Anyone who works with professional audio has probably at least once encountered integrated background sound systems. After all, it’s no secret that such small and medium-sized projects can consist of almost O The majority of sales are from the equipment distributor, the dealer, and the installer. And, unlike large systems, “distribution” does not require complex calculations, creation of acoustic models and other routine pre-sale work. An experienced specialist can draw up a standard specification “in his head”, knowing only the overall dimensions of the room. And, of course, such a system will work, but, as the famous joke says, there is one caveat...
Thanks to the successful work of marketers and salespeople, owners and franchisees of cafes, restaurants, shops and shopping centers around the world, and in our country, now fully understand that the right sound is important both for the mood and loyalty of the client, and for the effectiveness of the same advertising content. And, even though I’m speaking now with excerpts from the colorful catalogs of any manufacturer of ceiling speaker systems, we see the results of the work of marketers - all the serious global brands have long entered the Russian market and converted the client to their faith. And a competent business leader in this area has finally stopped neglecting sound quality, as was the case not so long ago.
It would seem that the job is done - create a standard offer and change the number of speaker systems in it depending on the configuration of the room. But it's not that simple. Or rather, it is relatively simple if you approach the construction of systems from the position of the least time spent per unit of goods. And there is logic in this. And the most undeniable argument is “this is not a philharmonic!” - has already become practically textbook, and it is ideally applicable to any object, except, strictly speaking, that very Philharmonic.
Probably some of you will say: “This is idle reasoning about nothing,” so I will finally move on to the main thing.
The main goal of the article is precisely to debunk the widespread opinion that designing a background sound system is not worth any serious time and mental investment. As for time, I partially agree - few of us have enough of it to afford to spend an hour or two choosing one of two adjacent ceiling sections for a loudspeaker. But connecting engineering will help us get better results from the same products as our competitors. And the result, with the right approach, will please both the client and your sales department. Agree that with the current assortment of very similar audio equipment from different manufacturers intended for commercial systems, the main, if not the only, way to attract and retain a client is to offer the most attractive price. And since the rare buyer will be in awe of the sound quality and will be able to objectively evaluate it, in most cases the winner will be the one who offers a more economical solution.
But let's try to abstract from all the commercial components and concentrate on what is near and dear to our hearts - the engineering part.
Engineer, your way out!
There are a thousand and one recommendations for calculating the same ceiling acoustic systems. Let's start with them. What manufacturers don’t offer us to simplify our work... One vendor distributes Talmuds with calculation recommendations among partners, another offers “user-friendly” acoustic simulators in which anyone can draw the desired speaker configuration, a third writes calculator applications in which enough enter the linear dimensions of the room, and you will receive a generated report with a layout diagram. Among the latter, for example, is JBL, which offers its own calculator for almost every product series. This, I admit, is the most convenient, and when used correctly it gives a quick result that is close to reality. But first things first.
I consider it necessary to “take apart” the pros and cons of existing methods.
A method that is undoubtedly autonomous and energy-independent is graphic, similar in principle to constructing a ray sketch. It requires knowing the nominal loudspeaker opening angle and ceiling height. This is what the result looks like:
Rice. 1. Graphic calculation of the pitch of ceiling speakers. A is the distance from the floor to the listener’s ears; B - distance from ears to ceiling; C - loudspeaker opening angle; D is the point of intersection of the beams of neighboring loudspeakers.
Everything is quite simple. The opening angle of the loudspeaker, the height of the listener's ears are graphically depicted (it is customary to take 1-1.2 meters for a person in a sitting position and 1.5 meters for a person in a standing position), and the point of intersection of the horizontal and the rays of the opening angle is considered the critical point that the ray from the neighboring one must intersect. loudspeaker. In this way, the pitch of the acoustic systems is determined.
Now let's dig a little deeper. It is known that the opening angle value indicated in the loudspeaker passport is nominal, i.e. averaged over a frequency band determined by the manufacturer at its discretion. And it’s no secret that the directional properties of any real emitter vary significantly in different frequency bands. As a result, we perform calculations, sometimes without even knowing in what range we received the correct coverage. So, colleagues, be careful - having made such a calculation using the nominal opening angle, you may well get “holes” in frequency bands, for example, above 8-10 kHz.
Now one more nuance. The nominal opening angle is usually calculated from the polar diagrams in such a way that when deviating away from the radiation axis by ½ of the declared opening angle, the pressure level drop will be 6 dB. Moreover, again, attention, at an equal distance from the emitter.
Rice. 2. Graphic calculation of the pitch of ceiling speakers. A is the distance from the floor to the listener’s ears; B - distance from ears to ceiling; C - loudspeaker opening angle; D is the point where the sound pressure level drops by 6 dB
It turns out that at the point of intersection of the horizontal and the beam, the drop will no longer be 6 dB, but more. Well, it’s okay, we arm ourselves with a compass and solve the problem.
However, this is also not all. What do you think, when we cross the beams from neighboring loudspeakers at the right point, what pressure will we get there? Having 2 waves with a pressure level of -6 dB SPL relative to the radiation axis, we can add them according to the rule of energy summation (L1, p. 33) as two equal pressures and get a sum equal to -3 dB relative to the axis. However, this rule works in the case of incoherent addition, i.e. for example, at unequal distances from the sources, but at the point of intersection of the rays the waves are coherent (in phase), and only at this point do they add up in the entire spectrum, giving a doubling of the pressure, i.e. it will be almost the same as on the radiation axis. The figure below shows the calculation result in a model with two closely spaced ceiling speakers.
Rice. 3. Calculation of sound pressure level using two ceiling speakers in an octave band centered at 500 Hz.
As a result, this is the picture we get: coherent addition of waves exactly between the loudspeakers always exists and gives an increase of up to +3 dB over a fairly small area, and literally centimeters from this “seam” the waves are added incoherently and a drop in pressure is observed. And I’ll immediately explain that it will not be possible to completely get rid of this “seam”. Below are the results of acoustic simulations with different speaker pitches.
Rice. 4. Sound pressure diagram when the speakers are located at a height of 3 meters from the floor with a step of 1.5 meters. The calculation is made in third-octave bands of 10 kHz (lower diagram) and 400 Hz (upper diagram).
Rice. 5. Sound pressure diagram when the speakers are located at a height of 3 meters from the floor with a step of 3 meters. The calculation is made in third-octave bands of 10 kHz (lower diagram) and 400 Hz (upper diagram).
Rice. 6. Sound pressure diagram when the speakers are located at a height of 3 meters from the floor with a step of 4.5 meters. The calculation is made in third-octave bands of 10 kHz (lower diagram) and 400 Hz (upper diagram).
Awl or soap?
Well, the result of the simulation showed that a negative result for coverage uniformity is produced by either too large a speaker pitch or too small one. And just too small a distance is perhaps a more serious problem, because there is a common misconception that by placing speaker systems with a minimum pitch, we will get uniform coverage over the entire frequency range. For the high-frequency region, this thesis is true, since any loudspeaker has a narrower radiation pattern in the high-frequency region. As for the incoherent addition of waves, thanks to interference in the low-frequency region, the pressure at the points where the beams intersect will be guaranteed to be greater than directly under the loudspeaker, no matter how paradoxical this may sound. Moreover, the interference pattern will change at each point, and the closer the speakers are located to each other, the more dramatic these changes will be. So is uniform coverage in the high frequencies worth such sacrifices? Don't think.
To make it a little clearer, I’ll make some clarifications. As is known, the direction of a wave depends on its length - long waves (with a frequency of 160 Hz and below) are omnidirectional, i.e. The opening angle of any loudspeaker at a frequency of, for example, 80 Hz will be equal to 360 degrees. In the case of ceiling systems, of course, 180 degrees. Short waves have a narrower directionality, which is due to the physics of the wave propagation process. Thus, in the 16 kHz octave band, an average ceiling loudspeaker can have an opening angle (at -6 dB) of 45-60 degrees with a nominal nominal 120 degrees, averaged over the range 1 kHz-8 kHz. It turns out that in order to avoid “sound holes”, the calculation should be carried out taking as a basis the opening characteristic of the loudspeaker at high frequencies. Right. Only not so narrowly directed long waves will create incomparably greater pressure, add and subtract many times, creating the sums and differences illustrated above in order to O The greater the pressure spread, the closer to each other their sources are located.
Based on what you have read, you have every right to blame me for not giving an obvious answer to how exactly to position the loudspeakers correctly. True, but if there was a clear answer, there would be no need for our services and anyone could design a sound system. This is precisely where the mastery of, as it is now called, “system design” lies - in finding a compromise solution, in balancing between mutually exclusive requirements and conditions.
