The performance effect of the sound system is jointly determined by the sound source equipment and the subsequent stage sound reinforcement, which consists of sound source, tuning, peripheral equipment, sound reinforcement and connection equipment.

1. Sound source system

The microphone is the first link of the entire sound reinforcement system or recording system, and its quality directly affects the quality of the entire system. Microphones are divided into two categories: wired and wireless according to the form of signal transmission.

Wireless microphones are particularly suitable for picking up mobile sound sources. In order to facilitate the sound pickup of various occasions, each wireless microphone system can be equipped with a handheld microphone and a Lavalier microphone. Since the studio has a sound reinforcement system at the same time, in order to avoid acoustic feedback, the wireless handheld microphone should use a cardioid unidirectional close-talking microphone for the pickup of speech and singing. At the same time, the wireless microphone system should adopt diversity receiving technology, which can not only improve the stability of the received signal, but also help eliminate the dead angle and blind zone of the received signal.

The wired microphone has a multi-function, multi-occasion, multi-grade microphone configuration. For the pickup of language or singing content, cardioid condenser microphones are generally used, and wearable electret microphones can also be used in areas with relatively fixed sound sources; microphone-type super-directional condenser microphones can be used to pick up environmental effects; percussion instruments are generally used Low-sensitivity moving coil microphones; high-end condenser microphones for strings, keyboards and other musical instruments; high-directivity close-talk microphones can be used when environmental noise requirements are high; single-point gooseneck condenser microphones should be used considering the flexibility of large theater actors .

The number and type of microphones can be selected according to the actual needs of the site.

2. Tuning system

The main part of the tuning system is the mixer, which can amplify, attenuate, and dynamically adjust the input sound source signals of different levels and impedance; use the attached equalizer to process each frequency band of the signal; After adjusting the mixing ratio of each channel signal, each channel is allocated and sent to each receiving end; control the live sound reinforcement signal and recording signal.

There are a few things to pay attention to when using the mixer. First, choose input components with greater input port bearing capacity and wide frequency response as much as possible. You can choose either microphone input or line input. Each input has a continuous level control button and a 48V phantom power switch. . In this way, the input part of each channel can optimize the input signal level before processing. Second, due to the problems of feedback feedback and stage return monitoring in sound reinforcement, The more equalization of input components, auxiliary outputs and group outputs, the better, and the control is convenient. Third, for the safety and reliability of the program, the mixer can be equipped with two main and standby power supplies, and can switch automatically.Adjust and control the phase of the sound signal), the input and output ports are preferably XLR sockets.

3. Peripheral equipment

On-site sound reinforcement must ensure a sufficiently large sound pressure level without generating acoustic feedback, so that the speakers and power amplifiers are protected. At the same time, in order to maintain the clarity of the sound, but also to make up for the shortcomings of the sound intensity, it is necessary to install audio processing equipment between the mixer and the power amplifier, such as equalizers, feedback suppressors, compressors, exciters, frequency dividers, Sound distributor.

Frequency equalizer and feedback suppressor are used to suppress sound feedback, make up for sound defects, and ensure sound clarity. The compressor is used to ensure that the power amplifier will not cause overload or distortion when encountering a large peak of the input signal, and can protect the power amplifier and speakers. The exciter is used to beautify the sound effect, that is, to improve the sound color, penetration, and stereo Sense, clarity and bass effect. The frequency divider is used to send the signals of different frequency bands to their corresponding power amplifiers, and the power amplifiers amplify the sound signals and output them to the speakers. If you want to produce a high-level artistic effect program, it is more appropriate to use a 3-segment electronic crossover in the design of the sound reinforcement system.

There are many problems in the installation of the audio system. The improper consideration of the connection position and sequence of the peripheral equipment results in the insufficient performance of the equipment, and even the equipment is burned. The connection of peripheral equipment generally requires order: the equalizer is located after the mixer; and the feedback suppressor should not be placed before the equalizer. If the feedback suppressor is placed in front of the equalizer, it is difficult to fully eliminate the acoustic feedback, which is not conducive to Feedback suppressor adjustment; the compressor should be placed after the equalizer and the feedback suppressor, because the main function of the compressor is to suppress excessive signals and protect the power amplifier and speakers; the exciter is connected in front of the power amplifier; The electronic crossover is connected before the power amplifier as needed.

