【 Weak current college 】 computer head light and scanning light use of the similarities and differences

Computer lamp structure features and similarities
Now popular PC lamp by structure type to generally there are two types of lenses scanning computer lamp, the other is shaking-intelligent lights.
Lens light scanning computer is on the front burner of the lamp on a reflective piece swing to projection beam. Lenses by pitch and position of the two motors, complete vertical and horizontal swing. Its biggest advantage is that the lens is very light, it is very easy to control, to produce a very rapid changes of the light beam movement. The drawback is: reflector, beam axis of range of motion. Therefore, more suitable for hanging.
"Ecstasy"-type computer lamp, is the origin of intelligent lights originally envisaged, it has the advantage of driving beam lamp rotation movement, rotation of a wide range to be rotated 360 °. This motion effect to produce flavor on the stage full visual experience. The disadvantage is driven by motor power shake of larger, heavier caused light body. But as scientific and technological progress, this disadvantage has gradually been overcome, makes such lamps in recent years been leaps, has become the mainstream TV arena intelligent lights. Its volume is already able to do very small, the weight to be very light, it is very easy to use. This kind of computer it is function of the lamp is more and more full, from the initial subject to technical limitations can only do a pure color effects, has developed to the same lenses as scanning computer lamp, can produce very rich artistic computer lamp. Suspension and are suitable for display.
The use of occasions
Beam scanning lamp is small and therefore more suitable to the suspension or bar hanging, and far, subject to the limitations of the scope. For example, a relatively large theatre, a large concert has professional stage steel frame and light bridge, Rod is ideal and scanning light lifting, as these local use.
Now the head light technology has been considered by leapsand bounds, small size, light weight, features more and more, but also more convenient to use, display, hanging can, in some such as KTV, disco, diffuse shake, fan club, the small outdoor summary performance, can be used.

【 Weak current College 】 stage machinery safety design (2)

2. safety equipment
Device security refers to the stage machinery equipment in working condition long-term use no accident; in the event of a temporary failure could reduce the technical parameters after working capacity; stage machinery equipment on non-functional State of perception, ability to display and alarm. This capability or performance are usually mechanical design itself and completed with electric control; consider performance stage machinery cannot be ruled out as soon as possible of a temporary failure of the stage machinery of the failure as possible without affecting the performance of the work.
Involves device security to a number of factors, mainly has following several aspects:
1. adequate factor of safety
All mechanical parts of the selection and design must guarantee at rated load and inertia loads under the combined action, be reliable and have certain security reserve, i.e. sufficient safety factor. Safety factor is defined as: all the materials of the ultimate stress and part of the maximum operating stress ratio. Part of the working conditions should consider maximum static load and dynamic load (emergency braking, collision, and so on reboot inertial load) under stress. For example: hanging weights or traction wire rope, the safety coefficient should be greater than or equal to 10; lifting chain of safety should be greater than or equal to 12; transmission chain of safety should be greater than or equal to 10; all parts of the transmission system in the selection should be able to withstand two times the load rating; initial brief calculations, transmission parts and mechanical parts of the safety coefficient should be greater than or equal to 6, precise calculation of the safety coefficient should be consistent with the relevant standards or specifications on the part of the class.
2. appropriate stiffness
The stage machinery of major force widget should have appropriate stiffness in order to adapt to the needs ofthe stage performance. Typically, the provisions in force in the Member deflection should be less than the widget span of 1/1000 ~ 1/800, sometimes also the absolute value of the deflection, these components must meet both requirements. Transmission parts of torsional rigidity, stability, etc. compressed should comply with the standard norms. On the platform, in order to obtain sufficient lateral rigidity, which calculates the horizontal load of not less than the vertical movement of the load of 1/20.
3. reasonable structure and the dimensional parameters
To avoid or reduce certain structural dimensions of machine parts as unreasonable and excessive additional stress or premature fatigue failure, the design-time structure should be given enough attention. Winch system parts typical, such as: winch drive drum diameter should be greater than the diameter of the wire rope and pulley diameter 18 times larger than the diameter of the wire rope 20 times, friction drive (traction transmission) of the driving wheel diameter should be greater than 40 times the diameter of the wire rope, etc., all of the accessories are ropes and wire rope diameter to be matched, rope connector can only use those delivery loads greater than 80% of the wire rope breaking load of the coupling type, such as: splicing connector, alloy cast casing joints, wedge-shaped connectors, extruded aluminium alloy casing and comply with the requirements and proper use of steel wire rope clips, etc. Drum flange height should be greater than 2 times the diameter of the wire rope, single-layer winding of the drum, leaving at least two ring friction fixed ring, steel rope and reel or pulley of deflection angle should be less than 4 °.
4. reliable security appliance
The stage machinery and equipment must have better security protection devices to ensure the safety of the operation of equipment. General devices are equipped with stroke limit switches and hyper-limit switch. Similar to the boom winchor supposes, winch rope protection, job hopping or composite rope protection, overload protection and speed (over current) protection and other security measures and in the very context display and alarm in the event of a dangerous situation, the control system only allows devices to reduce the dangerous movement. When the device returns to normal status in order to continue the original campaign. For security purposes, the stage machinery of the brake must be: two independently controlled brakes (or hydraulic lock), or institutions self-locking (including reducer self-locking) together with a brake. Moreover, each actuator is take all the transmission torque. In a two-stage brake when one is working brakes, another is the safety brake.

【 Weak current college 】 computer head light and scanning light use of the similarities and differences

Computer lamp structure features and similarities

Now popular PC lamp by structure type to generally there are two types of lenses scanning computer lamp, the other is shaking-intelligent lights.

Lens light scanning computer is on the front burner of the lamp on a reflective piece swing to projection beam. Lenses by pitch and position of the two motors, complete vertical and horizontal swing. Its biggest advantage is that the lens is very light, it is very easy to control, to produce a very rapid changes of the light beam movement. The drawback is: reflector, beam axis of range of motion. Therefore, more suitable for hanging.

"Ecstasy"-type computer lamp, is the origin of intelligent lights originally envisaged, it has the advantage of driving beam lamp rotation movement, rotation of a wide range to be rotated 360 °. This motion effect to produce flavor on the stage full visual experience. The disadvantage is driven by motor power shake of larger, heavier caused light body. But as scientific and technological progress, this disadvantage has gradually been overcome, makes such lamps in recent years been leaps, has become the mainstream TV arena intelligent lights. The audio, video, introduction of relevant international standards in the field of volume has been able to do very small, the weight to be very light, it is very easy to use. This kind of computer it is function of the lamp is more and more full, from the initial subject to technical limitations can only do a pure color effects, has developed to the same lenses as scanning computer lamp, can produce very rich artistic computer lamp. Suspension and are suitable for display.

The use of occasions

Beam scanning lamp is small and therefore more suitable to the suspension or bar hanging, and far, subject to the limitations of the scope. For example, a relatively large theatre, a large concert has professional stage steel frame and light bridge, Rod is ideal and scanninglight lifting, as these local use.

Now the head light technology has been considered by leaps and bounds, small size, light weight, features more and more, but also more convenient to use, display, hanging can, in some such as KTV, disco, diffuse shake, fan club, the small outdoor summary performance, can be used.

