2}\rho U_3^2$$ -EQN 2

Note that the pressure and the velocity at position 3 cannot change. This makes the right hand side of EQN 2 constant. Observation of EQN 2 reveals that in order to have oscillating pressure at 2, we must have oscillation velocity at 2. The flow velocity inside the glottis can be studied through the theories of the orifice flow.

The constriction of airflow at the vocal folds is much like an orifice flow with one major difference: with vocal folds, the orifice profile is continuously changing. The orifice profile for the vocal folds can open or close, as well as change the shape of the opening. In Figure 1, the profile is converging, but in another stage of oscillation it takes a diverging shape.

The orifice flow is described by Blevins as:

$$ U = C\frac{2(P_1 - P_3)}{\rho}$$ -EQN 3

Where the constant C is the orifice coefficient, governed by the shape and the opening size of the orifice. This number is determined experimentally, and it changes throughout the different stages of oscillation.

Solving equations 2 and 3, the pressure force throughout the glottal region can be determined.

The Collision Force
As the video of the vocal folds shows, vocal folds can completely close during oscillation. When this happens, the Bernoulli equation fails. Instead, the collision force becomes the dominating force. For this analysis, Hertz collision model was applied.

$$ F_H = k_H delta^{3/2} (1 + b_H delta')$$ -EQN 4

where

$$ k_H = \frac{4}{3} \frac{E}{1 - \mu_H^2} \sqrt{r}$$

Here delta is the penetration distance of the vocal fold past the line of symmetry.

Simulation of the model
The pressure and the collision forces were inserted into the equation of motion, and the result was simulated.

Figure 2 shows that an oscillating volumetric flow rate was achieved by passing a constant airflow through the vocal folds. When simulating the oscillation, it was found that the collision force limits the amplitude of oscillation rather than drive the oscillation. Which tells us that the pressure force is what allows the sustained oscillation to occur.

The acoustic output
This model showed that the changing profile of glottal opening causes an oscillating volumetric flow rate through the vocal folds. This will in turn cause an oscillating pressure past the vocal folds. This method of producing sound is unusual, because in most other means of sound production, air is compressed periodically by a solid such as a speaker cone.

Past the vocal folds, the produced sound enters the vocal tract. Basically this is the cavity in the mouth as well as the nasal cavity. These cavities act as acoustic filters, modifying the character of the sound. These are the characters that define the unique voice each person produces.

Related links

 * FEA Model
 * Two Mass Model

title: Acoustics/Human Vocal Fold


text:
Acoustics/Piezoelectric Transducers

=Introduction=

Piezoelectricity from the Greek word "piezo" means pressure electricity. Certain crystalline substances generate electric charges under mechanical stress and conversely experience a mechanical strain in the presence of an electric field. The piezoelectric effect describes a situation where the transducing material senses input mechanical vibrations and produces a charge at the frequency of the vibration. An AC voltage causes the piezoelectric material to vibrate in an oscillatory fashion at the same frequency as the input current.

Quartz is the best known single crystal material with piezoelectric properties. Strong piezoelectric effects can be induced in materials with an ABO3, Perovskite crystalline structure. 'A' denotes a large divalent metal ion such as lead and 'B' denotes a smaller tetravalent ion such as titanium or zirconium.

For any crystal to exhibit the piezoelectric effect, its structure must have no center of symmetry. Either a tensile or compressive stress applied to the crystal alters the separation between positive and negative charge sights in the cell causing a net polarization at the surface of the crystal. The polarization varies directly with the applied stress and is direction dependent so that compressive and tensile stresses will result in electric fields of opposite voltages.

=Vibrations & Displacements=

Piezoelectric ceramics have non-centrosymmetric unit cells below the Curie temperature and centrosymmetric unit cells above the Curie temperature. Non-centrosymmetric structures provide a net electric dipole moment. The dipoles are randomly oriented until a strong DC electric field is applied causing permanent polarization and thus piezoelectric properties.

