Famers Audio: Articles about Audio, Acoustics and Sound

Equalizer

2009-08-31T07:43:00.002+07:00

Equalizer, or EQ is sophisticated frequency control, which allow you to boost or minimize a range of frequencies, At its simplest, an equalizer divides audio signals into low, mid, and high frequencies. For example, you could increase the gain on the bass (the low frequencies) in your favorite dance music.

A common type of equalizer is the 1/3 octave equalizer, sometimes called a graphic equalizer. The graphic equalizer has 31 control points over the 10 octave to raise and lower the gain of the center frequency labeled above the control.

Limiter

2009-08-18T13:21:00.002+07:00

Limiter are the same as compressor except that signals exceeding the threshold level are reduced at ratios of 10:1 or greater. Limiter can be set to prevent all signal from going above the given amplitude. Prevent the audio signal from being too intense, causing distortion or damage to the loudspeakers.

An example showing the use of a limiter:
The sound of a dropped microphone may be too intense, and cause damage to the audio system's loudspeaker.

Compressor

2009-08-18T13:04:00.002+07:00

Compressor control the overall amplitude of a signal by reducing that part of the signal which exceeds an adjustable level (threshold) se by the user. When the signal exceeds the threshold level, the overall amplitude is reduced by a ratio, also usually adjustable by the user. For example, if the ratio of compression is set for 2:1, any part of the signal exceeding the threshold level is reduced by one-half. Compressor are used for reducing the overall dynamic range between the loudest and quietest signals. Compressor has ratio settings that are typically anything above 1:1 up to 10:1

Audio Signal Distortion

2009-08-13T07:18:00.002+07:00

In any part of a system (amplifier, equalizer, cable, etc), there is an ideal, expected output signal waveform. Any false reproduction of audio is considered signal distortion.

The most common example of distortion occurs in analog equipment when the capabilities of the gear that receives of delivers the signal are exceeded. The amplitude of a signal must fall in a certain range to be accurately received by the device. The better quality the device, the more accurately it will handle the signal (less distortion).

The original signal is beyond the threshold, so the resulting signal is squared off.

If the amplitude of the sugnal is too low or to high for the device, it may be less inteleligible because of signal distrortion. The distorted sound is actually the odd harmonics of the signal being amplified and then added back into the signal.

Manufacturers generally specifify their product to be within a certain allowable tolerance of total harmonic distortion (THD). THD is generally given as a small percentage of the total signal, and measured at the maximum specifies output of the device.

Telephone conversations, not known for their audio quality, generally tolerate up to 10 percent distortion. In professional AV systems, a 1 pecent THD is noticeable to most.

It is up to the AV systems professional to assure that the installed system adheres to this tolerance by assuring the system is designed, set up and adjusted so that each item of equipment is getting and delifering the signal levels within the limits stated by the manufactures.

Noise Masking Systems

2009-08-11T08:32:00.002+07:00

The purpose of noise masking systems is, simply, to mask noise. In the quiet, nonreverberant office spaces, conversations can be unintentionally overheard, invading privacy and mechanical or man-made noise can become disruptive to the work effort. When properly installed and balanced, the noise masking system makes these noises less noticeable. This can increase efficiency of the workforce as well as privacy.
Direct sound, so important for speech intelligibility in paging or reinforcement systems, is undersirable in masking systems. Loudspeakers, completely separate from paging loudspeakers, are throughtout the area. They are sometimes oriented horizontally to improve dispersion and avoid a spatial concentration of volume. When used properly, employees will not notice the installation at all, just that distruptive noise seem to be less noticable

High Pressure and Distributed systems

2009-08-10T17:22:00.000+07:00

Feedback

2009-08-08T14:36:00.002+07:00

Feedback is generated between a microphone and a loudspeaker when the same tone is caught in a loop an amplified again and again. This may happen when the volume of a certain sound is too loud; the audio system may be too loud or the source (the person talking) may be too far away from the microphone.

the avoid feedback poblems:
1) place microphone near the person, and physically as far from and behind the loudspeakers as possible.
2) Avoid placing microphones directly underneath ceiling mounted-speaker systems. Also, coach your presenters so that they cab avoid walkin underneath ceiling -mounted speakes
3)the presenter should stand at an appropriate distance from the microphoneor about 6 in. (15cm) away for a wires microphone. If the presenter is too far away, you may be empted to turn up the volume of the microphones output and that may cause feedback.
4) Beware of having too many open microphones at one time. Utilize the mute butons on a mixer to manage the microphones beter. You can also keep the gain (volume) lowered unused microphones.
5) Avoid placing too many microphone too close to each other.
6) Proper placement of a lavalier microphone on the presenter is critical to good microphone performance.