Otherwise, beautiful Marquise, everything is fine, everything is fine!
Perfectionism isn't a bad trait, but sometimes being productive requires an achievable benchmark. And we have it too. In quantitative assessment of the uniformity of the sound field, the so-called so-called in statistics helps a lot. Standard Deviation (STDev). I will not go deep into the explanation of this concept - there is a high chance of going too deep.
Rice. 7. Standard deviation
Before us is a graph of the distribution of certain random variables within a standard deviation from the mathematical expectation. Let's take it as a basis, using the distribution of sound pressure levels in the room as values.
Now let’s agree that the value of μ on the horizontal scale is the average value of the sound pressure level throughout the entire room, namely, our mathematical expectation. We take the value of σ as 2 dB (-20% +25% in absolute value), since the probable spread of values relative to the expected may be different. Now our task is to understand which spread will satisfy us and which will be considered unacceptable. If the pressure is the same over the entire measured area, the graph will turn into a straight line. The greater the spread of values, the steeper the rise and fall of the graph of this function will be. So, with a fairly uniform sound field, most quantities are concentrated near the average value. And we can consider this fairly uniform coverage to be a zone within the 1st standard deviation, i.e. if on 68% of the entire area of the room the pressure level fluctuates within +-2 dB from the average over the full frequency range, then the requirement is met. True, such pressure distribution statistics can only be seen by performing an acoustic calculation.
Despite the fact that such an interpretation is not fixed in the ISO or AES standards, it is often used in practice and generally reflects reality, therefore it can serve as a good guide and starting point for you in determining the uniformity of area coverage.
But do not forget that the average value over the entire range does not always describe the complete picture.
Black box
Well, the ceiling speakers seem to have been sorted out, to the extent possible in this format. What about wall systems? Is everything as simple with them as we used to think? In general, it is much simpler simply because, as a rule, we are extremely limited in the placement of cabinet speaker systems - walls, corners, columns. And at the same time, not every point on the wall is accessible for installing a loudspeaker - somewhere there is designer stucco, somewhere there is a TV, somewhere there is ventilation, and so on.
And it’s one thing when you need to voice 100 square meters. meters - I selected the opening angle, scattered 4 loudspeakers in the corners, and that's it, the system is ready - but what to do with a larger area? We look for load-bearing columns in the middle of the room, rejoice at their presence and cover them with loudspeakers. Well, what to do - there are no options. I agree, but with clarifications. For the answer, as always, you should turn to science.
Here is an example of the placement of speaker systems in a room.
Rice. 8. Position of wall speakers on columns
In a general sense, everything is fine, and with the right choice of speakers and proper installation there will be no problems. Looking ahead, I will say that all of the layout plans I presented below have the right to exist, but with some reservations.
If the speakers are full-range, with an opening of a crazy 150 degrees (and this happens), placing them in close proximity to each other will create a very interesting interference pattern for you. In order not to rant for a long time, this time I will immediately demonstrate the acoustic calculation, since it is difficult to come up with something more visual and easy to understand.
Rice. 9. Diagram of sound pressure level when loudspeakers are located on columns in an octave band centered at 500 Hz
Pay attention to the resulting “petals” - this is precisely the result of the addition and subtraction of two coherent waves, and their location, of course, changes depending on the wavelength. The same picture can be observed when placing loudspeakers in clusters - for the correct addition of waves, a number of measures need to be taken both during design and during setup, but that is a completely different story. Just in case, I will point out one obvious consequence of this fact: as a result of interference, the timbre of a sound program can be seriously distorted due to the subtraction of certain frequency components. Unfortunately, many specialists are confident that any timbral distortion can be corrected using a measuring microphone, spectrum analyzer and equalizer, and are sincerely surprised when they try to “pull out” the frequency lost due to interference when adjusting the frequency response of the system. But nothing happens on the graph, no matter how much you increase the filter gain - by +6 dB, by +12 dB, or even turn on two equalizers in series. There is simply no pressure at this frequency, and there is nowhere for it to come from if, for one of many reasons, subtraction of waves has occurred in this range.
Now let’s take it and try to get rid of these problems, and even make the system cheaper by reducing the number of speakers.
Rice. 10. Position of wall speakers on columns
Rice. 11. Diagram of sound pressure level when loudspeakers are located on columns in the full frequency range.
It turns out quite well: interference problems have been solved, the coverage in the area between the columns is close to ideal, coherent addition of waves is also not critical. As a budget option, this design is quite viable - the main thing is that the pitch of the columns allows you to meet the standard deviation. But there is still a certain nuance. And its root is buried deep in fundamental science.
Thanks to the physiology of hearing and, probably, evolution, humans are able to localize sound events, i.e. determining where a sound wave came from - this ability simply had to be developed for survival. But what about when there are a lot of sound waves, as, for example, in a primeval cave, where in addition to the direct sound from the source, there are countless reflections coming from all sides? Very simple. It was enough to develop the ability to determine the direction of the first wave, which would definitely arrive along the shortest path directly from the conditional mouth of the predator, and any reflection would definitely travel a longer distance and arrive with some delay. This phenomenon is described by the Law of the First Wavefront (aka Precedence Effect). In the presence of several identical waves arriving with a delay, the brain determines the direction solely by the first wave, even if the second and subsequent ones have a higher level (exceeding up to 10 dB) and arrive with a delay of up to 30 ms. You can read more about this interesting effect and its description in the literature on psychoacoustics.
So what's all this for? Now let's simulate a listener moving along the length of the room along a straight path, and see how the localization of sound will change for him. While moving past the first loudspeaker, a person will clearly hear the sound on the left; as he approaches the conventional opening boundary, the ratio of wave intensities on the left and right changes, since the second loudspeaker appears in the field of view. Our object reached the point of equal distance between the loudspeakers and both waves coherently added up, giving it +3 dB to the pressure level, and the sound localization instantly jumped to the point of equal distance between the sources, i.e. exactly in the place where the object’s head is currently located. And the next step will sharply shift the sound event to the right, since the wave from the second source will now arrive first.
In principle, there is nothing critical about this. But if customers are expected to constantly move around the area, as, for example, in a store, will they be comfortable listening to the sound jumping from point to point? Not every listener analyzes the causes of his discomfort and associates them with sound; the perception of the environment for him is formed unconsciously and consists of the totality of all sensations - visual, auditory, tactile and others. And it is enough for at least one of them to cause discomfort for the rest to turn out to be insignificant, and the subjective impression is spoiled.
At the finish line
Perhaps, the main issues of calculating the location of loudspeakers have been considered, but it would not be entirely fair on my part not to mention that almost all of these calculations take into account the energy of the direct wave from the emitter. And in real rooms, filled not only with direct sound, but also with numerous reflections, interference subtractions, of course, will not create points with zero sound pressure. The reflected waves will somewhat level out the dips and rises, of course, without eliminating them completely, and will significantly improve the uniformity of the coverage, compensating for the lack of direct sound at points remote from its source.
By the way, one of the interesting methods of creating non-localizable background sound of a system is based on the use of room reverberation to benefit the background sound. It consists in placing all speaker systems “facing” the ceiling. This arrangement almost completely relieves the listener of direct sound from the loudspeaker; all the energy he receives is a multitude of reflected waves from all directions. The resulting effect is extremely interesting in terms of spatial sound. The only disadvantage of this solution is the content limitation. Fast pop or rock music that isn't designed to take that much reverberation is unlikely to sound good from such a system.
P.S. What, it won’t sing without a cable?
Despite the seeming minor issue of cable routes, it is difficult to overestimate the importance of speaker (acoustic) cable for any sound system. I say this with complete confidence, because, unfortunately, in my practice it is not always possible to dictate to a client which cable to purchase, and this sometimes leads to silent scenes in the style of Chekhov’s Inspector General, when the site learns that a cable has been laid for the sound system ShVVP cable. In response to my question, I receive a completely reasonable answer - “Well, it works!” Works. It just works that way, it would be better if it didn't work. In general, you understand...
And that is why I present a method for calculating the cable cross-section. Those of you for whom it is obvious, and who know perfectly well how such calculations are made, can safely skip this part of the article - I will not bring anything new and hitherto unknown to science. But if suddenly you are faced with the need for calculation for the first time, then this information will be useful due to its practical applicability.
Effective current calculation:
Calculation of the effective power allocated to the load:
100V line.
Calculation of the total resistance of loudspeakers in a line:
,Where
Number of speakers per line
- rated power of one loudspeaker (Tap setting)
The remaining calculations are performed similarly to low-impedance lines.