【 Weak current College 】 read KOSMOS PRO actuator

In sound reinforcement systems, peripheral devices on the audio system is also critical, on-site auditory effect is good or bad, is closely linked with the peripheral equipment. While in many peripheral processing equipment, the actuator is an influential processing equipment. Its application areas, theatres, Conference Hall, multi-function Hall, DISCO dancing and singing etc can all see it. Audio system to properly use the actuator may enhance audio clarity, can be about sex and expressive and increase loudness, sound more enchanting, reduce the auditory fatigue; increases the sound image of three-dimensional sense and sound isolation, improve the sound level, thereby enhancing the quality of the sound playback, more beautiful to listen to the result.
For members about a company from PEAVEY's low frequency Exciter-KOSMOSPRO. The equipment is original Kosmos adds new sound control options such as STRATOS (increased luminance); Barometric switch control (regulation XPANSE range). Other new controls include a modified range of ultra-low frequency stimulation THUD switches, is used to modify QUAKE in compact or loose the dynamic control of the response, as well as high-precision band switch input/output level tables, as an innovative design, powerful low-frequency actuator. It can produce results in excellent low frequency output, thus increasing the clarity of any sound system and accuracy.
1. front panel features
1) bypass switch
When this switch is turned on, in addition to the low frequency suspension frequency level control (when off the kosmos-bass), other features disabled, signal output directly, while a red led is lit.
2) input level control
The knobs adjust the input gain. In order to have a good-to-noise ratio and performance regulation should monitor the input level meter. Adjustment to 0dB, led most of the time flashing, and the clipping indicator LEDs light does not shine as well. Because the center anchor is unified gain points,if the input level is too high, heavy bass may lead to clipping and distortion effects signal effect.
3) level selector switch
This machine level meter can display after level control after the input level, output level can also be displayed by level selector switch to switch the two-level display status. When this switch is pressed display output level; when the lift is display the input signal level. LED level indicator with yellow and red colours, yellow led light indicates warning, while a red led is lit indicating how there is 4dB cut off. During testing, when you see the yellow led is lit for a long time, it is necessary to note the level adjustment. In order to avoid signal overload, whenever possible, do not let the red light is lit, so as to ensure the best signal.
4) output level control
The knob to adjust the output level of the device. The centre point is unified gain points, 10dB gain can be used. When the bypass switch is pressed, the button does not work. The knobs are also adjustable S/PDIF output level.
5) subwoofer cutout switch
The switch from left and right main output removal after QUAKE and THUD-treated low frequency. It will not be removed from the program's normal bass. When you use 3-way speaker system, you can make all the increase of low frequency output from the Subwoofer port only. When the left and right speakers cannot produce low frequency processing extra, this is the best way to increase low-frequency. By XPANSE and STROATS signal is always sent to the primary output and are not affected by the impact of the switch.
6) pitch switch
Used to change the two different sizes of low-frequency harmonics between speakers of the base frequency. The "OUT" location has a higher centre frequency, which is more suitable for small speakers. "IN" position the center frequency down, in order to allow big speakers lower bass. If there is no significant difference, maybe the speaker does not exhibit reduced bass, it is recommended to use small speaker settings. If you modify the minimum frequency, the power amplifier has been working to bring to the speakers.
7）QUAKE
This switch can produce an even lower than the sound source a octaves of integrated low-frequency signal. It and DYNAMICS, corresponds to a specific frequency (frequency and amplitude) produces a natural low-frequency harmonics. When the input signal is beyond the scope of the set, it cannot go beyond this frequency effects processing. It also can enhance, increase the proportion of low-frequency frequency of music, particularly the bass drum. Because it also push for low-frequency section, care must be taken not to produce extra, extra low-frequency level so that the power amplifier is distorted or damaged speakers, especially when using small speakers.
8）QuakeActive
The reverse gear is used to indicate the QUAKE activity. As a reference, in the light does not light, if available, low frequency information exist, although the level may be very small, low-frequency harmonics are still in production. If there are no bass component sources, this lamp is not lit.
9）Dynamics
Some want low-frequency bass natural harmonic followed closely; there are times when you want it follows some loose, QUAKEDYNAMICS control can be adjusted to it. Follow tight, the increase hit effect; follow pine increases more bass, like the impact of a suppressed the bass drum, there is no impact on non-rejection of bass drum sound.
10）THUD
This control is the low-frequency harmonics QUAKE by adding a special chrome (natural instead of synthetic), it's about regulating a octaves above the low-frequency harmonic and balanced manner, using low-end. THUD and QUAKE on tweak the wave, the phase change, it will effectively modify waveform and amplitude, sound may and what you would expect.
11）DEEPER
It narrows down the THUDCircuit, delete parts of the bass frequencies. This makes it much more low frequency adjustment.
12）XPANSEbr>Use this switch in conjunction with BAROMETRICS, you can increase the frequency and bandwidth, minimum stereo position is the flatness of the settings. Clockwise rotation, left/right sound like a wider, while at the same time increase the sharpness. It is designed to maximize the sound of tension (elevating) mixed, and decrease the speaker's dressing filtering effect. It does not put the Mono sources into stereo, but it will improve the signal.
13）BAROMETRICS
This knob control XPANSE circuit between the two process phase. When counterclockwise (intensive) phase of dominant, clockwise rotation (thin) intensity high frequency increase. A typical setting close to the middle position.
14）STRATOS
This knob to the sound source to increase high-frequency harmonics, the frequency range above XPANSE range. It gives boring sources increase crisp effect.
15) bass
KosmosPRO internal has a low frequency divider 90Hz. Heavy low signal is QUAKE, THUD and from 90Hz divider of the aggregate of the low-frequency signal; this knob will change the output level. When the bypass, the increase of QUAKE and THUD portion will be dropped, leaving only the low frequency divider for low frequency signal. In this mode, is a standard bass control.
2. the back-panel features
1）S/PDIF
This is a stereo digital input output interface, the connection to other digital devices, such as computers or DAT recorder. If the default is no, then enter 44.1kHz will sync to the current frequency (up to 48kHz). If it and enter the clock locked, syncs indicator light; if there are errors of data err led light. If you select S/PDIF KosmosPro input modes, digital inputs is the only connect to the internal analog processing, but it will always synchronize internal sampling rate (if it has a clock). S/PDIF digital output has always been there, and about the same as the main output and. Digital full scale (0) is equivalent to an analog device pointer position (+ 22dBu).
2) subwoofer output
This is a balanced output 6.35mmTRS e, for balancing (6.35mmTRS plugs) or non-equilibrium (standard 6.35 plugs). Output also provides a signal to the subwoofer amplifier (line signal).
3) left and right output
Left and right output is electronic balance type, the Add method and the bass output as [② feet (or "tip" for cathode]. Card Lennon and 6.35mmTRS plugs directly parallel wiring. As balanced and unbalanced mixed into the same channel output, it becomes unbalanced. In mono mode, have left/Mono to mono output signal.
4) input
Left and right input is the balance of ② feet (or "tip" as the cathode. Card Lennon and 6.35mmTRS plugs directly parallel wiring. When you use a Mono signal, received left/Mono interface (card Lennon or 6.35mmTRS) and press the MONO selection switch.
5) input selection switch: you can select the simulation (card Lennon and 6.35mmTRS) input or digital input; select one another will be disconnected. This switch affects the left and right input. When the switch is selected in the S/PDIF digital input, MONO switch will not work.
6) MONO selection
If you use only one input channel KosmosPro, this switch should be pressed. It guarantees between left and right signal processing in the appropriate range. This switch does not S/PDIF unit.
3. the application of the system
KosmosPro in system connection generally use the threaded mode, and before the power amplifier, but then after the other peripheral equipment.
String into left and right channels, compensation, comprehensive voice signal because the left and right channels in various sound sources are mixed together, only for the overall compensation. Before the power amplifier, because power amplifier frequency response curve best, generally within 100kHz 10Hz — its gain change 0-0.2dB and actuators on the treble and overtone of compensation only 0.5-1dB, through the power amplifier is capable of varying success rate signal, direct drive speaker. If the add-in before other peripheral equipment, most likely because these devices are ideal frequency response performance, compensates the quantity is small in the treble and overtone constituents and lost, in this connection, it is usually a normal drive level.

【 Weak current College 】 mixer's basic operation principle

Sound mixer: Studio dedicated and dedicated two stage dance hall. 
Mixer's role is to:
1, pick up the signal, to zoom in;
2. press the required high, mid, bass tone Equalization;
3, will signal about bus as needed into or control group;
4, to feed into the secondary bus signal processing for art;
5. press the required output control. 
Mixer can be divided into input and output unit. 
(A) input cell input cell is an important component of a mixer, the input cell is tap parallel lines that are roughly the same per, generally can be divided into the following sections. 
A, input, select the portion of the TAPE: tape 1, 2, 3 MIC: microphone, LINE: line
B, the input Attenuator (PAD) if microphone or line input signal level is too high, while the gain control is unable to adjust, the attenuation switches to open when the preamplifier and input socket on the inserted a 20dB attenuator, to avoid overload. 
C, input gain control (GAIN) mixer sources are: microphones, musical instruments, tape, effects, sound amplifying equipment.
Because of their output level, in order to be able to match with them, we should use the mixer to gain control of the input sensitivity adjustment. If the input signal is too high can lead to clipping distortion, and vice versa if the input signal is too small, the noise will be unable to control the gain control is used to guarantee the sound mixer in fixed dynamic range. In the Panel gain control level size representation is 0dB = 775mV basis, depending on the size of the audio source output level, set in a different location. The input signal and gain level is shown in the following table. Gain (dB) 〓 input signals-60 to-50 〓 low-level microphone-microphone 35 〓 high (capacitance), electronic musical instruments-20 〓 low-level line (audio)
D, signal input jack is divided into low impedance balanced input (LO-Z card Lennon) and high-impedance unbalanced input (HI-Z 2 cores). General musical instruments and audio equipment in connection with non-balance, signal the "+" and "-" at one end and signal line screen public. For example: a core shielded cord, core wire is signal "+", shielded wire is signal and ground "-". This is better than no shielding of parallel lines induced noise, not belonging to the tube easy-masking. Professional audio device input and output are balanced, signal the "+" and "-" transport and then screened cables, "+" and "-" use independent of ground and plug XLR Plug me in using the card. 
E, overload (CLIP) overload is used to indicate the input signals instantaneous overload warning, the LEDs will peak (signal too large distortion level occurred) level when the glow below 3dB, easy to help set the gain switch location. 
F, input balance section input channel Equalizer is used on the input signal tone for the corrections to achieve a standard effect. As a result of a single control, so the mixer can make all the way to each balance control, but do not interfere with each other, the balance is divided into: HF (HIGH), medium (MID), LF (LOW). 0 location is flat; + direction (gain), + 15dB (enhanced 5 times);-direction (attenuation),-15dB (attenuation 5 times). Continuously tunable. Equalizer generally use the treble (10kHz), baritone (equalizer center frequency can move freely between 5kHz 350Hz — set), bass (100Hz) three equalizer. Because of the frequency section has a separate control, so you can enter a careful adjustment of the signal, which in turn can also adjust the tone for the bold attempt and squealing noise, noise, and so on not necessary ingredients to be effective.
1, high-frequency: 10kHz ± 15dB/slope impact zone: musical harmonics of the treble zone. Gain effect: metal sound increase, apical timbre comparison, gain too much noise can be clearly heard. Attenuation effect: can effectively remove hiss, attenuation is too much treble district of sense will be lost. 
2. intermediate: 3kHz ± 15dB/peak impact zone: musical instruments, voice and treble. Gain effect: the timbre brighter, harder texture, gain too much auditory fatigue susceptibility. Attenuation effect: music balance tends to bass, including sound will feel the same way. MF: 1kHz ± 15dB/peak impact zone: musical instruments, the human voice in tempo. Gain a clear effect: contour tone, sound phase forward sound projection, drum head modulated. Attenuation effect: after acoustic facies. MF: 500Hz ± 15dB/peak impact zone: the musical instrument, the bass voice. Gain powerful effect; timbre rich, gain too much telephone tone appears. Attenuation effect: tone head hard, balance tends too much treble, attenuation thin texture. 
3. low frequency: 100Hz ± 15dB/slope impact zone: musical bass area. Gain effect: a rich timbre, gain too much, the teeth tone is not clear. Attenuation effect: sound more relaxed sound good, tooth, background noise and loud hum noise effectively remove. 
G, acoustic phase knob for adjusting the crosstalk signal left, right balance, location in the channel after the level adjustment potentiometer. And each input channel signal on the Group 1-2 and 3-4 group rooms pan position location is determined by the knob. If the knob position in the Middle, pan position also in the middle. Knob to the left, navigate to the group in 1 or 3. Rotary tuning to discussions, positioned to the right in 2 or 4 groups. 
H, sniff (MON/SEND) sniff used to control the monitor bus input signal level value, this control in addition to gain control, is not subject to any control on the channel switch control (including the channel volume control). Therefore send signals and relatively independent of main bus bar signal. [fenye]
I, effects send (EFX/SEND) it includes all peripheral devices, used to determine the internal or external effect of the number of letters in theNumber into the input signal. It is balanced and volume attenuator, because each channel has its own effects sends, so by adjusting thenumber of channels, can make a difference, while other channels do not produce results. But be aware that internal and external effect one send control, so they should have the same audio source. 
J, listen switch (PFL/CUE) when this switch is "ON", the input channel signal to earphone to listen and to confirm the level meter, listens to switch on the order of precedence to firmly in mind.
Mixer operation terms and for the control of
GAIN: input signal HIGH-gain control: treble level control
MID-HIGH: treble level control LOW: bass level control
PAN: phase control MON.SEND: tap listens to signal control
EFX.SEND: tap effect signal control LIMIT (LED): signal amplitude indicator
LEFT.: left RIGHT signal level control: the right road signal level control
MONITOR: monitor system MON.OUT: monitor output
MASTER: General road level control EFX.MASTER: effect of the output level control
EFX.PAN: effect of phase control EFX.RET: effect return level control
Effect send EFX.MON: monitoring system level control DISPLAY: level meters
ECHO: reverb HIGHIIN: high-impedance input
LOWIIN: low resistance to enter OUT/IN: output/input jack
AUX.IN: auxiliary input multi-output MASTEROUT: total
EFX.OUT: effect of output results returned input EFX.RETURN:
LAMP: special lighting power supply POWER: total power switch
BALANCEOUTPUT: balanced output FUSE: a fuse
PEL: pre-listening (audition) press the EFF: effect of level control
MAIN: main LEVEL: channel balance control
HEADPHONE: Headphone Jack PHANTOMPOWER: phantom power switch
SIGNALPROCESSOR: signal processor EQUALIZER: equalizer
SUM: total output group switch LOWCUT: low frequency cut switch
HIGHCUT: high frequency switch PHONOINPUT resection: player input
STEREOOUT: stereo output ACTIVITY: dynamic indicator
CUE: selected to switch MONOOUT: mono output
PROGRAMBALANCE: primary output sound image control MONITORBALANCE: listen to output sound image controlr>EQIN (OUT): Equalizer access/exit key FTSW: foot switch
REV.CONTOUR: reverb profile regulates PA: fixed value decay, attenuator