A polarized ceramic may be subjected to stress causing the crystal lattice to distort changing the total dipole moment of the ceramic. The change in dipole moment due to an applied stress causes a net electric field which varies linearly with stress.

=Dynamic Performance=

The dynamic performance of a piezoelectric material relates to how it behaves under alternating stresses near the mechanical resonance. The parallel combination of C2 with L1, C1, and R1 in the equivalent circuit below control the transducers reactance which is a function of frequency.

Frequency Response
The graph below shows the impedance of a piezoelectric transducer as a function of frequency. The minimum value at fn corresponds to the resonance while the maximum value at fm corresponds to anti-resonance. Superscript textItalic text

=Resonant Devices=

Non resonant devices may be modeled by a capacitor representing the capacitance of the piezoelectric with an impedance modeling the mechanically vibrating system as a shunt in the circuit. The impedance may be modeled as a capacitor in the non resonant case which allows the circuit to reduce to a single capacitor replacing the parallel combination.

For resonant devices the impedance becomes a resistance or static capacitance at resonance. This is an undesirable effect. In mechanically driven systems this effect acts as a load on the transducer and decreases the electrical output. In electrically driven systems this effect shunts the driver requiring a larger input current. The adverse effect of the static capacitance experienced at resonant operation may be counteracted by using a shunt or series inductor resonating with the static capacitance at the operating frequency.

=Applications=

Mechanical Measurement
Because of the dielectric leakage current of piezoelectrics they are poorly suited for applications where force or pressure have a slow rate of change. They are, however, very well suited for highly dynamic measurements that might be needed in blast gauges and accelerometers.

Ultrasonic
High intensity ultrasound applications utilize half wavelength transducers with resonant frequencies between 18 kHz and 45 kHz. Large blocks of transducer material is needed to generate high intensities which is makes manufacturing difficult and is economically impractical. Also, since half wavelength transducers have the highest stress amplitude in the center the end sections act as inert masses. The end sections are often replaced with metal plates possessing a much higher mechanical quality factor giving the composite transducer a higher mechanical quality factor than a single-piece transducer.

The overall electro-acoustic efficiency is:

Qm0 = unloaded mechanical quality factor QE = electric quality factor QL = quality factor due to the acoustic load alone

The second term on the right hand side is the dielectric loss and the third term is the mechanical loss.

Efficiency is maximized when:

then:

The maximum ultrasonic efficiency is described by:

Applications of ultrasonic transducers include: Welding of plastics Atomization of liquids Ultrasonic drilling Ultrasonic cleaning Ultrasonic foils in the paper machine wet end for more uniform fibre distribution Ultrasound Non-destructive testing etc.

=More Information and Source of Information= MorganElectroCeramics

Ultra Technology

title: Acoustics/Piezoelectric Transducers


text:
Acoustics/Microphone Technique

General technique

 * 1) A microphone should be used whose frequency response will suit the frequency range of the voice or instrument being recorded.
 * 2) Vary microphone positions and distances until you achieve the monitored sound that you desire.
 * 3) In the case of poor room acoustics, place the microphone very close to the loudest part of the instrument being recorded or isolate the instrument.
 * 4) Personal taste is the most important component of microphone technique.  Whatever sounds right to you, is right.

Dynamic microphones
These are the most common general-purpose microphones. They do not require power to operate. If you have a microphone that is used for live performance, it is probably a dynamic mic.

They have the advantage that they can withstand very high sound pressure levels (high volume) without damage or distortion, and tend to provide a richer, more intense sound than other types. Traditionally, these mics did not provide as good a response on the highest frequencies (particularly above 10 kHz), but some recent models have come out that attempt to overcome this limitation.

In the studio, dynamic mics are often used for high sound pressure level instruments such as drums, guitar amps and brass instruments. Models that are often used in recording include the Shure SM57 and the Sennheiser MD421.