The total load resistance in a 100-volt line, as you can see, is usually at least 1000 ohms. With such high resistance, unit ohms of cable resistance have little effect on the overall line resistance, and therefore increase power losses only slightly compared to a low resistance connection.
Now a little about the interpretation of the results. How to determine how much power loss is acceptable? In general, the threshold value for the drop in power level on a cable is considered to be 0.5 dB. This corresponds to a loss of 10% relative to the rated power. For example, for an 8-ohm loudspeaker with a permissible rating of 1 kW, the maximum power drop according to these standards reaches a line with a cross-section of 2.5 sq. mm and a length of 30 meters. Whether this is a lot or a little, of course, is up to you to decide, and the decision here depends on the specific situation, but practice shows that increasing the cable cross-section from 2.5 sq. mm to, for example, 4 sq. mm will not significantly increase the cost of installation. Therefore, I always recommend keeping within 0.5 dB, because it is not at all difficult to do. And why should we waste precious Watts on the line when we have the opportunity to achieve maximum system efficiency?
And, despite the fact that the requirements for broadcast lines are significantly lower, using the right cable will help you make the system work more efficiently. Moreover, if in your practice you have not conducted experiments to evaluate the sound quality on different cables (other things being equal), then take my word for it, the influence of the cable cross-section on the sound is really noticeable by ear. This is especially true for the low-frequency region - the range during the transmission of which the greatest power develops, and which is most demanding in terms of current and damping factor.
Therefore, using the analogy so beloved by many, let’s not fill the Mercedes S-Class with 92-octane gasoline and then wonder why the declared performance is not achieved.
As you can see from the formulas, the only quantity that remains unknown for calculating the cable is its resistance, expressed in Ohm/km. Its meaning can be found in the cable specification. To do this, you will first have to select the cable cross-section offhand, take the corresponding resistance value, substitute it into the formula and carry out the calculation. If you get an excess power drop, or vice versa, the cross-section turns out to be excessive, you will have to select a cable of a different cross-section and return to the starting point of the calculation. I usually recommend starting the calculation with a cross section of 2x2.5 sq.mm (7.5-8 Ohm/km) for low-impedance lines and 2x1.5 sq.mm (about 13 Ohm/km) for transformer lines. Of course, this will force you to spend some time on calculations, but for convenience you can create a calculator for yourself in Excel, entering formulas and resistance values for cables of different cross-sections - this will take some time one-time, but will eliminate the need for manual calculations in the future.
We thank DIGIS for the materials provided
Sound system
(Greek sustnma, German Tonsystem) - high-altitude (interval) organization of music. sounds based on k.-l. a single principle. At the heart of Z. s. There is always a series of tones that are in certain, measurable relationships. The term "Z.s." used in various meanings:
1)
sound composition, i.e. a set of sounds used within a certain interval (often within an octave, for example, five-sound, twelve-sound systems);
2)
a certain arrangement of the elements of the system (the sound system as a scale; the sound system as a complex of sound groups, for example, chords in the tonal system of major and minor);
3)
a system of qualitative, semantic relationships, functions of sounds, developing on the basis of a certain principle of connection between them (for example, the meaning of tones in melodic modes, harmonic tonality);
4)
structure, mathematical expression of relationships between sounds (Pythagorean system, equal-tempered system).
Basic the meaning of the concept of z.s. associated with the sound composition and its structure. Z.s. reflects the degree of development, logical. coherence and orderliness of music. thinking and historically evolves with it. Evolution of the earth's system, in real history. a process carried out in a complex way and replete with internal contradictions, on the whole definitely leads to a refinement of sound differentiation, an increase in the number of tones included in the system, strengthening and simplifying the connections between them, and the creation of a complex branched hierarchy of connections based on sound kinship.
Logical scheme of development of z. s. only approximately corresponds to the concrete historical. the process of its formation. Z.s. in own In a sense, it is genetically preceded by primitive glissanding, devoid of differentiated tones, from which reference sounds are just beginning to stand out.
The chant of the Kubu tribe (Sumatra) is the love song of a young man. According to E. Hornbostel.
The lower form of Z. s. that replaces it. represents the singing of one reference tone, foundation (
), adjacent (
) above or below.
RUSSIAN FOLK PRACTICE
KOLYADNAYA
The adjacent tone may not be stable at a certain height or may be approximate in pitch position.
Further growth of the system makes possible the progressive, cantilinal movement of the melody (in the conditions of a five-, seven-step system or some other structure of the scale) and ensures the coherence of the whole due to the reliance on sounds that are in the highest relationship with each other. Therefore, the next most important stage in the development of agricultural systems. - “the era of the fourth,” filling the gap between the sounds of the “first consonance” (the fourth turns out to be the sound least distant from the original reference tone, being in a relationship of perfect consonance with it; as a result, it gains an advantage over other, even more perfect consonances - octave, fifth) . Filling the fourth forms a series of sound systems - semitone trichords and several tetrachords of various structures:
TRICHORDS
TETRACHORDS
LULLABY
EPIC CHUNT
At the same time, adjacent and passing tones are stabilized and become supports for new adjacent ones. On the basis of the tetrachord, pentachords and hexachords arise:
MASLENICNA
ROUND DANCE
From the combination of trichords and tetrachords, as well as pentachords (in a fused or separate way), composite systems are formed that differ in the number of sounds - hexachords, heptachords, octachords, which in turn are combined into even more complex, multi-component sound systems. octave and non-octave:
PENTATONIC
UKRAINIAN VESNYANKA
DANCER
PLACE OF THE Znamenny CHANT
RUSSIAN FOLK SONG
FOR THE Nativity OF THE VIRGIN, ZNAMENNY CHANT
HEXACHORD SYSTEM
Theoretical generalization of the practice of introducing tone in Europe. music of the late Middle Ages and the Renaissance (“musica ficta”), when whole-tone conclusions and whole-tone sequences were increasingly systematically replaced by half-tone ones (e.g., instead of
c-d
e-d
move
cis-d
e-d),
expressed in the form of chromatic-enharmonic. seventeen-step scale (by Prosdocimo de Beldemandis, late 14th - early 15th centuries):
The development of polyphony and the emergence of consonant triads as the main element of the sound system. led to its complete internal reorganization - the grouping of all the tones of the system around this supporting consonance, which acts as a center, tonic. triad (tonic), and in the form of its animations on all other levels of diatonic. scales:
The role of the constructive factor of Z. s. gradually transitions from fret melodious. models to chord-harmonic ones; in accordance with this Z. s. begins to be presented not in the form of a scale (“ladders of sounds” - scala, Tonleiter), but in the form of functionally related sound groups. As at other stages of the development of the Z. system, all the most important features of the earlier forms of the Z. system. are also present in more highly developed systems. - melodic energy. linearity, microsystems of the reference tone (foundation) and adjacent ones, filling the fourth (and fifth), animation of tetrachords, etc. Complexes belonging to a single centralization. the whole sound groups - chords at all levels - together with certain scales become a new type of chords - harmonious. tonality (see note above), and their ordered combination constitutes a “system of systems” of major and minor keys at each of the chromatic levels. scale. The total sound volume of the system theoretically extends to infinity, but is limited by the capabilities of pitch perception and represents a chromatically filled range ranging from approximately A2 to C5. The formation of the major-minor tonal system in the 16th century. required the replacement of the Pythagorean tuning in pure fifths (for example, f - c - g - d - a - e - h) with fifth-third tuning (the so-called pure, or natural, tuning of Fogliani - Zarlino), using two tunings. interval - a fifth of 2:3 and a major third of 4:5 (for example, F - a - C - e - G - h - D; large letters indicate primes and fifths of triads, small letters indicate thirds, according to M. Hauptmann). The development of the tonal system (especially the practice of using different keys) created the need for an equal-tempered system.
Contact of different elements tonalities leads to the establishment of connections between them, to their rapprochement and further to fusion. Together with the counter process of growth of intra-tonal chromaticity (alteration), the merging of multi-tonal elements leads to the fact that within the limits of one key any interval, any chord and any scale from each degree are fundamentally possible. This process prepared a new reorganization of the structure of the Z. s. in the works of a number of composers of the 20th century: all levels of chromatic. their scales are emancipated, the system turns into a 12-step one, where each interval is understood directly (and not on the basis of fifths or fifth-terts relations); and the original structural unit of the earth's system. becomes a semitone (or major seventh) - as a derivative of the fifth and major third. This makes it possible to construct symmetrical (for example, terzochromatic) modes and systems, the emergence of a tonal twelve-step, the so-called. “free atonality” (see Atonal music), serial organization (in particular, dodecaphony), etc.