【 Weak current College 】 audio cable in the audio signal transmission in the importance of (a)

Whether it's music production equipment or a sound playback device, audio cable has always been an important part, but quite a lot of people do not pay attention to it. This or short or long lines, the rough, has a small core, and all kinds of joints, play audio data transfer, and if a good music playback devices, does not match a proper audio cable, you will not be able to achieve the best playback effect, sometimes even bear to listen to "miserable." This article describes audio cables and related issues, there is only one purpose — to make the best transmit sound signals.
Digital audio cable
In General, because many digital audio cables use the same with analog audio connectors (such as cards, Lotus mouth agricultural XLR, RCA), many people with analog cable temporary place of digital cable. Although they can improvise, but you must realize that this is very wrong.
Analog cable and digital cable is a completely different analog cable impedance requirements, because the length, in cable points, impedance in 30 to 90 Ohm impedance fluctuation between changes, and will not affect the sound quality of analog audio. And this is for digital audio, digital audio signal frequency is very high (approximately 3MHz) of pulse wave, in order to accurately transmit signals, cable must be sending and receiving equipment match, the entire cable impedance must be consistent. For example, AES/EBU cable must be the other end from one end to show constant 110 Ohm impedance, which is also the AES/EBU cable than appearance almost microphone cable expensive many causes.
If you are using analog cable temporarily replace a digital cable? first because the impedance mismatch, the cable will have a standing wave reflection, "pollution" signal, the outline of the pulse wave. Pollution also from cable distribution capacity, it can reduce cable of high-frequency response, impact pulse rise time. Pulse waveform in high and low voltage conversion defined out signal 0 and 1,if you were not the correct impedance and capacitive effects has been polluted, pulse signal, the receiver of the signal interpretation error occurs, a time before and after the offset (called a flutter, Jitter), which reduces the audio quality, even wrong code appears.
Making computer music cannot be separated from the rest of the audio cable, they do not have a microphone, synthesizers, sound mixer, za, and other equipment that has a striking appearance, often looks like spread all over the floor. The average user on the device, compare careful careless, connecting cable comparison on the line, not much attention to quality. If all devices are generally lower, cable digital out insufficient also; if Studio equipment is advanced, low-quality cable became the sound of the "bottleneck", we will follow it degraded. Then replace the cable investment is necessary.
Analog audio cable
Analog audio cable can roughly be divided into three categories: microphone cable, guitar/line cable and loudspeaker cables. Typically, the center part of the audio cable is a conductor, is the carrier of the audio signal. Conductor outside covered with non-conductive plastic or rubber, and the outside is made up of a conductor, which on the one hand, the shield is isolated from the external interference, on the other hand, constitute a signal ground wire (loop). The outermost layer of the skin can protect the internal layers so that the cable is durable. Audio cable for general use copper conductor, because it cost low, the electrical conductivity is good, more flexible. However exposed to air oxidation of copper easily and become poor conductor copper oxide, affect current conduction.
Analog audio cables use the typical form of several patch, microphone typically use card administrative (XLR), line connections using large three-core (1/4 inch) or Lotus mouth (RCA), speaker cable connector often bare copper wire, sometimes with banana plugs or other plugs. Plug and socket binding, full ductility of gold-plated layer can fill the gap in the connection, keep the good connections.

【 Weak current College 】 woofer speaker setup and settings---Power By 【 China power house network 】

When you carefully adjusting a home theater system, you want to find a woofer speakers the best location to home theater systems to achieve optimum combination and sound effects for home theater users this has always been a great challenge.
In this challenge, there are two major factors are indispensable, surrounded by the listening area room boundaries and the boundaries around air energy storage capacity. These two factors combine to determine the listening area of unique acoustic properties.
Repeated testing is the key to getting the best sound and practical way. Reality does not exist a perfect set-up locations, you can only find those closer to the perfect location. You need to draw many sketch, and in some ways to make some compromises. Compromise is not only listening room acoustic properties of the result, but also for the beautiful, practical and other considerations, such as the wife of preference, and you cannot fail when back to business. In some families, wife of preference may be the most decisive factors.

The Bible of the woofer Setup cheat codes will be introduced to you, in one listening room how to display two woofer speaker. The trick here is to teach the technique does not require any type of measuring instrument, you only need a pair of ears can take care of everything. Of course, you also have the most basic test signal, such as a test CD albums and hit, or contains a large number of low-frequency content of music CD album. If you have a can of accurately measuring the sound pressure instrument, of course, the best; but if you haven't, don't worry, we have tricks up their sleeves.

A, room control

Many people have for their room the size of the "ideal" and proud. In General, the area of the listening room, low frequency waves can be heard, the fewer problems. In order to get smooth and balanced low-frequency sound, room ceiling to not less than 3 m, width to 3.6 m, length to 7.5 m (if you want more deep low-frequency, 9.1 m). Room of all sides (including the diagonal line) should not be an integer multiple of Sonic wavelength, the resonance phenomenon will occur. Of course, the larger the listening room, the more you need sound facilities to reduce acoustic reflex, but standing wave is generally not in the formation of large listening room. Another point is very important to use the absorption material, this time-frequency attenuation in General and to the same frequency. This is what people call "neutral room".
Light and flexible walls can act as a very good sound-absorbing walls, where low frequency sounds will go through the wall, there is no reflection. The gypsum board, timber and dual layer structure with the correct stone built walls can achieve this effect. A world class standard room usually require the use of sound-absorbing materials, powder tones/to eliminate resonance or echoes. To watch out for the holes in the walls, they also can cause resonance. Build a good low-frequency response, proper internal acoustic features and State-of-the-art external noise effects of large room is a professional audio Division.


Second, the speaker's place — use the feet or shock pad?

Speakers can be installed in a hole in the wall, the speaker of the front panel and flat to the wall. This installation requires the skills of the professional sound engineer, we don't do too much of the discussion. Main speaker must be coupled with the removal of the floor. Heavy and solid tripod at the top to the bottom of not greater than the speaker in order to avoid the sound diffraction (this is a kind of comb-filtering). Feet stable effect is good, but is easy when using the carpet punctured carpet.

Some people recommend heavy speakers following pad lock-mat in place of the feet. No matter which Hina, its goal is to reduce the synchronous resonance, or reduction in the box, floor and wall between the conduction of sound. Cabinet and the tripod's resonant frequency should be very low. Fist on a knock box, to confirm that it does not have a resonance characteristics to your audio system to input a sine wave signal, listen to the watch case is there any vibration.


3. positioning of the speakers

The woofer speaker set-up, we assume that: (1) you have a pair of healthy ears, you can assess your room's acoustic characteristics; (2) unless otherwise noted, you are using two woofer speaker; (3) do you have enough test signal software to test your listening position heard sound field characteristics at different locations, as well as the room is how it affects the sound effects; (4) sound system all necessary correction work already completed.
Woofer speaker set-up and correction can be divided into five steps:
1. woofer speaker set-up adjustments; 2. listener seat adjustment; 3. the main speakers and woofer speaker phase adjustment between; 4. the balance of the various parameters; 5. woofer speaker and main speakers match of gain.
I have seen in a room in the system to adjust the sound system to determine if more than one woofer speakers the best set-up of different methods. Here is one method:
This is a process of trial and error, you may want to once again listen to the same content. When you're bored you can sit down and have a rest, to auditory fatigue to a minimum.
In each test every step to make notes, to take advantage of some of the chart, try to write down some details, such as speaker set-up, and set the parameters, and so on, so that when you feel the sound is one hour earlier, you can then adjust back to a previous state. Remember, here you have to deal with the many variables, a record of each parameter can make adjustment process faster more efficient.

【 Weak current College 】 Studio knowledge-Power By 【 China power house network 】

Studio this objective environment, from the beginning, there will continue to progress and development of its various changes to make our audio production environment and play space are increasing, and this is how one thing? here is not just as simple as historical

Design by early scattering theory of control room

For how to design the control room, recording of demand is not high, so long as to be able to hear the voice, the so-called "right" to it to complete its work on the line, this is a question of accuracy. This is the most important, of course, you may request it's comfortable, able to work eight hours, and the fresh air, sufficient for the whole band member Ambassador can sit in a room in shrink and concrete, of arbitrary corner listening to sound like the noise better, in the control room in shrink out things, get another Studio, home, car audio to hear all the same, and so on.


