Some problems that should be paid attention to in the use of noise 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. According to the form of signal transmission, microphones are divided into two categories: wired and wireless.
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 tie 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 proximity microphone for 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, generally cardioid condenser microphones are 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 audio signals of different levels and impedances; use the attached equalizer to process each frequency band of the signal; adjust the mixing of the signals of each channel After proportioning, each channel is allocated and sent to each receiving end; the live sound reinforcement signal and recording signal are controlled.
There are several points to note 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. Secondly, due to the problems of feedback feedback and stage return monitoring in sound reinforcement, the more balanced the input components, the more 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 mains and backup power supplies and can switch automatically. Adjust the phase of the control 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 defects of the sound intensity, the audio processing equipment between the mixer and the power amplifier must be installed, such as equalizers, feedback suppressors, compressors, exciters, crossovers, 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. In order to produce high-level artistic effects, a 3-segment electronic crossover is more appropriate 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 frequency divider is required Connect before the power amplifier.
To make the recorded program achieve 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 based on 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 also extremely wide, it is completely possible to connect the exciter 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 on-site 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. Both ends of the signal have the same resistance to ground, but the polarity is opposite, which is a balanced input or output. Since the values of the interference signals received by the two balanced terminals are basically the same, and the polarity is also the same, 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 uses 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. Absolute losslessness is also impossible. In the case of limited loss, the signal transmission will not produce distortion and the distortion condition 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, you can debug the system frequency characteristics.
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 audio system, replay it by the speaker, and measure the sound pickup with the test microphone 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 housing of each device should be well grounded to prevent hum; signal input and output should be balanced; prevent loose wiring and irregular welding.