Audio Service Knowledge: The most detailed debugging method for professional audio systems

Professional audio systems are expensive and technically complex, often requiring audio rental companies to send professional debugging personnel to the site for assistance. In practical applications, the debugging of professional audio systems cannot rely solely on theoretical knowledge from books. Therefore, it is not recommended that non-professionals operate them.

On the contrary, it is more necessary for audio rental companies to provide professionals with rich experience in equipment debugging and even maintenance. They can flexibly apply their knowledge in complex environments. This article summarizes the most detailed professional audio system debugging method in history, which is worth collecting.

Audio Service Knowledge 1: Preparation before Debugging

1. Arrangement of the Main Sound Reinforcement System

The main stage speakers are arranged symmetrically, pointing at a 45° angle towards the central axis of the audience area. The split speakers adopt a vertical stacking structure of the bass unit (directivity < 30°) → the midrange unit (directivity 60°) → the tweeter unit (directivity 90°). The radiation areas of the left and right channels should maintain a sound energy overlap area of 15 - 20%.

2. Arrangement of the Auxiliary Sound Reinforcement System

The supplementary speakers on both sides point to the rear audience area, forming an angle of 75° with the main sound reinforcement system. Avoid installing speakers in the reflection sound focusing area (such as the back wall), and control the sound image positioning error within the range of ±15°.

3. Cable Connection Specifications

The speaker system uses 200-core oxygen-free copper wire (cross-sectional area ≥ 4mm²), and the output impedance of the power amplifier ≤ 1.5 times the nominal impedance of the speaker.

4. Signal Link

Use XLR male and female connectors (2 + 3) for balanced connection. Use an isolation transformer for unbalanced to balanced conversion. Keep a distance of 30cm between the signal cable and the power cable.

Audio Service Knowledge 2: System Startup Process

Power Supply Sequence: Turn on the power in the order of "mixing console → peripheral equipment → power amplifier"

Initial Level Settings:

Input channel fader: at the 2/3 position

Input gain knob: at the 4/5 position

Master fader: at the 0dB reference level

Noise Detection:

Monitor the static background noise. If it is abnormal, turn on the noise gate of the limiter to the audible threshold. Input the test signal and gradually increase the master fader to the normal operating level.

Audio Service Knowledge 3: Adjustment Steps of the Electronic Crossover

1. Setting of the Low-frequency Crossover Point: Turn off the high-frequency power amplifier, adjust the low-end crossover frequency to the reference of 800Hz, and take the sound balance of the mid-low frequency speakers as the adjustment standard.

2. Setting of the High-frequency Crossover Point: Turn off the low-frequency power amplifier, adjust the high-end crossover frequency to the reference of 2kHz, and take the natural connection of the mid-high frequency speakers as the adjustment standard.

3. Verification of Channel Consistency: After independently adjusting the left and right channels, a sound field coverage test is required.

Audio Service Knowledge 4: Adjustment Guide for the Limiter Function

1. Adjustment of the Compression Mode:

Symphony: Compression ratio 3:1 - 6:1, attack time 5 - 20ms, release time 100 - 500ms, compression threshold -10dB to -5dB.

Language Programs: Compression ratio 2:1 - 3:1, attack time 20 - 50ms, release time 300 - 800ms, compression threshold -15dB to -10dB.

2. Adjustment of the Limiting Mode:

Set the compression ratio to ∞:1 (hard limit), the upper limit of the limiting threshold = the maximum distortion-free level of the system - 3dB, and compensate the output gain to the level after limiting.

Precautions:

Avoid an attack time < 5ms or a release time > 1s

The compression threshold and output gain need to be adjusted in conjunction

An overload protection test is required after adjustment

Audio Service Knowledge 5: Adjustment Process of the Equalizer System

1. Room Equalization Adjustment of the Main Sound Reinforcement System

Test Preparation: Load the full-frequency pink noise test signal, return the master fader of the mixing console to zero, set the PAN sound image to separate the left/right channels, adjust the gain so that the peak indicator light is in a -3dB critical state, and the channel fader is at the 2/3 position.

Frequency Band Compensation Adjustment: Adjust in the order of 500Hz (mid-high frequency) → 2kHz (high frequency) → 100Hz (mid-low frequency) → 20Hz (low frequency). Make a fine adjustment of ±3dB at each octave point, keep the Q value between 1.0 - 1.4, and achieve a flatness of 20Hz - 20kHz ± 2dB.

Verification Standard: Compare the frequency response consistency between headphone monitoring and on-site sound reinforcement, and record the equalizer compensation curve.

2. Adjustment of the Auxiliary Sound Reinforcement System

Signal Routing: Turn off the main sound reinforcement system, group and output the test signal to the auxiliary channel, and adjust the group fader to the target sound pressure level (3dB lower than the main sound reinforcement).

Compensation Adjustment: Adopt the same process as the main sound reinforcement system, focus on compensating the sound image positioning deviation area, and establish an independent compensation curve file.

3. Adjustment of the Monitor System

Monitoring Settings: Adopt the singer's in-ear monitoring mode, set an independent auxiliary output channel, and activate the pre-fader monitoring mode.

Compensation Adjustment: Focus on optimizing the vocal frequency band of 200Hz - 8kHz, and reserve a dynamic adjustment margin of ±6dB.

Audio Service Knowledge 6: Debugging Specifications of the Feedback Suppressor

1. System Preparation

Microphone Placement: Maintain a 180° pointing angle with the speaker, and the distance should be at least 2 meters. Use a pop filter to reduce the low-frequency proximity effect.

Channel Settings: Set the input gain to the 0dB reference, reset the parametric equalizer to 0dB, position the sound image in the center, and start the fader level at -10dB.

2. Debugging Process

Mode Selection: Activate the dynamic filtering mode, and set 6 groups of fixed filters and 3 groups of dynamic filters.

Filter Parameters: Set the activation threshold to -20dBFS, the attenuation amount to 12dB/octave, and the response time to 50ms.

Step-by-step Debugging: Increase the fader of the main sound reinforcement system to -5dB, trigger the automatic filtering and lock 6 fixed filter points; repeat the process for the auxiliary system; use manual filtering for the monitor system, and focus on processing the frequency band of 1kHz - 5kHz.

3. Safe Operation Specifications

The gain increase speed ≤ 3dB/s

Set the limiter threshold to -3dBFS and turn on the power amplifier clipping protection

Precautions:

1. Wear monitoring headphones during debugging.

2. Check the phase consistency after completing each system.

3. It is recommended to calibrate the system every 2 hours.

4. Add a 3dB safety margin in extreme environments (high temperature/high humidity)

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