Feedback is best reduced through a combination of good loudspeaker placement and processing equipment. If the placement cannot be corrected, turn down the microphone or loudspeaker. Once the problem is corrected the result isnt instantaneous: it might take a second for the feedback o complete a cycle and die out.

Feedback also occurs uring audio and video conferencing. It manifests itself in the form of an echo or more feedback. The solution to this problem is often simple. Move the microphone away fom he loudspeaker or turn down the incoming or outgoing audi. There are many ways to resolve this problem, speak to a qualified professional.

best practices for controllong feedback include;
1) Keeping the microphone as close to the sound source as possible.
2) Keeping the loudspeakers in front of, and as far from the microphones as possible.
3) Selecting microphones with pickup patterns that complement the application.
4) Selecting loudspeakers with sound patterns that complement the application.

These practices will help increase your headroom - the maximum gain possible before feedback.That means how much power, which is translated to volume, can be achieved by the system before feedback occurs. Why does feedback occur with the power? Well, if you increase the volume too much, the sound waves from the loudspeaker have enough power toreach the microphone, causing the re-amplification of the same signal. If your loudspeakers output 70dB for a good listening level, abd feedback occurs in your

Sound System : Basic Device

2008-01-14T21:20:00.000+07:00

Sound system for the live performace and the recorder had two main components that is: microphone and mixer. Microphone was used to convert voice signal to electric signal. Mixer was used to combine all these signals.
The signal from mixer output is connected with amplifier.
Amplifier will strengthen this signal in being given to speaker.
Speaker will change this signal into the acoustic wave as the sound.

Afterwards in the system of the voice order also was gotten the processor that generally is connected around mixer and amplifier.
The processor was used to increase the quality of the signal audio.
The most general processor was used that is equaliser, effect, and the compressor.
Equaliser basically is the control collection of the tone that could raise or reduce the frequency of the signal audio specifically.
Effect is used to give special effect of the sound, like reverb (the voice reverberated) and delay (gave the postponement of the signal).
The signal that was too strong until passing the limit that was given to the compressor will be reduce.
In this case, the compressor could help prevented damage in speaker.

It's inline frame from http://audiofarmers.blogspot.com

RECORDING: the basic Equipment that was needed for the recording

2008-01-13T19:36:00.000+07:00

The process of the recording one music show had the aim of producing the original recorder cassette/the master. Usually after live show, the band group will send this recorder cassette to be processed further in the mastering before being doubled in the form of CD or the cassette. To make the good recording have several parts that must be considered:

1. The first,, and most important, is the microphone. Choose the microphone with the best quality for guaranteed the accuracy pickup of the desired sounds. Try several microphones and placements before settling on one best choice for each instrument.
A cardioid microphones are a good type for the beginning. Try to get the same type of microphone for all singers. It’s to avoid the difference of the timbre character because of using the different microphone.

2.Next, consider to the recording device that wanted to be used. The recording device could consist of various forms and size, from simple cassette tape recorders to the recording implement multitrack digital that more went up. The standard of the recording media to professional at this time was Digital Audio Tape (DAT) that will be introduced by Sony mid the 80 's. This media had the shape of the magnetic tape that could record from 2 to 22 of GB. DAT had sample rates from 44,1 kHz, CD audio the standard, to 48 kHz.

3.Finally, depending on the capabilities of the recording device, may need a
mixer. A mixer had the dual function; apart from being utilised to combine several microphones, mixer also had a function of changing the level signal of the microphone to the level of the recording device. Most consumer-quality cassette tape recorders of the consumer class did not permit the microphone directly to be connected to the record inputs. A microphone has the low output level that would result in little or no signal actually making it to tape.Mixer had the function to strengthen the signal from the level signal of the microphone to the level that could be accepted the recorder.

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SOUND SYSTEM & RECORDING: The introduction for the Process of the recording and the Reinforcement System of the Voice

2008-01-10T20:10:00.000+07:00

Sound the system was the vital part in one showed modern music, although in this case often we ignored him. Possibly as a Manager the performance must look for a person professional that was responsible for arranging this. But in fact, a person professional that was capable in handled sound system not always was available. In the long run this responsibility was shifted to the person that not specially could handle sound system like for example the musician. You think that this person could handle sound system was the same his matter when you installed some equipment sound system hi-fi in the house and you got the clear vote. You were kept buying several microphones and speaker furthermore showed could be begun. And you too hoped that showed this definitely could be recorded well.

