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Sound effects

Let’s dwell on the consideration of those manipulations with sound that allow you to achieve the appearance of effects such as, for example, echo, reverb, etc. There are various ways of transforming sound (amplitude, frequency, etc.). Based on these transformations, sound effects are realized. Basically, the goal of sound processing is to give the existing sound some new qualities or eliminate undesirable ones. Sound effects refer to those sound transformations that give the sound new forms or completely change the sound information.

The hardware implementation of sound effects is found in digital signal processors (DSPs). Any more or less decent MIDI synthesizer has a built-in effect processor of one or another complexity (the effect processor is one or more DSPs). Complex effect processors “know how” to apply several different effects to the sound signal at once, moreover, separately for each channel, allowing you to adjust the parameters of the effects in real time. However, the cost of such effect processors is extremely high (like the cost of any other high-performance microprocessor), so professional DSPs are installed only on high-quality musical equipment. On more or less cheap sound cards, DSP is often installed with a simplified set of features: applying one or more effects to all channels simultaneously.
A hardware effect processor is certainly good, but you can also process sound at a high level programmatically. There are many different sound editors that allow you to do much more complex things with sound than even the most complex effect processors allow. In addition, effect processors are often emulated in virtual WT synthesizers, and also find software implementation in special programs for real-time sound processing.

So, back to the description of the effects. These effects are obtained mainly in four ways: using delay, changing amplitudes, filtering and changing the frequency components.

Delay use

Delay. Actually, the delay effect (from the English “delay” – delay) is used more often in cases where a mono signal needs to be converted into something like a pseudo stereo. If a mono signal is fed into both channels of a stereo speaker system, then by a certain delay of the signal in one of the channels it is possible to achieve a stereo effect. If the signal arrives at the same time in both channels, then it will seem to the listener that the sound source is located in the middle. By changing the signal delay in one of the channels within 8 ms, you can get the effect of moving the sound source in a stereo panorama.

Echo Using the delay method, the creation of the echo effect is built. In fact, to obtain an echo, it is necessary to impose a time-delayed copy on the original input signal. In order for the human ear to perceive the second copy of the signal as a repetition, and not as an echo of the main signal, it is necessary to set the delay time to approximately 50 ms. In addition, the main signal can be superimposed not just one copy, but several, which will allow the output to obtain the effect of repeated repetition of sound (polyphonic echo). In order for the echo to appear damped, it is necessary to impose on the original signal not just delayed copies of the signal, but also muted in amplitude.

Reverberation. Using the delay, you can achieve the appearance of another interesting effect – reverb (from the English. “Reverberation” – repetition, reflection). The reverb effect is to give the sound volume that is characteristic of a large hall, where each sound generates a corresponding, slowly fading echo. Thus, using reverb, you can “revive”, for example, a phonogram made with a muffled room. The reverberation differs from the “echo” effect in that a delayed in time, not a copy of it, but an output signal is superimposed on the input signal. Such a process occurs as follows. At the first moment of time, the input signal passes to the output unchanged. Then, after the delay time has elapsed, it is removed from the output, its amplitude is multiplied by some coefficient A (usually having a value less than 1, which actually muffles the signal) and is summed with the input signal. And again, after the next lapse of the delay time, the already mixed signal is removed from the output, again multiplied by the coefficient A, and once again summed with the input signal.

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