Sound systems. Sound system

Anyone who works with professional sound, for sure at least once faced with integrated background sound systems. After all, it is no secret that from such small and medium-sized projects maybe consist of hardly aboutmore expense of sales and distributor of equipment, and at the dealer, and at the installer. And, unlike large systems, "distribution" does not require complex calculations, creating acoustic models and other routine pre-sale work. An experienced specialist can make a model specification "in the mind", knowing only the overall dimensions of the room. And, of course, such a system will work, but, as they say in a famous joke, there is one nuance ...

Due to the successful work of marketers and sellers, owners and franchisees cafes, restaurants, shops and shopping centers around the world, and in our country, now quite understand that the correct sound is important both for the mood and loyalty of the client and for the efficiency of the same advertising content. And, let me now show excerpts from the colorful catalogs of any manufacturer of ceiling acoustic systems, the results of the labor of marketers we see - all serious world brands have long reached the Russian market and turned the client in their faith. And the competent business leader in this area finally ceased to neglect the quality of the sound, as it was not so long ago.

It would seem that the case is done - form a typical proposal and change the number of acoustic systems in it, depending on the configuration of the room. But everything is not so simple. Rather, relatively simply, if approaching the construction of systems from the position of the smallest time costs per unit of goods. And there is logic in it. And the most indisputable argument - "This is not a philharmonic!" - Already became practically hosted, and it is ideally applicable to any object, except for actually, the very Philharmonic.

Probably, one of you will say: "These are idle reasoning about anything", so I will go, finally, to the main one.

The topsack of the article is precisely in debunking a common opinion that the design of the background sound system is not worth any serious temporary and mental costs. As for the time, I partially agree - few of us has it in such a quantity to allow yourself to spend a watch-another to choose from one of two neighboring ceiling sections for the loudspeaker. But the connection of engineering thought will help us get the best result from the same products as competitors. And the result, with the right approach, please both the client and your sales department. Agree that at the current assortment very similar to each other of sound equipment of different manufacturers intended for commercial systems, still the main one, if not the only way to attract and keep the client - to offer the most attractive price. And since a rare buyer will be trepidated to the quality of sound and will be able to objectively appreciate it, in most cases he will benefit, will offer a more economical decision.

But let's try to abstract from all commercial components and concentrate on the native and close heart - on the engineering part.

Engineer, your way out!

There is a thousand and one recommendation for the calculation of the same ceiling acoustic systems. Let's start with them and begin. What do not offer us manufacturers to simplify our work ... One Vendor distributes the Talmuda partners with recommendations on the calculation, the other offers "user-friendly" acoustic simulators in which anyone can draw the necessary configuration of loudspeakers, the third writes the calculators that are enough Enter linear size of the room, and you will receive a generated report with the location scheme. Among the latter, for example, JBL, offering its calculator almost for each product series. This, I confess most conveniently, and with proper use gives a quick and approximate reality. But first things first.

I consider it necessary to "disassemble the bones" pluses and disadvantages of existing methods.

The method that is no doubt an autonomy and non-volatile - graphic, similar on its own principle to build a radiation sketch. It is necessary to know the nominal opening angle of the loudspeaker and the height of the ceiling. Here is what the result looks like:

Fig. 1. Graphic calculation of the layout of the ceiling loudspeakers. A - distance from the floor to the ears of the listener; B - the distance from the ears to the ceiling; C - the opening angle of the loudspeaker; D - point of intersection of rays of neighboring loudspeakers.

Everything is simple enough. Graphically depicts an angle of opening the loudspeaker, the height of the listener's ears (it is customary to take 1-1.2 meters a man in a sitting position and 1.5 meters - in standing), and the point of intersection of the horizontal and the rays of the disclosure angle is considered a critical point that the ray must be crossed from the neighboring loudspeaker. In this way, the step of the arrangement of acoustic systems is determined.

Now the shn is more deeper. It is known that the magnitude of the angle of disclosure, indicated in the passport of the loudspeaker is nominal, i.e. Averaged on the frequency band, determined by the manufacturer at his discretion. And it's no secret that the directional properties of any real emitter will seriously differ in various frequency bands. As a result, we carry out the calculation, sometimes not even knowing, in which range the correct coating was obtained. So, colleagues, be careful - making such a calculation using the nominal disclosure angle, you may well get "pits" in the frequency bands, for example, above 8-10 kHz.

Now another nuance. The nominal disclosure angle is usually calculated from the polar diagrams in such a way that with the deflection of the declining angle to the ½ of the declared angle of disclosure, the drop in the pressure level will be 6 dB. Moreover, again attention, at an equal distance from the emitter.

