LO1-Article-PASS-describe.set up heading-describe it.leave a gap/section for merit and distinction-set up the microphone and explain on what you are doing, you can get this information from Lo2/Lo3
Microphones
-History
A microphone is a device that transforms sounds into an eletrical signal
-Polar pattern
The polar pattern of a microphone is a sensitivity to sound relative to the direction or angle from which the sound arrives.
Omni microphonesalso called 'pressure microphones', because they measure sound pressure in a space point. A diaphragm is fixed across the mouth of a sealed cavity, so in effect the microphones behaves like a very small barometer capable of following audio frequency pressure changes — but it has no means of detecting the direction of the sound waves, hence it has an omnidirectional polar pattern. Because this arrangement only senses pressure, it doesn't matter what direction the sound approaches from. All that matters is the change in pressure at that point in space, so it is more or less equally sensitive to sounds from all directions.
A figure-of-eight microphone uses a diaphragm open to the air on both sides, so, rather than responding directly to pressure, it responds to the difference (or gradient) in pressure between the front and the rear of the diaphragm — hence the generic term 'pressure gradient' microphone (sometimes also referred to as a 'velocity' microphone, because it detects the velocity of sound waves). This arrangement of the diaphragm makes the microphone very sensitive to sounds approaching from either the front or rear axis, while sounds approaching from the side cause no diaphragm movement at all, as the pressure each side of the diaphragm always remains equal. The practical outcome is a microphone that is essentially 'deaf' 90 degrees off-axis but is equally sensitive to both front and rear. Sound picked up by the rear of the diaphragm also produces an inverted electrical signal compared with the same sound picked up by the front of the diaphragm
if you wanna learn more about omni microphone check this site out!-https://www.soundonsound.com/sos/mar07/articles/micpatterns.htm
-Frequency response
A microphone's frequency response pattern is shown using a chart like the one below and referred to as a frequency response curve. The x axis shows frequency in Hertz, the y axis shows response in decibels. A higher value means that frequency will be exaggerated; a lower value means the frequency is attenuated. In this example, frequencies around 5 - kHz are boosted while frequencies above 10 kHz and below 100Hz are attenuated. This is a typical response curve for a vocal microphone.
http://www.mediacollege.com/audio/microphones/frequency-response.html
Condenser microphones have flattery responses than a dynamic microphone.
There are different frequency in a microphone such a flattery response it means no frequencies would be exaggerated or reduced. making the response to produce the purest audio,but sometimes even the best flattery response can have some problems.
a microphone which is said to have a frequency response of 20 Hz to 20 kHz can reproduce better Frequencies with this range.
-sensitivity
Microphone sensitivity is typically measured with a 1 kHz sine wave at a 94 dB sound pressure level (SPL), or 1 pascal (Pa) pressure. The magnitude of the analog or digital output signal from the microphone with that input stimulus is a measure of its sensitivity. This reference point is but one characteristic of the microphone, by no means the whole story of its performance.
http://www.analog.com/library/analogDialogue/archives/46-05/understanding_microphone_sensitivity.html
-Types- condenser, dynamic, Ribbon,PZM
-Dynamic
The dynamic microphone works with a electromagnetic induction to generate an output signal voltage . The diaphragm is attached to a coil of fine wire. The coil is mounted in the air gap of the magnet such that it is free to move back and forth within the gap. When the sound wave strikes the diaphragm, the diaphragm vibrates in response. The coil attached to the diaphragm moves back and forth in the field of the magnet. As the coil moves through the lines of magnetic force in the gap, a small electrical current is induced in the wire. The magnitude and direction of that current is directly related to the motion of the coil, and the current then is an electrical representation of the sound wave.
http://www.crownaudio.com/kb/entry/76/
-Condenser
Condenser microphones are usally the most famous microphone used in recording studios and radio stations because its an ideal microphone to capture a voice or an acoustic instrument for a recording. They are capable better athan dynamic microphones because they have wider, flatter frequency response.
