This page explains the DSPs lined up on the input side. These are DSPs that mainly process input signals such as microphones and CDs. 

​Furthermore, it is assumed that the system will be used primarily for announcements and speeches, and for music as BGM. The purpose of this system is to deliver a sound that is uniform in volume and quality, making it easy for the listeners to hear the content of announcements and speeches. The purpose of using this system is different from that of a music production studio or concert, and the expectations for the DSP are also different.
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Trim

Trim is used to first equalize the volume of the incoming signal. Depending on the type of microphone, the output volume will differ even if you speak at the same distance and at the same volume. Also, the volume of sound sources such as CDs differs depending on the disc. The purpose of using Trim is to equalize these volumes before performing any processing.

This is an example of Trim's software GUI. It looks like a lot of faders lined up.

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The volume of each signal input to Trim varies, but please imagine that the volume of each signal after passing through Trim is the same.

Filter

Filters change the volume of each frequency in a signal. For example, a filter can reduce the low frequencies or emphasize the high frequencies of a signal. When used with a microphone (MIC) for speech, it is mainly used to reduce low frequencies to make it easier to hear the content of the speech. MIC has the property of amplifying low frequencies when the speaker brings his or her mouth close to the speaker (Please refer to this page​). Also, low-frequency reverberation tends to remain, making it difficult to hear the content of the speech. Therefore, it is meaningful to reduce the low frequency range.

The GUI shown on the left is used to set the filter. In the graph, the horizontal axis represents frequency, and the vertical axis represents the volume changed by the filter.

For example, in the case of the settings shown on the left, the vertical axis has a value of 0 when the signal exceeds 300Hz, so the filter has no effect and the input signal is output as is. On the other hand, below 300Hz, the value on the vertical axis is negative, so the filter lowers the volume of the input signal before outputting it. As a result, only the low frequency range of the input signal is reduced and output.

You can create filters of various shapes as long as the functionality allows. Filters that create standard shapes have names. The left one in the above figure is called the HPF (High-pass filter), the center one is called the PEQ (Parametric Equalizer), and the right one is called the LPF (Low-pass filter). PEQ has a function that changes the volume of the frequency band around a specific frequency.

Comp

Comp (compressor) has the function of limiting the maximum volume (strictly speaking, it suppresses the volume to around the specified maximum volume).
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In a PA system, it is important that the volume of speeches and announcements remains constant. This is because the louder or softer the volume, the harder it becomes to hear. Especially in a crowded environment, it becomes difficult to hear small sounds. 

By suppressing loud sounds using Comp, a signal with a constant volume is output. A signal with a constant volume is easy to handle, and even if the volume is increased, it can be increased evenly, making it less likely that local crackling will occur.

On the GUI, the Comp status is expressed using a graph like this. In this graph, the horizontal axis represents the volume of the entrance signal, and the vertical axis represents the volume of the output signal.

In the example shown on the left, the signal level changes as shown below as Comp passes through.

-30dB --> [Comp] --> -30dB
+20dB --> [Comp] --> 0dB

Click here to edit.

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If you create a horizontal section on the graph, that will be the target value for the maximum volume. In this figure, the output value (vertical axis) is set horizontally at -20dB. This means that any input signal between -20dB and +20dB will be at -20dB.

In addition to Comp, there are also functions called "Leveler" and "Gate". The leveler mainly increases the level of signals smaller than the threshold to near the threshold. Gate is a function that does not output signals below the threshold. The figure below is a simple representation of level changes.

Gain (Input Gain)

Input gain is used when you want to intentionally give different volumes to each input.

It is similar to Trim in that it changes the volume, but the purpose of changing the volume is completely different. 

​Trim is used to eliminate volume differences due to input. In other words, when Input Gain is the same value, Trim will be adjusted so that all output signals have the same level.

In short, use Trim to make all channels the same level, and use Input Gain to vary the volume if necessary.

Others

There are also DSPs on the input side that have functions other than those listed above.

Ducker
A function that lowers the BGM volume during MIC broadcasts. In short, decide which channel will be muted when outputting priority, and then automatically link the two operations.

Feedback suppression
This feature automatically detects feedback and suppresses it before it destroys equipment. If the position of each MIC with respect to the speaker (SP) is expected to be different, it is desirable to have a feedback suppression function that operates independently for each input channel.