Fig. 2. Graphic calculation of the layout of the ceiling loudspeakers. A - distance from the floor to the ears of the listener; B - the distance from the ears to the ceiling; C - the opening angle of the loudspeaker; D - Sound pressure drop point for 6 dB

It turns out, at the point of intersection of the horizontal and the ray, the fall will not be 6 dB, but more. Well, nothing terrible, arming a circulation and solve the problem.

However, this is also not all. What do you think when we cross the rays from neighboring loudspeakers in the right point, what pressure do we get there? Having 2 waves with a pressure level of 6 dB SPL relative to the radiation axis, we can fold them according to the rule of energy summation (L1, p.33) as two equal pressure and get an amount equal to -3 dB relative to the axis. However, this rule works in the case of incoherent addition, i.e. For example, with a different distance from sources, but at the point of intersection of the waves of the wave coherent (syphase), and only in it are inserted throughout the spectrum, giving doubling pressure, i.e. It will be practically the same as on the radiation axis. Figure below shows the result of the calculation in the model with two closely located ceiling loudspeakers.

Fig. 3. Calculation of sound pressure level using two ceiling loudspeakers in an octave strip from centers at 500 Hz.

As a result, it turns out that picture: coherent addition of waves exactly between loudspeakers always exists and gives rise to +3 dB on a rather small area, and literally in centimeters from this "seam" waves are summed by incoherent and there is a pressure drop. And I will immediately explain that it will not be possible to completely get rid of this "seam". Below are the results of acoustic modeling with a different step of loudspeakers.

Fig. 4. Sound pressure diagram when the loudspeakers are located at an altitude of 3 meters from the floor with a 1.5 meter increments. The calculation is made in third-octave strings of 10 kHz (lower diagram) and 400 Hz (upper chart).

Fig. 5. Sound pressure diagram when the loudspeakers are located at an altitude of 3 meters from the floor with a step of 3 meters. The calculation is made in third-octave strings of 10 kHz (lower diagram) and 400 Hz (upper chart).

Fig. 6. Sound pressure diagram when the loudspeakers are located at a height of 3 meters from the floor with a pitch of 4.5 meters. The calculation is made in third-octave strings of 10 kHz (lower diagram) and 400 Hz (upper chart).

Shilo or soap?

Well, the result of the simulation showed that the result is negative for uniformity, the result gives both too much speakers, and too small. And just too small distance is almost a more serious problem, because the misconception is common, which placing speakers with a minimum step, we get a uniform coating throughout the frequency region. For the high-frequency area, this thesis is valid, since any loudspeaker has a narrower pattern of directivity in the field of high frequencies. As for non-coherent additions of waves, due to the interference in the field of low frequencies, the pressure at the beam intersection points will be guaranteed more than right under the loudspeaker, as if paradoxically sounded it. Moreover, the interference picture will change at every point, and the closer to each other there are loudspeakers, the more such changes will be. So is it worth a uniform coating in the field of high frequencies of such victims? I do not think.

So that it becomes slightly clearer, make a refinement. As is known, the direction of the wave depends on its length - long waves (frequency of 160 Hz and below) are omnidirectional, i.e. The angle of disclosure of any loudspeaker at frequency, for example, 80 Hz will be 360 \u200b\u200bdegrees. In the case of ceiling systems, by itself, 180 degrees. A short waves have a narrower orientation, which is due to the physics of the wave propagation process. Thus, in the octave strip of 16 kHz, the average ceiling loudspeaker may have an angle of disclosure (per -6 dB) 45-60 degrees with passport nominal 120 degrees averaged by a range of 1 kHz-8 kHz. It turns out to be avoided by the "sound holes", the calculation should be carried out by taking as a basis the characteristic of the loudspeaker disclosure in high frequencies. Right. Just not so narrowly-directed long waves will create incomparably greater pressure, repeatedly develop and deduct, creating illustrated amounts and differences in the one aboutlongly scattering pressures, the closer to each other their sources are located.

Based on the read you have a complete right to blame me in the fact that I did not give an obvious answer, as it is precisely correctly to have loudspeakers. So, it is, but if an unequivocal answer existed, in our services would not have any needs and to design the sound system could be any. This is how the workshop is, as it is called, "System Design" - in finding a compromise solution, in balancing between mutually exclusive requirements and conditions.

And the rest, beautiful marquise, everything is fine, everything is fine!

Perfectionism is not such a bad feature, but sometimes it requires an achievable reference point for productive work. And he also has. In a quantitative assessment of the uniformity of the sound field, it helps used in statistics of the so-called. Standard deviation (STDEV). I will not delve into an explanation of this concept - a great chance to deepen too much.

Fig. 7. Standard deviation

We have a schedule for the distribution of some random variables within the standard deviation from the mathematical expectation. Take it as a basis using the distribution of sound pressure levels as quantities.