-Ribbon
A ribbon microphone were very simple designed with a thin sheet of metal in a field of a magnet. Ribbons tend to emphasise the warm low-mids and gradually roll off at the top end, giving them what at first listen can seem a dull sound when compared with modern condensers. However, on repeated listening your ears adjust and you start to find that ribbons seem to extract the musicality from a performance without getting hung up on detail.
http://www.soundonsound.com/sos/nov07/articles/ribbonmics1.htm
-PZM
PZM mics are most unusual-looking. They are flat and don't really look like microphones at all. There working place is against a hard, flat surface for example, a tabletop or wall.
http://www-personal.umich.edu/~hkumao/courses/audio.html
-Input, outputs
The XLR connector is a style of electrical connecter, primarily found on professional audio, video, and stage lighting equipment. The connectors are circular in design and have between 3 and 7 pins. They are most commonly associated with balanced audio interconnection, including AES3 digital audio, but are also used for lighting control, low-voltage power supplies, and other applications. XLR connectors are available from a number of manufacturers and are covered by an international standard for dimensions, IEC 61076-2-103. They are superficially similar to the older and smaller DIN connector range, but are not physically compatible with them.
http://en.wikipedia.org/wiki/XLR_connector
Microphones
-History
A microphone is a device that transforms sounds into an eletrical signal
-Polar pattern
The polar pattern of a microphone is a sensitivity to sound relative to the direction or angle from which the sound arrives.
Omni microphonesalso called 'pressure microphones', because they measure sound pressure in a space point. A diaphragm is fixed across the mouth of a sealed cavity, so in effect the microphones behaves like a very small barometer capable of following audio frequency pressure changes — but it has no means of detecting the direction of the sound waves, hence it has an omnidirectional polar pattern. Because this arrangement only senses pressure, it doesn't matter what direction the sound approaches from. All that matters is the change in pressure at that point in space, so it is more or less equally sensitive to sounds from all directions.
A figure-of-eight microphone uses a diaphragm open to the air on both sides, so, rather than responding directly to pressure, it responds to the difference (or gradient) in pressure between the front and the rear of the diaphragm — hence the generic term 'pressure gradient' microphone (sometimes also referred to as a 'velocity' microphone, because it detects the velocity of sound waves). This arrangement of the diaphragm makes the microphone very sensitive to sounds approaching from either the front or rear axis, while sounds approaching from the side cause no diaphragm movement at all, as the pressure each side of the diaphragm always remains equal. The practical outcome is a microphone that is essentially 'deaf' 90 degrees off-axis but is equally sensitive to both front and rear. Sound picked up by the rear of the diaphragm also produces an inverted electrical signal compared with the same sound picked up by the front of the diaphragm
if you wanna learn more about omni microphone check this site out!-https://www.soundonsound.com/sos/mar07/articles/micpatterns.htm
-Frequency response
A microphone's frequency response pattern is shown using a chart like the one below and referred to as a frequency response curve. The x axis shows frequency in Hertz, the y axis shows response in decibels. A higher value means that frequency will be exaggerated; a lower value means the frequency is attenuated. In this example, frequencies around 5 - kHz are boosted while frequencies above 10 kHz and below 100Hz are attenuated. This is a typical response curve for a vocal microphone.
http://www.mediacollege.com/audio/microphones/frequency-response.html
Condenser microphones have flattery responses than a dynamic microphone.
There are different frequency in a microphone such a flattery response it means no frequencies would be exaggerated or reduced. making the response to produce the purest audio,but sometimes even the best flattery response can have some problems.
a microphone which is said to have a frequency response of 20 Hz to 20 kHz can reproduce better Frequencies with this range.