Over the years, a variety of efforts to control room is a real voice, but in fact not true, because you listen to anything or depend on the listening environment. To this end, room designer for this invention many tips, eventually became early acoustic scattering solution.

Concentration of history--control room design

1960s, dry sound room (DeadRoom)

At that time, most music is rock music, producer requires direct sound heard that other not very bad, the finished product, the recording time can only work for 20 minutes, the whole room no short reverb, the human ear can not afford. May not be so bad that an overall impression.

The 70 's

Then have a stereo, people began to pay attention to the design of the control room, the best control room-step full wall with rough stone, speakers embedded in the wall, wall and ceiling made of bevel prevent echo. Hard wall provides sound reflection, produces acoustic perspective, it sounds no longer like put your head into the hair to do box, the General decay time, 100 m to 200 m to 380mg, 430mg, but no two rooms sounds like, a room and two local also sounds different.

Experienced sound recording engineer on the amplifier before, so that both Heng sound spectrum analyzer that look the same, but still does not resolve the problem.

The 80 's:

LEDE and lighting control

Mr. LEDE (LiveEndDeadEnd) first off the popularity in the 1980s, the United States were in the control room DonDavis step surface area of almost no reaction. His design enables direct access to sound and reflections of the time difference is long enough for the first time, the human brain can feel the room acoustic reflection, and can be ignored, this can be restored in the control room listening environments, and in the same LEDE control room to hear things are the same, in order to make the recording more comfortable to work, back wall or hard materials, but not cause slap. The theory that scattering MwfredSchroeder no Echo (ECHO), but there are some short reflex Decay.


Using short delay produced by the environmental effects before

The 60 's, 70 's wet reverb effect today is not too much. 1980s to low chorus, the 1990s were very clear sound localization. Recording with less reverb, listen to point the red pepper.

Only with a dry sound without reverb can have that kind of effect, but our musical instruments in an environmental performance, musical instrument itself. If you do, the lack of natural and suction line. Old like a drum machine, clean but no vitality, synthesizer and sampler instrument is the same, but if you joined the reverberation, makes sound distance to farther, do to make the recording sounds very real, and winds coming?

Some recording with speakers play the sound, electronic musical instruments and microphones to pick up, not as a method, but the better approach, with a short delay to create this kind of "room microphone".

Sound models, an electronic reverb for RI is a variety of room sound reflection of felt-, first in small, hard, sound-absorbing dry room recorded the sound of the "face" of the early people use digital delay, the delay time is set to 1, mixing these-10mg delayed sound to create a "comb wave" effect. With today's effects, reverb with reflection instead of long decay. Instead of a short delay.

Many effects provides a short Delay skills, multitone heavy chorus algorithm. Many cheap multiple effects to a stereo input signal into Mono, long delayed by lntialDelayFeedbacklevelpan to control, multi-tone modulation of Chorus can also Delay the total human time or each delay code.

Note:

This created a small, flat room effect

Very full modulation (Modulaton) join the dynamic of angry

If you want more static sound, the depth in ooo,

To change the properties of a room. Adjusting the predelay time (005, 007, 010). The next preset operation with the Tapped Midiverb4, although it only three Tap, as four tone chorus, each more flexible.


Time1005007003MasterGrrfnsvk:50%

Levell:999999Lowcetfilter:177Hz

Tan:-505000Hicnt:15.1KHz

Feedback:000000

MasterFeed50%, this makes it easier to hear each of the break. Can lower the cutting the drums independent environment have strong blowTo play sound.


A short delay in sound and image in the middle of the long sides. If you adjust to obtain additional processing signals, be careful with phase offset, listen to Mono determine sound and correct.

This method may cause the stereo field gets smaller, but I think this is worth it for the drum's overall sound field of the environment.

【 Weak current College 】 phase and time delay adjustment---Power By 【 China power house network 】

You may find that the adjustment phase settings (usually in the woofer speaker adjustment, if necessary, can also be adjusted on the decoder) will cause the primary speaker/sub woofer speaker integrated sound more balanced. Test will immediately display is improved: you can be in a certain phase to hear more or fewer low-frequency. Change the phase you hear listening room acoustics sound effects arising both sides, so that you are listening to the location of the sound effects will be different. To carry out phase adjustment, you must clear this would bring joys and half results.

Like I said before, you sit in a listening area where the audition of different, let your assistant to adjust a woofer speaker phase (the other one is closed), audition repeatedly until you get the best sound. Then another woofer speaker for the same test, you must first turn off the speakers. Finally, two woofer speakers are turned on, if necessary and then they are more adjustment, you will find this process using a different test software would be more helpful.

Despite the delay adjustment for different purposes, I still use it with phase adjustment back together, because when you adjust the delay parameter, you can also change the phase. Delay can be regarded as a system to display the sub woofer speaker or reduce the woofer speaker signal routing differences between electronic method.
Delay adjustment to different objectives for the professional audio and home audio fields are frequently used. Some people find ignored phase settings, put the effort in the delay would be better organised. In the woofer speaker sound effects of the integration process, the delay can be used to make from the woofer speaker and main speakers at the arrival of direct sound to hear. Know the speed of the voice, you can quickly figure out a variety of positioning mode signal transmission distance of difference (so even if the delay of many), but how do you figure out the figure number 39aof delay?
For this particular situation, suppose front woofer speaker and friendly speaker of a substantial mode enables its audio at the same time get to listen to the location. Calculation of sound from the left, Center, right, and the former woofer speakers to listen to the location of the time (in ms) or distance (in feet or metres). Then calculate the rear woofer speaker delay.

For example, suppose your left and right, in front of the main speaker/subwoofer speakers from the listening position 5 meters, while the rear left overload listen subwoof position 1 m, 5m – 1m = 4m. Therefore you need to make the rear woofer speaker 4m delay. In this way, from the main left and right, front woofer in/sound and rear woofer speaker sound can also reach the listening position.

【 Weak current College 】 multimedia listening environment and power mix---Power By 【 China power house network 】

In the old shaoyou spread in such a view: listening to music, one third of the equipment, seven environment. Many of my friends might think that the words were overkill, however, often heard that an Internet user in 10 sqm cabin bought 12-inch subwoofer, ready to use 200 watts of power to enjoy intensively, tearing pants of bass, it cannot help heart hair tightly, inhaled gasped, at this point, to think of the "one third of the equipment in seven environments", deeply embedded learning them.

This secret, to hear beautiful moving relationship, it is necessary to put the person, the box, the relationship between the environment. And this one, the size of the room and the speaker's power match, is our most neglected issues.

On the environment: a little experiment

To understand the environmental impact of multimedia sound, a little experiment.

A first step, the experiment room size on effect of sound, the sound box moved to the toilet in a song Hotel California, listen to the water Curtain Cave-like sound effects. The second step, the experiment room open on effect of sound, the speaker to move to places no empty furniture House on a piece of celery CAI, listen to what is called. The third step, the speaker to move to put a song in the plain area, to hear him Jay thin throat much color. You can also put the speaker to move to the people's Hall, Notre Dame de Paris ...

Of course, is a joke, not moving anywhere, a sound card capable of simulating various environment sound effect, then, with the extreme sound to hear me, for my part, very often, then you can appreciate the taste of the die.

Then, listen to the music room much? follow now room conditions, it might be a 10 square meters, 20 square meters and the lower limit is a limit of 10 square meters, in particular the following 5 square meters, according to the principles of acoustics, users almost impossible to get good bass, a low frequency band extruding, sounds dull, turbid in the chest, the heart beat of the side effects of stimulus. Therefore, for this condition, do not recommend to invest in multimedia speaker, using headphones. 25 square meters, more than 25 square metres in particular, could not support most of the multimedia speakers, their actual power not enough. These sections will be expanded later.

Random House's listening. Things messy often absorbs excess wave reflection, make the room close to the correct reverb index. Similarly, on the walls to hang some tapestry wool products such as are valid through the cudgel furniture decorations let become irregular, also have immediate effect.

The high roof is fortunate to have some, perceived sense of listening to the sound field of three-dimensional sense on and in depth, but unfortunately real estate ' and does not pay attention to these, in order to save costs, community building storey in 2.6 meters, raise your hands almost chuding, wretched so chilling. The solution is to put the roof also become irregular, such as the use of more complex style chandelier (careful), in addition, hanging various strange service drops must be young people live, in pursuit of a given a voice like that, you have the courage to leave their rooms into a Moleskine small gift show?

For a room on a variety of factors influence to listen, as follows:


The best worst condition settlement
15 ~ 20 m 5 m following discard multimedia speaker
Top high 2.8 meters 2.4 m to improve reflection at the top, such as the pendant lamp, ornaments
Irregular rectangular shape, due to the extremely narrow or a corner to adjust the computer location, set the partition
Display slightly confusing large glass or smooth products linked to cotton or wool shop products
Corner empty corner little empty corner lot, and solid cubic pile up incidentals or acoustical materials

Power and the environment: give me true digital

Discussed the size of the room, the next question naturally fell in power, more power to meet the needs of our stupid?

Every day we face merchants "800 Watt, 1000-Watt" bombing, even a small plastic loudspeaker can also play on 180 Watts of labels. This fool's trick was Japan speakers in Chinese sales of invention, and now is people give as "market without borders", "Bacchus". In fact, just a small concept trick – peak power or music power, 800, 1000, digital marking is speaker at the moment to be able to withstand the maximum power, and the speaker of the General relationship between the power output should be 10: 1 to conversion, and may even reach 16: 1.

What is a peak power, look at the lifter will know that they can break out of the hundreds of pounds of heavy lift the barbell, can only support a few seconds. While the speaker even a few seconds are not supported.

In accordance with real power, our room is about how much is enough?, or sentence, to be an experiment, movers to 1900T2 nominal power is 15W * 2, in the course of the nominal power M200 is 20W * 2, 15 square meters room listening to suffice. If enough sound pressure is not enough, then try to buy United States accurately, 2.1 speakers actually achieve 260W, 5.1 speakers 500W is reached. In the General House of the living-room, the ground shakes. At the recent emerging from the new multimedia, dew A100 callout 50W * 2 of output power, but they deal with the 20 square meters or more room or to some degree. In effect, doubling the power, sound pressure only three DB, therefore, the power of 50W * 2 and 20W * 2 power in most cases, the difference between the sound pressure.

