Installation and debugging of professional sound system

The debugging of audio engineering is a job that requires both technical experience and carefulness. Debugging is the only means to make the audio system meet reasonable design requirements. If the debugging is not careful, not only can the design effect of the project not be achieved, but the equipment may also work in an abnormal state. Therefore, before debugging, we must fully realize the importance of this work. Before debugging,
Fae W!C we must carefully confirm whether each device is installed and connected correctly, and carefully ask the construction personnel about the relevant problems left over from the construction; before debugging, we must carefully read all the equipment manuals, and carefully check the markings and connection methods of the design drawings; before debugging, we must make sure that there are no problems with the power supply line and power supply voltage; and prepare the corresponding instruments and tools. h+w'L Q pq h'K1
. System power-on
'F2s0u:c6I n After the installation of the audio system is completed, it can be powered on and debugged. System power-on is to power on each device to verify whether each unit is intact, whether the connection is correct, and whether the system can make sound. Only on this basis can detailed adjustments and debugging be carried out. Although system power-on is not complicated, some problems in the project must be verified in this work. System power-on is the first step to ensure the quality of the project. The instruments and tools that need to be prepared are: phase meter, noise generator, spectrum analyzer (including sound pressure level meter), multimeter, etc.
l'f F ['?5l+R 1. Inspection before power on Inspection
before power on is very important. If there are serious problems with the equipment or lines and they are not discovered early, blindly turning on the power will cause a wider range of system failures and damage. Before powering on, you must make full preparations, carefully check the quality of the pipeline project and conduct a preliminary inspection of each single piece of equipment. Only when it is confirmed that there is no short circuit fault can the system be powered on.
1Z2U b)w!\9d+Q (1) Inspection of pipeline project quality
1A2J t Z'] h7} T+G B'P The pipeline project of the audio system should be constructed and installed in accordance with the building electrical specifications and accepted according to this standard. Before the system is powered on, it must be carefully inspected to prevent problems with the pipeline project from harming the valuable audio equipment. Here, we would like to emphasize a few key issues: ① Modern audio equipment is powered by single-phase AC
. After the pipeline project is completed, you should check whether the power supply voltage of the power distribution board that supplies power to the audio equipment is 220V. If the wiring is wrong, if two phase lines are connected to a single-phase power socket, there will be a 38OV voltage, which will burn the machine. ②
Check whether the signal line input to the mixer is short-circuited with the power line. If high voltage is mistakenly sent to the input end of the mixer, the mixer will burn. ③ The output end of the power amplifier
must not be short-circuited, so you should focus on checking the speaker feeder, plug, and socket to ensure that there is no short circuit. You can unplug the speaker first, and use a multimeter to measure the resistance of the speaker wire at both ends at the end of the sound control room. At this time, it should be an open circuit. Then connect the speaker plug and measure its resistance at the end of the sound control room. This resistance is generally about 1.1 times the impedance of the speaker. If the resistance of the speaker wire is considered, its resistance will be larger. Plug short circuit is the most common malignant accident and should be paid attention to. "d$N k"z @ ry(UF ^1\
(2) Equipment inspection
J)r\ q${.Y B"d There are many devices in the audio system. If a single device fails, it will often cause damage to a large area of ​​equipment. For example, if the power amplifier is damaged, there may be a high DC voltage at the output end, which will cause damage to the speaker system. Although professional audio equipment and equipment are strictly inspected before leaving the factory, these equipment often have to go through long-distance transportation, and sometimes they have to be transferred several times before they finally reach the hands of the user. During the loading and unloading process, collisions are sometimes inevitable, causing damage to the equipment, and poor storage environment may cause the equipment to get damp. Therefore, before the system is powered on, each piece of equipment must be powered on and tested one by one.