Unfortunately in composing a set sound system in one showed not was as simple as that. Indeed in this case not was difficult like apparently, but when you came to a shop audio, there were many choices that could make you confused.

When you were asked the kind microphone what was needed by you, of Cardioid or Dynamic? Speaker the kind what was wanted by you, of Low Impedance or High Impedance? You really had difficulty answering him, because you did not understand would that.

With few basic guides, you could study what was needed by you, till finally you knew how sound system that was bought by you could produce the good voice good in showed and when in the process of the recording.

That must be paid attention to as the foundation in the process of the recording adalahketika feedback happened (feedback), also the characteristics of the acoustics in this recorder space.

Many of various microphone kinds and mixer that possibly to be chosen in the process of the recording, was the same his matter with the strengthening system of the voice to live show.

If the voice canned be heard good by you, that already a good step.
And became capital of the foundation to become better and finally produced the voice that excellent.

A little knowledge and some logic will help you to choose the system was good with the budget that be enough and avoided disappointment.

It's inline frame from http://audiofarmers.blogspot.com

SOUND SYSTEM: Church Sound System Configurations (1)

2007-12-15T08:32:00.000+07:00

Church Sound Reinforcement systems include the following item:
1. Microphones for preacher and choir
2. Mixer / Amplifier or separate mixer, equalizer, and amplifier
3. Speakers
4. Cables to interconnect the components

for additional items for church sound reinforcement systems such as:
1. Tape deck
2. CD player
3. Monitor speakers or Headphones
4. Fill-in speakers for under balcony seats
5. Rear Fill-in speakers for long rooms

Configuration of Church Sound System:

The basic sound system consist of a few microphones, a mixer amplifier and a couple of speakers. Although the speakers with two driver are designed primarily for speech and recorded music. This speakers can handle limited live music, but they lack the bass response to produce a rich natural sound. This speakers are made for a small room because they can't project the sound or control it very well.

It's inline frame from http://audiofarmers.blogspot.com

ACOUSTICS: Sanctuary Shape

2007-12-11T20:34:00.000+07:00

It's inline frame from http://audiofarmers.blogspot.com"

MICROPHONE: A Transducer

2007-12-11T19:42:00.000+07:00

MICROPHONE: Operating Principle

2007-11-20T07:41:00.000+07:00

How does the microphone change sound into an electrical signal? The operating principle describes the type of transducer inside the microphone. It is the part of the microphone that actually picks up sound and converts it into an electrical signal. The operating principle determines some of the basic capabilities of the microphone. There are six key types of microphone:

Electret

Piezoelectric

All employ different mechanisms to convert sound energy to electrical energy. Hence all have different advantages and disadvantages. You will hence need to choose the right type of microphone for the right type of application.

MICROPHONE: The transducer which transforms sound energy into electrical energy

2007-11-18T16:41:00.000+07:00

Microphone is a generic term that is used to refer to any element which transforms acoustic energy (sound) into electrical energy (the audio signal). A microphone is therefore one type from a langer class of elements called transducers (device that changes energy from one form into another, in this case, acoustic energy into electrical energy). The microphone is the first link in the audio chain and is therefore critical to the overall performance of a sound system. The fidelity with which a microphone generates an electrical representation of a sound depend, in part, on the method by which it performs the energy conversion. Historycally, a number of different methods have been developed for varying purposes, and today a wide variety of microphone types may be found in everyday use.Improper selection of microphones may prevent the rest of the system from functioning to its full potential. Proper selection of microphones depends on an understanding of basic microphone characteristics and on a knowledge of the intended application. To be most effective, a microphone must be matched both to the desired sound source (voice, musical instrument, etc.) and to the sound system (PA system, tape recorder, etc.) with which it is used. There are five areas of microphone characteristics that must be considered when selecting a microphone for a particular application. They are:
1) the operating principle of the microphone
2) Frequency response
3) Directional characteristics
4) the electrical output of the microphone
5) the physical design of the microphone

Sound Wave

2007-10-05T19:41:00.000+07:00

In today’s technologically sophisticated world, sound systems of various types are a part of almost everyone’s daily life. We must to be familiar with some general aspects of sound helpful Good sound quality is the goal of any sound system: how it is produced, transmitted, and received. In addition, it is also useful to describe or classify sound according to its acoustic behavior. Finally, the characteristics of “good” sound should be understood.