【 Weak current College 】 importance of acoustic design

In recent years, there is increasing concern of acoustics, International Journal of all major forums, discussions on acoustic design is becoming more common. When listening and microphone and flat frequency response, when digital recording solution to the problem of noise of analog era, a new era had arrived for audio. Equipment gap shrinking, but the quality of the recording works basically depends on the recording technology and recording studios and final mix room acoustic design. Therefore, from the objective conditions, acoustical design for recording, than any other factors are important.
Ensure that the control room of the frequency response is flat for mixing and monitoring is very important, but a large portion of the control room did not deal with this. Worse, many people know nothing about it, not knowing ways to solve this problem. Listen for the sound produce distortion, in any case will not be able to make high quality recordings of works. The Studio of design and control room is completely different according to the recording studio working on need-the need for sound field accordingly adjusted, not all of the strong sound absorption. In order to meet the various needs of acoustic source recording, so that the effect of early recordings more perfect.
About sound insulation
In daily life, building or per far away from the street noise pollution sources doubled distance can be reduced noise, 6dB community trees can reduce the noise around 4dB.
(A) the noise propagation
Noise propagation can be in two forms: airborne noise, solid contained noise.
1. airborne noise
Sonic's most basic mode of transmission is airborne, and airborne noise is ambient noise which spreads through the air into the room noise. A fabric with small holes thick metal plate, small area of approximately 13% of metal plates, when a voice against the surface, there will be almost 97% of sound through. Sound propagation through the pore is quite amazing, asubtle cracks can destroy an entire seal acoustic system. Therefore, the room is isolated from the strict seal on airborne noise emitted is very necessary.
2. solid noise load
Sound through the vibration propagation in solid of machinery such as wood, metal, concrete, etc. Because the air transmission of inefficient to bring solid vibration, so solid containing the main source of noise is due to the vibration source contact solid media surface, such as underground generator or some electrical appliances. Solid sound transmission loss ratio of air transmission of small, can spread very far in distance but no major loss of energy. Wood, concrete and other longitudinal vibration about 304.8 m loss around 2dB. If the transmission medium for the iron and steel materials, the loss will be reduced to 20 times.
(Ii) sound processing
Acoustic treatment has many methods, Studio due to its strict on this floor noise, so the use of suspension structure. Suspension structure can greatly improve the sound isolation degree, especially for the Studio in the city. Suspension structure can add rooms and other structural separation, it can be suspended wall and the wall between a cavity and in the cavity fill soft acoustic material, acoustic using cavity absorption, so that the air, solid noise reduction to a minimum. But suspended structure to meet the requirements of the load-bearing structures.
Regarding the diffusion and absorption
Many people get plastered walls of sound-absorbing materials, thought this would be enough to eliminate reflections, because they feel when they clap your hands, so there is no echo, but this reflection on the control of low-frequency has no meaning. The brick and concrete walls, the problem is particularly serious. The more rigid wall, the more reflective.
Most people who blindly believe that a good Studio should do to minimize the reflection of sound absorption. But the shaping of sound field in no way merely refers to the uptakeof reflections, but rather on the room for listening and recording of optimization, that is, for the recording of work to do optimal consideration.
(A)
Diffusion and its role
Diffusion on improvement of sound field and listening to interested has an important role, quadratic residue diffusion body because of its good diffusion capacity and convenient installation, now widely used. Quadratic residue diffusion of formula for hn = (λ 0/2 N)? ' Sn, Sn to n-squared divide by N remainder, λ 0 for diffusion center frequency wavelength, n is the number of diffusion of lattice (must be a prime number), h is the height of a lattice, n = 0, 1, 2, 3, 4, 5, 6 ....
Quadratic residue diffusion of diffusion frequency threshold approximately in the center of the lower frequency-doubling, limit depends on the number of squares, can reach the center frequency n-1 times.
Diffusion and smooth reflector, which can effectively avoid sound focus. When the sound wave through a smooth wall reflection, all of the sound energy is reflected along the same direction, its reflection direction fixed, depending on the location of the sound source. When sound waves through the diffusion of reflection, the sound energy is scattered reflection in different directions, and have different phase.
The formation of uniform energy of irregular reflection sound makes one ear subjective produces a sense of space, at the same time use in high frequency will improve the sound of the proliferation of "bright". Its reflection direction roughly a semicircle, sound energy average spread. Diffuser has another effect, when the sound source, the acoustic reflection through the back wall, if the reflector is a smooth wall, a band only fixed reflection path points to the location of the recording. But when the reflector is a diffuser, due to the proliferation of Sonic to semicircle direction, there are countless different band reflection path converges in the recording location, and so on, there are numerous points of convergence of the same nature, which translates into expanded the best listening to tempo.
(2)
Sound-absorbing material (structure) and its role
Sound absorption can be divided into two situations, low frequency and high frequency absorption and absorption. Sound absorption can reduce echo, but also can effectively control the reverberation time, improve listening to tempo frequency response also play a very important role. So, you can create a listen for more clear and more standard mix environment.
1. high frequency absorption
High-frequency absorption is mainly directed against sound field design, early reflection sound wave interference, reverb and ECHO's control. Usually you can use full-band sound-absorbing glassFibreboard for sound absorption and conducted jointly with the diffusion of sound shaping.
2. low frequency absorption (low frequency sound,)
(1) role
The role of low frequency sound, is to avoid low-frequency interference in wave and sound of the low frequency sound distortion. Shows, when za sound impact wall surface, after reflection of sound waves and za continue to issue of Sonic interference effects. Depending on the wavelength, the sound pressure or be strengthened or be offset, and different locations with different frequency response. In an untreated rooms, Sonic mutual effect of reversed phase interference occurs, most will produce 25dB even more.
Many people mistakenly believe that the use of near-field monitor speakers can exclude acoustic defect. In fact, as there will be a standing wave, just listening position to the sound energy is large. Although the human ear close to the speakers when, because of the high-frequency acoustic reflection will gradually decrease the masking effect, but the low-frequency interference still exists.
Another misconception is to use the equalizer to change because of acoustic phase cancellation caused by the change of frequency response. Sonic interference androom structure, is objective and existent, unless you change the direction of sound transmission, otherwise use equilibrium cannot be changed because the sound wave interference induced frequency distortion. And the different locations of the frequency response curve is totally different, so it is not possible to rely on the equalizer to compensate acoustic defect.
Although the use of headphones you can avoid the room acoustic problems caused by, but the headphones sound only direct sound, making it very difficult for us to go to control the volume of some tracks. When we use the headphones, the main play music or song in a very small volume, you can even listen to very clear, so our final mix, lead singer of the level will be slightly lower than the desired level of value. Similarly, reverb and delay also lack accurate judgment.
Without sound, treatment room, some frequencies decay time reached 300ms, this will give other low frequency tone produces a big impact, damage, or even make voice clarity is harmonious.
In General, a room need as many sound depression. Although he may make the room became the "dead", but the rooms can never have too many mixed low frequency. Sound, can be installed in a room corner, walls, floor and ceiling, the greater the processing area, ideal for low-frequency effects.
(3) commonly used in low-frequency sound
A. Hershey Helmholtz resonator
Helmut Helmholtz resonator and a low frequency sound, that is different from the composition of the sound of glass wool, it can absorb more low-frequency component. It has adjustable hollow structure, a band of absorption is very effective. Absorption frequency range and the quality factor Q, Helmut Helmholtz resonator cavity absorption bandwidth formula is: f2-f1 = fr/Q, fr for resonance frequency is the maximum absorption frequency. By adding glass wool or add several openings of different sizes, you can make the absorption band of variable width. Helmut Helmholtz resonator of many species, usually is uses a large box of internal fill glass wool, front end cover a series of different dimensions, different spacing veneer. This design is known as narrow plates resonator. Although the hull Helmholtz resonator can absorb a certain band, but it can absorb a limited range, and the use of multiple resonator frequency absorption of broadening its scope be on sound field of activity, so it must be very careful.
B. Panel sound,
Panel of the sound in a narrow band noise, which can absorb bandwidth for approximately a octaves, it can use a series of a square around the Panel sound, to cover the entire frequency, without having to use a very heavy materials to increase the acoustic range. Because of the low-frequency components are close to four octave, so you can pass a different thickness of the Panel of the sound absorption in a combination of different frequency of low frequency and high frequency components can be its Panel reflection, therefore install multiple Panel sound, nor will it makes sound absorption can be all too silent sound field caused. Front panel can also be other shapes or combination, if placed in the back wall can be up to the required role diffusion.
C. acoustic suspension of sound-absorbing structure
Due to the wide range of low-frequency radiation, energy intensity, so when the low frequency sound wave into suspension system, the suspension of affected with swing (Visual observation), so will its kinetic energy is converted to its kinetic energy and heat. And because the suspension of the angle is different, so the transmission of sound direction also changed, so that the equivalent of a voice of the labyrinth. Acoustic in which constantly crash and be absorbed up to eliminate almost all energy.
Acoustic design to achieve the following four objectives:
1.
Prevent standing wave and wave interference, and ensure the flat frequency response.
2.
Reduction of small resonant frequency of the room, even distribution of the sound field.
3.
Reduce large room reverberation time of low-frequency, the better the reverberation time frequency characteristics.
4.
By absorption or diffusion, prevent echo and improve stereo positioning capabilities.
5.
Sound needs to meet the requirements of control rooms.
Important factors to consider
1, early reflection
Initial delay gap refers to the direct sound and the first indoor reflection i.e. head early reflection of the time difference, as shown in the figure. In subjective listening to interested, initial delay gap will bring the room characteristics of perception. For example, when we are in a closed room darkness, applaud, we can know the approximate size of room. This is due to the perception of auditory system to its initial delay clearance and transmitted to the brain room information. When the initial delay clearance greater than 20ms, the room will have sense and produced open sound about the same 7m 20ms. Large control room of the side wall of the dots in the early reflection at least listening points 3-4 meters, so make sure you are on the side walls of absorbing or diffusion treatment.