Non-European Z. s. (for example, countries of Asia, Africa) sometimes form varieties that are far removed from European ones. Thus, the more or less usual diatonic scale of Indian music is decorated with intonation. shades, theoretically explained as the result of dividing the octave into 22 parts (the shruti system, also interpreted as the totality of all possible heights).
In Javanese music, the 5- and 7-step "equal" octave divisions (slendro and pelog) do not coincide with the usual anhemitonic pentatonic scale, nor with the fifth or fifth-tert diatonic scale.
Literature : Serov A. N., Russian folk song as a subject of science (3 articles), “Musical Season”, 1869-70, No. 18, 1870-71, No. 6 and 13, reprint. in his book: Selected articles, vol. 1, M.-L., 1950; Sokalsky P. P., Russian folk music?, Khar., 1888, Peter V. I., On compositions, tunings and modes in ancient Greek music, K., 1901 Yavorsky B., The structure of musical speech, vol. 1-3, M., 1908, Tyulin Yu. N., The doctrine of harmony, L., 1937, M, 1966; Kuznetsov K. A., Arabic music, in: Essays on the history and theory of music, vol. 2, L., 1940; Ogolevets A. S., Introduction to modern musical thinking, M.-L., 1946; Musical acoustics. General Ed. N. A. Garbuzova, M, 1954; Jami A., Treatise on Music. Ed. and comments by V. M. Belyaev, Tash., 1960; Pereverzev N.K., Problems of musical intonation, M., 1966; Meshchaninov P., Evolution of pitch fabric (structural-acoustic justification...), M., 1970 (manuscript); Kotlyarevsky I., Diatonics and chromatics as a category of musical thinking, Kipv, 1971; Fortlage K., Das musikalische System der Griechen in seiner Urgestalt, Lpz., 1847, Riemann H., Katechismus der Musikgeschichte, Tl 1, Lpz., 1888, Russian. lane - Catechism of the history of music, part 1, M., 1896), by him, Das chromatische Tonsystem, in his book: Präludien und Studien, Bd I, Lpz., 1895, Emmanuel M., Histoire de la langue musicale, v . I-II, R., 1911; Haba A., Harmonische Grundlagen des Vierteltonsystems, Prag, 1922; Ellis A. J., ber die Tonleitern verschiedener Völker, in the book: Abhandlungen zur vergleichender Musikwissenschaft Munch., 1922; Stumpf C., Tonsystem und Musik der Siamesen ibid., Abraham O., Hornbostel E. M., Tonsystem und Musik der Japaner, ibid Hornbostel E. M., ber die Musik der Kubu, ibid., Musikalische Tonsysteme, in the book: Handbuch der Physik hrsg. von H. Geiger und K. Scheel, Bd VIII. Akustik, B., 1927; Farmer H. G., A history of Arabian music to the XIII century, L., 1929; Hornbostel E. M., Lachmann R., Das indische Tonsystem bei Bharata und sein Ursprung "Zeitschrift für vergleichende Musikwissenschaft", Jahrg. 1, No. 4, 1933; Gombosi O. J., Tonarten und Stimmungen der antiken Musik, Kph., 1939; Strunk O., The tonal system of Byzantine music, "MQ", v. XXVIII, 1942, No. 2 Danckert W., Der Ursprung der halbtonlosen Pentatomk, in the book: Fes schritt Z. Kodbly, Bdpst, 1943; Szabolcsi B., Five-tone scales and civilisation, "Acta musicologica", XV, 1943, p. 24-34; Handschin J., Der Toncharakter, Z., 1948; Kunst J., Music in Java, v. 1-2, The Hague, 1949; Hood M. , The nuclear theme as a determinant of Patet in Javanese music, Groningen (Djakarta), 1954; Schneider M., Die Entstehung der Tonsysteme, in: Kongress-Bericht Hamburg. 1956, Kassel-Basel, 1957; Wiora W., Alter als Pentatomk, in the book: Studia memoriae Belae Bartuk Sacra, Bdpst, 1957, p. 185-208, Bardos L., Natürliche Tonsysteme, ibid., p. 209-48, Avasi B., Tonsysteme aus Intervall-Permutationen, ibid., p. 249-300, Smits van Waesberghe J., Antike und Mittelalter in unserem Tonsystem, "Musica", Jahrg. XII, 1958, H. 11, Sachs S., Vergleichende Musikwissenschaft. Musik der Fremdkulturen, Hdlb., 1959; Spiess L. B., The Diatonic "Chromaticism" of the Enchiriadis treatises, "Journal of the American Musicological Society", v. XII, 1959, No. 1, Husmann H., Grundlagen der antiken und orientalischen Musikkultur, B., 1961; Vogel M., Die Entstehung der Kirchentonarten, in the book: Kongress-Bericht Kassel 1962, (Kassel, 1962), his, An den Grenzen des Tonsystems, "Musica", Jahrg. XVII, 1963; H. 4, Kraehenbuehl D., Schmidt Chr., On the development of musical system, "Journal of Music Theory", v. VI, 1962 No. 1, Apfel E., Spatmittelalterliche Klangstruktur und Dur-Moll-Tonalitat, "Die Musikforschung", Jahrg. XVI, 1963, H. 2 Dahlhaus K., Untersuchungen ьber die Entstehung der harmonischen Tonalltät, Kassel - (u.a.), 1968; Manik L., Das arabische Tonsystem im Mittelalter, Leiden, 1969. Yu. N. Kholopov.
Musical encyclopedia. - M.: Soviet Encyclopedia, Soviet composer. Ed. Yu. V. Keldysh. 1973-1982 .
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for IBM PC
INTRODUCTION
The interaction of a person with a computer must first of all be mutual (that’s why it is communication). Reciprocity, in turn, provides for the possibility of communication between both a person and a computer, and a computer with a person. It is an indisputable fact that visual information, supplemented by sound, is much more effective than simple visual influence. Try, covering your ears, to talk with someone for at least a minute; I doubt that you will get much pleasure, as well as your interlocutor. However, so far many orthodox programmers/designers still do not want to admit that sound influence can play the role of not only a signaling device, but an information channel, and accordingly, due to inability and/or reluctance, they do not use in their projects the possibility of non-visual communication between a person and a computer, but even they never watch TV without sound. Currently, any large project that is not equipped with multimedia tools (hereinafter, by the word “multimedia tools” we will primarily understand a set of hardware/software tools that complement the traditionally visual ways of human interaction with a computer) is doomed to failure.
BASIC SOUNDING METHODS
There are many ways to make a computer talk or play.
1. Digital to Analogue (D/A) conversion. Any sound (music or speech) is contained in the computer memory in digital form (in the form of samples) and, using a DAC, is transformed into an analog signal, which is fed to amplifying equipment, and then to headphones, speakers, etc.
2. Synthesis. The computer sends musical notation information to the sound card, and the card converts it into an analog signal (music). There are two synthesis methods:
a) Frequency Modulation (FM) synthesis, in which the sound is reproduced by a special synthesizer that operates on the mathematical representation of a sound wave (frequency, amplitude, etc.) and from the totality of such artificial sounds, almost any necessary sound is created.
Most systems equipped with FM synthesis show very good results when playing “computer” music, but trying to simulate the sound of live instruments does not work very well. The disadvantage of FM synthesis is that with its help it is very difficult (almost impossible) to create truly realistic instrumental music, with a large presence of high tones (flute, guitar, etc.). The first sound card to use this technology was the legendary Adlib, which used a Yamaha YM3812FM synthesis chip for this purpose. Most Adlib-compatible cards (SoundBlaster, Pro Audio Spectrum) also use this technology, only on other more modern chip types, such as the Yamaha YMF262 (OPL-3) FM.
b) synthesis using a wave table (Wavetable synthesis), with this synthesis method, a given sound is “drawn” not from the sines of mathematical waves, but from a set of actually voiced instruments - samples. Samples are saved in RAM or ROM of the sound card. A special sound processor performs operations on the samples (using various kinds of mathematical transformations, the pitch and timbre are changed, the sound is supplemented with special effects).
Since samples are digitizations of real instruments, they make the sound extremely realistic. Until recently, this technique was only used in high-end instruments, but it is becoming increasingly popular now. An example of a popular card using WSGravis Ultra Sound (GUS).
3. MIDI. The computer sends special codes to the MIDI interface, each of which indicates an action that the MIDI device (usually a synthesizer) should perform (General) MIDI is the basic standard for most sound cards. The sound card independently interprets the sent codes and matches them with sound signals (or patches) stored in the card’s memory. The number of these patches in the GM standard is 128. On PC-compatible computers, there have historically been two MIDI interfaces: UART MIDI and MPU-401. The first is implemented in SoundBlaster's cards, the second was used in early Roland models.