&V1n1Y0| _/eq;t:N1P The above preliminary tests on individual equipment mainly include the following aspects: *d#NEH { u U!F n1w
① Check the power supply of the equipment. Check whether the power supply voltage of the equipment is consistent with the mains voltage of 220V, and whether the power supply is set to 220V. If the equipment does not have a 220V voltage range, you should consider purchasing a transformer. Connect a single device to the power supply to observe whether there is any abnormality. Measure the output voltage without adding an input signal. At this time, the output voltage should be basically zero, and there should be no DC level output. The extremely small output voltage that exists is the output noise.
nt-M/x4Z!C o2D u ② Turn on the machine separately. Start from the sound source and gradually check the signal transmission. Only when the signal is well transmitted in each device, the power amplifier and the speaker will receive the correctly processed signal and have good sound quality. When doing this step, the speaker and the power amplifier should not be znk S p!F {
To connect, the peripheral processing equipment should also be placed in the branch state. When checking, follow the direction of the signal, and gradually check its level setting, gain, phase and smoothness to ensure that each device can get the best signal provided by the previous device and provide the best signal for the next stage. While checking the signal, you should also observe whether the equipment is working normally and stably one by one. The significance of this work is that if a single device fails or is unstable at this time, it is easier to handle and will not endanger the safety of other devices. Therefore, this inspection should not be carried out in the next step. A single device should pass the above inspections before being connected to the system.

2. Power on the system -
Based on the above inspection, it will be safe to power on the system. First, correctly connect the input and output cables of each device, lower the gain control of each level of equipment, and adjust the volume to the minimum. Then connect the power of each device from the front stage to the back stage. After the above is correct, connect the speakers and amplifiers to the system one by one. At a low volume, use a phase meter to first check whether the phase of all speakers is consistent, so as to prepare for the following debugging. And adjust according to the following steps until the program sound is heard in the speakers, and the system is turned on. 8J0U }3p$l7_#K:g
(1) Select a CD with a small dynamic range, play it with the corresponding signal source device, push the main fader on the mixer to 0, and push the sub-fader of the corresponding input channel to 0. On a standard mixer, the 0 position is around 70% of the stroke. At this time, the fader should be placed at a particularly obvious scale near the 70% stroke, and the input channel gain adjustment knob should be slowly turned up. Observe the VU meter reading and adjust it until the VU meter usually indicates below -6VU and the maximum reading does not exceed OVU. /pq p3j+M5k
(2) According to the order in which the signal flows through the equipment, adjust its working level and gain one by one. The general principle is to ensure that the sound signal processing equipment at all levels has zero gain, neither increasing nor attenuating the signal level. Unless the line level standards of the equipment in the system are inconsistent, it is generally necessary to control the input and output levels of the equipment to make a single device have a certain gain or attenuation to achieve the working level adaptation of each device in the system.
r3Y3u*i d+_ ?O p (3) The room equalizer is temporarily set to 0, neither increasing nor attenuating the frequency of each segment. ms B9s(T(r [
(4) Slowly turn up the power amplifier attenuator to gradually increase the volume. At this time, normal program sound should be heard from the speakers in the venue, the signal indicator of the power amplifier should flash, and the peak (clipping) indicator (Peak/clip) is only allowed to flash occasionally as a standard. } u*D k2x W2g1v [
II. Debugging of the sound system U f ] v g6U
After the system is powered on, further detailed adjustment and debugging are required. These debugging work generally requires the help of some special instruments and equipment to be well completed. Commonly used instruments and equipment mainly include: audio signal generator, millivoltmeter, noise generator, sound level meter, real-time spectrum analyzer; when reverberation needs to be measured, a level recorder is also required.
b1z y S u!| O,_ 1. Microphone phase
&~ o/Z#B)N O5Q'R e2N9y6d calibration Microphones used simultaneously in an audio system should generally be in phase. Before the project is delivered for use, the phases of all microphones in the system must be corrected to be in phase. When it is required to connect individual microphones in reverse phase due to special needs, the phase inversion switch on the tuning console can be used or a "phase inversion line" can be inserted. The method of checking the microphone phase is very simple. If two microphones are in phase, the volume will increase significantly when the two microphones point to the same sound source. If the two microphones are in reverse phase, the volume will decrease when the two microphones are used at the same time. When adjusting, you can choose any microphone as a benchmark and compare all microphones in the system with it. The microphones with the same phase are classified into one category, and the microphones with different phases are classified into another category. Adjust the phase of a smaller number of microphones, that is, swap the wiring of the 2nd and 3rd pins of the XLR plug to achieve phase adjustment.