Sound is produced by vibrating objects. These include musical instruments, loudspeakers, and, of course, human vocal cords. The mechanical vibrations of these objects move a physical medium-usually air which is immediately adjacent to them, alternately “pushing” and “pulling” the air from its resting state. Each back-and-forth vibration produces a corresponding pressure increase (compression) and pressure decrease (rarefaction) in the air. A complete acoustical pressure wave consists of one half-cycle of compression (higher pressure) of the molecules, followed by one half-cycle of rarefaction (lower pressure) of the air molecules. These cyclic pressure changes travel outward from the vibrating object, forming a pattern called a sound wave. A sound wave is a acoustical energy consists of fluctuating waves of pressure changes (cycles) moving through the air.

A simple sound wave can be described by its frequency and by its amplitude. The rate of air pressure fluctuation is called the frequency of the wave. The unit Hertz (Hz) is now used to indicate frequency in cycles per second (cps):

1 Hz = 1 cps

The range of frequencies audible to the human ear extends from a low of about 20 Hz to a high of about 20,000 Hz. In practice, a sound source such as a voice usually produces many frequencies simultaneously. In any such complex sound, the lowest frequency is called the fundamental and is responsible for the musical attribute of pitch. The higher frequencies are called harmonics and are responsible for the timbre or tone of the music. Harmonics allow us to distinguish one source from another, such as a piano from a guitar, even when they are playing the same fundamental note.

The amount of time required for one complete cycle of a sound wave is called the period of the wave. A wave’s period is expressed in seconds per cycle, and is found by using the equation:

where: P = Period
f=frequency

The wavelength of a sound wave is the physical distance covered from any point in one cycle to the same point in the next cycle of a given frequency sound as it passes though air. Wavelength is expressed by the equation:

where: λ = wavelength
c = speed of sound
f = frequency
Since each cycle is the same, the distance from any point in one cycle to the same point in the next cycle is also one wavelength: for example, the distance from one maximum pressure point to the next maximum pressure point. Wavelength is related to frequency by the speed of sound. The speed of sound is the velocity at which a sound wave travels. The speed of sound is constant and is equal to about 1130 ft/sec (344m/sec) in air at sea level on a standard temperature day (which is 59o Farenheit or 15o Celcius).

It does not change with frequency or wavelength, but it is related to them in the following way: the frequency of a sound, multiplied by its wavelength always equals the speed of sound. Thus, the higher the frequency of sound, the shorter the wavelength, and the lower the frequency, the longer the wavelength. The wavelength of sound is responsible for many acoustic effects.

The amplitude of a sound wave refers to the magnitude (strength) of the pressure changes and determines the “loudness” of the sound. Amplitude is measured in decibels (dB) of sound pressure level (SPL) and ranges from 0 dB SPL (the threshold of hearing), to above 120 dB SPL (the threshold of pain). The level of conversational speech is about 70dB SPL. A change of 1 dB is about the smallest SPL difference that the human ear can detect, while 3 dB is a generally noticeable step, and an increase of 10 dB is perceived as a “doubling” of loudness

Proportion of Direct Sound vs. Ambient Sound

2007-10-04T21:38:00.000+07:00

In contrast, ambient sound, such as reverberation, has a relatively constant level. Therefore, at a given distance from a sound source, a listener (or a microphone) will pick up a certain proportion of direct sound vs. ambient sound. As the distance increases, the direct sound level decreases while the ambient sound level stays the same. A properly designed sound system should increase the amount of direct sound reaching the listener without increasing the ambient sound significantly.

(from Audio System Guide for Houses of Worship, Shure Educational Publication)

Increase and decrease of sound level

2007-10-04T21:35:00.000+07:00

A very important property of direct sound is that it becomes weaker as it travels away from the sound source, at a rate governed by the inverse-square law. For example, when the distance increases by a factor of two (doubles), the sound level decreases by a factor of four (the square of two). This results in a drop of 6 dB in sound pressure level (SPL), a substantial decrease. Likewise, when the distance to the direct sound source is divided by two (cut in half), the sound level increases by 6 dB.

(from Audio System Guide for Houses of Worship, Shure Educational Publication)

Standing Wave

2007-10-04T21:32:00.000+07:00

One additional form of indirect sound is known as a standing wave. This may occur when the wavelength of a sound is the same distance as some major dimension of a room, such as the distance between two opposite walls. If both surfaces are acoustically reflective, the frequency corresponding to that wavelength will be amplified, by addition of the incoming and outgoing waves, resulting in a strong, stationary wave pattern between the two surfaces. This happens primarily with low frequencies, which have long wavelengths and are not easily absorbed.