Early reflections on listening quality have a critical role, rich early reflection can bring relative active sound field, and the appropriate early reflection to improve the clarity of the sound has a certain role. 50ms listening points within reach of early reflections will enhance the clarity of the sound, but the control room can not be too much to join their roomCharacteristics, so early reflection should try to control their energy. Selection of the diffuser is a good way to spread evenly, allowing early reflection enhance clarity and sound field with a certain sense of space. At the same time help to promote high-reflection, whereas the low frequencies loudness makes sound reflection, so non-roiling should only non-high-frequency components (center frequency is approximately 1kHz), reflection onthe low-frequency should be absorbed.
But early reflection will also bring comb filter, noise interference problems that affect the sound field of positioning and listening spot frequency response. Many people early reflections are very inconsistent, reflections on the direct sound serious interference, can affect the forward sound image orientation, and then it will be very dogmas and all of the reflex points all covered with absorption material, and to top the whole room filled with sound-absorbing materials! this will only be more attenuated sound energy that we in the mix is constantly upgrading and enhancing the reverb loudness.
We are in the control room as well as shrinkage concrete strip processing, you need a standard space environment, the band reverberation time and room of frequency response as straight. At the same time, we want to ensure that the reflection of sound does the formation of mutual interference or influence the comb filter frequency response.
2. room dimensions and shapes
Room structure to listen to the sound effects are:
Parallel walls of stationed in wave resonance problems
Asymmetric structure on the positioning of
(1) in wave and resonance
Standing wave is composed of two columns in the opposite direction, with the frequency of the sound wave overlapping. When parallel wall space is an integer multiple of half-wavelength, that is, standing wave generated axial. At the same time, there is a ramp and tangential standing wave. When standing wave continues, will have resonance. The room of the resonance frequency of the general formula is:



Where c is the speed of sound, lx, ly, lz-room length, width and height, n is 0, 1, 2, 3, 4, ….
When several resonance mode resonance frequency of the same, the phenomenon of resonance frequency degenerate. A degenerate resonance frequency on the same frequency and resonance of sound is greatly enhanced, this could cause serious distortion of frequency. In the low frequency range, this phenomenon is particularly serious.
(2) symmetric structure
Thanks to produce stereo positioning two speakers to the sound of the volume and the time is different, therefore, in order to ensure accurate positioning of the original signal, Binaural accept to room information to maintain a certain balance. That is, when the speaker plays a tone when the test signal, Binaural should accept to almost the same reflection sound signal. If the room is a rectangle, the speaker should call set in a relatively short length of the front end, making listening location away from the back wall, so that it can be avoided in the back wall, near the most complex sound interference. The following figure is a set of comparison chart, on the left is the correct symmetric structures, the right to not ideal layout structure. On the right, the left speaker's high frequency components can be directly reflected to the listening point, while the right side of the left part of the sound through speakers wall reflection, reflection through the back wall part, leading to listen and point on either side of the reflex sound different, resulting in the sound and image positioning offset.


4. measurement of acceptance
Standards are as follows:
(1) above Acoustics: 55dB
(2) the frequency of the reverberation time: T = 0.4s (125Hz? 4kHz allows +-0.1s error)
(3) listen to the tempo of the frequency response: no uniformity of control within the 3dB

【 Weak current College 】 fuse in the use of sound

On some needed to buy relatively expensive electronic limiter, you can try to use a standard AGC fuse to protect your speakers. AGC fuse those internal metal, ends with metal caps small glass tube. While electronic limiter is more advanced, it can "read" the content of an audio signal, the limiter of creation and the content of the signals in accordance with the need to protect the speakers. In comparison, the fuse is a relatively stupid devices. They are on the current and one fixed time constant response (for example: fast current or slow currents).
You need to do work
Here comes to fuse the hardware can very simply from RadioShack or other electronic components supply sources. You need to find the AGC fuse holder, fuse and take some artificial. A good AGC fuse holder should have that kind of block type. It's two "U"-shaped metal card and can be used to stuck in its previous fuse. There is another relatively slight narrowly AGC fuse holder that is an embedded with like pig tail-like wire retainer. This type of fixation is generally used in automobile aftermarket electronic component, you may be in the auto parts supplier found. The third option is that fixed on the back is equipped with a removable cap mosaic plate fixation, in many electronic devices can see such a fixation. They can be placed in the vast majority of Jack Panel, and makes all the connections are done internally. The capacity of its current capacity is perfectly fine. Any one can provide guitar amplifier stores will have this fixed.
AGC fuse often are more packed up together to sell, so naturally you "forced" into buying spare parts!
Installing fuse holder
The fuse holder installed in one convenient place. You can set the fixed installation in the speakers on the back of the input panel recessed, facing the back of the grid, the internal loudspeaker cabinet, or the location of the amplifier output. Remember, if a fuse is burnt, you must be able to easily replace one, but sometimes you need to act quickly (for example, in a performance of the process).
Regardless of the wire is to be received on or direct sound received Horn, fuse holder must press series welding. As for the end is the cathode or anode wire that is not important. Also, the fuse and it's fixed for there is no polarity. If you cannot use the welding method, then you should choose with conductor connection bolts embedded fixer, of course, this is not the best way.
Fuse in protecting what
Protection of an inner divider speakers the best approach is to fuse the divider in front of current insurance tube should follow each numerical sums for the Horn. This requires that the speakers are turned on and access to each of the speakers on the construction of the wire. If you choose to fuse holder on the outside of the box, then you may need to extend the speaker wires. You can wire up the holes from the body of a received either from the input panel wear out to you want to place the fuse holder in place. You must be this small hole closed to avoid wire around the air flow and leak. In this case you'd better be able to use the embedded form factors of holder, so that you can avoid the problem of sealing body and makes the skin more beautiful with speakers. Usually when you want to use a fiber gasket used to seal the hole is completed.
Of course for a multi-channel speaker system you can also use a signal insurance, to avoid for each loudspeaker installation of insurance work, however, such protection would not be so precise. For multilayer amplification system, you must have for every loudspeaker installed fuse.
Insurance choices
Now the only remaining problem is to choose what type and how much rated amperage fuse. If you are in for the entire system installed fuse, you should use fast current fuse. If you are a single horn installed fuse, it should be the treble (compression) drive select fast current fuse, tapered speaker select slow current fuse. This is because the drive than treble cone loudspeaker for instantaneous excessive current have a certain tolerance.
You have to calculate and to determine the energy ratio. RMS will give more protection, but may also be very easy to burn. If the audio content will not be very full bass, then you can safely use the ratio of energy, which is usually twice the RMS ratio.
You can use the following formula to calculate a numerical rating of insurance:
Amps = squareroot (watts/Ohms) (power/impedance after square)
For example: If a speaker (or speaker) of power is the 300 Watt (RMS or long power), the nominal impedance of 8 Ohms. Then
Amps=squareroot(300/8)
Amps=squareroot(37.5)
Amps=6.1
This numerical rating of insurance should be identified as 6 amps. Once the value of more than 3 amps, then you will be very difficult to find in a 1/4 amperes for the change of insurance, so you should use this number of recent fuse: use more than this number of fuse relatively less insurance, while the use of numerical base than this fuse is more conservative. If you want to install the entire system from the beginning, choose use speaker's RMS ratio. When you find that your insurance will be very easy to burn and you're sure you do not have excessive driving your system, you can swap the values higher than 1/2-amp fuse.
If you have a parallel in the two speakers, for example, two bass Horn, and when you want to use for both of them with a fuse, then you have to be represented by a single horn calculated rated insurance value doubled. If two Horn-line,Then use by a single horn calculated rated insurance value.

【 Weak current College 】 audio technology law and effect-14--Power By 【 China power house network 】

1. the frequency domain of subjective feeling

Frequency domain most important subjective feeling tone, like loudness as tone is also a kind of subjective mental capacity of the hearing, it is the level of hearing judge voice tone.

Psychology of tone and music scales is the difference between the former is a pure tone, while the latter is the music that combined the pitch of the voice. Composite sound tones not merely frequency resolution, is also the role of the auditory nervous system, listen, listen and experience and learning.

2. the time domain of subjective feeling

If the length of the sound over approximately 300ms, increase or decrease the length of time that the sound on auditory threshold change has no effect. For the pitch of the sound of feeling and the length of time. When sound duration is very short, did not recognize your voice, just hear "a la". Sound duration lengthen, you can have the tone of the feeling, only sound continuously for more than 10 MS, feeling tone to stability.

Time domain characteristics of another subjective feeling is echo.

3. spatial domain of the subjective feeling

Human ear ears listening than using single-ear listening has obvious advantages, its high sensitivity, listen to the valve, the sound source with a sense of direction, and has a strong anti-interference ability. In stereo, with speaker and stereo headphone listen and get a sense of space is not the same, the former sound seems to be located in the surrounding environment, and the latter hear sound position at the head of the Interior, in order to distinguish these two kinds of space, the former is referred to as directed, the latter is called localization.

4. hearing of Weber's law

Weber's law indicates that the person hearing the sound of the subjective experience and objective stimulation amount proportional to the logarithm of a. When the voice is small, increasing Sonic amplitude, the humanear's subjective experience increased a lot larger volume; when the sound intensity is greater, increasing the same Sonic amplitude, ear subjective experience significant increases of the volume is small.

According to the human ear in the listening, the volume control on the design circuit require exponential type potentiometers as volume control, so that uniform Rotary potentiometers turn handle, the volume is linear increase.

5. hearing of Ohm's law

Famous scientist ohm electrical was discovered, at the same time in the Ohm's law, he also discovered the human auditory Ohm's law, on the law of disclosure: ear hearing only the sound of all sounds in frequency and intensity, and the distinction between independent sound phase. According to this law, the sound system to record, replay, process control can not go to consider complex sounds in the min audio phase relationship.

Ear is a frequency Analyzer, you can convert the homophonic in tremolo, ear to frequency resolution sensitivity is high, at this point the ear than eyes high resolution, the human eye cannot see the white colour of light weight.

6. masking

Environment in other sound makes listening to a voice of listening, referred to as cloaking. When a sound intensity far louder than the other, when a certain level and that two sounds at the same time, people can only hear the sound of the voice and unnoticeable another sound exists. Masking and masking the sound of the sound pressure, masking the sound of the sound pressure level increases, the amount of increase as cloaking. In addition, the low-frequency sound masking range is greater than the range of high frequency sound masking.