"] m8V i"DB y2I0] W 2. Room equalizer adjustment8J ,]&r Q v&n3eRoom
equalizers generally require the help of pink noise generators and real-time spectrum analyzers to accurately adjust. Room equalizers are mainly used to correct and compensate for room frequency characteristics. Therefore, during debugging, the consistency between the hall environment and the actual listening environment should be ensured. In addition, the adjustment of the room equalizer sometimes needs to be combined with the adjustment of the speaker layout.
4N#l H#`+ua The room equalizer compensates for the frequency characteristics of the environment by changing the frequency characteristics of the signal. Changes in frequency characteristics will inevitably lead to changes in phase characteristics, causing phase distortion. When the adjustment amount of the room equalizer is too large, especially in a narrow frequency band, a large adjustment amount must be used to achieve the equalization effect. Although the frequency characteristics of the room are corrected, the listening experience will become very poor due to phase distortion, which will be more prominent for stereo systems. In the case of poor building acoustic conditions, the adjustment of the room equalizer can sometimes only compromise between frequency characteristics and listening experience. Forcing the flatness of frequency characteristics sometimes backfires. The best way is to improve the acoustic characteristics of the room itself. 4g ~ d MV o @9n F g
The principle of room equalizer adjustment is shown in Figure 8-8. 6L*v![ SR j Z |@"A0f n
Figure 8-8 Principle of room equalizer adjustment
C H5x c q6I6O/T (1) Debugging process
b ?*r*S d&@;a ① Use pink noise as the system input test signal. This noise is obtained by passing white noise through a -6dB/oct filter. Compared with white noise, pink noise has greater low-frequency energy. Because the energy distribution of pink noise is closer to that of real music signals, it is often used as a test signal for sound engineering and audio equipment. The power capacity of speakers is generally measured using pink noise. If no pink noise occurs, a CD with pink noise recorded on it can also be used to play pink noise. Generally, the frequency response of mid-range and above laser players can be 2OHz~20kHz +0.5dB, which can meet the test requirements. @ | gc J,W
② Input pink noise into the mixer and adjust the mixer to the standard output level, usually OVU, output level +4dB. It should be noted that the equalizer EQ on the mixer is adjusted to a flat line, that is, all at zero position, and the frequency of each segment of the test signal is neither increased nor attenuated. The frequency adjustment potentiometers of each point of the room equalizer are also temporarily set to zero position. Slowly increase the power amplifier volume adjuster to hear the pink signal sound, and use a sound pressure meter to monitor until the sound pressure level of the pink noise signal in the hall reaches about 85dB. 5N(D!E)FD `%C2l
③ Place its measurement microphone in the center of the hall and set the selector switch on the spectrum analyzer to the "OCT" position (this position is the octave filter position, corresponding to the characteristics of pink noise). At this time, the LED display on the real-time spectrum analyzer is the frequency characteristic curve of the listening environment. The flatter it is, the better the frequency characteristics of the room sound. vjr)r H0v O6K
④ Adjust the frequency boost/attenuator at each point on the equalizer to make the frequency characteristic curve on the spectrum analyzer a straight line. PF B+S a N
After the above debugging is completed, it is generally necessary to "smooth" the equalizer curve on the equalizer. This is mainly to prevent excessive phase distortion when the equalizer is adjusted to a sawtooth frequency characteristic. g s+r0I S7{
(2) Key points for adjusting the room equalizer
sY gR ? ] i'{ ① In the low frequency band of about 20 to 50 Hz and the high frequency band above 14kHz, the frequency characteristics do not need to be forced, especially in the low frequency band. Because it is difficult for ordinary speakers to extend to 20Hz, it is considered good to be able to reach 40Hz. Forcing the flatness of the low frequency band characteristics and enhancing the ultra-low frequency will cause the speaker to "lose control" due to excessive extension of the low frequency, and the distortion will be aggravated.