Condenser microphones
These microphones are often the most expensive microphones a studio owns. They require power to operate, either from a battery or phantom power, provided using the mic cable from an external mixer or pre-amp. These mics have a built-in pre-amplifier that uses the power. Some vintage microphones have a tube amplifier, and are referred to as tube condensers.

While they cannot withstand the very high sound pressure levels that dynamic mics can, they provide a flatter frequency response, and often the best response at the highest frequencies. Not as good at conveying intensity, they are much better at providing a balanced accurate sound.

Condenser mics come with a variety of sizes of transducers. They are usually grouped into smaller format condensers, which often are long cylinders about the size of a nickel coin in diameter, and larger format condensers, the transducers of which are often about an inch in diameter or slightly larger.

In the studio, condenser mics are often used for instruments with a wide frequency range, such as an acoustic piano, acoustic guitar, voice, violin, cymbals, or an entire band or chorus. On louder instruments they do not use close miking with condensers. Models that are often used in recording include the Shure SM81 (small format), AKG C414 (large format) and Neumann U87 (large format).

Ribbon microphones
Ribbon microphones are often used as an alternative to condenser microphones. Some modern ribbon microphones do not require power, and some do. The first ribbon microphones, developed at RCA in the 1930s, required no power, were quite fragile and could be destroyed by just blowing air through them. Modern ribbon mics are much more resiliant, and can be used with the same level of caution as condenser mics.

Ribbon microphones provide a warmer sound than a condenser mic, with a less brittle top end. Some vocalists (including Paul McCartney) prefer them to condenser mics. In the studio they are used on vocals, violins, and even drums. Popular models for recording include the Royer R121 and the AEA R84.

Close miking
When miking at a distance of 1 inch to about 1 foot from the sound source, it is considered close miking. This technique generally provides a tight, present sound quality and does an effective job of isolating the signal and excluding other sounds in the acoustic environment.

Bleed
Bleeding occurs when the signal is not properly isolated and the microphone picks up another nearby instrument. This can make the mixdown process difficult if there are multiple voices on one track. Use the following methods to prevent leakage:

 * Place the microphones closer to the instruments.
 * Move the instruments farther apart.
 * Put some sort of acoustic barrier between the instruments.
 * Use directional microphones.

A B miking
The A B miking distance rule (ratio 3 - 1) is a general rule of thumb for close miking. To prevent phase anomalies and bleed, the microphones should be placed at least three times as far apart as the distance between the instrument and the microphone.

Distant miking
Distant miking refers to the placement of microphones at a distance of 3 feet or more from the sound source. This technique allows the full range and balance of the instrument to develop and it captures the room sound. This tends to add a live, open feeling to the recorded sound, but careful consideration needs to be given to the acoustic environment.

Accent miking
Accent miking is a technique used for solo passages when miking an ensemble. A soloist needs to stand out from an ensemble, but placing a microphone too close will sound unnaturally present compared the distant miking technique used with the rest of the ensemble. Therefore, the microphone should be placed just close enough to the soloist so that the signal can be mixed effectively without sounding completely excluded from the ensemble.

Ambient miking
Ambient miking is placing the microphones at such a distance that the room sound is more prominent than the direct signal. This technique is used to capture audience sound or the natural reverberation of a room or concert hall.

Stereo
Stereo miking is simply using two microphones to obtain a stereo left-right image of the sound. A simple method is the use of a spaced pair, which is placing two identical microphones several feet apart and using the difference in time and amplitude to create the image. Great care should be taken in the method as phase anomalies can occur due to the signal delay. This risk of phase anomaly can be reduced by using the X/Y method, where the two microphones are placed with the grills as close together as possible without touching. There should be an angle of 90 to 135 degrees between the mics. This technique uses only amplitude, not time, to create the image, so the chance of phase discrepancies is unlikely.

Surround
To take advantage of 5.1 sound or some other surround setup, microphones may be placed to capture the surround sound of a room. This technique essentially stems from stereo technique with the addition of more microphones. Because every acoustic environment is different, it is difficult to define a general rule for surround miking, so placement becomes dependent on experimentation. Careful attention must be paid to the distance between microphones and potential phase anomalies.