Ear of the auditory characteristics to design reduce noise circuits provide important insights. Tape playback, so the listening experience, when the music and sounds in continuous change is large, we will not hear the noise floor of tape, when the music ends (blank section of the tape), it can feel to tape the "' ..." noise exists.

In order to reduce the noise impact on program sound, signal to noise ratio (SN) concept, which requires that the signal strength than the noise intensity enough big, so listen would not have felt the presence of noise. Some of the noise reduction system is the use of masking effect of the principle of design.

7. dual ear effect

Binaural effect principle is this: If sound comes from listening to the front, as the sound source to the left and right ear, so the same distance from the left and right ear sound waves reach the time difference (phase), tone difference is zero, you feel that the sound comes from listening to the front, and not biased towards one side. Sound intensity, you can feel the noise source and listen to.

8. Haas effect

Test proves: Harris in two sound sources at the same time, the sound, a sound source with another sound source of delay quantity, binaural listening experience is different and can be divided into the following three scenarios to illustrate:
(1) two sound sources in a sound source with another sound source of delay amount within 5 ~ 35mS as if the two sound sources into one, listen and can only feel ahead of the existence of a sound source and direction, not another sound source.
(2) If a sound source delay another sound source 30 ~ 50mS, already feel two acoustic source exist, but the direction is set by the leading.
(3) If an amount greater than the sound source delay another sound source is 50mS, you can feel the two sound sources at the same time, direction of individual sound sources to determine the latency is clear and echo.
Haas effect is stereo system one directional basis.

9. de? wave El effect

De? wave El effect is stereo system directed another Foundation. De? wave El effect is: placed left and right channel two speakers, listen to two speaker symmetric line listen and give two speakers-infeed different signal,you can get the following findings:
(1) If you give two speakers into the same signal, that is, the intensity levelPoor Δ L = 0, t = 0 Δ time difference, you only feel a sound, and from two speakers on the line of symmetry.
(2) if the only speaker of intensity differentials Δ L is not 0, you listen and feel the sound preference more loud a sound box, if the intensity differentials Δ L is greater than or equal to 15dB, feel the sound entirely from a sound that only speakers.
(3) if the intensity differentials Δ L = 0, but the time difference of the two speakers Δ t is not 0, you feel the sound to be reached by the speaker orientation. If the time difference is greater than or equal to Δ t 3ms, feel sound entirely from first arrived only speaker orientation.

10. Lloyd's effect

Lloyd's effect is a stereo range of psycho-acoustic effects. Lloyd's effect revealed: If the delay of signal then RP superimposed on direct-to-signal, will produce a clear sense of space, the sound seems to come from all directions, listen to it in the band.

11. Keyhole effect

Mono recording system using a microphone recording signals in a track, all the way to the amplifier is used for playback and a speaker, the sound source is a point source, like listening to through the door of the keyhole to listen to the Interior of the Symphony, this is the so-called Keyhole effect.

12. bathroom effect

Confront the bathroom there is a personal feel and the sound in the bathroom, the reverberation time is too long and too much, this phenomenon in the electroacoustic technology sound effect called bathroom in the description. When a low, intermediate frequency, resonance, exaggeration of the frequency response is flat, 300Hz lifting excess, bathroom effect.

13. the Doppler effect

Doppler effect reveals mobile sound for listening features: when the sound source and listen to the relative motion between, will feel a frequency determined by thesound of his voice changed when the sound source to listen who approached is frequency at a slightly higher pitched, when the sound source left is slightly lower tone frequency. The frequency of change called Doppler frequency shift. Closer to the sound source in listening to the same distance than does not move when the intensity is high, and remove the source of strength to smaller, often sound source to the direction of movement.

14. likai test

Likai test proves: two sound sources in phase contrast, sound like you can out of two sound sources, or even jump to listen.

Likai test also prompts, as long as proper control of two sound sources (left and right channel speakers) intensity, phase, you can access a wide range (rotation, depth) of the mobile field.

【 Weak current College 】 car audio noise removal method

When using a car audio or replace the audio, the audio will produce more noise, noise can affect the sound quality, the lower the audio indicator, or even the entire sound system cannot listen for noise issues, the introduction of noise sources and basic troubleshooting.
First, the source of noise
Noise from the following channels into car audio system.
1, invade the power cord (through the host and the amplifier power cord into the system).
2, through the Earth's current (via xxx ground and power amplifier of the ground).
3. subject to other wire harness induction (via xxx receive and original car harness induction).
4. electrical interference (electric Horn, motors, etc.).
1, 2, 3, there is a correlation.
2. meet noise countermeasures
Generally deal with the noise-to-use components, automotive capacitor or coaxial motor capacitors (particularly effective to deal with high-frequency noise), Earl flow ring (ICP), LC filters, grounding, diodes, etc.
1, against ignition system noise
Check ignition coil cathode on the capacitor is installed, such as installation check capacity is reduced, if size reduces Platinum-contact easily spark erosion, interference, you need to replace the electrical capacity of 0, 5UF/400V promise of capacitance.
Check ignition voltage is using carbon fine line, if you use wire-prone to interference, which is a radio interference, so the parts must be replaced.
You can increase the damping resistance, noise suppression sparks by 1 megohm resistance concatenated output main high-voltage ignition coils, reduce interference.
2, the motor noise removal
First audio equipment and signal cable from the motor and motor line, available 1 only promise of capacitor in parallel at both ends of the motor, you can also only with 2-line in motor inductor respectively and cathode ray, then 2-polarity motor Capacitor respectively in the positive and negative cables, on the other side of the Earth form filtercircuit that spark of absorbing carbon brushes so that the noise reduction.
3, no relays electric Horn noise, exclude method has the following main categories:
In one of the speaker terminals on the ground and then a capacitor.
-In one of the Horn of a Terminal first-line to ground inductance, and received a capacitor.
In the two speaker terminals on the use of method-respectively.
In the steering wheel of the loudspeaker button-contact a capacitor between parallel.
4, the relay electric Horn noise, exclude method has the following main categories:
Electric Horn bracket and body should contact good.
-In relay contacts at both ends, a capacitor in parallel, or in contact with both ends of a capacitor in parallel.
? Use 3,? and ˉ method.
5 grounding bad will produce noise.
If the hood failed to secure the entire earth, bonnet becomes a xxx, the car noise radiation produced by ministries to the surrounding space and from xxx and the circuit into the audio system. Bonnet and body equipped with a connector, required to land on paint, grease, dirt, etc. are completely eliminated.
Engines (engines) and the body, or between front wheel suspension, body, between exhaust pipe and the body should have a good connection.
6. power cable noise suppression:
To effectively eliminate power line noise, should put Earl flow ring * near power amplifier installed. If there is a lot of power amplifiers, each power amplifier is installed near a stream circle because Earl noise from a power amplifier to another, so that a single failure of Earl flow ring.
7, to the signal cable (RCA) incoming noise suppression:
Signal cable should be located away from power cables and power cables are installed on both sides of the body, so that you can avoid the possibility of interference. The signal cable is double shielded network, you should put the outer shielding network attached to the chassis.
In practice also encounter many problems, but the important thing is to find the source of interference, then the remedy to solve it, so that the car audio system for playback of each program are clear and pure perfection.
Noise is the car audio fault because most and more difficult to troubleshoot because the vehicle's structure and lines of more complex, easy to place a lot of noise, so you should have some auto circuit and car audio knowledge, troubleshooting can be handy.
Solution: first of all, according to the noise analysis and judgement by principle identify failure is part of, and then take different exclusion method.
Simple exclusion approach, first listen to the audio of the radio, tape and CD part are the noise, if you have to check the power cord, Earth wire, signal line, the alignment of the parts is reasonable, the signal line whether with or without loosing ground breaking the skin, the power amplifier should be insulated with the body. In response to different noise sources, take different measures for processing, if the noise cannot be excluded, is to generate noise signal source disposed before the measures taken against noise, so that we can resolve the noise.