-sensitivity
Microphone sensitivity is typically measured with a 1 kHz sine wave at a 94 dB sound pressure level (SPL), or 1 pascal (Pa) pressure. The magnitude of the analog or digital output signal from the microphone with that input stimulus is a measure of its sensitivity. This reference point is but one characteristic of the microphone, by no means the whole story of its performance.
http://www.analog.com/library/analogDialogue/archives/46-05/understanding_microphone_sensitivity.html
-Types- condenser, dynamic, Ribbon,PZM
-Dynamic
The dynamic microphone works with a electromagnetic induction to generate an output signal voltage . The diaphragm is attached to a coil of fine wire. The coil is mounted in the air gap of the magnet such that it is free to move back and forth within the gap. When the sound wave strikes the diaphragm, the diaphragm vibrates in response. The coil attached to the diaphragm moves back and forth in the field of the magnet. As the coil moves through the lines of magnetic force in the gap, a small electrical current is induced in the wire. The magnitude and direction of that current is directly related to the motion of the coil, and the current then is an electrical representation of the sound wave.
http://www.crownaudio.com/kb/entry/76/
-Condenser
Condenser microphones are usally the most famous microphone used in recording studios and radio stations because its an ideal microphone to capture a voice or an acoustic instrument for a recording. They are capable better athan dynamic microphones because they have wider, flatter frequency response.
-Ribbon
A ribbon microphone were very simple designed with a thin sheet of metal in a field of a magnet. Ribbons tend to emphasise the warm low-mids and gradually roll off at the top end, giving them what at first listen can seem a dull sound when compared with modern condensers. However, on repeated listening your ears adjust and you start to find that ribbons seem to extract the musicality from a performance without getting hung up on detail.
http://www.soundonsound.com/sos/nov07/articles/ribbonmics1.htm
-PZM
PZM mics are most unusual-looking. They are flat and don't really look like microphones at all. There working place is against a hard, flat surface for example, a tabletop or wall.
http://www-personal.umich.edu/~hkumao/courses/audio.html
-Input, outputs
The XLR connector is a style of electrical connecter, primarily found on professional audio, video, and stage lighting equipment. The connectors are circular in design and have between 3 and 7 pins. They are most commonly associated with balanced audio interconnection, including AES3 digital audio, but are also used for lighting control, low-voltage power supplies, and other applications. XLR connectors are available from a number of manufacturers and are covered by an international standard for dimensions, IEC 61076-2-103. They are superficially similar to the older and smaller DIN connector range, but are not physically compatible with them.
http://en.wikipedia.org/wiki/XLR_connector
Polar patttern
Mixing desks
-History
The best thing to use in the music industry to creat soundtracks are the mixing desks.Sounds can be combined into more channels, also mixing desks change the sound format and it can add the frequency,dynamics and paronamic position.This are done to creat a more interesting soundtrack that is more appeling to the listeners.Before the mixing desks sounds were recorded during a live performance,if a musician made a mistake the song had to be balanced back so the perfomance was obtain in a good standard.
-routing
The output from one channel has to be combined with that from other channels. In the simplest desk, all channels may be permanently routed to a master stereo output, but more typically channels are routed through groups and from there to the main outputs.
Depending on the intended role of the desk, there may be anything from two to 48 groups, with varying levels of sophistication in terms of additional equalisers and auxiliary sends. Commonly, the groups are allocated in pairs, with the channel pan-pot providing the means of restricting a signal to a single group, and image positioning within a pair of groups for stereo working. On the subject of stereo, it's always better to use a dedicated stereo channel for a stereo source rather than a pair of mono channels panned left and right, because channel gains, fader positions and equaliser settings must be matched between the two sides of a stereo signal -- awkward to do with separate channels, but very easy with a dedicated stereo channel.
http://www.soundonsound.com/sos/1997_articles/apr97/mixeranatomy.html
A useful point to note: unused channels should not be left routed to groups or main outputs because this often degrades the noise performance of the mixing stages (although this will depend on the precise detail of the circuit topology used).
-channels
In a mixing desk in total there are 24*4*2 channels=24 input channels, 4 subgroup channels and two output channels.All the channels are quite the same but control each different input. They also the out most of the space in a mixing desk.There are different channels in a mixing desk:
-Input gain/Attenuation
-Phantom Power
-Equalization
-Auxilary channels
-Pan and assignment
-solo/mute/pfl
-channel on/ off
-slider
-busses
A bus is basically a path in which you can route one or more audio signals to a particular destination. Destinations can include groups, auxiliary sends, stereo mix, foldback or monitor. Commonly busses are used to route channel signals to a master group fader, a multitrack recorder, or the main stereo master fader (or all).