Amplifiers
When miking an amplified speaker, such as for electric guitars, the mic should be placed 2 to 12 inches from the speaker. Exact placement becomes more critical at a distance of less than 4 inches. A brighter sound is achieved when the mic faces directly into the center of the speaker cone and a more mellow sound is produced when placed slightly off-center. Placing off-center also reduces amplifier noise.

A bigger sound can often be achieved by using two mics. The first mic should be a dynamic mic, placed as described in the previous paragraph. Add to this a condenser mic placed at least 3 times further back (remember the 3:1 rule), which will pickup the blended sound of all speakers, as well as some room ambience. Run the mics into separate channels and combine them to your taste.

Brass instruments
High sound-pressure levels are produced by brass instruments due to the directional characteristics of mid to mid-high frequencies. Therefore, for brass instruments such as trumpets, trombones, and tubas, microphones should face slightly off of the bell's center at a distance of one foot or more to prevent overloading from wind blasts.

Guitars
Technique for acoustic guitars is dependent on the desired sound. Placing a microphone close to the sound hole will achieve the highest output possible, but the sound may be bottom-heavy because of how the sound hole resonates at low frequencies. Placing the mic slightly off-center at 6 to 12 inches from the hole will provide a more balanced pickup. Placing the mic closer to the bridge with the same working distance will ensure that the full range of the instrument is captured.

A technique that some engineers use places a large-format condenser mic 12-18 inches away from the 12th fret of the guitar, and a small-format condenser very close to the strings nearby. Combining the two signals can produce a rich tone.

Pianos
Ideally, microphones would be placed 4 to 6 feet from the piano to allow the full range of the instrument to develop before it is captured. This isn't always possible due to room noise, so the next best option is to place the microphone just inside the open lid. This applies to both grand and upright pianos.

Percussion
One overhead microphone can be used for a drum set, although two are preferable. If possible, each component of the drum set should be miked individually at a distance of 1 to 2 inches as if they were their own instrument. This also applies to other drums such as congas and bongos. For large, tuned instruments such as xylophones, multiple mics can be used as long as they are spaced according to the 3:1 rule. Typically, dynamic mics are used for individual drum miking, while small-format condensers are used for the overheads.

Voice
Standard technique is to put the microphone directly in front of the vocalist's mouth, although placing slightly off-center can alleviate harsh consonant sounds (such as "p") and prevent overloading due to excessive dynamic range.

Woodwinds
A general rule for woodwinds is to place the microphone around the middle of the instrument at a distance of 6 inches to 2 feet. The microphone should be tilted slightly towards the bell or sound hole, but not directly in front of it.

Sound Propagation
It is important to understand how sound propagates due to the nature of the acoustic environment so that microphone technique can be adjusted accordingly. There are four basic ways that this occurs:

Reflection
Sound waves are reflected by surfaces if the object is as large as the wavelength of the sound. It is the cause of echo (simple delay), reverberation (many reflections cause the sound to continue after the source has stopped), and standing waves (the distance between two parallel walls is such that the original and reflected waves in phase reinforce one another).

Absorption
Sound waves are absorbed by materials rather than reflected. This can have both positive and negative effects depending on whether you desire to reduce reverberation or retain a live sound.

Diffraction
Objects that may be between sound sources and microphones must be considered due to diffraction. Sound will be stopped by obstacles that are larger than its wavelength. Therefore, higher frequencies will be blocked more easily than lower frequencies.

Refraction
Sound waves bend as they pass through mediums with varying density. Wind or temperature changes can cause sound to seem like it is literally moving in a different direction than expected.

title: Acoustics/Microphone Technique


text:
Acoustics/Sealed Box Subwoofer Design

A sealed or closed box baffle is the most basic but often the cleanest sounding sub-woofer box design. The sub-woofer box in its most simple form, serves to isolate the back of the speaker from the front, much like the theoretical infinite baffle. The sealed box provides simple construction and controlled response for most sub-woofer applications. The slow low end roll-off provides a clean transition into the extreme frequency range. Unlike ported boxes, the cone excursion is reduced below the resonant frequency of the box and driver due to the added stiffness provided by the sealed box baffle.

