The construction of modern audio systems is simulating the transition to digital, fiber and network go hand in hand, and the pattern of digital audio processing equipment and media matrix is ​​emerging. Audio professional engineers have greatly expanded the choice of design and architecture systems, and the flexibility and control of the system and the extended coverage area can almost reach the desired level. But at the same time, it should be soberly realized that as a good standard for measuring audio system engineering, it is not just a matter of how many high-tech equipments are stacked and how expensive it is, but it should be reasonable and flexible from the practical level. Changed structure concept and equipment matching skills, while improving cost performance, but also get good results and quality.
A good audio system should have the following characteristics: stable and reliable system architecture, convenient operation and detection functions, and good overall results. Low-cost audio systems can be as good as they are, and high-input audio systems don't necessarily work well, which is a common occurrence in the industry. It is very difficult to make major changes after the completion of the general system. In many cases, it is only necessary to make more efforts in debugging and operation. It is the solution to all the functions and overall potential of the equipment. Different systems have their own unique characteristics. The advantages and disadvantages of the system are the basic principles of system debugging. That is to say, the flexible and versatile system architecture should be combined with flexible and varied operation, and the local conditions should be avoided.
Diversification of function realization
For example, a multi-functional conference hall that is currently more common, to achieve multi-channel microphone pickup and sound reinforcement in the podium and audience area, to achieve the basic functions of meeting and entertainment compatibility. Depending on the amount of investment, there are a number of different equipment matching options to choose from, and the corresponding system operation methods are also diverse, and the howling suppression when using multiple microphones in a multi-function conference system is usually the primary consideration. of. The following is a brief description of different system configurations and corresponding feedback suppression methods.
1 Low cost system configuration
Simulate equipment should be used as much as possible when the amount of capital invested is small.
Taking into account the suppression of the feedback of the conference system, an automatic microphone mixer can be used, serially connected in the system or using the grouping GROUP function of the analog mixer to group the multi-way conference microphones, and then inserting the feedback suppressor into the corresponding In the INSERI port of the marshalling channel, the occurrence of howling can be effectively suppressed. If the operator also masters certain system equalization debugging techniques and setting methods of dynamic and static parameters of the feedback suppressor, the effect will be more obvious and the sound quality will be further improved. If the automatic mixer and feedback suppressor are not considered in the system design, then the system debugging and speaker placement should be more difficult, the box can be hung or extended along the platform to the ends, the whole The sound reinforcement system preferably uses the PINK signal corresponding to the spectrum tester to adjust the indoor sound field spectrum to a relatively flat state.
system debugging
For this relatively simple system configuration, in order to obtain a better sound reinforcement effect, we must first make full use of the system equalizer, try to make the frequency of multiple points in the hall in a straight state, especially the frequency band showing the peaks and valleys must be It is equalized until a flatter spectral curve is presented. Secondly, the sound quality of the microphone is trimmed by the mixer channel equalization. For the voice vocal, it is better to appropriately attenuate the high-frequency and low-frequency frequency bands to remove the mouth-tooth and nasal sounds to obtain a clear and transparent sound quality. After the above steps are adjusted, fine-tune the system equalizer.
Another control technique for this kind of streamlined sound reinforcement system (that is, there is no automatic microphone mixer and no feedback suppressor) is a safe control technique. When the scene is tuned, the eyes focus on the speaking seat, and the heart accurately predicts who is the speaker. The finger gently pushes the corresponding channel fader and pulls down the remaining microphone channels, and so on, only pushes the speaker's channel fader at a time. By using the effective sound gain of the system for the most needed channel, the sound pressure of the speaker of the corresponding channel can be maximized, thereby eliminating the necessary conditions for generating feedback. The system's sound gain cannot be increased indefinitely. It is more constant at a time, and it is more feasible to increase the availability of the system within its maximum allowable range. But this method is not so easy to control for beginners.
2 Medium-input system configuration
A mode in which digital devices are shared with conventional analog devices can be used with moderate capital investment.