In live sound applications it is favourable to have a number of busses available to use. This is due to the fact that as audio channels can be bussed together and controlled by one group fader, it frees up the hands of the engineer and gives him/her less faders to have to control on the fly. It should be noted that although the master group fader does have control over the level of the summed audio in the group, the individual levels will remain with the offset set on the channel fader.
In Studio applications, busses can be used to group signals together for recording when there are too many channels of audio for them all to be sent to your multitracker/interface/soundcard. Busses are also commonly used to create a foldback headphone mix for the musician(s) to listen to. If you are using a smaller mixing desk for home recording there is not great need for a great number of busses - although it is good to have an auxilliary send and/or in built effects.
http://www.dolphinmusic.co.uk/article/103-what-is-a-mixer-bus-and-why-do-i-need-them-.html
-frequency
The high and low shelving EQ’s have a set frequency point. Your mid-range peaking EQ’s have a baseline frequency,For example, in my mixer, the frequency point of the High EQ knob is 10 kHz and the Low EQ knob is 100 Hz.All frequencies below 80 Hz are significantly cut with a shelving ratio of around 18 dB per octave. Considering my Low EQ frequency point is 100 Hz, then when I boost or cut my lows with the HPF engaged, I’m affecting a very small range of frequencies. If I really wanted to get some thump out of my kick drum, I wouldn’t want that HPF engaged
-polar pattern
Mixing desk
This are channels inputs
XLR
Microphones and some audio devices. Usually balanced audio, but XLRs can also accommodate unbalanced signals.
6.5mm Jack
Musical instruments such as electric guitars, as well as
various audio devices. Mono jacks are unbalanced, stereo jacks can be either
unbalanced stereo or balanced mono.
RCA
Musical devices such as disc players, effects units, etc.
XLR
Microphones and some audio devices. Usually balanced audio, but XLRs can also accommodate unbalanced signals.
6.5mm Jack
Musical instruments such as electric guitars, as well as
various audio devices. Mono jacks are unbalanced, stereo jacks can be either
unbalanced stereo or balanced mono.
RCA
Musical devices such as disc players, effects units, etc.
Channels
Recording devices
-History
there were alot of recording devices but the first one that was ever made was the phonograph made in 1857 by Leon scott.this device was created with a horn to direct the waves to the flexible diaphragm place at the small end.The phonograph also had a stylus and a lever to allow scratch a on the rotating cylinder beneth it.the cylinder was coated with a lamp black you could probably applied by allowing carbon acquire to.In 1874 before Leon finished creating the phonograph alexander bell experimented with the phonograph to see how the ear of a human being can detect a sound by experimenting it he used a cadaver attaching the stylus to the ear drum and using it to make a recording he was abled to use this experiment to make a recording of the sounds that camed out of the horn.this recorded on a moving glass strip coated with a film carbon so there was probably no original recording.
-Tracks
Multitrack recording is the same as sound recording the idea of multitrack recording was to have the track in the same tape but using different audio channels to seperate the tracks.In the 1980`s to 1990`s computers were provided with all of this softwares so sound recording and reproduction could be digitalized.Korgs are also another device that were made for tracks it can be used for multitracks and it can also record different instruments in these tracks.
-Input
each korg has about more than 4 inputs Audio input devices allow a user to send audio signals to a computer for processing, recording, or carrying out commands. Devices such as microphones allow users to speak to the computer in order to record a voice message or navigate software. Others are made to interface a computer with a CD audio source, digital audio, or MIDI instrument such as a synthesizer.
http://www.wisegeek.com/what-are-audio-input-devices.htm#didyouknowout
-Output
Any device that outputs information from a computer is called, not surprisingly, an output device. Since most information from a computer is output in either a visual or auditory format, the most common output devices are the monitor and speakers. These two devices provide instant feedback to the user's input
http://pc.net/glossary/definition/outputdevice
-Format-standeler or computer
A recording format is a format for encoding data for storage on a storage medium. The format can be container information such assectors on a disk, or user/audience information (content) such as analog stereo audio. Multiple levels of encoding may be achieved in one format. For example, a text encoded page may contain HTML and XML encoding, combined in a plain text file format, using either EBCDIC or ASCII character encoding, on a UDF digitally formatted disk.
http://en.wikipedia.org/wiki/Recording_format
In electronic media, the primary format is the encoding that requires hardware to interpret (decode) data; while secondary encoding is interpreted by secondary signal processing methods, usually computer software.