Closed baffle boxes are typically constructed of a very rigid material such as MDF (medium density fiber board) or plywood .75 to 1 inch thick. Depending on the size of the box and material used, internal bracing may be necessary to maintain a rigid box. A rigid box is important to design in order to prevent unwanted box resonance.

As with any acoustics application, the box must be matched to the loudspeaker driver for maximum performance. The following will outline the procedure to tune the box or maximize the output of the sub-woofer box and driver combination.

Closed baffle circuit
The sealed box enclosure for sub-woofers can be modelled as a lumped element system if the dimensions of the box are significantly shorter than the shortest wavelength reproduced by the sub-woofer. Most sub-woofer applications are crossed over around 80 to 100 Hz. A 100 Hz wave in air has a wavelength of about 11 feet. Sub-woofers typically have all dimensions much shorter than this wavelength, thus the lumped element system analysis is accurate. Using this analysis, the following circuit represents a sub-woofer enclosure system.

where all of the following parameters are in the mechanical mobility analog

 * $$V_e$$ - voltage supply
 * $$R_e$$ - electrical resistance
 * $$M_m$$ - driver mass
 * $$C_m$$ - driver compliance
 * $$R_m$$ - resistance
 * $$R_{Ar}$$ - rear cone radiation resistance into the air
 * $$X_{Af}$$ - front cone radiation reactance into the air
 * $$R_{Br}$$ - rear cone radiation resistance into the box
 * $$X_{Br}$$ - rear cone radiation reactance into the box

Driver parameters
In order to tune a sealed box to a driver, the driver parameters must be known. Some of the parameters are provided by the manufacturer, some are found experimentally, and some are found from general tables. For ease of calculations, all parameters will be represented in the SI units meter/kilogram/second. The parameters that must be known to determine the size of the box are as follows:

 * $$f_0$$ - driver free-air resonance
 * $$C_{MS}$$ - mechanical compliance of the driver
 * $$S_D$$ - effective area of the driver

Resonance of the driver
The resonance of the driver is usually either provided by the manufacturer or must be found experimentally. It is a good idea to measure the resonance frequency even if it is provided by the manufacturer to account for inconsistent manufacturing processes.

The following diagram shows verga and the setup for finding resonance:

Where voltage $$V_1$$ is held constant and the variable frequency source is varied until $$V_2$$ is a maximum. The frequency where $$V_2$$ is a maximum is the resonance frequency for the driver.

Mechanical compliance
By definition compliance is the inverse of stiffness or what is commonly referred to as the spring constant. The compliance of a driver can be found by measuring the displacement of the cone when known masses are place on the cone when the driver is facing up. The compliance would then be the displacement of the cone in meters divided by the added weight in Newtons.

Effective area of the driver
The physical diameter of the driver does not lead to the effective area of the driver. The effective diameter can be found using the following diagram:

From this diameter, the area is found from the basic area of a circle equation.

Acoustic compliance
From the known mechanical compliance of the cone, the acoustic compliance can be found from the following equation:

$$V_{as} = P C^2 C_{ms} S_d^2$$

Where $$P$$ is air density and $$C$$ the speed of sound at a given temperature and pressure.