.yq J a!K } ② The adjustment of the room equalizer should always take into account the contradiction between the flat frequency characteristics and the minimization of phase distortion, and make a compromise.
-| H;cq:_ f1K c } _ ③ In the case of obvious "peaks" and "valleys" in the frequency characteristics of the building acoustic environment, it is necessary to consider changing the speaker position and trying to change the building acoustic characteristics. ④
The adjustment of the room equalizer is a very delicate work, and it needs to be adjusted repeatedly for many times
,gB'c |8G8d before it can be finally adjusted. This is because in the adjustment process, it is often necessary to make some adjustments to the speaker placement and building acoustic environment, and the equalizer will have mutual restraint during adjustment.
;HQ a T0 Objectively speaking, the role of the room equalizer is limited, and the defects of the building acoustic environment cannot be expected to be completely solved by the room equalizer. The smaller the equalization amount, the better the sound quality will be. In the absence of a pink noise generator and a real-time spectrum analyzer, the frequency signals of each point with the same amplitude can be sent to the system by an audio signal generator according to the frequency points on the selected room equalizer, and the sound pressure in the field can be tested with a sound pressure meter, and the room equalizer can be adjusted. Make the input signals of each frequency point produce the same sound pressure level in the field. The actual effect of this debugging method is worse than using standard pink noise. Therefore, professional units should configure pink noise generators and real-time spectrum analyzers as much as possible. 3. Debugging of electronic crossovers The debugging of electronic crossovers can be divided into high, medium and low frequencies for separate debugging. The purpose of the crossover in the system is different, and the debugging method is also different. If the crossover is only used for the crossover of the bass speaker, the bass crossover point of the crossover should be set between 150 and 300 Hz when the bass speaker works alone, and the gain of the bass clear signal should be adjusted appropriately. It is OK to feel that the bass volume is appropriate, and then listen to it together with the full-frequency system, and then balance the bass and full-frequency volume; if the crossover is used in the full-frequency system, it is required to accurately set the high, medium and low frequency crossover points according to the parameters provided by the speaker manufacturer, and then repeatedly adjust the signal gain of each frequency band until the listening experience of each frequency band is relatively balanced, and then refer to the sound pressure conditions tested by the spectrum analyzer at each test point for further fine-tuning. 4. Adjustment of the delay device As mentioned above, the purpose of using the delay device in the sound reinforcement system, in addition to producing some "special effects" of sound, is mainly used to prevent accentuation and echo and improve the clarity of the sound. The adjustment of the delay device used for this purpose should be based on the principle of eliminating the time difference between the direct sound radiated by different speakers and reaching the listener. However, in actual engineering applications, it is often not required to compensate this time difference to zero. First of all, it is difficult to do so, because if the time difference is zero at a certain point, there will inevitably be a time difference at the surrounding positions. Secondly, it will be unnatural to listen to the time difference of the direct sound radiated by different speakers completely compensated to zero. Because in the case of relying entirely on the natural sound of the building acoustic structure, the uniform distribution of the sound pressure level is mainly achieved by the enhancement of the direct sound by the near sub-reflected sound. At this time, the time difference between the near sub-reflected sound and the direct sound reaching the audience reflects the sense of space in the hall. Of course, the time difference between the near sub-reflected sound with strong energy and the direct sound cannot exceed 50ms indicated by the Hass effect, otherwise the clarity will be greatly affected. Proper adjustment can obtain a more realistic and natural sound effect. ]*D7P qhd:fv H 5. Adjustment of the compressor 2t k/UPB l o&U4vFor the adjustment of the compressor, it should be carried out after the above equipment of the system is basically adjusted. Generally, in the project, the role of the compressor is to protect the power amplifier and the speaker and make the sound change smoothly. Therefore, when debugging, the compression start