-History
there were alot of recording devices but the first one that was ever made was the phonograph made in 1857 by Leon scott.this device was created with a horn to direct the waves to the flexible diaphragm place at the small end.The phonograph also had a stylus and a lever to allow scratch a on the rotating cylinder beneth it.the cylinder was coated with a lamp black you could probably applied by allowing carbon acquire to.In 1874 before Leon finished creating the phonograph alexander bell experimented with the phonograph to see how the ear of a human being can detect a sound by experimenting it he used a cadaver attaching the stylus to the ear drum and using it to make a recording he was abled to use this experiment to make a recording of the sounds that camed out of the horn.this recorded on a moving glass strip coated with a film carbon so there was probably no original recording.
-Tracks
Multitrack recording is the same as sound recording the idea of multitrack recording was to have the track in the same tape but using different audio channels to seperate the tracks.In the 1980`s to 1990`s computers were provided with all of this softwares so sound recording and reproduction could be digitalized.Korgs are also another device that were made for tracks it can be used for multitracks and it can also record different instruments in these tracks.
-Input
each korg has about more than 4 inputs Audio input devices allow a user to send audio signals to a computer for processing, recording, or carrying out commands. Devices such as microphones allow users to speak to the computer in order to record a voice message or navigate software. Others are made to interface a computer with a CD audio source, digital audio, or MIDI instrument such as a synthesizer.
http://www.wisegeek.com/what-are-audio-input-devices.htm#didyouknowout
-Output
Any device that outputs information from a computer is called, not surprisingly, an output device. Since most information from a computer is output in either a visual or auditory format, the most common output devices are the monitor and speakers. These two devices provide instant feedback to the user's input
http://pc.net/glossary/definition/outputdevice
-Format-standeler or computer
A recording format is a format for encoding data for storage on a storage medium. The format can be container information such assectors on a disk, or user/audience information (content) such as analog stereo audio. Multiple levels of encoding may be achieved in one format. For example, a text encoded page may contain HTML and XML encoding, combined in a plain text file format, using either EBCDIC or ASCII character encoding, on a UDF digitally formatted disk.
http://en.wikipedia.org/wiki/Recording_format
In electronic media, the primary format is the encoding that requires hardware to interpret (decode) data; while secondary encoding is interpreted by secondary signal processing methods, usually computer software.
-monitors
-History
There are different monitors based on audio sounds that makes a great effect on music industry the most impotant one is Studio monitors that are designed just for audio productions like recording studios,filmaking,television studio and radio stations.Where as other monitors are made for loud speakers.
-Speakers
The first speaker ever made was a loud speaker called the standard dynamic loud speaker made in the 1920`s that uses a magnetic fiel to move the coil that is connected to the diaphragm so it can produce a sound.The first speaker ever made was a horn made by Thomas Edison, Magnavox, and Victrola that was created in the 1880 to 1920`s the problem with horns were that it dint capture the sound so much.Horns dint use eletricity it used Audio Recording for entertainment and recorded keeping, later
on for voice radio.
monitor speakers are used for audio mixing and mastering tasks this will creat a track that is pleasing for a range of playback systems that are played in radios,car stereo and boom boxes.To mix well you have to hear well if your about to mix a track at home and have the most rubbish speakers ever the track that you just created will sound rubbish the better speakers you have the better sound your mixing track will became.Studio monitor speakers are designed accurated mixing sound as possible.
how do speakers work?
To make a speaker work an eletrical sound has to pass through a wire coil,attached to a diaphragm and with a magnet fixed within the coil.The current in the coil within the magnetic field causes the coil to move ,which causes the diaphragm to move – at the same frequency as the original sound vibrations.
