Nowadays, most conference rooms, theaters, etc. have a standard configuration, that is, audio, which has many uses, so there is no need to say more about this issue. But in addition to these uses, will the audio experience interference? The conference venue equipment includes conference terminals and related external equipment, and the types and functions of the equipment are basically the same. Therefore, the audio signal transmission process is divided into two parts:

The sound of the local end is picked up by the microphone, and the human voice is converted into an electrical signal and sent to the mixer. After amplification, mixing, distribution, sound quality modification and effect processing, it is divided into two channels: one signal is sent to the power amplifier of the local end for amplification. After the speaker is restored to sound; the other end is amplified by the mixer and then output by the speaker (AUX).

The other part: After the signal sent by the remote end is processed by the conference terminal, the audio signal enters the mixer for processing, and then is sent to the power amplifier at the local end for amplification, and is restored to sound through the speaker. The above analysis shows that the mixer is the meeting point of each signal processing process. How to adjust each button and function key of the mixer is the key to debugging.

Looking at the routine debugging and holding of meetings of the national organization department conference system, the quality of sound reproduction has been significantly improved, but there are still some problems in individual provinces and cities, mainly manifested in large noise, large distortion, irregular level, uneven frequency response, sound Disproportionate, mixing delays, etc.

loud noise

This will affect the clarity, softness and brightness of the sound, and in severe cases it will mask the sound of the venue.

① Background noise. When the gain is too large, the automatic gain control method is adopted, and the impedance is not matched, background noise will be generated.

②The microphone "flicks" out. The microphone is the first link in the amplification system, and its signal quality directly affects the overall amplification effect. Therefore, it should be selected and set reasonably according to the characteristics and performance of the amplification system, reproduction characteristics and the relative relationship between various sound sources. For speakers with strong broken voices, microphones that prevent "flashing" should be used.

Large distortion

This is related to the nonlinear deformation of the equipment and human factors. Symptoms are hoarseness, brokenness, sharpness, and in severe cases, it can affect the clarity, suppleness, brightness, fullness and presence of the sound. There are many reasons for distortion, such as excessive microphone sensitivity, improper placement, abnormal phantom power supply, excessive level adjustment of the mixer, improper adjustment of the equalizer, and impedance mismatch between devices.

non-standard level

The mixer, equalizer and other equipment were not adjusted before adjustment, resulting in the signal level being too high or too low. Also, connecting unbalanced outputs and balanced line inputs directly results in a drop in signal level.

uneven frequency response

This is related to frequency response indicators and human factors, such as microphone pointing offset, excessive mid-frequency attenuation, and long microphone transmission lines, which will affect the clarity, layering, and fullness of sound reproduction.

intonation

If the sound ratio of the three sub-venues including the main control room, the main venue, and the sub-venues is out of balance, it will affect the restoration of the long-distance sound balance in other sub-venues, resulting in a decrease in the overall effect of the conference.

false alarm delay

Due to the different requirements on the reverberation delay time of various conference types and audio environments, the reverberation delay parameters should be adjusted according to the acoustic characteristics.

①The size of the field affects the time sense and auditory time sense of sound reflection during reverberation.

②The reverberation time is the process in which the sound diffuses from the sound source through the surrounding absorption and reflection. The reverberation time depends on the size of the venue. The larger the distance, the longer the reverberation time. Audio processing equipment compensates for and retouches certain sound imperfections, but not properly adjusted can backfire. If the reverberation time is too long, the sound will have a "turbid" feeling, so the reverberation effect should be added reasonably according to the actual situation to enhance the depth of the sound. The same applies to delayed processing.

The frequency characteristics of reverberation reflect the reverberation effect and sound quality of the sound at different frequencies. To be familiar with the frequency characteristics of reverberation, adjust the ratio of the microphone to the direct sound and reflected sound to improve the realism and clarity of the sound.

Sound diffusion is a parameter that reflects the acoustic characteristics of the scene. The microphone should be set up reasonably to make up for the defects in the sound diffusion conditions of the venue. At the same time, the vibration echo area and strong reflection direction should be avoided.

During the transmission of audio signals, a lot of interference occurs, such as power supply interference, inter-device interference, and lighting interference.

power disturbance

Poor power supply grounding, poor grounding contact between devices, mismatched impedance, unpurified power supply, audio cables and AC cables in the same pipe, in the same trench or in the same bridge, etc., will cause clutter interference to audio signals and form low-frequency AC " buzzing.

Inter-device interference

"Howling" is caused by positive feedback between the speaker and the microphone, mainly because the microphone is too close to the speaker, or the microphone is pointed at the speaker. "Empty sound" is generated when the sound source is delayed. For example, the microphone both picks up the sound source signal and takes the signal restored by the amplification, or two microphones at different distances from the sound source pick up the signal of the same sound source, or one microphone. The signal will have a corresponding delay. After these signals are superimposed, some frequency components cancel each other out, resulting in "empty sound".

lighting disturbance

Lighting lamps that use ballasts to start intermittently at the scene will generate high-frequency radiation when the lamp is excited, and will be connected through the microphone and its leads, resulting in a "da-da" sound; if the microphone line is too close to the lamp line, there will also be a "da-da" sound. "Squeak" sound; in addition, high-frequency electromagnetic interference will also occur.

To restore the sound of the video conference, according to the theoretical knowledge of acoustics and different actual situations, the mixer and equalizer should be flexibly adjusted to process and beautify the sound, make up for the defects of the sound field, and create a more ideal acoustic environment; appropriately adjust the compression limit Amplifier, when encountering sudden large peak signals, it will not overload and unbalance the device, so as to make up for the lack of sound field and create a more ideal acoustic environment.