level should be set first. Usually, it should not be set too low. The specific setting should depend on the adjustment range and signal conditions of various compressors; secondly, the compression start and recovery time should be set. Usually, the start time should not be too long to avoid untimely protection action; for the protection of equipment, a shorter start time will be more beneficial. In order to maintain a good sense of dynamics in the sense of hearing, the recovery time should not be too short to avoid damage to the sound effect. In general, the compression ratio is set at about 4:1 in the project. In general, the adjustment of these two parameters should be based on the specific situation of the program, and the natural sound should be heard without obvious changes in the sound. Pay special attention to the setting of the noise gate in the compressor. If the system does not have a lot of noise, the noise gate can be closed; if there is a certain amount of noise, the threshold level of the noise gate can be set lower to avoid the phenomenon of intermittent sound reinforcement signal; if the system noise is large, it should be analyzed from the construction technology aspect, and the noise gate cannot be used alone. Other settings can be determined according to different requirements. _.GF } H(e 6. Measurement of hall sound pressure level #k#\ G9l Q2j5FBased on the above debugging, the sound pressure level of the hall is measured by sound pressure meter test. A pink noise generator is used as the noise source, and several frequency points are selected in the high, medium and low frequency bands for testing. The test goal is: under the premise of ensuring the best dynamic of the signal, the sound pressure of the system's sound reinforcement should reach the designed sound pressure level at each point after adjustment. At the same time, the situation of each point in the high, medium and low frequency bands should be referred to, and the equalizer and electronic crossover should be slightly adjusted respectively. If the result of the pressure level of each test point is large, that is, the uniformity of the sound field is not good, it should be carefully analyzed and improved accordingly. First of all, we should start with the construction process of building decoration. If there are major defects in this aspect, which affects the quality of the sound field, feasible rectification measures should be proposed: if there are no obvious defects in the decoration, the speaker should be analyzed from the aspects of placement, direction and installation form. The content of the analysis includes: the distance between the speaker and the four sides of the building, the installation position requirements between the speakers, the direction and frequency characteristics of the speakers, etc. 3. Acoustic feedback suppression Acoustic feedback directly affects the sound quality of the sound reinforcement system. In serious cases, it will undermine the stability of the entire system. In actual engineering, necessary measures should be taken to suppress it. This section focuses on the method of suppressing acoustic feedback in conjunction with the debugging of the sound reinforcement system. 1. Formation of acoustic feedback (1) Generation and influence of acoustic feedback In the sound reinforcement system, in addition to the forward electrical transmission channel from microphone to amplifier system to speaker , there is also a feedback channel from the speaker directly feeding back to the microphone. The feedback sound is then sent back to the speaker through the microphone-amplifier system. This cycle is repeated, and the entire system will produce self-excitation, which will make the system unable to work normally. In outdoor sound reinforcement systems, acoustic feedback is mainly caused by the direct sound of the speaker. In indoor sound reinforcement systems, the factors that cause acoustic feedback include the direct sound of the speaker and the reflected sound from the various wall interfaces in the indoor sound field. In fact, the value of the acoustic feedback coefficient is related to the phase between the sound pressure of the feedback sound and the sound pressure of the sound source. Usually, the operating frequency range of the audio system is 100~8000Hz. Therefore, in the entire sound reinforcement system, the acoustic feedback will have positive feedback and negative feedback as the frequency changes. When positive feedback occurs, the system will produce self-excitation and cause howling. To make the entire system work stably , the method is to overcome positive feedback, provided that the acoustic feedback coefficient β＜＜1. In order to prevent self-excitation and reduce the impact on the sound quality of the sound reinforcement system, the acoustic feedback coefficient allowed in actual work is only 0.2~0.3.


