Professional stage audio equipment includes: monitor mixer; power amplifier mixer; portable mixer; power amplifier; dynamic microphone; condenser microphone; wireless microphone; speaker; monitor speaker; power amplifier; subwoofer; equalizer; reverberator; effector; time delay device; compressor; limiter; splitter; noise gate; CD player; VCR; projector; tuner; tuner; music player; music player; audio-visual equipment; audio-visual equipment; audio-visual equipment; audio-visual equipment; audio-visual equipment; audio-visual equipment; audio-visual equipment; audio-visual equipment; audio-visual equipment; audio-visual equipment; audio-visual equipment; audio-visual system; audio-visual system; audio-visual system; audio-visual system; Headphones and many other devices. A tuner is a professional who is responsible for all sound equipment on the stage. First of all, all the sound sources need to be centralized to the mixer, and then they are divided into various sub channels, marshalling and tuning, and then the output audio of the main channel is evenly sent to the power amplifier and speaker box to make sound to adapt to the sound characteristics of the site.
In order to control the volume of each song and balance them, the tuner needs to know the music characteristics of each link, the frequency of the most easy feedback and the resonance point of the venue. In addition, they should be able to improve the voice and instrument sound without using the effector, and be familiar with the functions and limitations of each equipment, so as to avoid distortion and other adverse effects. The quality and reliability of the selected equipment should be guaranteed. The possible faults should be prepared in advance and effective remedial measures should be taken in time. Sometimes it is necessary to finish the preparation work a few hours before the performance. Although the time is very tight, for the sake of safety, each signal line must be connected in person.
This article can be read and reference materials for beginners, amateur audio enthusiasts and music hall sound engineers. If there are any shortcomings, please give criticism and correction.
1、 Power amplifier
Professional Amplifiers need to work continuously for a long time in large-scale activities, and can also withstand the vibration and impact during transportation. Therefore, compared with the amplifier used in general audio, the professional amplifier pays more attention to the durability and structural reliability of long-term use in design. Amplifier has an important impact on the sound quality of sound reinforcement, accounting for about 30% of the whole set of audio equipment. Therefore, in order to give full play to the performance and role of audio equipment, we should pay attention to the quality of amplifier. Otherwise, high-quality sound reinforcement system will not work. There are three types of power amplifiers:
1. Single type power amplifier;
2. Mixer + power amplifier integration;
3. Speaker + power amplifier integration.
Single amplifier: this kind of power amplifier is an independent component, which can combine the sound system according to its own plan. Generally, a power amplifier consists of two channels.
Mixer + power amplifier integration: this kind of power amplifier is simple to connect and easy to operate. It is widely used in small and medium-sized sound reinforcement systems.
Integration of power amplifier and speaker: considering the matching of power amplifier and speaker, it is easy to use and is mostly used for monitoring speaker and keyboard instrument speaker. When the amplifier is connected with the amplifier or between the amplifier and the transmitter, the impedance matching between them must be considered (the impedance is in Ohm). Impedance matching means that the rated output impedance of the power amplifier should be equal to the rated impedance of the speaker. At this time, the power absorbed by the speaker is the maximum. If the rated impedance of the speaker is much smaller than the rated output impedance of the power amplifier, the working current will increase sharply and the loudspeaker and amplifier will be damaged. When one channel of the power amplifier drives two speakers, the total impedance of the speaker will be reduced, and then the load impedance of the amplifier will be reduced, and the power amplifier will be over driven in the case of near short circuit. Therefore, when the power amplifier is connected with the speaker, the input impedance value of the loudspeaker must be within the load impedance range of the power amplifier.
Power matching: in principle, the rated output power of the power amplifier should be equal to the rated power of the speaker. However, due to the serious nonlinear distortion of the amplifier tube after overload, it is usually intended to increase the rated output power of the amplifier to make it greater than the rated power of the loudspeaker. The correct connection should be: the output power of the amplifier is 30% higher than the nominal power of the speaker. If the power of the speaker is too small than that of the power amplifier, you should be extra careful when using the amplifier. The volume should be adjusted gradually from small to large, and not too high, otherwise the speaker will be damaged. In practical work, the output power of power amplifier is relatively large, which is beneficial to improve the sound quality. In addition, the dynamic range of the sound source is very large, so we should pay attention to the damage caused by the instantaneous overload of the power amplifier.