To make the recorded program get the best results, the compressor parameters must be adjusted appropriately. Once the compressor enters the compressed state, it will have a destructive effect on the sound, so try to avoid the compressor in the compressed state for a long time. The basic principle of connecting the compressor in the main expansion channel is that the peripheral equipment behind him should not have the signal boost function as much as possible, otherwise the compressor cannot play a protective role at all. This is why the equalizer should be located before the feedback suppressor, and the compressor is located after the feedback suppressor.

The exciter uses human psychoacoustic phenomena to create high-frequency harmonic components according to the fundamental frequency of the sound. At the same time, the low-frequency expansion function can create rich low-frequency components and further improve the tone. Therefore, the sound signal produced by the exciter has a very wide frequency band. If the frequency band of the compressor is extremely wide, it is perfectly possible for the exciter to be connected before the compressor.

The electronic frequency divider is connected in front of the power amplifier as needed to compensate for the defects caused by the environment and the frequency response of different program sound sources; the biggest disadvantage is that the connection and debugging are troublesome and easy to cause accidents. At present, digital audio processors have appeared, which integrate the above functions, and can be intelligent, simple to operate, and superior in performance.

4. Sound reinforcement system

The sound reinforcement system should pay attention to that it must meet the sound power and sound field uniformity; the correct suspension of the live speakers can improve the clarity of sound reinforcement, reduce sound power loss and acoustic feedback; the total electric power of the sound reinforcement system should be reserved for 30%-50 % Of reserve power; use wireless monitoring headphones.

5. System connection

Impedance matching and level matching should be considered in the issue of device interconnection. Balance and unbalance are relative to the reference point. The resistance value(Impedance value) of both ends of the signal to the ground is equal, and the polarity is opposite, which is a balanced input or output. Since the interference signals received by the two balanced terminals have basically the same value and the same polarity, the interference signals can cancel each other out on the load of the balanced transmission. Therefore, the balanced circuit has better common-mode suppression and anti-interference ability. Most professional audio equipment adopts balanced interconnection.

The speaker connection should use multiple sets of short speaker cables to reduce line resistance. Because the line resistance and the output resistance of the power amplifier will affect the low frequency Q value of the speaker system, the transient characteristics of the low frequency will be worse, and the transmission line will produce distortion during the transmission of audio signals. Due to the distributed capacitance and distributed inductance of the transmission line, both have certain frequency characteristics. Since the signal is composed of many frequency components, when a group of audio signals composed of many frequency components passes through the transmission line, the delay and attenuation caused by different frequency components are different, resulting in so-called amplitude distortion and phase distortion. Generally speaking, distortion always exists. According to the theoretical condition of the transmission line, the lossless condition of R=G=0 will not cause distortion, and absolute losslessness is also impossible. In the case of limited loss, the condition for signal transmission without distortion is L/R=C/G, and the actual uniform transmission line is always L/R<C/G. In the audio engineering, since the path is not too long, the resistance R is reduced by increasing the line length, so it is better to use a thicker audio line. The microphone cable and some external audio signal lines are relatively long. Because the audio signal is relatively weak, the transmission line used must be shielded in a balanced manner to reduce interference. The main contradiction of the connection cable from the final amplifier to the speaker is the large current. Therefore, the speaker cable should be of good quality, thicker, and the linear path design should be as short as possible, so that it is possible to create a good sound field.

6. System debugging

Before adjustment, first set the system level curve so that the signal level of each level is within the dynamic range of the device, and there will be no non-linear clipping due to too high signal level, or too low signal level to cause signal-to-noise comparison Poor, when setting the system level curve, the level curve of the mixer is very important. After setting the level, the system frequency characteristic can be debugged.

Modern professional electro-acoustic equipment with better quality generally has very flat frequency characteristics in the range of 20Hz-20KHz. However, after multi-level connection, especially the speakers, they may not have very flat frequency characteristics. The more accurate adjustment method is pink noise-spectrum analyzer method. The adjustment process of this method is to input the pink noise into the sound system, replay it by the speaker, and use the test microphone to pick up the sound at the best listening position in the hall. The test microphone is connected to the spectrum analyzer, the spectrum analyzer can display the amplitude-frequency characteristics of the hall sound system, and then carefully adjust the equalizer according to the results of the spectrum measurement to make the overall amplitude-frequency characteristics flat. After adjustment, it is best to check the waveforms of each level with an oscilloscope to see if a certain level has clipping distortion caused by a large adjustment of the equalizer.

System interference should pay attention to: the power supply voltage should be stable; the shell of each device should be well grounded to prevent hum; signal input and output should be balanced; prevent loose wiring and irregular welding.