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Bn b2Z2z y Y5G5b The impact of acoustic feedback on the sound reinforcement system is as follows: ① The coexistence of positive feedback and negative feedback directly destroys the frequency response of the system, produces distortion, and in severe cases affects the sound quality of the system; ② Under certain conditions, the self-excitation phenomenon causes howling, which destroys the stability of the system; ③ In the indoor sound field, the delay of acoustic feedback will prolong the reverberation time, produce regenerative reverberation interference, and affect the clarity of the language in the listening area. (2) Maximum power gain In the sound reinforcement system, when β≤1 and the feedback channel signal transmission is in phase with the forward transmission channel signal transmission, the system will be unstable and self-excitation will occur. Therefore, when β=1 is defined, the sound power output by the sound reinforcement system is called critical power, represented by WC. In actual engineering, in order to avoid frequency distortion, system self-excitation and regenerative reverberation interference caused by acoustic feedback, and to enable the system to work stably, β < < 1 should be satisfied, that is, the actual sound power output of the sound reinforcement system should be lower than the critical power. The sound power output by the speaker of the sound reinforcement system under actual use conditions is defined as the maximum sound power, expressed as WM. The maximum power gain can be used as a basis for calculating the power capacity of speakers and amplifiers when designing a sound reinforcement system . (3) Transmission gain In actual engineering, the degree of acoustic feedback can also be evaluated by transmission gain. The definition of transmission gain is: "When the sound reinforcement system reaches the highest available gain, the difference between the steady-state sound pressure level generated by the speakers at each audience seat and the steady-state sound pressure level generated by the sound source in the sound reinforcement system."



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2. Acoustic feedback suppression method
(|{+j)I ["~ @5^ T According to the characteristics of the sound field, the acoustic feedback suppression of the sound reinforcement system should start from system design, sound field layout, equipment selection and sound field adjustment. Every link should do a good job in preventing acoustic feedback. The specific methods are: ps xO1i5m e6l q
(1) Suppress the peak of acoustic feedback3 { O7p k6|#U |
Use equalization, frequency shifting, phase adjustment and other methods to suppress the peak of feedback sound energy, thereby ensuring the stability of the system.
1C ]:G u'o FLI r (2) Control the acoustic conditions of the hall ^-U j(VH qFor
indoor sound reinforcement systems, the longer the indoor reverberation time and the greater the reverberation sound energy, the more opportunities for acoustic feedback. Therefore, appropriately reducing the indoor reverberation time can effectively reduce acoustic feedback. In addition, the uniformity of the indoor sound field is good, which is conducive to improving the stability of the sound reinforcement system. The formulation of hall acoustic indicators and the control of acoustic conditions should be fully considered at the beginning of hall sound quality design.
'n T)J!O+N (3) Eliminate acoustic feedback channels
-b9V ih#y By using the directional characteristics of the microphone and the speaker, we can adjust the spatial position between them, handle the position relationship between the microphone and the speaker, and eliminate the acoustic feedback channel. The ideal state should be that the speaker is far away from the microphone, so that the direct sound emitted by the speaker cannot enter the microphone at all, and the acoustic feedback in the system can be handled very small. The actual working position of the speaker cannot be completely away from the microphone, especially for a speaker system with a centralized layout of the indoor sound field. Therefore, the position relationship between the speaker and the microphone should ensure that the sound supply of the speaker allows the entire audience to obtain sufficient sound energy, and minimize the impact of the speaker on the microphone. This requires analysis and processing based on the distance between the speaker and the microphone, the relevant position and the directional characteristics. ①
Eliminate the acoustic feedback channel through device selectionv IJ,R"L g!n |
a. Directional microphones can suppress the direct sound from other directions of the speaker, reduce the indoor reverberation sound energy, and improve the stability of the system. Therefore, in the sound reinforcement system, especially the indoor sound reinforcement system, its selection should be completely dominant. Usually, compared with the omnidirectional microphone, the cardioid directional microphone can improve the stability of the system by 5dB, and compared with the cardioid directional microphone, the supercardioid and super-directional microphones can further improve the system gain.
X([#Q T5W O5u/g V a f3a b. On the basis of satisfying the sound field coverage of the auditorium, the speaker directivity should use a device with narrow directivity.
#nn E.D+o%Y t a&X c. The sound column has a small signal in the low-frequency band, and its vertical directivity is narrow, which is easy to control the sound feedback. It is often used as the main sound device for language sound reinforcement.
-[9g'I"C:v,ED d. The constant directivity horn speaker has a variety of constant directivity angles to choose from, and there is no side lobe of the secondary sound beam outside the directional &C/v3s t Z n H
main sound beam, which is beneficial to reducing sound feedback. Therefore, it is applicable to sound reinforcement systems in various occasions.