And now we agree that the value of μ on a horizontal scale is the average value of the sound pressure level throughout the room, namely our mathematical expectation. The value of σ take 2 dB (-20% + 25% by absolute value), since the likely scatter of the values \u200b\u200brelative to the expected can be different. Now our task is to understand which scatter will satisfy us, and what will be considered unacceptable. If the pressure is the same throughout the measured area, then the schedule will turn into a straight line. The greater the scatter of the magnitudes, the more steep will be the rise and decline in the graph of this function. So, with a fairly uniform sound field, most values \u200b\u200bare concentrated near the average value. And with this fairly uniform coating, we can consider the zone within the 1st standard deviation, i.e. If 68% of the entire area of \u200b\u200bthe room, the pressure level varies within + -2 dB from the middle frequency range, then the requirement is made. True, it is possible to see such statistics for the distribution of pressures only by spending an acoustic calculation.

Despite the fact that in ISO or AES standards, this interpretation is not recorded, in practice it is often used and generally reflects reality, therefore it can serve as a good guideline and a starting point in determining the uniformity of the area coverage.

But do not forget that the value averaged over the entire range does not always describe the full picture.

Black box

Well, with ceiling loudspeakers seem to have figured out how much it was possible in this format. And what about the wall systems? Is everything so easy with them, how are we used to thinking? In general, it is much easier simply because, as a rule, we are extremely limited in placing cabinet acoustic systems - walls, angles, columns. And at all, not any point of the wall is available for the installation of the loudspeaker - somewhere the designer stucco, somewhere TV, somewhere ventilation and so on.

And one thing when you need to voice 100 square meters. meters - picked up the angle of disclosure, scattered in the corners of 4 loudspeakers, and everything is ready for the system - and how to do with a larger area? We are looking for bearing columns in the middle of the room, we rejoice in their presence and riding them with loudspeakers. Well, what to do - no options. I agree, but with clarifications. For the answer, as usual, it is worth contacting science.

Here is an example of the location of acoustic systems indoors.

Fig. 8. Location of wall loudspeakers on columns

In general, everything is fine, and with the right choice of loudspeakers and the proper installation there will be no problems. Run ahead, I will say that all of the location schemes presented by me further have the right to exist, but with some reservations.

In the event that full-range loudspeakers, with disclosure into crazy 150 degrees (and it happens), their location in close proximity to each other will create you a very interesting picture of interference. In order not to raise for a long time, this time immediately demonstrate an acoustic calculation, because something more visual and accessible to understand it is difficult.

Fig. 9. Sound pressure level diagram when the loudspeakers are located on columns in an octave strip with a 500 Hz center

Pay attention to the resulting "petals" - this is the result of the addition and subtraction of two coherent waves, and the location of them, of course, changes depending on the wavelength. The same picture can be observed when the loudspeakers are arranged in clusters - for the correct addition of the waves, it is necessary to take a number of measures both when designing and configured, but this is a completely different story. Just in case, I designate one obvious consequence of this fact: as a result of the interference, the voice program can be seriously distorted due to the subtraction of some frequency components. Many specialists are unfortunately confident that any collected distortion is corrected using a measuring microphone, spectrum analyzer and equalizer, and sincerely surprise, trying when setting up the frequency "pull" the frequency lost during interference. And on the chart nothing happens, how much to increase the heine filter - by +6 dB, by +12 dB, and even two equalizers are consistently turned on. The pressure at this frequency is simply absent, and it is necessary to undertake it that, if one of the many reasons in this range, the waves occurred.

Now let's take and try to get rid of these problems, and even a needed system, reducing the number of loudspeakers.

Fig. 10. Location of wall loudspeakers on columns

Fig. 11. Sound pressure level diagram when the loudspeakers are locating in the columns in the full frequency range.

It turns out quite decently: the interference problems are solved, the coating in the zone between the columns is close to the perfect, coherent addition of the waves is also not critical. As a budget version, such a design is quite viable - the main thing is that the pitch of columns allow you to put in a standard deviation. But a certain nuance is still there. And his root is swapped deep in fundamental science.

Due to the physiology of hearing and, probably, the evolution is capable of localizing sound events, i.e. To determine where the sound wave arrived - this ability was simply necessary to work out for survival. And what to be when there are a lot of sound waves, such as in a primitive cave, where in addition to direct sound from the source, there are countless reflections arriving from all sides? Very simple. It was enough to develop the ability to determine the direction of the first wave, which is definitely for the shortest path will arrive directly from the conditional mouth of the predator, and any reflection will accurately go through a larger way and come with some delay. This phenomenon describes the law of the first wave front (it is Precedence Effect). In the presence of several identical waves coming with a delay, the brain determines the direction exclusively on the first wave, even if the second and subsequent has a higher level (excess to 10 dB) and comes with a delay of up to 30 ms. You can read more about this entertaining effect and its description in the literature on psychoacoustics.

