On the question of sound processing – PART 2

Choir (chorus). As a result of its application, the sound of the signal becomes the sound of a choir or the simultaneous sound of several instruments.

The scheme for obtaining such an effect is similar to the scheme for creating an echo effect with the only difference that the delayed copies of the input signal are subject to weak frequency modulation (on average 0.1 to 5 Hz) before mixing with the input signal. . Increasing the number of voices in the chorus is achieved by adding copies of the signal with different delay times.

Of course, as in all other areas, signal processing also has problems that are something of a hindrance. So, for example, when decomposing signals into a frequency spectrum, there is an uncertainty principle that cannot be overcome. The principle states that it is impossible to obtain an accurate spectral image of the signal at a specific moment in time: either to obtain a more accurate spectral image, it is necessary to analyze a larger time section of the signal or, if we are more interested At the moment when this or that change in the spectrum occurred, we must sacrifice the precision of the spectrum itself … In other words, it is impossible to get the exact spectrum of the signal at one point: the exact spectrum for a much of the signal, or a very rough spectrum, but for a small portion.

Signal processing mechanisms exist in both software and hardware versions (so-called effect processors). For example, voice coders and guitar processors, choruses, and reverbs exist both as hardware and as programs.

Practical signal processing can be divided into two types: on-the-fly processing and post-processing … On-the-fly processing involves instantaneous signal conversion (i.e. with the ability to output the processed signal almost simultaneously with its input ). A simple example is guitar devices or reverb during a live performance on stage. Such processing occurs instantaneously, that is, the performer sings into a microphone and the effects processor transforms his voice and the listener hears the already processed version of the voice. Post-processing is the processing of an already recorded signal. The processing speed can be much slower than the playback speed. Such processing pursues the same objectives, that is, to give the sound a certain character, or to change the characteristics, but it is used in the stage of mastering or preparing the sound for replication, when rush is not required, but quality and study. scrupulous. of all the nuances of sound are most important.

Signal processing is a complex procedure and, most importantly, it is very resource consuming. It has started relatively recently to be carried out on digital devices; Previously, various sound and other effects were achieved by processing sound on analog devices. In analog equipment, sound in the form of electrical vibrations passes through various paths (blocks of electrical elements), thus changing the phase, spectrum and amplitude of the signal. However, this processing method has many disadvantages. First, the quality of the processing suffers, because each analog element has its own error and several dozen elements can critically affect the precision and quality of the desired result. And secondly, and this is perhaps the most important, almost all the effects are achieved by using a separate device, when each of these devices can be very expensive. The possibility of using digital devices has undeniable advantages. The quality of signal processing in them depends much less on the quality of the equipment, the main thing is to qualitatively digitize the sound and be able to reproduce it qualitatively, and then the quality of processing falls only on the software mechanism. In addition, for various manipulations with sound, a constant change of equipment is not required. And, most importantly, since the processing is carried out programmatically, it opens up simply incredible opportunities for you, which are limited only by the power of computers (and it is increasing every day) and human imagination. However, (at least today) there are some problems here. For example, often, even for simple signal processing, it is necessary to decompose them into a frequency spectrum. In this case, on-the-fly signal processing can be difficult precisely because of the resource intensity of the decomposition stage.

On the question of sound processing

Sound processing should be understood as various transformations of sound information to change some characteristics of sound.

Sound processing includes methods for creating various sound effects, filtering, as well as methods for cleaning the sound of unwanted noise, changing the timbre, etc. This whole huge set of transformations ultimately boils down to the following basic types:

Amplitude transformations. They are carried out on the amplitude of the signal and lead to its amplification / attenuation or change according to some law in certain parts of the signal.

Frequency conversions. They are performed on the frequency components of the sound: the signal is presented in the form of a frequency spectrum at regular intervals, the necessary frequency components are processed, for example, filtering and reverse “folding” of the signal from the spectrum. on a wave.

Phase transformations. Phase shift of the signal in one way or another; for example, such transformations of a stereo signal allow the effect of rotation or “three-dimensional” sound to be realized.

Temporary transformations. Implemented by overlaying, stretching / compressing signals; They allow you to create, for example, echo or chorus effects, as well as to influence the spatial characteristics of the sound.

The discussion of each of the named types of transformations can become a complete scientific work. It is worth giving several practical examples of the use of this type of transformation when creating real sound effects:

Echo (echo). It is implemented through temporary transformations. In fact, to get an echo, it is necessary to superimpose a delayed copy of 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. On the main signal, you can superimpose not one copy, but several, which will allow you to obtain the effect of multiple sound repetitions (polyphonic echo) on the output. In order for the echo to appear muffled, it is necessary to superimpose on the original signal not only delayed copies of the signal, but also dull in amplitude.

Reverberation (repetition, reflection). The effect is to add the spaciousness of a large room, where each sound generates a corresponding sound that slowly fades away. In practice, with the help of reverb, you can “revive”, for example, a phonogram made in a muffled room. Reverb differs from the “echo” effect in that a delayed output signal is superimposed on the input signal and not a delayed copy of the input signal. In other words, the reverb block is simplified as a loop, where the output of the block is connected to its input, so that the already processed signal is fed back to the input every cycle, mixing with the original signal.