As the diaphragm moves, it will cause the air around it to vibrate as well, and will transmit sound waves the same as the original sound waves.
http://www.soundware.co.uk/pages/speakers.php
-Mid field and nearfield the loudness of course, having a big monitor nearby would require you to lower the volume a lot, so its a bit overkill. Second, most midfields are 3-way or more-way speakers, meaning you have 3 or more speakers to produce the sound, bigger speakers are designed so the output of all the speakers in conjunction with the cabinet mix up together and reach up to you, if you are very close to the speaker (say like a near field) you could probably hear stuff detached, instead of a single "ball of sound". Thats a determining factor for me besides size, the amount of speakers inside the cabinet, for instance Genelec 1032s are big speakers but they are 2 way, so to me, they are kinda in the limbo between mid-fields and big near fields, but say Genelec S30Ds are exactly the same size of the 1032s but they are 3 way speakers, so i would consider those mid-fields.
Which brings me to the next point, and thats size, if you have a big speaker near you, you will probably wont be able to sit at the correct height or distance in order to get all three (or more) speakers to deliver the sound to you equally, meaning if you are sitting at subwoofer height you wont hear enough highs, or viceversa, it would be kinda like having a movie theater screen very close to you, you wont be able to see the whole picture.
Dont be fooled with the terminology, loudspeakers are loudspeakers, the nomenclature is just used to describe a better usage for the speaker, a mid field will sound better if you are not very close to it, just like a nearfield sounds better if you are not very far. And of ccourse, the same principle applies, the bigger the speaker, the bigger the room should be, you wouldnt have a movie theater screen in a tiny room, would you?
And also, and this is my own personal opinion, after a certain size, the bigger the speaker, the less useful it is for mixing, recording, etc.. and the more useful it is to impress the clients.
http://www.gearslutz.com/board/studio-building-acoustics/643234-nearfield-vs-midfield-monitors.html
-Frequency response
The chunk of response missing around 2kH is simply caused by destructive interference between the output of the two drive units. Over the region of the speaker's bandwidth, where both drive units contribute to the output (this overlap is illustrated in Figure 2), there will be points in space where the path lengths from each driver differ by multiples of half a wavelength, and, in a throwback to school physics 'ripple tank' experiments, silence breaks out. The second obvious change between the response at two different microphone positions -- the faster roll-off above 10kHz -- is caused by the directional characteristics of the tweeter. Any radiating diaphragm will begin to become directional as the wavelength of the radiated energy approaches the size of the diaphragm, and when we moved the microphone 'off axis' the tweeter directionality began to show. With phenomena such as interference and directivity at play, each different microphone position will have a unique frequency response, and the arbitrary choice of "microphone at 1m on tweeter axis" for a specification is just that -- arbitrary.
http://www.soundonsound.com/sos/nov00/articles/ustandingmons.htm
-Two way - Bass driver and tweeter. Three way - Bass driver, tweeter and mid range driver.
The near field and mid field terms relate to the proximity that the speakers are designed for. Near field would be on your desk/meter bridge. Mid field a few feet further away. The output of each being incrementally louder.
The term full range is applied to single speakers designed to play back a wide range of frequencies, like you would normally get in a car or it can refer to a system (of two speakers or more) that plays the full range
http://www.hijackbristol.co.uk/board/the-desk/2-way-3-way-studio-monitors/
-Passive-speakers will require a separate
amplifier and -actives
Actives are a cheap way of buying a cheap amp. Do you really want a cheap amp?