From the driver acoustic compliance, the box acoustic compliance is found. This is where the final application of the sub-woofer is considered. The acoustic compliance of the box will determine the percent shift upwards of the resonant frequency. If a large shift is desire for high SPL applications, then a large ratio of driver to box acoustic compliance would be required. If a flat response is desired for high fidelity applications, then a lower ratio of driver to box acoustic compliance would be required. Specifically, the ratios can be found in the following figure using line (b) as reference.

$$C_{AS} = C_{AB}r$$

$$r$$ - driver to box acoustic compliant ratio

Volume of box
The volume of the sealed box can now be found from the box acoustic compliance. The following equation is used to calculate the box volume

VB= CAB&gam

Box dimensions
From the calculated box volume, the dimensions of the box can then be designed. There is no set formula for finding the dimensions of the box, but there are general guidelines to be followed. If the driver was mounted in the center of a square face, the waves generated by the cone would reach the edges of the box at the same time, thus when combined would create a strong diffracted wave in the listening space. In order to best prevent this, the driver should be either be mounted offset of a square face, or the face should be rectangular.

The face of the box which the driver is set in should not be a square.

title: Acoustics/Sealed Box Subwoofer Design


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Acoustics/Basic Room Acoustic Treatments

Introduction
Many people use one or two rooms in their living space as "theatrical" rooms where theater or music room activities commence. It is a common misconception that adding speakers to the room will enhance the quality of the room acoustics. There are other simple things that can be done to increase the room's acoustics to produce sound that is similar to "theater" sound. This site will take you through some simple background knowledge on acoustics and then explain some solutions that will help improve sound quality in a room.

Room sound combinations
The sound you hear in a room is a combination of direct sound and indirect sound. Direct sound will come directly from your speakers while the other sound you hear is reflected off of various objects in the room.

The Direct sound is coming right out of the TV to the listener, as you can see with the heavy black arrow. All of the other sound is reflected off surfaces before they reach the listener.

Good and bad reflected sound
Have you ever listened to speakers outside? You might have noticed that the sound is thin and dull. This occurs because when sound is reflected, it is fuller and louder than it would if it were in an open space. So when sound is reflected, it can add a fullness, or spaciousness. The bad part of reflected sound occurs when the reflections amplify some notes, while cancelling out others, making the sound distorted. It can also affect tonal quality and create an echo-like effect. There are three types of reflected sound, pure reflection, absorption, and diffusion. Each reflection type is important in creating a "theater" type acoustic room.

Reflected sound
Reflected sound waves, good and bad, affect the sound you hear, where it comes from, and the quality of the sound when it gets to you. The bad news when it comes to reflected sound is standing waves.

These waves are created when sound is reflected back and forth between any two parallel surfaces in your room, ceiling and floor or wall to wall.

Standing waves can distort noises 300 Hz and down. These noises include the lower mid frequency and bass ranges. Standing waves tend to collect near the walls and in corners of a room, these collecting standing waves are called room resonance modes.

Finding your room resonance modes
First, specify room dimensions (length, width, and height). Then follow this example:

There are some room dimensions that produce the largest amount of standing waves.

 * 1) Cube
 * 2) Room with 2 out of the three dimensions equal
 * 3) Rooms with dimensions that are multiples of each other

Absorbed
The sound that humans hear is actually a form of acoustic energy. Different materials absorb different amounts of this energy at different frequencies. When considering room acoustics, there should be a good mix of high frequency absorbing materials and low frequency absorbing materials. A table including information on how different common household absorb sound can be found here.

Diffused sound
Using devices that diffuse sound is a fairly new way of increasing acoustic performance in a room. It is a means to create sound that appears to be "live". They can replace echo-like reflections without absorbing too much sound.

Some ways of determining where diffusive items should be placed were found on this website.

 * 1) If you have carpet or drapes already in your room, use diffusion to control side wall reflections.
 * 2) A bookcase filled with odd-sized books makes an effective diffuser.
 * 3) Use absorptive material on room surfaces between your listening position and your front speakers, and treat the back wall with diffusive material to re-distribute the reflections.

How to find overall trouble spots in a room
Every surface in a room does not have to be treated in order to have good room acoustics. Here is a simple method of finding trouble spots in a room.

 * 1) Grab a friend to hold a mirror along the wall near a certain speaker at speaker height.
 * 2) The listener sits in a spot of normal viewing.
 * 3) The friend then moves slowly toward the listening position (stay along the wall).
 * 4) Mark each spot on the wall where the listener can see any of the room speakers in the mirror.
 * 5) Congratulations!  These are the trouble spots in the room that need an absorptive material in place.  Don't forget that diffusive material can also be placed in those positions.