(from Audio System Guide for Houses of Worship, Shure Educational Publication)

Echo dan Reverberation

2007-10-04T21:31:00.001+07:00

Echo occurs when an indirect sound is delayed long enough (by a distant reflecting surface) to be heard by the listener as a distinct repetition of the direct sound. If indirect sound is reflected many times from different surfaces it becomes “diffuse” or non-directional. This is called reverberation, and it is responsible for our auditory perception of the size of a room. Reverberant sound is a major component of ambient sound, which may include other non-directional sounds, such as wind noise or building vibrations. A certain amount of reverberant sound is desirable to add a sense of “space” to the sound, but an excess tends to make the sound muddy and unintelligible.

Direct Sound and Indirect Sound

2007-10-04T21:28:00.002+07:00

Sound can be classified by its acoustic behavior; for example, direct sound vs. indirect sound. Direct sound travels from the sound source to the listener in a straight line (the shortest path). Indirect sound is reflected by one or more surfaces before reaching the listener (a longer path). Since sound travels at a constant speed, it takes a longer time for the indirect sound to arrive, and it is said to be “delayed” relative to the direct sound. There are several kinds of indirect sound, depending on the “acoustic space” (room acoustics).

(from Audio System Guide for Houses oh Worship, Shure Educational Publication)

Sound Picked Up

2007-10-04T21:26:00.000+07:00

Once a sound has been produced and transmitted, it is received by the ear and, of course, by microphones. In the ear, the arriving pressure changes “push” and “pull” on the eardrum. The resulting motion of the eardrum is converted (by the inner ear) to nerve signals that are ultimately perceived as “sound”. In a microphone, the pressure changes act on a diaphragm. The resulting diaphragm motion is converted (by one of several mechanisms) into electrical signals which are sent to the sound system. For both “receivers”, the sound picked up is a combination of all pressure changes occurring just at the surface of the eardrum or diaphragm.

Absorbed and Reflected

2007-10-04T21:21:00.000+07:00

After it is produced, sound is transmitted through a “medium”. Air is the typical medium, but sound can also be transmitted through solid or liquid materials. Generally, a sound wave will move in a straight line unless it is absorbed or reflected by physical surfaces or objects in its path. However, the transmission of the sound wave will be affected only if the size of the surface or object is large compared to the wavelength of the sound. If the surface is small (compared to the wavelength) the sound will proceed as if the object were not there. High frequencies (short wavelengths) can be reflected or absorbed by small surfaces, but low frequencies (long wavelengths) can be reflected or absorbed only by very large surfaces or objects. For this reason it is easier to control high frequencies by acoustic means, while low frequency control requires massive (and expensive) techniques.

(from Audio System Guide for Houses of Worship, Shure Educational Publication)

Wavelength and Speed of Sound

2007-10-03T21:20:00.000+07:00

Another characteristic of a sound wave related to frequency is wavelength. The wavelength of a sound wave is the physical distance from the start of one cycle to the start of the next cycle, as the wave moves through the air. Since each cycle is the same, the distance from any point in one cycle to the same point in the next cycle is also one wavelength: for example, the distance from one maximum pressure point to the next maximum pressure point. Wavelength is related to frequency by the speed of sound. The speed of sound is the velocity at which a sound wave travels. The speed of sound is constant and is equal to about 1130 feet-per-second in air.

It does not change with frequency or wavelength, but it is related to them in the following way: the frequency of a sound, multiplied by its wavelength always equals the speed of sound. Thus, the higher the frequency of sound, the shorter the wavelength, and the lower the frequency, the longer the wavelength. The wavelength of sound is responsible for many acoustic effects.

(from Audio System Guide for Houses of Worship, Shure Educational Publication)

Amplitude

2007-10-03T21:13:00.000+07:00

The amplitude of a sound wave refers to the magnitude (strength) of the pressure changes and determines the “loudness” of the sound. Amplitude is measured in decibels (dB) of sound pressure level (SPL) and ranges from 0 dB SPL (the threshold of hearing), to above 120 dB SPL (the threshold of pain). The level of conversational speech is about 70dB SPL. A change of 1 dB is about the smallest SPL difference that the human ear can detect, while 3 dB is a generally noticeable step, and an increase of 10 dB is perceived as a “doubling” of loudness

(from Audio System Guide for Houses of Worship, Shure Educational Publication)

Frequency of Sound

2007-10-03T21:05:00.000+07:00

A simple sound wave can be described by its frequency and by its amplitude. The frequency of a sound wave is the rate at which the pressure changes occur. It is measured in Hertz (Hz), where 1 Hz is equal to 1 cycle-persecond. The range of frequencies audible to the human ear extends from a low of about 20 Hz to a high of about 20,000 Hz. In practice, a sound source such as a voice usually produces many frequencies simultaneously. In any such complex sound, the lowest frequency is called the fundamental and is responsible for the pitch of the sound. The higher frequencies are called harmonics and are responsible for the timbre or tone of the sound. Harmonics allow us to distinguish one source from another, such as a piano from a guitar, even when they are playing the same fundamental note.