SOUND CAPABILITIES OF THE IBM PC FAMILY
Already on the very first IBM PC models there was a built-in speaker, which, however, was not designed for accurate sound reproduction: it did not provide reproduction of all frequencies in the audible range and did not have sound volume controls. And although the PC speaker has been preserved on all IBM clones to this day, it is more a tribute to tradition than a vital necessity, because the speaker has never played any serious role in communication between a person and a computer.
However, already in the PCjr model a special sound generator TI SN76496A appeared, which can be considered a harbinger of modern sound processors. The output of this sound generator could be connected to a stereo amplifier, and it itself had 4 voices (not an entirely correct statement - in fact, the TI chip had four independent sound generators, but from the programmer’s point of view it was one chip with four independent channels ). All four voices had independent control of volume and frequency. However, due to marketing errors, the PCjr model never became widespread, was declared unpromising, was discontinued and support for it was discontinued. From that moment on, IBM no longer equipped its computers with sound tools of its own design. And from that moment on, the place Sound cards have firmly occupied the market.
SOUND CARDS REVIEW
A kind of “bastard son” of the PC and a person’s desire to hear decent sound with a minimum of financial costs. Covox is not without reason called the “SoundBlaster for the poor” because its cost is an order of magnitude lower than the cheapest sound card. The essence of Covox is extremely simple - any standard IBM-compatible machine must have a parallel port (usually it is used for a printer). 8-bit codes can be sent to this port, which, after simple mixing at the output, will give a completely satisfactory mono sound.
Unfortunately, due to the fact that the main software manufacturers ignored this simple and ingenious device (collusion with sound card manufacturers), covox never received any software support. However, it is not difficult to write a driver for covox"a yourself and replace with it the driver of any 8-bit sound card that is used in DAC mode, or slightly change the program code by redirecting 8-bit digitization, say, to the 61st port of the PPI .
The SoundBlaster Pro (SB-pro) The Creative Labs "SoundBlaster (SB) was the first Adlib-compatible sound card that could record and play 8-bit samples, supported FM synthesis using the Yamaha YM3812 chip. The original SB mono model was equipped with one such chip, and the newer stereo model was equipped with two. The most advanced model of this family is the SB-pro. 2.0, this card contains the most advanced FM synthesis chip (OPL-3 standard). The SB-pro is capable of digitizing/ playback of real sound with a frequency of up to 44.1 Hz (CD player frequency) in stereo mode. Also, with the help of external drivers, this card supports the General MIDI interface. Contains a built-in 2-Watt preamplifier and a CDD controller (usually Matsushita).
External line in.
SB compatible MIDI,
SB CD-ROM interface.
The SB-pro was fully compatible with the Adlib card, which made it a stunning success in the low-cost home audio market (primarily for games). And although professionals were dissatisfied with the unnatural “metallic” sound, and the MIDI simulation left much to be desired, this card appealed to numerous fans of computer games, who encouraged developers to add support for SundBlaster cards into their games, which finally cemented Creative Labs’ leadership in the market . And now any program that claims to produce sound on something other than a PC speaker is simply obliged to support SB, which has become the de facto standard. Otherwise, she risks simply not being noticed.
SoundBlaster 16 (SB 16) is an improved version of SB-pro, which is capable of recording and playing back 16-bit stereo sound. And of course SB16 is fully compatible with Adkib & SB. The SB-16 is capable of playing 8- and 16-bit stereo samples at frequencies up to 44.1 KHz with dynamic audio filtering (this card allows you to suppress unwanted frequency ranges during playback). The SB16 can also be equipped with a special ASP (Advanced (Digital) Signal Processor) chip, which can perform audio compression/decompression on the fly, thereby relieving the CPU for other tasks. Like the SB-pro, the SB-16 performs FM synthesis using a Yamaha YMF262 (OPL-3) chip. It is also possible to additionally install a special WaveBlaster expansion card, which provides higher quality sound in General MIDI mode.
Pro Audio Spectrum Plus and Pro Audio Spectrum 16 The Media Vision's
Pro Audio Spectrum Plus and -16 (PAS+ and PAS-16), this is one of many attempts to expand the family of SB-like cards. Both cards are almost identical, except that the PAS-16 supports 16-bit sampling. Both cards are capable of increasing the playback frequency to 44.1 KHz and dynamically filtering the audio stream. Like the SB-pro and SB-16, the PAS provides FM synthesis via a Yamaha YMF262 (OPL-3) chip.
Supported input devices:
External line in.
PC speaker (wow!).
Supported output devices:
Audio line out (headphones, amplifier),
SCSI (not just for CD-ROM, but also for tape-streamers,
optical drives, etc),
General MIDI (requires optional MIDI Mate),
Although Media Vision claims that its products are fully SB compatible, this is not entirely true and many people have had unpleasant surprises with this card when trying to use it as an SB. However, this is somewhat compensated for by excellent stereo sound and very low noise levels.
The Gravis UltraSound
The Advanced Gravis"
Gravis UltraSound (GUS) is the undoubted leader in the field of WS synthesis. A standard GUS has 256 or 512 kilobytes of memory “on board” for storing samples (also called patches), with the help of playing which GUS generates all sound effects and music. GUS can operate at sampling rates up to 44.1 KHz and can provide 16-bit stereo audio. Recording is somewhat more complicated - initially standard GUS models only provided 8-bit audio recording, but new models (GUS MAX) are capable of 16-bit recording. In general, the sound reproduced by GUS is more realistic (due to the use of WS synthesis instead of FM), and of course GUS provides excellent support for General MIDI due to the fact that it does not need to “construct” all the variety of sounds from the set sine waves - he has at his disposal a special library of about 6M in size, from which he can load instruments during playback.
Supported input devices:
Audio Line In.
Supported output devices:
Audio Line Out
Amplified Audio Out
Speed compensating joystick (up to 50 Mhz),
General MIDI (requires optional MIDI adapter),
SCSI CD-ROM (requires optional SCSI interface card).
GUS is not an SB compatible card and does not support the SB or Adlib standard. Some compatibility, however, can be achieved by software emulation using special SBOS (Sound Board Operating System) drivers supplied with GUS. However, in practice, satisfactory operation of SBOS is more an accidental phenomenon than a natural one. In addition, SBOS significantly slows down the processor , which makes the GUS practically unsuitable for running multimedia applications written exclusively for SB. Yet the exceptional sound qualities of the GUS have forced software manufacturers to include drivers for this card in their products. And although support for the GUS standard has not yet become as common as support for the SB standard, there is no doubt that the second most important card after SB is the GUS card.
The problems of promoting GUS to the modern gaming market are complicated by the fact that currently 45% of games are written in Miles Design AIL 2.0 - 3.15, 50% in HMI SOS 3.0 - 4.0, and the remaining 5% in homemade sound libraries. I only learned how to support GUS with AIL 3.15 and then only almost. Before this (AIL 3.0-, HMI 4.0-), before loading the game, LOADPATS.EXE or something similar (MEGAEM...) was launched, which loads all (!!!) timbres that this game uses (and in total there are 512 -and the GUS kilobyte memory can accommodate 30-50 timbres), in AIL 3.15 it is a little more humane - timbres are loaded as needed (almost) but not unloaded (!!), so the situation is reduced to the previous one. I’m not saying that original timbres are used by rare units of manufacturing companies and I understand the rest very well - for the sake of one GUS, there is no point in buying timbres and “pulling” the music. Not to mention the problems manufacturers have with creating music for standard tones and figuring out how to cram them into 512/256K.
The Roland LAPC-1 and SCC-1
The Roland LAPC-1 is a semi-professional sound card based on the Roland MT-32Module. LAPC is identical to the MIDI interface on PC cards. It contains 128 instruments. LAPC-1 uses a combined method of constructing the sound of a note: each note consists of 4 "partials", each of which can be a sample or a simple sound wave. The total number of partials is limited to 32, therefore only 8 instruments can play simultaneously, and there is also a 9th channel for percussion. In addition to 128 instruments, LAOC-1 contains 30 percussion sounds and 33 sound effects. The SCC-1 is a further development of LAPC-1. Like the LAPC-1, it contains an MPU-MIDI interface, but in turn is a full-fledged WS synthesis card. It contains 317 samples (patches) hardwired into the internal ROM memory. A patch can consist of 24 partials, but most patches consist of one partial. 15 instruments and one percussion can be played at the same time. Although there is no ability to change internal samples, this is to some extent compensated by the presence of two sound effects: hall and echo. One of the biggest drawbacks of the Roland family of cards is that none of them are equipped with a DAC/ADC or a CD-ROM controller, which makes them impossible to use in multimedia systems that meet the MPC standard.