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title: False Friends of the Slavist/Map palьcь


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Acoustics/Sonic Boom

A sonic boom is the audible component of a shock wave in air. The term is commonly used to refer to the air shocks caused by the supersonic flight of military aircraft or passenger transports such as Concorde (Mach 2.2, no longer flying) and the Space Shuttle (Mach 27). Sonic booms generate enormous amounts of sound energy, sounding much like an explosion; typically the shock front may approach 100 megawatts per square meter, and may exceed 200 decibels.

Cause of sonic booms
As an object moves through the air it creates a series of pressure waves in front and behind it, similar to the bow and stern waves created by a boat. These waves travel at the speed of sound, and as the speed of the aircraft increases the waves are forced together or 'compressed' because they cannot "get out of the way" of each other, eventually merging into a single shock wave at the speed of sound. This critical speed is known as Mach 1 and is 1,225 km/h (761 mph) at sea level.

In smooth flight, the shock wave starts at the nose of the aircraft and ends at the tail. There is a sudden increase in pressure at the nose, decreasing steadily to a negative pressure at the tail, where it suddenly returns to normal. This "overpressure profile" is known as the N-wave due to its shape. We experience the "boom" when there is a sudden increase in pressure, so the N-wave causes two booms, one when the initial pressure rise from the nose hits, and another when the tail passes and the pressure suddenly returns to normal. This leads to a distinctive "double boom" from supersonic aircraft. When maneuvering the pressure; distribution changes into different forms, with a characteristic U-wave shape. Since the boom is being generated continually as long as the aircraft is supersonic, it traces out a path on the ground following the aircraft's flight path, known as the boom carpet.

A sonic boom or "tunnel boom" can also be caused by high-speed trains in tunnels (e.g. the Japanese Shinkansen). In order to reduce the sonic boom effect, a special shape of the train car and a widened opening of the tunnel entrance is necessary. When a high speed train enters a tunnel, the sonic boom effect occurs at the tunnel exit. In contrast to the (super)sonic boom of an aircraft, this "tunnel boom" is caused by a rapid change of subsonic flow (due to the sudden narrowing of the surrounding space) rather than by a shock wave. In close range to the tunnel exit this phenomenon can causes disturbances to residents.

Characteristics
The power, or volume, of the shock wave is dependent on the quantity of air that is being accelerated, and thus the size and weight of the aircraft. As the aircraft increases speed the shocks grow "tighter" around the craft, and do not become much "louder". At very high speeds and altitudes the cone does not intersect the ground, and no boom will be heard. The "length" of the boom from front to back is dependent on the length of the aircraft, although to a factor of 3:2 not 1:1. Longer aircraft therefore "spread out" their booms more than smaller ones, which leads to a less powerful boom.

The nose shockwave compresses and pulls the air along with the aircraft so that the aircraft behind its shockwave sees subsonic airflow.

However, this means that several smaller shock waves can, and usually do, form at other points on the aircraft, primarily any convex points or curves, the leading wing edge and especially the inlet to engines. These secondary shockwaves are caused by the subsonic air behind the main shockwave being forced to go supersonic again by the shape of the aircraft (for example due to the air's acceleration over the top of a curved wing).

The later shock waves are somehow faster than the first one, travel faster and add to the main shockwave at some distance away from the aircraft to create a much more defined N-wave shape. This maximizes both the magnitude and the "rise time" of the shock, which makes it seem louder. On most designs the characteristic distance is about 40,000 ft, meaning that below this altitude the sonic boom will be "softer". However the drag at this altitude or below makes supersonic travel particularly inefficient, which poses a serious problem.

Abatement
In the late 1950s when SST designs were being actively pursued, it was thought that although the boom would be very large, they could avoid problems by flying higher. This premise was proven false when the North American B-70 Valkyrie started flying and it was found that the boom was a very real problem even at 70,000 ft (21,000m). It was during these tests that the N-wave was first characterized.