The arguments given for using active speakers are the fact that the speaker wires are very short, and the fact that they can using bi-amping instead of a passive crossover. This means they can tweak the response to compensate for whatever deficiencies in the speakers and cabinet.
the amplifiers in active monitors are compromised to a certain degree. This isn't to say there aren't good active speakers out there . Generally, active monitors don't use really high end amplifiers because the power supplies would make them too heavy.If you look at a decent power amp, they are heavy and expensive. With few exceptions, the amps in active speakers are compromised.
http://www.gearslutz.com/board/high-end/172646-compare-contrast-active-vs-passive-monitors.html
-Power rating power is the rate at which energy is converted from one form to another. The basic international unit of energy is called a joule and is defined as the amount of work energy performed by applying a certain amount of force (called a NEWTON) through a distance of one meter. The number of Joules of energy converted each second defines the amount of Power in watts . 1 Watt = 1 Joule/second; 5 watts = 5 Joules/second, etc. For instance, if your Exercycle had a generator connected to a lamp, the brightness of the lamp would depend on how hard and fast you pedaled. The faster and harder you pedal, the more power you generate and the brighter will be the lamp.
http://www.prestonelectronics.com/audio/Speakers.htm
The purpose of an amplifier and speaker system is to convert the raw electrical energy from the AC power line into acoustic (sound) energy that you can hear. During this process, energy is converted from matter (such as coal or gas) into heat, then mechanical energy, then (possibly through magnetic energy) into electricity. From your AC power outlet the electrical energy is controlled by the amplifier in order to build a weak electrical signal into a strong signal, which is fed to the speaker to create magnetic energy, which pushes the speaker cone back and forth (mechanical energy), which moves air molecules in order to convert some of that mechanical energy into sound energy. The more energy (per second) is converted into sound, the louder the sound will be. So volume depends on the amount of electrical POWER used to create the sound by the speaker.
http://www.prestonelectronics.com/audio/Speakers.htm
-History
There are different monitors based on audio sounds that makes a great effect on music industry the most impotant one is Studio monitors that are designed just for audio productions like recording studios,filmaking,television studio and radio stations.Where as other monitors are made for loud speakers.
-Speakers
The first speaker ever made was a loud speaker called the standard dynamic loud speaker made in the 1920`s that uses a magnetic fiel to move the coil that is connected to the diaphragm so it can produce a sound.The first speaker ever made was a horn made by Thomas Edison, Magnavox, and Victrola that was created in the 1880 to 1920`s the problem with horns were that it dint capture the sound so much.Horns dint use eletricity it used Audio Recording for entertainment and recorded keeping, later
on for voice radio.
monitor speakers are used for audio mixing and mastering tasks this will creat a track that is pleasing for a range of playback systems that are played in radios,car stereo and boom boxes.To mix well you have to hear well if your about to mix a track at home and have the most rubbish speakers ever the track that you just created will sound rubbish the better speakers you have the better sound your mixing track will became.Studio monitor speakers are designed accurated mixing sound as possible.
how do speakers work?
To make a speaker work an eletrical sound has to pass through a wire coil,attached to a diaphragm and with a magnet fixed within the coil.The current in the coil within the magnetic field causes the coil to move ,which causes the diaphragm to move – at the same frequency as the original sound vibrations.
As the diaphragm moves, it will cause the air around it to vibrate as well, and will transmit sound waves the same as the original sound waves.
http://www.soundware.co.uk/pages/speakers.php
-Mid field and nearfield the loudness of course, having a big monitor nearby would require you to lower the volume a lot, so its a bit overkill. Second, most midfields are 3-way or more-way speakers, meaning you have 3 or more speakers to produce the sound, bigger speakers are designed so the output of all the speakers in conjunction with the cabinet mix up together and reach up to you, if you are very close to the speaker (say like a near field) you could probably hear stuff detached, instead of a single "ball of sound". Thats a determining factor for me besides size, the amount of speakers inside the cabinet, for instance Genelec 1032s are big speakers but they are 2 way, so to me, they are kinda in the limbo between mid-fields and big near fields, but say Genelec S30Ds are exactly the same size of the 1032s but they are 3 way speakers, so i would consider those mid-fields.
Which brings me to the next point, and thats size, if you have a big speaker near you, you will probably wont be able to sit at the correct height or distance in order to get all three (or more) speakers to deliver the sound to you equally, meaning if you are sitting at subwoofer height you wont hear enough highs, or viceversa, it would be kinda like having a movie theater screen very close to you, you wont be able to see the whole picture.