.l VE \(v VW [.G ② Eliminate the sound feedback
[.e S ? {#zme channel If conditions permit, the distance L between the speaker and the microphone should be as large as possible. For indoor sound fields, the distance between the speaker and the microphone should be greater than the critical distance DC.
hl$r"r6Q6Q+A;_(e ^ DC = 0.057 (VQL/T60) 1/2
kfs(I7S K5g X px Where: V is the volume of the room; QL is the directivity factor of the speaker; T60 is the indoor reverberation time.
\eb;E9P h A2C Beyond the critical distance, the direct sound is smaller than the reflected sound, which can basically reduce the impact of the speaker's direct sound on the microphone, but it should be noted that the distance of the speaker's diffusion field in each direction is different. For halls with a long reverberation time and mainly used for language amplification, a dispersed layout is adopted, which is conducive to increasing the distance between the speaker and the microphone, blocking the sound feedback channel, strengthening the direct sound in the audience seats, and improving language clarity. z"b t2p0W$|:^
③ Related positions HXC qv(U z9t
a. The sound radiation "dead angle" of the speaker, that is, the direction of the weakest sound radiation, faces the direction of the worst sensitivity of the microphone, so that the effect of suppressing sound feedback is better. -dw Q gt F {
b. It is better to arrange the sound column above the microphone. On the one hand, it can expand the distance and range of sound radiation, and on the other hand, it can use the narrow vertical directivity of the sound column to suppress sound feedback. Figure 8-9 shows the two positional relationships between the sound column and the microphone. In Figure (a), the sound column is installed low and is in a horizontal state with the microphone. Due to the wide horizontal directivity of the sound column, it has a great impact on the microphone and is very likely to cause sound feedback. In Figure (b), the sound column is placed in front of the microphone, which is obviously beneficial to suppressing sound feedback. f Z%d"`:t'wFigure
8-9 Positional relationship between the sound column and the microphone2k { f!I)s T v G
(4) Use frequency equalization technology to suppress sound
;B oc L u+[ \ \ feedback For indoor sound field, the transmission response of the sound reinforcement system is not only related to the direct sound field of the speaker, but also to the physical characteristics of the hall itself. Sometimes, it is not completely effective to deal with the position of the speaker and the microphone alone. For example, in the low frequency band, the wavelength of the sound wave is longer, and the low-frequency radiation of the speaker is non-directional. For another example, when the phase of the acoustic feedback coefficient is in phase with the sound source, it is also very easy to produce self-excitation and howling. At this time, an equalizer can be inserted into the sound reinforcement system, and the frequency response equalization technology can be used to suppress the peak of the acoustic feedback transmission response and improve the transmission gain of the system. The use of frequency response equalization technology can not only suppress acoustic feedback, but also improve the sound quality of the hall, and improve the fullness, clarity and naturalness of the sound reinforcement. m e7e YZ
In order to control the irregularity of the room transmission response in the sound reinforcement system, the sound reinforcement system can be self-excited first, and each self-excitation frequency of the system in the hall that has a greater impact on the quality of language and music can be measured. Then, the equalizer inserted into the system can be accurately tuned to each self-excitation frequency, and sufficient damping can be added to ensure the required increase in gain without self-excitation. During the debugging process, new self-excited frequencies will be generated. At this time, the above process must be repeated until the system sound transmission gain reaches the required index.