Average output power refers to the power that works continuously for a long time. Peak power is the maximum power that can be sustained in a short time, which is much larger than the rated power. The output of sound amplification is determined by the power amplifier. For a concert of a certain scale, there must be a certain power. The standard is one watt per person. Power varies depending on the type of concert, the size of the venue, the reverb and the number of speakers.
Total power / output power of one power amplifier = required number of power amplifiers
Bridge output: bridge output is a way to amplify a stereo amplifier to mono channel. It is to obtain high power output circuit form, also known as BTL mode.
The principle of bridge connection method: the positive half cycle signal is amplified by a channel, and the negative half cycle signal is amplified by way B, so that the output power is doubled.
Bridge connection and specific steps: when one power amplifier works separately, the rated output power of each channel is 400W / 4?, which is the common stereo connection. When higher rated power output is required (below 400W), bridge connection method can be adopted
1. Turn the mode switch to the "bridge" position;
2. The signal is input from a channel;
3. The power is output from the "+" end of the two channels, the output "+" for channel A and "- output" for channel B.
Power amplifier output level display: the display is a color LED trapezoidal group, which is used to display the level height of power amplifier immediately. The normal level is in green; the level signal is in yellow when the amplifier is required to transmit a continuous high pitched signal; the red LED flashes (sometimes flashes) when the audio signal of the music peaks or drums are playing. All of the above are normal.
If the red LED is on all the time, it means that the amplifier may be overloaded. This kind of situation often occurs when a power amplifier drives multi-channel speakers. Therefore, the system should be reconfigured to eliminate this overload phenomenon.
Peak display (peak): when peak diode is flashing, the gain control should be lowered.
Protection of power amplifier: in case of some wrong operation, the built-in protection circuit of power amplifier will be automatically disconnected, and then the protection display will flash. After the error operation is eliminated, the protection display lamp will go out.
There are two types of mixer: recording room and stage dance hall.
The functions of the mixer are as follows:
1. Pick up the signal and amplify it;
2. According to the needs of high, medium and low tone balance;
3. Send the signal to the left and right bus or group control as required;
4. The signal sent to the auxiliary bus is processed artistically;
5. Output control as required.
The mixer can be divided into input unit and output unit.
（1） Input unit
The input unit is an important part of the mixer. The input unit is a shunt parallel circuit, each of which is roughly the same. Generally, it can be divided into the following parts.
A. Input selection section
1. Tape: tape
2. Mic: microphone
3. Line: Line
B. Input attenuator (PAD)
If the microphone or line input signal level is too high and the gain control cannot be adjusted, turn on the attenuation switch, and then a 20dB attenuator is inserted between the preamplifier and the input socket to avoid overload.
C. Input gain control (gain)
The sound sources of the mixer include: microphone, musical instrument, tape, effector, sound reinforcement equipment, etc. Because their output levels are different, in order to match them, the gain control is used to adjust the input sensitivity on the mixer. If the input signal is too large, it will produce clipping distortion; otherwise, if the input signal is too small, the noise will not be controlled. Gain control is used to ensure that the mixer works in a fixed dynamic range. On the panel, the expression of gain control level is based on 0dB = 775mv, which is set at different positions according to the output level of sound source.
The input signal and gain level are shown in the table below.
Gain (DB) 〝 input signal? - 60 ~ - 50 〓 low level microphone? - 35 〓 high level microphone (capacitor), electronic instrument
-20 〓 low level circuit (general audio)
D. Signal input socket
It is divided into low resistance balanced input (lo-z XLR) and high resistance unbalanced input (Hi-Z two core).