Dont be fooled with the terminology, loudspeakers are loudspeakers, the nomenclature is just used to describe a better usage for the speaker, a mid field will sound better if you are not very close to it, just like a nearfield sounds better if you are not very far. And of ccourse, the same principle applies, the bigger the speaker, the bigger the room should be, you wouldnt have a movie theater screen in a tiny room, would you?
And also, and this is my own personal opinion, after a certain size, the bigger the speaker, the less useful it is for mixing, recording, etc.. and the more useful it is to impress the clients.
http://www.gearslutz.com/board/studio-building-acoustics/643234-nearfield-vs-midfield-monitors.html
-Frequency response
The chunk of response missing around 2kH is simply caused by destructive interference between the output of the two drive units. Over the region of the speaker's bandwidth, where both drive units contribute to the output (this overlap is illustrated in Figure 2), there will be points in space where the path lengths from each driver differ by multiples of half a wavelength, and, in a throwback to school physics 'ripple tank' experiments, silence breaks out. The second obvious change between the response at two different microphone positions -- the faster roll-off above 10kHz -- is caused by the directional characteristics of the tweeter. Any radiating diaphragm will begin to become directional as the wavelength of the radiated energy approaches the size of the diaphragm, and when we moved the microphone 'off axis' the tweeter directionality began to show. With phenomena such as interference and directivity at play, each different microphone position will have a unique frequency response, and the arbitrary choice of "microphone at 1m on tweeter axis" for a specification is just that -- arbitrary.
http://www.soundonsound.com/sos/nov00/articles/ustandingmons.htm
-Two way - Bass driver and tweeter. Three way - Bass driver, tweeter and mid range driver.
The near field and mid field terms relate to the proximity that the speakers are designed for. Near field would be on your desk/meter bridge. Mid field a few feet further away. The output of each being incrementally louder.
The term full range is applied to single speakers designed to play back a wide range of frequencies, like you would normally get in a car or it can refer to a system (of two speakers or more) that plays the full range
http://www.hijackbristol.co.uk/board/the-desk/2-way-3-way-studio-monitors/
-Passive-speakers will require a separate
amplifier and -actives
Actives are a cheap way of buying a cheap amp. Do you really want a cheap amp?
The arguments given for using active speakers are the fact that the speaker wires are very short, and the fact that they can using bi-amping instead of a passive crossover. This means they can tweak the response to compensate for whatever deficiencies in the speakers and cabinet.
the amplifiers in active monitors are compromised to a certain degree. This isn't to say there aren't good active speakers out there . Generally, active monitors don't use really high end amplifiers because the power supplies would make them too heavy.If you look at a decent power amp, they are heavy and expensive. With few exceptions, the amps in active speakers are compromised.
http://www.gearslutz.com/board/high-end/172646-compare-contrast-active-vs-passive-monitors.html
-Power rating power is the rate at which energy is converted from one form to another. The basic international unit of energy is called a joule and is defined as the amount of work energy performed by applying a certain amount of force (called a NEWTON) through a distance of one meter. The number of Joules of energy converted each second defines the amount of Power in watts . 1 Watt = 1 Joule/second; 5 watts = 5 Joules/second, etc. For instance, if your Exercycle had a generator connected to a lamp, the brightness of the lamp would depend on how hard and fast you pedaled. The faster and harder you pedal, the more power you generate and the brighter will be the lamp.
http://www.prestonelectronics.com/audio/Speakers.htm
The purpose of an amplifier and speaker system is to convert the raw electrical energy from the AC power line into acoustic (sound) energy that you can hear. During this process, energy is converted from matter (such as coal or gas) into heat, then mechanical energy, then (possibly through magnetic energy) into electricity. From your AC power outlet the electrical energy is controlled by the amplifier in order to build a weak electrical signal into a strong signal, which is fed to the speaker to create magnetic energy, which pushes the speaker cone back and forth (mechanical energy), which moves air molecules in order to convert some of that mechanical energy into sound energy. The more energy (per second) is converted into sound, the louder the sound will be. So volume depends on the amount of electrical POWER used to create the sound by the speaker.
http://www.prestonelectronics.com/audio/Speakers.htm