The sound quality of the LAPC-1 is very high. Some patches (like a piano or pipe) are superior in quality to similar GUS instruments. The quality of the reproduced sound effects is also very high. The sound quality of the SCC-1 can be considered simply outstanding. Which makes Roland cards one of the best for creating professional instrumental music, however They are completely unsuitable for use in multimedia systems, and Roland cards are not compatible with any modern audio standards.
Other cards
Adlib and SB compatible card with SCSI and MIDI interface.
Based on the Yamaha OPL-3 FM chip. 20 channels.
Improved sound quality compared to the original Adlib.
12-bit sampling and play at frequencies up to 44.1 KHz.
Similar to Adlib Gold 1000, but performs 16-bit sampling.
Based on the Yamaha YMF3812 FM chip. 11 channels.
8-bit mono sound at frequencies up to 22 KHz. Compatible with SB standard. Contains a MIDI interface.
Adlib and SB compatible card based on the Yamaha YM3812FM chip. 11 channels. 8-bit stereo sound up to 44.1 KHz. Contains a MIDI interface.
Turtle Beach MultiSound
Based on Motorola 56001 DSP chip. Contains 384 16-bit samples. 15 channels. Special effects. Stereo sound up to 44.1 KHz. Not compatible with any other standard.
AudioBahn 16 from Genoa Systems
Based on the Arial from Sierra semiconductor chip.
Adlib and SB compatible card with SCSI and MIDI interface. Contains 1M samples in ROM. 32 channels. 16-bit stereo sound at frequencies up to 44.1 KHz.
TXX SOUND CARDS: BASIC CONCEPTS
Before moving on to the next section, which addresses practical issues of purchasing a sound card, it is necessary to clarify a number of terms:
Frequency Response
Shows how well the sound system reproduces sound across the entire frequency range. An ideal device should equally transmit all frequencies from 20 to 20,000 Hz. And although in practice at frequencies above 18000 and below 100 there may be a decrease in performance by -2 dB due to the presence of a high/low pass filter, it is considered that a deviation below -3 dB is unacceptable.
Signal to noise ratio (S/N Ratio)
It is the ratio of the values (in dB) of the undistorted maximum signal of the board to the level of electronic noise that occurs in the board’s own electrical circuits. Because humans perceive noise at different frequencies differently, a standard A-weighting grid has been developed that takes into account noise annoyance levels. This number is usually what is meant when talking about S/N Ratio. The higher this ratio, the better the sound system. Reducing this parameter to 75 dB is unacceptable.
Noise quantization
Residual noise characteristic of digital devices, which arises due to imperfect signal conversion from analog to digital form. This noise can only be measured in the presence of a signal and is shown as a level (in dB) relative to the maximum permissible output signal. The lower this level, the higher the sound quality.
Total harmonic distortion + noise Reflects the influence of distortions introduced by sound amplification equipment and noise generated by the board itself. It is measured as a percentage of the undistorted output level. A device with an interference level of more than 0.1% cannot be considered high quality.
Channel Separation
Simply a number indicating the extent to which the left and right channels remain mutually independent. Ideally, the channel separation should be complete (absolute stereo effect), but in practice there is penetration of signals from one channel to another. On a high-quality stereo device, the channel separation should not be less than 50 dB.
Dynamic range
The difference, expressed in dB, between the max and min signal that the board can transmit. Typically dynamic range is measured at a frequency of 1Khz. An ideal digital audio system should have dynamic range close to 98dB.
Intermodulation distortion
Potential Gain
The maximum gain provided by the sound card preamplifier. It is desirable to have high potential gain at low input voltage. A voltage of 0.2V is considered low, which corresponds to the typical output signal of a household tape recorder.
WHICH BOARD SHOULD I CHOOSE?
As you can see above, at the moment there is simply a huge number of sound systems for personal computers released onto the market. Therefore, choosing a sound card is not an easy task, because each of them has its own advantages and disadvantages, and there are no absolute favorites, as well as absolute outsiders. And yet, let us take the liberty, in conclusion, to give some advice to those who are planning to equip their computer with a modern sound system.
1. In any case, you should opt for a 16-bit sound card that supports a sampling rate of at least 44Khz. This will give you the potential to listen to CD-quality sound.
2. If you are going to equip your computer with a CD-ROM drive, then it is advisable that the sound card you choose already carries a CD-ROM controller of the design you have chosen.
3. And finally, you should decide for what purposes you need a sound system, how high the demands are on your sound card and how much money you can sacrifice. All this forces you to divide the entire set of sound cards into several classes. Within each class, sound systems have approximately the same quality, which makes the choice much easier.
LIST OF REFERENCES USED
1. P. Norton "Programmer's guide to the IBM PC" - Microsoft Press 1985
2. Explanatory dictionary of computing systems / edited by V. Illingworth et al. - M, Mechanical Engineering, 1989
3. PC Magazine/Russian edition, 07.95- SK Press, Moscow
4. Sound Card review by Jerry van Waardenberg- comp.sys.ibm.pc.soundcard
Sound system a personal computer is used to reproduce sound effects and speech accompanying the reproduced video information, and includes:
- recording/playback module;
- synthesizer;
- interface module;
- mixer;
- sound system.
The components of the sound system (excluding the speaker system) are structurally designed in the form of a separate sound card or partially implemented in the form of microcircuits on the computer motherboard.
As a rule, the signals at the input and output of the recording/playback module are in analog form, but the processing of audio signals occurs in digital form. Therefore, the main functions of the recording/playback module are reduced to analog-to-digital and digital-to-analog conversion.
To do this, the input analog signal is subjected to pulse code modulation (PCM), the essence of which is to discretize time and represent (measure) the amplitudes of the analog signal at discrete moments in time in the form of binary numbers. It is necessary to select the sampling frequency and bit depth of binary numbers so that the accuracy of the analog-to-digital conversion meets the requirements for the quality of sound reproduction.
According to Kotelnikov’s theorem, if the time sampling step separating adjacent samples (measured amplitudes) does not exceed half the oscillation period of the highest component in the frequency spectrum of the converted signal, then time sampling does not introduce distortions and does not lead to information loss. If for high-quality sound it is enough to reproduce a spectrum with a width of 20 kHz, then the sampling frequency should be at least 40 kHz. Personal computer (PC) audio systems typically adopt a sampling rate of 44.1 or 48 kHz.
The limited capacity of binary numbers representing signal amplitudes determines the sampling of signal magnitudes. In most cases, sound cards use 16-bit binary numbers, which corresponds to 216 quantization levels or 96 dB. Sometimes 20- or even 24-bit analog-to-digital conversion is used.
It is obvious that improving sound quality by increasing the sampling frequency f and the number k of quantization levels leads to a significant increase in the volume S of the resulting digital data, since
S = f t log2k / 8,
where t is the duration of the sound fragment, S, f and t are measured in MB, MHz and seconds, respectively. With stereo sound, the data volume doubles. Thus, at a frequency of 44.1 kHz and 216 quantization levels, the amount of information to represent a stereo sound fragment lasting 1 minute is about 10.6 MB. To reduce the requirements for both memory capacity for storing audio information and the throughput of data transmission channels, information compression is used.
The interface module is used to transmit digitized audio information to other PC devices (memory, speaker system) via computer buses. The bandwidth of the ISA bus, as a rule, is not enough, so other buses are used - PCI, a special musical instrument interface MIDI, or some other interfaces.
Using a mixer, you can mix sound signals, creating polyphonic sound, add musical accompaniment to speech accompanying multimedia fragments, etc.
A synthesizer is designed to generate sound signals, most often to imitate the sound of various musical instruments. Frequency modulation, wave tables, and mathematical modeling are used for synthesis. The source data for synthesizers (note codes and instrument types) is usually presented in MIDI format (MID extension in file names). Thus, when using the frequency modulation method, the frequency and amplitude of the summed signals from the main generator and the overtone generator are controlled. According to the wave table method, the resulting signal is obtained by combining digitized sound samples obtained from real musical instruments. In the method of mathematical modeling, instead of experimentally obtained samples, mathematical models of sounds are used.
know:
PC sound system. Composition of the PC sound system. Operating principle and technical characteristics of sound cards. Directions for improving the sound system. The principle of processing sound information. Specification of sound systems.