0i vnr `/q H W1g ]C4q Usually, there are multiple self-excited frequencies in the room transmission response, and there is no regularity, which requires repeated and meticulous debugging. In recent years, real-time analyzers have gradually become popular. It is convenient to observe during adjustment, intuitive to debug, and easy to grasp the overall tone balance. After repeated adjustments, the sound transmission gain of the system can be improved. 6S;A _ i2\:CI
(5) Using the frequency shift method to reduce acoustic feedback
} ] C;L&}#p The basic idea of ​​using the frequency shift method to reduce acoustic feedback is to use the offset frequency method to destroy the in-phase condition between the feedback sound and the original signal and suppress the self-excited oscillation of the system. In the sound reinforcement system, insert a frequency shifter to make the output signal of the speaker offset an appropriate amount relative to all frequencies of the microphone signal. This method can effectively suppress acoustic feedback and reduce frequency distortion and regenerative reverberation interference. K*XI C3w5S wmWhen
the sound reinforcement system has no frequency shift, the system will generate self-excitation when the maximum value of the loop gain exceeds 0dB. Therefore, the maximum allowable value of the system gain depends on the maximum value of the transmission response, and the corresponding maximum value of the loop gain must be lower than 0dB, otherwise the system will be unstable. @9p&[ {0Y U a UQAfter
inserting the frequency shifter, the stability of the system no longer depends on the maximum value of the loop gain, but on the average gain of the transmission response. As long as the average gain is lower than 0dB, the system is stable. Therefore, the frequency shift method allows the sound reinforcement system to increase the gain by the difference between the maximum gain and the average gain on the frequency response. The optimal frequency shift is equal to the average distance between each peak and the adjacent valley on the transmission response, because at this time the excess energy generated by the gain peak will be quickly "absorbed" at the valley. Practice has shown that the optimal frequency shift is related to the reverberation time T60 of the hall, which is approximately 1/T60. Although a larger frequency shift can also increase the gain of the sound reinforcement system, when the frequency shift exceeds 7Hz, it will affect the sound quality. 5f q X;g!\ QTXSet
the operating frequency range of the sound reinforcement system to B. Then, when BT60＞5000, after the sound reinforcement system is inserted with a frequency shifter, the increment of its stable gain ΔG should be:
M%f` _ o*n7q on ΔG = 10Lg(Lg(BT60/2(2))+6.3There
)uj I9|7l.W k are several methods to realize frequency shifting. The single-sideband modulation method and the basic frequency band offset method have been made into frequency shifters. The
+w Y%P2v+N wjn single-sideband modulation method is to make the signal undergo modulation and demodulation. If the two carrier frequencies maintain a given frequency offset, the output signal can be moved on the frequency axis.
'H0K l(g4^ k4w J The basic frequency band offset method is to pass the signal through a phase-splitting network to give two signals with phases offset by 90 degrees relative to each other. Through the two orthogonal primary windings of the rotating transformer, the frequency shift is obtained by using the rotating output of the transformer. {;g%} i ~ L:u%QThe
sound reinforcement system with a frequency shifter has a transition process when the sound feedback is close to self-excitation. This phenomenon is particularly obvious for simple sound feedback systems. During the transition process, the sound reinforcement system is unstable, but it will not suddenly oscillate, which is beneficial to the use and maintenance of the sound reinforcement system. *a2`V8_ Z0W
(6) Using phase modulation to improve acoustic feedback
c4u4x"d `:HQ yx The main factors causing acoustic feedback are the acoustic feedback coefficient and its phase. If the parameters of the sound reinforcement system remain unchanged, then the phase is the only condition that determines whether the system will produce self-excitation. Inserting a phase modulation device or frequency modulation device that continuously changes according to a simple periodic function into the sound reinforcement system causes the phase of the feedback signal to deviate from the phase of the main channel signal, destroying the system self-excitation condition and improving the stability of the system. !Xt$O2w ^6? dThe
phase rotator of the phase modulation device consists of a resistor-capacitor bridge circuit. Changing one of the parameters can make the phase of the output voltage change evenly. When the variable resistance value is changed from 0 to ∞, the phase change between the output voltage and the input voltage changes from 0° to 180°. In the sound reinforcement system with the phase rotator inserted, the phase deviation value has the greatest stability within the range of 140°. At this time, the gain of the sound reinforcement system is allowed to increase by 7 to 8dB. In addition, the frequency of phase change has a great influence on the effectiveness of the phase rotator. When the audience does not notice the interference, the maximum allowable value of the phase change frequency is 1 to 4.5Hz. &]&g"n hw#G,[ j|
The frequency modulation device is an electronic phase modulator. When the phase modulation index is 1.4-2.4 and the phase modulation frequency is 1Hz, the system gain of the sound reinforcement system can be increased by 4dB. The characteristic of using this phase modulator to suppress acoustic feedback is that the perceptible distortion is very small, which is especially suitable for music sound reinforcement systems.