【 Weak current College 】 how to properly use VHF and UHF wireless microphone

Wireless microphone because there is no transfer cable, easy to use, flexible and widely used. But if improper selection, often occur with intermittent, sound choppy signal, noise, sound quality is poor and not far from the transmission distance, produce, such as the main reason is not based on field and use the correct selection of wireless radiation of wavelengths and the correct method of installation.
The widespread adoption of wireless microphones VHF (very high frequency) and UHF (ultra-high frequency) two transmission frequency band. VHF-band range 30MHz?? 300MHz, wavelength for 10m?? 1m, usually called "metric". UHF-band frequency range is 300MHz?? 3000MHz, wavelength for 1m?? 0.1m, commonly referred to as "decimeter". Both radio frequency is widely used for TV, FM radio, mobile phones, pagers, stock information machines, microwave communications and radar, etc. Channel extremely crowded.
A variety of radio wave propagation in the space of freedom, time and geographical constraints, frequency overlapping intersections, without constraints and regulations inevitably produce interactive, the world's radio waves using a uniform requirements, so that the interaction between them is minimal. Wireless microphone allowing use of the frequency range of requirements:
VHF frequency band for 169MHz?? Total occupancy 230MHz 61MHz frequency range, and 6-12 TV channels of the same frequency range. In 61MHz frequency range is divided into A, B, C three segments, namely: (A) for the VHF 169MHz?? 185MHz, VHF (B) for 185MHz?? 200MHz, VHF (C) to 200MHz.
UHF frequency band to 690MHz?? Total occupancy 960MHz 270MHz frequency range, and TV 35?? 68 channels can be set to the same extent as the hundreds of wireless microphone channels. If necessary, the frequency range you can also scale up, set more wireless radio channels.
VHF and UHF two band characteristics of radio wave propagation
VHF and UHF radio frequency signals of collectively (RF), but the UHF bandmaking greater use of direct radiation of electromagnetic waves, VHF band except the use of direct radiation of electromagnetic waves, the used part and diffraction of electromagnetic energy, therefore in the same transmitter power and transmission, the transmission distance can be further. The reason for this is:
(1) metal objects on electromagnetic wave propagation and reflection of the barrier
Metal objects on electromagnetic wave has a barrier and reflection effects. Barrier effect of electromagnetic waves emitted wavelength and the size of the metal objects. Electromagnetic waves of a wavelength less than the size of the metal objects, you will be subject to reflection, electromagnetic wave propagation. That is the higher the frequency, metal objects, the stronger the reflection of electromagnetic waves. If electromagnetic waves of a wavelength greater than the size of the metal objects, electromagnetic wave will bypass the metal barriers continue to spread (diffraction). Obviously UHF VHF of reflectance reflex more stronger.
(2) electromagnetic wave on the metal grid (or perforated metal plate) penetration capacity
Electromagnetic waves of a wavelength greater than metal hole distance between the grid, the diffraction of electromagnetic waves will pass, the lower the frequency, the longer wavelength longer wavelength, through the more metal grid.
(3) non-metallic body (such as the human body and walls) on the role of electromagnetic wave absorption
The higher the frequency of the electromagnetic wave, non-metallic body, the greater its absorption, electromagnetic wave propagation loss.
(4) the air temperature on the influence of electromagnetic wave propagation
The higher the frequency, the greater the electromagnetic wave propagation loss. Propagation loss is also proportional to the size of the humidity.
Wireless microphone signal quality and transmission distance
Wireless microphone signal quality directly affects the transmission distance. Signal quality (S/N) and transmit power and space field strength distribution, receivers and receiving antennas for quality, efficiency and outside interference sources is closely related to the situation.
(1) to solve the electromagnetic field intensity fluctuation of measures
Because indoor objects reflection and absorption of electromagnetic wave propagation in multiple ways, so that the space of the electromagnetic wave field strength promulgated very complex, some parts of the field strength is weak, some parts of the rendering area of the "shadow", or even become a corner. When the transmitter and receiving antenna distances frequently change, the undulation of the field is changed. When receiving antenna in a weak field locations, the "mute sound" points or poor sound quality. To do this, the wireless microphone receiver with dual antenna diversity reception technique (Diversity) and AGC AGC technology to give up. Diversity reception technique is set in a receiver with two sets of identical reception antennas and two receivers and one through built-in computer chip circuit automatically detects and automatically choose a stronger way to receive the signal as the received signal, in collaboration with AGC control, you can effectively resolve the "mute sound" point.
Unfortunately the market a lot of cheap fake double antenna wireless microphone receiver, these counterfeit products in addition to the two receive antenna, there are no diversity reception device, so you cannot solve the problem of the "mute sound".
(2) wireless microphone transmitter power
Increase transmit power to increase the transmission range for wireless microphones. But wireless microphone transmitter power from radio wave effects on the human body, power supply voltage of the battery life and interference to nearby devices, and other factors, market products are limited to transmitting power 2mW?? 50mW。 Due to technical reasons, of VHF frequency band of transmitting power than UHF transmitting power easier to do it, the price is cheaper.
(3), receive antenna
, Receive antenna type, length, and match status directly affect the antenna gain and efficiency. Efficient, receive antennas are 1/2 wavelength omnidirectional radiation dipole antenna (half-wave dipole antenna). If the length of the antenna and the wavelength is disproportionate, will directly affect the transmission distance. UHF of short wavelength, so the UHF antenna VHF antenna with than a shorter length, transmitter antenna easier to install.
(4) interference signal
Interference signal increases wirelessMicrophone output will make noise, serious wireless microphone does a for. Interference signals from industry interference, TV and radio signal interference and wireless microphone between adjacent channel interference and other radio frequency interference, etc.
Industry interference: including automobile ignition systems, radio and cleaners, glow discharge lamps (fluorescent lights, neon lights, high pressure mercury lamp). The majority of their interference in the VHF frequency band of spectrum.
TV, FM radio, information and mobile communications platform, transmit power, frequency range, VHF band interference is greater than on the UHF band interference. The solution is to first find the interference source frequency range, and then select use avoid these frequency wireless microphone channel.
More wireless microphone between adjacent channel interference: in many wireless microphone use, often encounter adjacent channel at the same time, if one channel of harmonics fall into another channel occurs adjacent channel interference, the solution is to choose to use a frequency interval of channels. VHF band can choose the number of channels, the UHF band can choose many intervals larger channel.
How to choose?
In the above two band wireless microphone performance compare, VHF-band wireless microphone advantages of 1?? 8 point disadvantage is: 9?? UHF-band 16 points; wireless microphone has the following advantages: 9?? 16 points, the drawback is: 1?? 8 point. Selection should be based on the use of occasions and their characteristics, the "strength", give full play to their greatest strengths.
In General, suitable for VHF hotels, Convention Center, Sports Hall (Hall), function rooms and teaching system suitable for radio and television; UHF, theatre performances and demanding, many group channel at the same time work wireless microphone system. But regardless of which band, choose dual antenna diversity reception technique of high-performance receiver will have better results and more clear sound. If you need to increase the transmission distance, the receiving antenna mounted to close off the transmitter, and (or) use the antenna amplifiers, not blind pursuit of increased transmitter power.

【 Weak current College 】 what is compression limiter?

Compression limiter is a compressor and limiter. It is an audio signal processing devices, you can connect the audio signal of dynamic compression or restrictions. Compressor for variable gain amplifier, its magnification (gain) can be input signal intensity and automatically change is inversely. When the input signal reaches a certain level (threshold or threshold), the output signal with the input signal increases, a condition known as compression (Compressor); no longer increase is called limit (Limiter). Past press adopts hard abduction (Hard-knee) technology, the input signal is a threshold value. Gain on the reduction, which immediately appears on the signal in the corner (gain change turning points) dynamic mutations in the human ear clearly feel a sudden strong signal is compressed. In order to address this deficiency, the modern novel from the compression device incorporates a soft corner (soft-knee) technology, the compressor in the compresses better than before and after the threshold is a balance, a gradient, change the compression, the sound quality is difficult to detect.
The compression used in the recording process to make musical instruments and singing to maintain a certain balance volume; ensure a balanced variety of signal strength. Sometimes also used to eliminate the singers ' articulate sounds, or by changing the compression and release time, produce sound by the small size of the "reverse sound" special effects.
In the broadcasting system is used to compress a large dynamic range of signal modulation in the prevention of distortion and prevent transmitter overload, increase average emission level.
In sound reinforcement system and dance hall, the compressor is the signal through the compressed while maintaining the original program appearance, reduce the dynamics of the music, to satisfy the amplification system and artistic activities.
Although compressor for a variety of purposes, the modern compressor ordinary adopts new technologies such as soft corner, can further reduce the compressor for the side effects of compressor, but does not mean that the compressor is the destructive effect of the sound quality is lost. Therefore, in sound reinforcement systems, not to abuse the compression, even if you want to use and should be used sparingly to reduce compressor for signal processing. This is not only protect the power amplifier, speakers, but also to improve the sound quality.
Compressor is also able to use the following process work:
(1), in the auditoria, as well as radio broadcasting, television broadcasting and recording etched, which limit the dynamic range of the signal, to adapt to the specific technical requirements. Limiter is sometimes also played the role of security protection, avoid transmitter and sound reinforcement system (especially the treble speakers) overload and even damage.
(2), when a player or singer suddenly change their distance between the microphone and makes the volume of the major changes occurred, use the compressor allows the volume changes smoothly.
(3), use the compressor can make the electric guitar, electric bass and other instruments volume stable. Often electric guitar bass chord volume than treble strings, compressor enables different range of volume level is balanced.
(4), manufacture of special sound effects, such as on a plucked instruments, carefully adjust the start time, recovery time, compression, threshold and compression ratio, in particular the recovery time is much shorter transfer, you can get a similar to the sound of the accordion.
(5), the limiter before joining area pass filter (can also use the equalizer) into the limiter of signals for the human voice in overweight wanna be tone and musical instruments of high frequency noise, which trigger the limiter can only due to the microphone or improperly tuned balance resulting from excessive hissing restrictive effect.






What is a DSP
DSP is a fast and powerful microprocessors, unique in that it can process data, it is immediately the instant capacity makes it the most suitable support DSP cannot tolerate any delay of the application. For example, you ever used a does not allow the handset to talk by both? you must wait until the other finish, you can then say; if you have two people talking at the same time, the signal will be cut off so that you do not hear each other's voices. Today's digital mobile phone allows you to talk to the normal way, because it uses a DSP.
Mobile phones within the DSP to ultra high speed processing of voice, so you can instantly hear each other, completely do not feel any delay. Then in the same application, for example, early mobile phones often appears echo, but digital mobile phone was able to answer and call a halt to completely eliminate the phenomenon. DSP will sound like the real-world signals, through mathematical operations to change its characteristics, in order to get better sound quality; DSP can compress data (your voice), elimination of background noise, make your voice more high-speed transmission, which in turn provides a clear and strong call quality, no annoying echo.
This is the most simple DSP applications. To improve the signal, you need a digital signal, and then process it, the result might be more clear sound and sharper picture or more quickly; and this signal capacity also brings breakthrough new applications such as Internet music and home broadband are therefore to be achieved.



What is an active speaker
Typically, the active sound box is equipped with a speaker internal amplifier mixes a class speakers. The power amplifier is designed to promote the sound of the Horn, as a specialized matching design, these amplifiers can be better used to promote the sound of the Horn, so that users do not need to take into account the amplifier's power is the impedance mismatch. In addition, because the speaker also equipped with the front of the amplifier will be frequency electronic dividers and each amplifier is only responsible for zoom a frequency audio signals amplifier efficiency often could be higher, distortion or relative can be as small as possible.
At present, except for home theater front main speakers, but also has specialized as a subwoofer, and surround speakers in the active speaker. As far as it used to VCD home theater with the active loudspeaker, recently saw some on the market, such as those who become the high-fidelity 3D active speaker is in theseSpeakers also with 3D surround sound decoder (processor). You only need to configure two such active speakers that can use dual Mono signal phase, delay and associated processing, only the original performance or stereo two channel signal into not just left, right, and can also have front and back of the other three dimensional sound. VCD's greatest advantage is the hardware and software is cheap, but to VCD for program source of "home cinema" device always is a popular type of cheap-home theater. With the previous reset, reset, and surround speakers, and so is not cost-effective, isn't necessary. If one of these active speakers, will be "temporary" home theater is used for a while, but do see it two or three years. Of course, a few years later, to buy genuine home theater, and the complete set of non-equipped DVD home theatre speaker system is not used.