Guidelines
PC sound system- a set of software and hardware that performs the following functions:
recording audio signals coming from external sources, such as a microphone or tape recorder, by converting input analog audio signals to digital ones and then storing them on a hard drive;
playback of recorded audio data using an external speaker system or headphones (headphones);
playback of audio CDs;
mixing (mixing) when recording or playing back signals from several sources;
simultaneous recording and playback of audio signals (Full Duplex mode);
processing of audio signals: editing, combining or separating signal fragments, filtering, changing its level;
audio signal processing in accordance with surround (three-dimensional - 3D-Sound) sound algorithms;
generating the sound of musical instruments, as well as human speech and other sounds using a synthesizer;
control of external electronic musical instruments via a special MIDI interface.
Figure 10 - Structure of the PC sound system
Classic sound system as shown in fig. 5.1, contains:
sound recording and playback module;
synthesizer module;
interface module;
mixer module;
sound system.
PC sound system designs are undergoing significant changes; There are motherboards with a Chipset installed on them for audio processing.
However, the purpose and functions of the modules of a modern sound system (regardless of its design) do not change. When considering the functional modules of a sound card, it is customary to use the terms “PC sound system” or “sound card.”
Questions for self-control:
PC sound system;
Composition of the PC sound system;
Operating principle and technical characteristics of sound cards;
Directions for improving the sound system;
The principle of processing sound information;
Specification of sound systems.
Topic 6.2 Audio information processing interface module
The student must:
have an idea:
about the PC sound system
know:
composition of the PC audio subsystem;
operating principle of the recording and playback module;
principle of operation of the synthesizer module;
operating principle of the interface module;
operating principle of the mixer module;
organizing the operation of the acoustic system.
Composition of the PC audio subsystem. Recording and playback module. Synthesizer module. Interface module. Mixer module. Operating principle and technical characteristics of acoustic systems. Software. Sound file formats. Speech recognition tools.
Guidelines
Sound system recording and playback module carries out analog-to-digital and digital-to-analog conversions in the mode of software transmission of audio data or transmission via DMA channels (Direct Memory Access - direct memory access channel).
Sound recording is the storage of information about sound pressure fluctuations at the time of recording. Currently, analog and digital signals are used to record and transmit sound information. In other words, the audio signal can be in analog or digital form.
In most cases, the sound signal is supplied to the input of the PC sound card in analog form. Due to the fact that the PC operates only with digital signals, the analog signal must be converted to digital. At the same time, the speaker system installed at the output of the PC sound card perceives only analog electrical signals, therefore, after processing the signal using a PC, it is necessary to reverse convert the digital signal to analog.
A/D conversion is the conversion of an analog signal to a digital signal and consists of the following main steps: sampling, quantization and encoding.
^ The pre-analog audio signal is fed to an analog filter, which limits the frequency band of the signal.
Signal sampling consists of sampling samples of an analog signal with a given periodicity and is determined by the sampling frequency. Moreover, the sampling frequency must be no less than twice the frequency of the highest harmonic (frequency component) of the original audio signal.
Amplitude quantization is the measurement of instantaneous amplitude values of a discrete time signal and converting it into discrete time and amplitude. Figure 11 shows the analog signal level quantization process, with instantaneous amplitude values encoded as 3-bit numbers.
^
Figure 11 - Scheme of analog-to-digital conversion of an audio signal
Coding consists of converting a quantized signal into a digital code. In this case, the measurement accuracy during quantization depends on the number of bits of the code word.
^
Figure 12 - Time sampling and quantization based on the level of the analog signal for quantizing the sample amplitude.
Analog-to-digital conversion is carried out by a special electronic device - an analog-to-digital converter (ADC), in which discrete signal samples are converted into a sequence of numbers. The resulting digital data stream, i.e. signal includes both useful and unwanted high-frequency interference, to filter which the received digital data is passed through a digital filter.
Digital-to-analog conversion generally occurs in two stages, as shown in Figure 12. At the first stage, signal samples are extracted from the digital data stream using a digital-to-analog converter (DAC), following the sampling frequency. At the second stage, a continuous analog signal is formed from discrete samples by smoothing (interpolation) using a low-frequency filter, which suppresses the periodic components of the discrete signal spectrum.
To reduce the amount of digital data required to represent an audio signal with a given quality, compression is used, which consists in reducing the number of samples and quantization levels or the number of bits per sample.
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Figure 13 - Digital-to-analog conversion circuit
Such methods of encoding audio data using special encoding devices make it possible to reduce the volume of information flow to almost 20% of the original one. The choice of encoding method when recording audio information depends on the set of compression programs - codecs (encoding-decoding) supplied with the sound card software or included in the operating system.
Performing the functions of analog-to-digital and digital-to-analog signal conversion, the digital audio recording and playback module contains an ADC, a DAC and a control unit, which are usually integrated into a single chip, also called a codec. The main characteristics of this module are: sampling frequency; type and capacity of ADC and DAC; audio data encoding method; ability to work in Full Duplex mode.
The sampling rate determines the maximum frequency of the signal that is recorded or played back. For recording and playback of human speech, 6 - 8 kHz is sufficient; music with low quality - 20 - 25 kHz; To ensure high quality sound (audio CD), the sampling frequency must be at least 44 kHz. Almost all sound cards support recording and playback of stereo audio at a sampling rate of 44.1 or 48 kHz.
^ The bit depth of the ADC and DAC determines the bit depth of the digital signal (8, 16 or 18 bits).
Full Duplex is a data transmission mode over a channel, according to which the sound system can simultaneously receive (record) and transmit (play) audio data. However, not all sound cards fully support this mode, since they do not provide high sound quality during intensive data exchange. Such cards can be used to work with voice data on the Internet, for example, during teleconferences, when high sound quality is not required.
Synthesizer module
An electromusical digital sound system synthesizer allows you to generate almost any sound, including the sound of real musical instruments. The principle of operation of the synthesizer is illustrated in Figure 14.
Synthesis is the process of recreating the structure of a musical tone (note). The sound signal of any musical instrument has several time phases. Figure 15, a shows the phases of the sound signal that appears when you press a piano key. For each musical instrument, the type of signal will be unique, but three phases can be distinguished in it: attack, support and attenuation. The set of these phases is called the amplitude envelope, the shape of which depends on the type of musical instrument. The attack duration for different musical instruments varies from a few to several tens or even hundreds of milliseconds. In the phase called support, the amplitude of the signal remains almost unchanged, and the pitch of the musical tone is formed during support. The last phase, attenuation, corresponds to a section of a fairly rapid decrease in the signal amplitude.
In modern synthesizers, sound is created as follows. A digital device using one of the synthesis methods generates a so-called excitation signal with a given pitch (note), which should have spectral characteristics as close as possible to the characteristics of the simulated musical instrument in the support phase, as shown in Figure 15, b. Next, the excitation signal is fed to a filter that simulates the amplitude-frequency response of a real musical instrument. The amplitude envelope signal of the same instrument is supplied to the other filter input. Next, the set of signals is processed to obtain special sound effects, for example, echo (reverberation), choral performance (chorus). Next, digital-to-analog conversion and filtering of the signal are performed using a low-pass filter (LPF).
Figure 15 - Operating principle of a modern synthesizer: a - phases of the sound signal; 6 - synthesizer circuit
Main characteristics of the synthesizer module:
sound synthesis method;
Memory;
possibility of hardware signal processing to create sound effects;
polyphony - the maximum number of simultaneously reproduced sound elements.
The synthesis method based on frequency modulation (Frequency Modulation Synthesis - FM synthesis) involves the use of at least two signal generators of complex shapes to generate the voice of a musical instrument. The carrier frequency generator generates a fundamental tone signal, frequency-modulated by a signal of additional harmonics and overtones that determine the sound timbre of a particular instrument. The envelope generator controls the amplitude of the resulting signal. The FM generator provides acceptable sound quality, is inexpensive, but does not implement sound effects. Therefore, sound cards using this method are not recommended according to the PC99 standard.
Sound synthesis based on a wave table (Wave Table Synthesis - WT synthesis) is performed by using pre-digitized sound samples of real musical instruments and other sounds stored in a special ROM, made in the form of a memory chip or integrated into the WT generator memory chip. The WT synthesizer provides high quality sound generation. This synthesis method is implemented in modern sound cards.
^ The amount of memory on sound cards with a WT synthesizer can be increased by installing additional memory elements (ROM) for storing banks with instruments.
Sound effects are generated using a special effect processor, which can either be an independent element (microcircuit) or integrated into the WT synthesizer. For the vast majority of cards with WT synthesis, reverb and chorus effects have become standard. Sound synthesis based on physical modeling involves the use of mathematical models of sound production of real musical instruments for digital generation and for further conversion into an audio signal using a DAC. Sound cards using the physical modeling method are not yet widely used because they require a powerful PC to operate.