(from Audio Systems Guide for Houses of Worship, Shure Educational Publication)

Sound Wave

2007-10-03T20:41:00.000+07:00

Because good sound quality is the goal of any sound system, it is helpful to be familiar with some general aspects of sound: how it is produced, transmitted, and received. In addition, it is also useful to describe or classify sound according to its acoustic behavior. Finally, the characteristics of “good” sound should be understood.
Sound is produced by vibrating objects. These include musical instruments, loudspeakers, and, of course, human vocal cords. The mechanical vibrations of these objects move the air which is immediately adjacent to them, alternately “pushing” and “pulling” the air from its resting state. Each back-and-forth vibration produces a corresponding pressure increase (compression) and pressure decrease (rarefaction) in the air. A complete pressure change, or cycle, occurs when the air pressure goes from rest, to maximum, to minimum, and back to rest again. These cyclic pressure changes travel outward from the vibrating object, forming a pattern called a sound wave. A sound wave is a series of pressure changes (cycles) moving through the air.

(from Audio System Guide for House of Worship, Shure Educational Publication)

Decibels

2006-08-25T10:32:00.000+07:00

In all phases of audio technology the decibel is used to express signal level differences. expressed by the same level, 10 dB. Any 10 dB level difference, regardless of the actual powers involved, will represent a 2-to-1 difference in subjective loudness. We will now expand our power decibel table: P1 (watts) Level in dB.

the decibel (abbreviated dB) is used to express differences between two values signal. The reason the decibel is such a useful measure is that it enables us to use a comparatively small range of numbers to express large and often unwieldy quantities. For convenience, we find the ratio between the two numbers and convert that into a logarithm. . If we measured one value as 10 watt and another as 1000 watt, we say that one is 20 dB greater than the other. If the softest audible sound has a power of about 0.000000000001 watt/sq. meter and the threshold of pain is around 1 watt/sq. meter, giving a total range of 120dB.

Power difference in dB = 10 log ( Power A / Power B)

The following tabulation illustrates the usefulness of the concept. Letting Power B = 1 watt:

Power A (watts) Level in dB
1 0
10 10
100 20
1000 30
10,000 40
20,000 43

The usefulness of all this becomes apparent when we think about how the ear perceives loudness. The decibel also makes sense from a psychoacoustical point of view in that it relates directly to the effect of most sensory stimuli. First of all, the ear is very sensitive. In the second place, our judgment of relative levels of loudness is somewhat logarithmic. If a sound has 10 times the power of a reference (10dB) we hear it as twice as loud. If we merely double the power (3dB), the difference will be just noticeable.

Converting voltage or pressure ratios to decibels

Remember that the dB is used to describe relationships of power. Power is not often conveniently measured, especially in electronic devices. We can use voltage, current, and pressure ratios as they relate to power. Most often we measure voltage and use the formula P = E2/Z to get power. Squaring a value doubles its logarithm, so our dB formula becomes:

Power divergence in dB = 20 log (voltage A / voltage B)

We now present a table useful for determining levels in dB for ratios given in voltage. Letting Voltage B = 1 volt:

Voltage A Ratios Level in dB
1 0
1.25 2
1.60 4
2 6
2.5 8
3.15 10
4 12
5 14
6.3 16
8 18
10 20

Sound Pressure Level (SPL)

For decibels of sound pressure level (SPL), the reference is the extremely low value of 20 x 10-6 newtons/m2. This reference pressure corresponds roughly to the minimum audible sound pressure for persons with normal hearing. SPL ranges from 0 dB SPL (the threshold of hearing), to above 120 dB SPL (the threshold of pain). Conversational speech is about 70dB SPL. A change of 1 dB is about the smallest SPL difference that the human ear can detect, while 3 dB is a generally noticeable step, and an increase of 10 dB is perceived as a “doubling” of loudness. As a convenient point of reference, note that an rms pressure of 1 pascal corresponds to a sound pressure level of 94 dB. We commonly use a sound level meter (SLM) to measure SPL

Directional Characteristic of a Microphone

2006-08-23T19:44:00.000+07:00

The directional characteristic of a microphone is defined as the variation of its output when it is oriented at different angles to the direction of the sound. It determines how best to place the microphone relative to the sound source(s) in order to enhance pickup of desired sound and to minimize pickup of undesired sound.
Many people have the misconception that microphones only pick up sound from sources they are pointed at, much as a camera only photographs what is in front of the lens. This would be a nice feature if we could get it, but the truth is we can only approximate that action, and at the expense of other desirable qualities.

