Procedures and methods of system debugging in professional audio engineering

The debugging of audio engineering is generally divided into system debugging and sound debugging. Here are the steps of system debugging. The following system debugging steps are generally adopted by me and introduced to everyone. Audio engineering debugging steps:

System level preliminary debugging

1. Check the circuit according to the system configuration connection drawings. Check the circuit connection carefully to confirm that there is no problem.

2. The initial state setting of the device sets the power amplifier input to small, and sets the input and output knobs of all peripheral devices to the 0 decibel position or the intermediate position. Turn on the power in the order from the front stage to the back stage (do not turn on the power amplifier first). After checking that all devices are properly powered, power on the power amplifier.

3. Initially check the system status and properly open the gain control of the power amplifier. Put a familiar piece of music in the CD and adjust the input level of the mixer to the basic normal position. Slowly push up the mixer fader to hear whether the sound from the speaker is normal or distorted. If it is not normal, immediately shut down and check.

4. After the polarity detection system of the speaker and the system is basically normal, turn on the power of all equipment, set the power discharge level to large, pull down the mixer output fader, the phase meter generator is connected to the mixer input channel, turn on the phase meter power adjustment The output gain and mixer input gain to the mixer indicator table are 0 dB. Slowly push up the mixer output fader, and wait for the "pop" sound from the speakers to reach a sufficient loudness (if the loudness is not enough, the test results are sometimes inaccurate), use the phase meter detector to check whether each speaker is in phase or The description of the speaker manual is consistent. It is better to turn off other speakers during detection to prevent interference, and it is more accurate to detect each one one by one. If something is abnormal, check if the speaker cable is reversed or if the system cable is reversed. Check again after transfer or replacement.

5. Phase adjustment If the combination of ultra-low frequency and full frequency is used at the same time, due to the existence of the crossover system and the installation location, there may be phase problems caused by cross-frequency interference or different delay times, so phase adjustment is required. The pink noise (PINKNOISE) generator is connected to the mixer input channel, and the level is adjusted to the normal position. The phase meter test microphone is placed in the middle of the venue, forming a regular triangle with the speaker. Push up the mixer output fader, and check whether the spectrum analyzer screen has a valley point in the frequency band near the crossover frequency of the full-frequency and ultra-low frequency speakers. If so, increase the corresponding frequency band of the equalizer. If it does not increase, there is a phase problem. The appearance of the phase problem will directly affect the sound quality, and it is just solved with an equalizer. To solve the phase problem, it is necessary to adjust the phase angle of the crossover or the delay time between the speakers. When adjusting, pay attention to the spectrum analyzer display. First adjust the phase angle of the low-frequency divider to see if there is any improvement. If there is improvement, adjust the delay time after determining a good value. The adjustment of the delay time depends on the scene. If the low-frequency speaker is close to the seat, you need to adjust the delay of the bass, also looking at the spectrum analyzer screen, adjust the delay time to make the curve as flat as possible. Reduce phase interference to low.

6. After adjusting the frequency balance, the frequency response curve of the system needs to be adjusted. Put the test microphone of the spectrum analyzer at a position in the seating area, play the pink noise source, watch the spectrum analyzer display, and use the equalizer to correct the defective places. Then put the test microphone in a different place, and then adjust the balance in reverse, change a few more positions, and repeatedly adjust the balance to make the frequency response curves of each area as straight as possible, even if the equalizer is roughly adjusted.

7. Dynamic control device adjustment Generally, the dynamic adjustment device is a compressor. If the compressor is connected in series with a chain connection, it is generally used as a protection system, so it mainly uses the limiter function of the compressor. I usually operate the limiter in this way: the limiter is set to pass-through (BYPASS), the input and output gain is set to 0 decibels, the compression ratio is set to "infinity", and then the mixer sends out a pink noise signal, gradually push it up, observe The input level indicator of the power amplifier, when the clip indicator (CLIP) lights up. Connect to the compressor and adjust the start level knob to turn off the power amplifier input clipping indication. Then increase the mixer output by 6 dB or let the output level indicator reach +6 dB, and then adjust the start level a bit to make the power amplifier clipping indicator just start flashing. So far, the level debugging of the system is completed. As for tone adjustment and effect processing (including effects, exciters, etc.), they generally belong to the task of the tuner. Of course, in audio engineering, this is generally done by the technicians of the audio company.