Richard Seebass and his colleague Albert George at Cornell University studied the problem extensively, and eventually defined a "figure of merit", FM, to characterize the sonic boom levels of different aircraft. FM is proportional to the aircraft weight divided by the three-halves of the aircraft length, FM = W/(3/2·L) = 2W/3L. The lower this value, the less boom the aircraft generates, with figures of about 1 or lower being considered acceptable. Using this calculation they found FM's of about 1.4 for Concorde, and 1.9 for the Boeing 2707. This eventually doomed most SST projects as public resentment, somewhat blown out of proportion, mixed with politics eventually resulted in laws that made any such aircraft impractical (flying only over water for instance).

Seebass-George also worked on the problem from another angle, examining ways to reduce the "peaks" of the N-wave and therefore smooth out the shock into something less annoying. Their theory suggested that body shaping might be able to use the secondary shocks to either "spread out" the N-wave, or interfere with each other to the same end. Ideally this would raise the characteristic altitude from 40,000 ft to 60,000, which is where most SST designs fly. The design required some fairly sophisticated shaping in order to achieve the dual needs of reducing the shock and still leaving an aerodynamically efficient shape, and therefore had to wait for the advent of computer-aided design before being able to be built.

This remained untested for decades, until DARPA started the Quiet Supersonic Platform project and funded the Shaped Sonic Boom Demonstration aircraft to test it. SSBD used a F-5 Freedom Fighter modified with a new body shape, and was tested over a two year period in what has become the most extensive study on the sonic boom to date. After measuring the 1,300 recordings, some taken inside the shock wave by a chase plane, the SSBD demonstrated a reduction in boom by about one-third. Although one-third is not a huge reduction, it could reduce Concorde below the FM = 1 limit for instance.

There are theoretical designs that do not appear to create sonic booms at all, such as the Busemann's Biplane. Nobody has been able to suggest a practical implementation of this concept, as yet.

Perception and noise
The sound of a sonic boom depends largely on the distance between the observer and the aircraft producing the sonic boom. A sonic boom is usually heard as a deep double "boom" as the aircraft is usually some distance away. However, as those who have witnessed landings of space shuttles have heard, when the aircraft is nearby the sonic boom is a sharper "bang" or "crack". The sound is much like the "aerial bombs" used at firework displays.

In 1964, NASA and the FAA began the Oklahoma City sonic boom tests, which caused eight sonic booms per day over a period of six months. Valuable data was gathered from the experiment, but 15,000 complaints were generated and ultimately entangled the government in a class action lawsuit, which it lost on appeal in 1969.

In late October 2005, Israel began using nighttime sonic boom raids against civilian populations in the Gaza Strip as a method of psychological warfare. The practice was condemned by the United Nations. A senior Israeli army intelligence source said the tactic was intended to break civilian support for armed Palestinian groups.

Media
These videos include jets achieving supersonic speeds.

[[Media:Yeager supersonic flight 1947.ogv|First supersonic flight]] (info)
 * Chuck Yeager broke the sound barrier on October 14, 1947 in the Bell X-1.

[[Media:F-14 Tomcat sonic boom.ogg|F-14 Tomcat sonic boom flyby (with audio)]] (info)
 * F-14 Tomcat flies at Mach 1 over the water, creating a sonic boom as it passes.

[[Media:F-14A Tomcat supersonic flyby, 1986.ogg|F-14A Tomcat supersonic flyby]] (info)
 * Supersonic F-14A Tomcat flying by the USS Theodore Roosevelt CVN-71 in 1986 for the tiger cruise.

[[Media:Shuttle Columbia sound barrier.ogg|Shuttle passes sound barrier]] (info)
 * Space shuttle Columbia crosses the sound barrier at 45 seconds after liftoff.

title: Acoustics/Sonic Boom


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title: False Friends of the Slavist/Map pisati