The polar pattern of a microphone is the graphical representation of the output produced vs. the angle of the sound source. The output is represented by the radius of the curve at the incident angle. The two most common directional types are omnidirectional and unidirectional.

Omnidirectional Microphone

A omnidirectional is simplest microphone design that will pick up all sound, exhibits the same output regardless of its orientation to the sound source. This indicates that the microphone is equally sensitive to sound coming from all directions.
An omnidirectional microphone can therefore pick up sound from a wide area, but cannot be “aimed” to favor one sound over another. Generally have good to outstanding frequency response.

Unidirectional Microphone

A unidirectional microphone is most sensitive to sound coming from only one direction. On a polar graph, this will appear as a rounded but noncircular figure.

Unidirectional microphone may be aimed at a desired, direct sound by orienting its axis toward the sound until the output of a unidirectional microphone is maximum for sound arriving at an angle of 0 degrees, or on-axis. It falls off only slightly for sound arriving from within a certain angle off-axis. For any microphone, the direction of least sensitivity (minimum output) is called the null angle. it may also be aimed away from an undesired, direct sound by orienting its null angle toward the sound. The total directional range for usable output is called the coverage angle or pickup area.

In addition, a unidirectional microphone picks up less ambient sound than an omnidirectional, due to its overall lower sensitivity at the sides and rear. The most common type of unidirectional microphone is called a cardioid, because of its heartshaped polar pattern.

Cardioid Microphone

A cardioid type is most sensitive to sound coming from in front of the microphone (the bottom of the “heart”). On the polar graph this is at 0 degrees, or “on-axis”. It is less sensitive to sound reaching the microphone from the sides (“off-axis”), and the direction of least sensitivity is toward the rear (the notch at the top of the “heart”). For a cardioid pattern, this is at 180 degrees or directly behind the microphone. Coverage angle for a cardioid microphone is about 130 degrees. A cardioid picks up only one-third as much ambient sound as an omnidirectional type.

Supercardioid and the Hypercardioid Microphone

Compared to a cardioid type, supercardioid and the hypercardioid have a progressively narrower coverage angle: 115 degrees for a supercardioid and 105 degrees for a hypercardioid. However, unlike the cardioid, they have some pickup directly behind the microphone. This is indicated in their polar patterns by a rounded projection, called a lobe, toward the rear of the microphone. The direction of least sensitivity (null angle) for these types is about 125 degrees for the supercardioid and 110 degrees for the hypercardioid. In general, any directional pattern that has a narrower front coverage angle than a cardioid will have some rear pickup and a different null angle. The supercardioid has the maximum ratio of on-axis pickup to ambient pickup, while the hypercardioid has the least overall pickup of ambient sound (only onequarter as much as an omni). These can be useful types for certain situations, such as more distant pickup or in higher ambient noise levels, but they must be placed more carefully than a cardioid to get best performance. The Hypercardioid pattern is very popular, as it gives a better overall rejection and flatter frequency response at the cost of a small back pickup lobe.

A "shotgun" mic carries these techniques to extremes by mounting the diaphragm in the middle of a pipe. The shotgun is extremely sensitive along the main axis, but posseses pronounced extra lobes which vary drastically with frequency. In fact, the frequency response of this mic is so bad it is usually electronically restricted to the voice range, where it is used to record dialogue for film and video.
Bi-directional Microphone
bidirectional type. As the name implies, it is equally sensitive to sound from two directions: directly in front of the microphone and directly behind it. Its polar graph consists of a front pickup area and an identical rear lobe and resembles a “figure 8” pattern. Although the front coverage angle of a bidirectional microphone is only 90 degrees, it has equal rear coverage. The null angle is at 90 degrees, which is directly at the side of the microphone. While the bidirectional microphone is not used by itself in any typical house of worship sound application, it is occasionally used in combination with other types for stereo sound reproduction.

Frequency Response of Microphones

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There is no inherent advantage in fidelity of one type of microphone over another. Condenser types require batteries or power from the mixing console to operate, which is occasionally a hassle, and dynamics require shielding from stray magnetic fields, which makes them a bit heavy sometmes, but very fine microphones are available of both styles. The most important factor in choosing a microphone is how it sounds in the required application.