The preferred solution is that the mixer is selected from small and medium-sized numbers, and the peripheral device is configured in the same manner as the compact sound reinforcement system, and the adjustment and variation are not made. At this time, the advantages of the digital mixer can be better played. The dynamic processing, equalization setting and channel parameter copying of the single channel of the digital mixer can improve the sound gain, eliminate noise and optimize the sound quality. And suppress the role of feedback.
If the noise gate of each microphone channel is set in an appropriate state, the common background noise such as footsteps, conversation sounds, coughing sounds, cups and cups can be effectively rejected, and the sound signal can be purified. To achieve the effect of improving speech intelligibility. Secondly, this setting can effectively suppress the generation of feedback. The basic principle is that only the sound pressure of the normal speaker can be transmitted beyond the threshold setting of the noise gate, and the remaining microphone channels of the speaker are now closed. This is actually the same effect as the above-mentioned combination of the hand and the eye. The former is controlled by people, and the latter is the automatic control of the noise control of the digital mixer, which reduces the operation intensity and improves the safety factor. Moreover, each input channel and output bus of the digital mixer is embedded with a parametric or graphic equalizer, so that even if the sound is not ideal, sudden feedback or sound field defects can be adjusted and corrected in real time. In addition, you can set several common conference modes, such as the number of different microphones used, different microphone use positions, etc., to store and name various conference states, so that it is more convenient and reliable when used in the field.
3 high cost system configuration
In the case of relatively abundant funds, a separate multi-function conference subsystem can be configured, which uses an analog or digital mixer to match the digital audio processor or media matrix.
In fact, digital audio processors and media matrices themselves have many commonalities, such as the use of network protocols and fiber optic ports for audio and control signals, internal processing adaptive digital-to-analog and analog-to-digital conversion. Their advantages are powerful, suitable for a wide range of venues, easy to expand and upgrade, and high operational reliability. Nowadays, many multi-function conference systems can be automatically controlled after unattended operation. A variety of conference modes (such as first-in, first-out, question-answering, master-control unit priority, etc.) have been pre-programmed into the system, and the operator only needs to The form of the meeting and the pre-selection of the corresponding mode are required. Since such system settings usually also speak for a single person, it is better to avoid the occurrence of howling under normal conditions. The parameter settings of the digital audio processor and media matrix need to be adjusted repeatedly to store after determining the best value. Their feedback suppression adjustment settings are very similar to digital mixers and are not described here. Generally, the related elements such as routing, usage area, usage mode, and user authority can be flexibly configured and combined according to actual use requirements, and the program number corresponding to the program can be quickly called when used in the field.
The advantages of the media matrix in the audio control of large venues and plazas are revealed. The optical interface function can realize the transmission of several thousand audio and control signals, and the remote control points can be operated by the desktop operating system in the main control room. Centralized control, including system working time control, on-site sound pressure automatic adjustment control, frequency equalization adjustment, system delay adjustment, channel gain and Q value adjustment, dynamic compression control and other functions and parameter settings. The calling of these functions and the setting of parameters correspond to different operating ranges according to different operating rights. For example, system architectures such as system path and route allocation can only be modified by technicians with higher authority, system balancing, system delay, feedback suppression, Dynamic processing control and other functions, due to the precise parameter requirements, the commissioning time should also be set to a high level after one-time adjustment to a suitable state, prohibiting arbitrary changes, for functions such as manual volume adjustment, system work timing control, etc. The situation is adjusted accordingly. These different permission settings fully reflect the strengths of the digital device, and better avoid the use failure caused by improper human operation and other factors.
Comparison of three modes
Generally speaking, the system with weak configuration function generally has certain requirements on the user's on-site handling ability, and skillful skills are used to make up for the lack of functions. The powerful control system is more complicated in the previous adjustment and setting, but it is more worry-free when it is set up. Some devices can even turn the system on and off regularly, replacing the manual operation with advanced technology. However, the basic working principle of the advanced digital audio system is also derived from the analog audio system. The signal routing and parameter setting are based on the mastery of the analog equipment, but only the display screen of the digital device. The desktop operating system makes the user feel more intuitive and convenient.
Therefore, playing simulation, mastering numbers, and familiarizing with the network are what modern audio workers should try their best. This is not only to meet the needs of daily work, but also the requirements of the development of modern audio technology for the practitioners in this industry.
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