The second step of sound effect adjustment: to avoid howling and debugging of the microphone sound

â‘ The monitoring system adjusts the level of a microphone according to the steps in the step. First, the microphone of the mixer is not balanced, and the channel fader of the mixer is placed at 0 decibels (if the signal for stage monitoring is taken from the fader This is done after the auxiliary output). Put the microphone in the main position on the stage, turn on the monitor output master (AUX) and gradually increase the output. After the microphone causes howling in a certain frequency band, fine-tune the AUX knob to stabilize the howling at a certain volume level, and then adjust the corresponding balance To eliminate the howling in this frequency band, and then continue to increase the volume, wait for the howling in another frequency band to occur, and then eliminate it by adjusting the equalizer, and so on, until the mixer output level fader or AUX knob is adjusted to In the normal position (such as 0dB), the microphone no longer produces howling, OK, pull down the mixer fader. This method is used to find howling points that easily cause resonance in the sound field, and then appropriately reduce the level of the microphone channel, find someone to speak to the microphone, and then gradually increase the microphone volume to the normal position, if there are still howling, then Eliminated by the equalizer. Key points of operation: Be sure to control the level so that the howling can be maintained at a stable level and then adjusted to be more accurate. The operation must be slow, otherwise it will be impossible to find the correct howling points one by one. The resonance points in the room are generally around 5-6 points. If there are too many feedback points, it is necessary to check whether the placement of the speakers is reasonable.

â‘¡After the main sound reinforcement system adjusts the monitoring howling point, then adjust the main sound reinforcement system in the same way. If the main sound reinforcement is a two-channel system, first turn off a channel and push the input fader of the mixer, gradually Increase the volume to find howling points. After adjusting one channel, turn off this channel and adjust the other. After adjusting both sides, push the two channels up at the same time, check whether there are other howling points, and then eliminate them through the equalizer.

2. After the problem of howling is resolved, adjust the tone of the microphone

â‘ The vocal microphone uses a high-quality microphone as a reference (I recommend using SHUREBETA87A) to adjust the balance of the microphone input channel so that the tone of the microphone used is as close as possible to the high-quality microphone used for reference. For detailed methods, see my previous Posts. After this step is completed, the microphone timbre generally meets the requirements of most actors, and then the actors are allowed to audition the microphones. According to the requirements of the actors, the channel balance of the mixer can be appropriately fine-tuned.

â‘¡ After adjusting the original tone of the effect microphone, you can add reverb, use the effect to select the appropriate effect type, turn on the corresponding AUX output of the mixer, and adjust the input level of the effect and the auxiliary output level of the mixer at the same time. Speak into the microphone to see if the signal output to the effector is too large or too small. Generally, this signal is controlled at 0 dB. Set the mixing ratio of the effector to 100%, then gradually increase the output level of the effector, check whether the level of the output callback console is normal, if it is normal, gradually increase the knob of the effector return, according to the requirements of the actor, put Set the volume of the effector to the appropriate level. After the effect is added, due to the existence of artificial reverberation, a new howling point may be generated. It is generally easy to have a howling point in the low frequency band. The return of the good effect device is connected to the line input of the mixer. Adjust the balance of this input channel to eliminate howling. soundman comment: As a formal sound engineering company, I think it is very necessary to equip professional testing instruments. After all, many objective technical indicators cannot be distinguished by ear alone, and the instrument can be used to quickly and accurately correct Debugging. Of course, many friends engaged in on-site debugging do not have this condition. Without the aid of the instrument, many people fully believe that there is a certain flaw in their own listening experience. The subjective judgment of experience accumulation can be used as a reference. After all, people and people feel different. Friends who have been engaged in live tuning for a long time have enough listening experience. If you add a correct understanding, you can also improve the debugging effect, but the premise is that you must have a deep understanding of the various objective technical indicators of the device generated in different sound fields. Variety. Subjective listening debugging is built on the basis of objective technical indicators, which allows you to maximize your tuning ability and reflect the advantages of the device.