Interface module Provides data exchange between the sound system and other external and internal devices.
The PCI interface provides high bandwidth (for example, version 2.1 - more than 260 Mbit/s), which allows you to transmit audio data streams in parallel. Using the PCI bus allows you to improve sound quality, providing a signal-to-noise ratio of over 90 dB. In addition, the PCI bus allows for cooperative processing of audio data, when data processing and transmission tasks are distributed between the sound system and the CPU.
MIDI (Musical Instrument Digital Interface - digital interface of musical instruments) is regulated by a special standard containing specifications for the hardware interface: types of channels, cables, ports through which MIDI devices are connected to one another, as well as a description of the order of data exchange - the information exchange protocol between MIDI devices. In particular, using MIDI commands, you can control lighting equipment and video equipment during the performance of a musical group on stage. Devices with a MIDI interface are connected in series, forming a kind of MIDI network, which includes a controller - a control device, which can be used as a PC or a musical keyboard synthesizer, as well as slave devices (receivers) that transmit information to the controller via its request. The total length of the MIDI chain is not limited, but the maximum cable length between two MIDI devices should not exceed 15 meters.
Connecting a PC to a MIDI network is done using a special MIDI adapter, which has three MIDI ports: input, output and pass-through, as well as two connectors for connecting joysticks.
^ The sound card includes an interface for connecting CD-ROM drives
Mixer module
The sound card mixer module does:
switching (connection/disconnection) of sources and receivers of audio signals, as well as regulation of their level;
mixing (mixing) several audio signals and adjusting the level of the resulting signal.
number of mixed signals on the playback channel;
regulation of the signal level in each mixed channel;
regulation of the level of the total signal;
amplifier output power;
availability of connectors for connecting external and internal
receivers/sources of audio signals.
Software control of the mixer is carried out either using Windows tools or using the mixer program supplied with the sound card software.
Compatibility of the sound system with one of the sound card standards means that the sound system will provide high-quality reproduction of sound signals. Compatibility issues are especially important for DOS applications. Each of them contains a list of sound cards that the DOS application is designed to work with.
The Sound Blaster standard is supported by applications in the form of DOS games, in which the sound is programmed with a focus on sound cards of the Sound Blaster family.
^ Microsoft's Windows Sound System (WSS) standard includes a sound card and software package aimed primarily at business applications.
Acoustic system (AS) directly converts the audio electrical signal into acoustic vibrations and is the last link in the sound-reproducing tract. A speaker system usually includes several audio speakers, each of which can have one or more speakers. The number of speakers in a speaker system depends on the number of components that make up the sound signal and form separate sound channels.
As a rule, the operating principle and internal structure of sound speakers for household use and those used in technical means of informatization as part of a PC speaker system are practically the same.
Basically, a PC speaker consists of two audio speakers that provide stereo playback. Typically, each speaker in a PC speaker has one speaker, but expensive models use two: for high and low frequencies. At the same time, modern models of acoustic systems make it possible to reproduce sound in almost the entire audible frequency range due to the use of a special design of the speaker or loudspeaker housing.
To reproduce low and ultra-low frequencies with high quality in the speakers, in addition to two speakers, a third sound unit is used - a subwoofer, installed under the desktop. This three-component PC speaker system consists of two so-called satellite speakers that reproduce mid and high frequencies (from approximately 150 Hz to 20 kHz), and a subwoofer that reproduces frequencies below 150 Hz.
A distinctive feature of PC speakers is the possibility of having its own built-in power amplifier. A speaker with a built-in amplifier is called active. Passive speakers do not have an amplifier.
The main advantage of active speakers is the ability to connect to the linear output of a sound card. The active speaker is powered either from batteries (accumulators) or from the electrical network through a special adapter, made in the form of a separate external unit or power module installed in the housing of one of the speakers.
The output power of PC speakers can vary widely depending on the specifications of the amplifier and speakers. If the system is intended for sounding computer games, a power of 15 - 20 W per speaker is sufficient for a medium-sized room. If it is necessary to ensure good audibility during a lecture or presentation in a large audience, it is possible to use one speaker with a power of up to 30 W per channel. As the power of the speaker increases, its overall dimensions increase and the cost increases.
^ Main characteristics of the speakers: reproduced frequency band, sensitivity, harmonic distortion, power.
Reproducible frequency band (FrequencyResponse) is the amplitude-frequency dependence of sound pressure, or the dependence of sound pressure (sound intensity) on the frequency of the alternating voltage supplied to the speaker coil. The frequency band perceived by the human ear is in the range from 20 to 20,000 Hz. Speakers, as a rule, have a range limited in the low frequency region of 40 - 60 Hz. The problem of reproducing low frequencies can be solved by using a subwoofer.
The sensitivity of a speaker (Sensitivity) is characterized by the sound pressure that it creates at a distance of 1 m when an electrical signal with a power of 1 W is applied to its input. In accordance with the requirements of the standards, sensitivity is defined as the average sound pressure in a certain frequency band.
The higher the value of this characteristic, the better the speaker conveys the dynamic range of the music program. The difference between the “quiest” and “loudest” sounds of modern phonograms is 90 - 95 dB or more. Speakers with high sensitivity reproduce both quiet and loud sounds quite well.
Total Harmonic Distortion (THD) evaluates nonlinear distortion associated with the appearance of new spectral components in the output signal. The harmonic distortion factor is standardized in several frequency ranges. For example, for high-quality Hi-Fi speakers this coefficient should not exceed: 1.5% in the frequency range 250 - 1000 Hz; 1.5% in the frequency range 1000 - 2000 Hz and 1.0% in the frequency range 2000 - 6300 Hz. The lower the harmonic distortion value, the better the speaker quality.
The electrical power (Power Handling) that the speaker can withstand is one of the main characteristics. However, there is no direct relationship between power and sound reproduction quality. The maximum sound pressure depends rather on sensitivity, and the power of the speaker mainly determines its reliability.
Often on the packaging of PC speakers they indicate the peak power of the speaker system, which does not always reflect the real power of the system, since it can exceed the nominal power by 10 times. Due to significant differences in the physical processes occurring during AS tests, the electrical power values may differ several times. To compare the power of different speakers, you need to know exactly what power the product manufacturer indicates and by what test methods it is determined.
Some models of Microsoft speakers are connected not to the sound card, but to the USB port. In this case, the sound arrives at the speakers in digital form, and its decoding is carried out by a small Chipset installed in the speakers.
Questions for self-control:
Composition of the PC audio subsystem;
Recording and playback module;
Synthesizer module;
Interface module;
Mixer module;
Operating principle and technical characteristics of acoustic systems. Software;
Sound file formats;
Speech recognition tools.
Practical work 8. PC sound system
The student must:
have an idea:
about the PC sound system
know:
principles of processing audio information;
composition of the PC audio subsystem;
main characteristics of sound cards
be able to:
connect and configure PC audio subsystems;
record audio files.
Section 7. Printing devices
Topic 7.1 Printer
The student must:
have an idea:
about devices printing information
know:
operating principle of dot matrix printer output devices. Main components and operating features, technical characteristics;
operating principle of inkjet printer information output devices Main components and operating features, technical characteristics;
operating principle of laser printer output devices Main components and operating features, technical characteristics.
General characteristics of printing devices. Classification of printing devices. Impact printers: principle of operation, mechanical components, operating features, technical characteristics, operating rules. Basic modern models.
^ Inkjet printers: principle of operation, mechanical components, operating features, technical characteristics, operating rules. Basic modern models.
Laser printers: principle of operation, mechanical components, operating features, technical characteristics, operating rules. Basic modern models.
Guidelines
Printers- devices for outputting data from a computer, converting ASCII information codes into corresponding graphic symbols and recording these symbols on paper.
Printers can be classified according to a number of characteristics:
the method of forming symbols (printing signs and synthesizing signs);
chromaticity (black and white and color);
method of forming lines (serial and parallel);
printing method (character-by-character, line-by-line and page-by-page)
print speed;
resolution.
When working in text mode The printer receives character codes from the computer, which must be printed from the character generator of the printer itself. Many manufacturers equip their printers with a large number of built-in fonts. These fonts are written to the printer ROM and can only be read from there.
To print text information, there are print modes that provide different quality:
draft printing (Draft);
typographic print quality (NLQ - Near Letter Quality);
print quality close to typographical (LQ - Letter Quality);
high-quality mode (SQL - Super Letter Quality).
Based on the method of applying an image to paper, printers are divided into impact, inkjet, photoelectronic and thermal printers.