The frequency response of a microphone is defined by the range of sound (from lowest to highest frequency) that it can reproduce, and by its variation in output within that range. It is the frequency response that determines the basic “sound” of the microphone.

The two general types of frequency response are flat and shaped. These terms refer to the graphical representation of frequency response or response curve. A microphone that provides a uniform output at every audible frequency is represented on a frequency response graph as an even, flat line, and is said to have a flat response. This means that the microphone reproduces all of the sound within its frequency range with little or no variation from the original sound. In addition, flat response microphones typically have an extended frequency range; that is, they can reproduce very high and/or very low frequencies as well. Widerange, flat response microphones have a natural, highfidelity, “uncolored” sound.

Flat Frequency Response

A flat frequency response has been the main goal of microphone companies for the last three or four decades. In the fifties, mics were so bad that console manufacturers began adding equalizers to each input to compensate. This effort has now paid off to the point were most professional microphones are respectably flat, at least for sounds originating in front. The major exceptions are mics with deliberate emphasis at certain frequencies that are useful for some applications. This is another part of the microphone mystique. Problems in frequency response are mostly encountered with sounds originating behind the mic.

By contrast, a shaped microphone response will appear on a frequency response graph as a varying line with specific peaks and dips. This shows that the microphone is more sensitive to certain frequencies than to others, and often has a limited frequency range. A shaped response is usually designed to enhance the sound of a particular source in a particular application, while at the same time minimizing the pickup of certain unwanted sounds. Shaped response microphones each have a “characteristic” sound.

Shaped Frequency Response

The selection of a flat or shaped response microphone involves consideration of both the sound source and the sound destination. The frequency range of the microphone must be wide enough to pick up the desired range of the sound source. This range must also be appropriate to the intended destination of the sound: that is, wider range for high-quality sound systems or recording/broadcast systems, narrower range for speechonly public address systems. Within its range the microphone should respond in such a way that the sound is reproduced either with no change (flat response) or with changes that enhance the sound in some desirable manner (shaped response).

Normally, microphones with flat, wide-range response are recommended for high-quality pickup of acoustic instruments, choral groups and orchestras, especially when they must be placed at some distance from the sound source. Flat response microphones are less prone to feedback in high gain, distant pickup applications because they do not have frequency response peaks that might trigger feedback at any specific frequency. The most common shaped response is for vocal use. Typically, this consists of limiting the range to that of the human voice and adding an upper mid-range response rise. Such a “presence rise”, coupled with controlled low- and high-frequency response can give a sound with improved vocal clarity. This is especially true for lapel or lavalier microphones. The pickup of certain instruments such as drums and guitar amplifiers may also benefit from a shaped response microphone. Finally, the frequency response of some microphones is adjustable, typically by means of switches, to tailor the microphone to different applications. Most common are low-frequency rolloff controls, which can help prevent “rumble”, and presence rise switches to enhance
intelligibility.

Dynamic and Condenser Microphones

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A microphone is an example of a transducer, a device that changes energy from one form into another, in this case, acoustic energy into electrical energy. Sound information exists as patterns of air pressure; the microphone actually picks up this sound information and converts it into patterns of electric current. The recording engineer is interested in the accuracy of this transformation, a concept he thinks of as fidelity.

A variety of mechanical techniques can be used in building microphones. These are the two most common types of microphones in the world:

Dynamic microphones
Dynamic microphones are the type most widely used in live sound, which refers to the design of the microphone capsule. This type are economical and rugged, with fairly low sensitivity, good for handheld or "close-miked" applications. Dynamics microphones are commonly used for solo vocalists and on drum kits. They can provide excellent sound quality and good specifications in all areas of microphone performance. In particular, they can handle extremely high sound levels: it is almost impossible to overload a dynamic microphone.

Dynamic microphones employ a diaphragm/voice coil/magnet assembly which forms a miniature sounddrivenelectrical generator. Sound waves strike a thin plastic membrane (diaphragm) which vibrates in response. A small coil of wire (voice coil) is attached to the rear of the diaphragm and vibrates with it. The voice coil itself is surrounded by a magnetic field created by a small permanent magnet. It is the motion of the voice coil in this magnetic field which generates the electrical signal corresponding to the sound picked up by a dynamic microphone.

Condenser microphones
Condenser microphones are typically (but not always) much more sensitive than dynamics and can provide a smoother, more naturalsound, particularly at high frequencies. Flat frequency response and extended frequency range are much easier to obtain in a condenser. In addition, condenser microphones can be made very small without significant loss of performance. Typically used for recording orchestras, choirs, and in other applications where you wish to capture the sound of the ensemble, versus individual sounds.

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