General musical instruments and audio equipment are connected in an unbalanced way, and one end of the signal "+", "-" is shared with the shielding layer of the signal line. For example: one core shielded wire, the core wire is the signal "+", and the shielded wire is the signal "-" and ground wire. This is less than that of the parallel line without shielding, which belongs to the cylinder type incomplete shielding.
The input and output of professional audio equipment are balanced, and the signals are transmitted by "+" and "-" respectively. In addition, the shielded wire is connected. The "+", "-" uses independent ground wire, and the plug uses XLR plug.
E. Overload (clip)
The overload indicator is used to warn the input signal of instantaneous overload, and the indicator lamp will light when it is 3dB below the peak value (the level of signal distortion due to excessive signal), which is convenient for setting the position of gain switch.
F. Input equalization part
The input channel equalizer is used to correct the timbre of the input signal to achieve the standard effect. Due to the single channel control, the mixer can carry out equalization control for each channel without mutual interference. The equalization is divided into high frequency (high), intermediate frequency (MID) and low frequency (low).
0 position is flat; + direction (gain), + 15dB (enhancement 5 times); - direction (attenuation), - 15dB (attenuation 5 times). Continuously adjustable.
The equalizer generally adopts treble (10kHz), midrange (the center frequency of equalizer can be freely set between 350hz-5khz) and bass (100Hz).
Due to the independent control of each frequency band, the input signal can be adjusted carefully, and then the timbre can be adjusted boldly, and the unnecessary components such as howling and noise can be effectively removed.
1. High frequency: 10kHz ± 15dB / slope
Influence area: high order harmonics in the treble region of the instrument.
Gain effect: the metal sound increases, the timbre is sharper, the gain is too much, the noise can be heard obviously.
Attenuation effect: it can effectively remove the hissing sound. If the attenuation is too much, the transparency of the treble area will be lost.
2. If: 3kHz ± 15dB / peak
Area of influence: instrument, high pitch area of human voice.
Gain effect: the timbre is bright, the texture is hard, the gain is too much, and the hearing is easy to be tired.
Attenuation effect: the balance of music tends to be low, including the sound.
If: 1kHz ± 15dB / peak
Area of influence: musical instruments, the midrange of human voice.
Gain effect: the timbre contour is clear, the sound phase is protruding forward, and the drum sound head is strengthened.
Attenuation effect: phase backward.
If: 500Hz ± 15dB / peak
Area of influence: musical instruments, mid bass area of human voice.
Gain effect: the tone is thick and powerful. If the gain is too much, the telephone voice will appear.
Attenuation effect: the sound head is hard, the balance tends to be high, and if the attenuation is too much, the texture will be thin.
3. Low frequency: 100Hz ± 15dB / slope
Area of influence: the bass area of an instrument.
Gain effect: the timbre is thick. If the gain is too much, the sound of teeth is not clear.
Attenuation effect: the sound is easy, the sound is good, and the background noise and buzz can be effectively removed.
G. Sound phase
The sound phase knob is used to adjust the left and right balance of the signal, and the position is after the level adjustment of the channel potentiometer. In addition, the position of each input channel signal between groups 1-2 and 3-4 is also determined by this knob. If the knob position is in the middle, the sound phase position is also in the middle. Turn the knob to the left and position it in group 1 or 3. Turn the cud to the right and locate in group 2 or 4.
H. Monitor send (Mon / send)
Monitoring transmission is used to control the level value of input signal on the monitoring bus. This control is not controlled by any control switch on the channel (including the control of channel volume) except for gain control. Therefore, the transmitted signal is relatively independent from the main bus signal.
1. Effect sending (EFX / send)
It includes all peripherals that determine how much of an internal or external effect is added to the input signal. It is affected by equalization and volume attenuator, because each channel has its own effect transmission, so by adjusting, some channels can produce effects, while others do not.
Note, however, that internal and external effects share the same transmission control, so they should have the same source.
J. Pre monitoring switch (PFL / cue)
When the switch is on, the signals of each input channel can be monitored in the headset and confirmed on the level meter. The priority of the monitoring switch should be firmly remembered.