So what is all this? Now let's simulate the listener moving along the length of the room in a straight trajectory, and see how the sound localization will change for it. In the process of movement by the first loudspeaker, a person will clearly hear the sound of the left, as it approaches it to the conditional border of disclosure, the ratio of the wave in the left and on the right is changed, since the second loudspeaker appears in the field of view. Our object reached a point of equal distance between loudspeakers and both waves were coherently developed, giving him +3 dB to the level of pressure, and the localization of the sound instantly rearranged to a point of equal distance between the sources, i.e. Just at that place where the head of the object is at the moment. And the next step will shift sharply the sound event to the right, since the wave from the second source will now come first.

In principle, there is nothing critical in it. But if constant movements of customers are assumed along the area, such as, for example, in the store, will they comfortably listen to the sound of the sound from the point to the point? Not every listener analyzes the causes of their discomfort and binds them with sound, the perception of the environment is consistent for it unconsciously and consists of a set of all sensations - visual, audible, tactile and others. And enough so that at least one of them caused discomfort so that the rest are insignificant, and the subjective impression was spoiled.

On the finish line are direct

Perhaps the main issues of calculating the arrangement of loudspeakers were considered, however, it will not be quite honest on my part not to mention that almost all of these calculations take into account the energy of a direct wave from the emitter. And in conditions of real premises, filled not only by direct sound, but also numerous reflections, interference subtraction, of course, will not create points with zero sound pressure. Reflected waves will somewhat level failures and lifts, of course, not to eliminate them completely, and significantly improve the uniform of the coating, compensating for the lack of direct sound in the points removed from its source.

By the way, one of the interesting methods for creating an unlocated background sound of the system is based on the use of reverb of premises for the benefits of background sound. It consists in the location of all the acoustic systems "face" to the ceiling. Such a location almost completely eliminates the listener from direct sound from the loudspeaker, all the energy obtained by them is a set of reflected waves from all directions. Extremely interesting is the effect in terms of the spatiality of the sound. The only minus of such a solution is to limit on content. Fast pop or rock music, which is not calculated on such a serious effect of reverb, is unlikely to sound well from such a system.

P.S. And what, without a cable does not fall?

Despite the seeming mining of the issue of cable tracks, it is difficult to overestimate the importance of speaker (acoustic) cable for any sound system. I'm talking about this with complete confidence, because, unfortunately, in my practice it is not always possible to dictate to the client, which cable it is to buy, and it sometimes leads to silent scenes in the style of the Chekhov auditor, when the object finds that for the sound system it was laid Cable ShVVP. In response to your question, I get a completely reasonable answer - "And what works!". Works. Just so works that it did not work better. In general, you understand ...

And that is why we bring the method of calculating the cable cross section. Those of you, for whom she is obvious, and who knows perfectly, how such calculations are made, can safely miss this part of the article - I will not give anything new and reach the science of the Unknown. But if suddenly you first encountered the need for calculation, then this information will be useful in view of its applied applicability.

Calculation of efficient current:

Calculation of the effective power allocated on the load:

100V line.

Calculation of the total resistance of loudspeakers in the line:
,Where

Number of loudspeakers on the line
- Rated Power of Single Loudspeaker (TAP Setting)

The remaining calculations are performed similar to low-voltage lines.

The total load resistance in the 100-volt line, as can be seen, is usually obtained at least 1000 ohms. With such a high resistance, the cable resistance unit is slightly affected by the overall line resistance, and, therefore, increase the power loss slightly compared to the low-voltage connection.

Now a little about the interpretation of the results. How to determine which power loss is permissible? In the general case, the threshold value of the power level on the 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, a permissible face value of 1 kW of the maximum drop in these norms reaches a cross section of 2.5 sq. M. Long in 30 meters. A lot or a little, of course, to solve you, and the solution here depends on the specific situation, but practice shows that an increase in the cable cross section from 2.5 sq. Mm to, for example, 4 square meters will not significantly increase the cost of installation. Therefore, I always recommend stacked at 0.5 dB, because it is not difficult to do it. And why should we lose the precious watts on the line when we have the opportunity to achieve maximum system efficiency?

And, despite the fact that the translation lines of the requirement is significantly lower, the use of the correct cable will help you make the system work more efficiently. Moreover, if in your practice you did not conduct experiments to assess the quality of sound on different cables (other things being equal), then believe me on the word, the effect of the cable cross section on the sound is really noticeable for rumor. This is especially true of the low-frequency region - the range, when the transmission of which is developing the highest power, and which is most demanding to the current and dumping factor.

Therefore, using so much loved by many analogy, let's not fill in the Mercedes S-class 92th gasoline, and then wonder why the stated performance is not achieved.

As can be seen by formulas, the only value that remains unknown to calculate the cable is its resistance, expressed in OM / km. Its value can be found in the cable specification. To do this, you will have to first select the crossing cable cross-section, take the appropriate resistance value, substitute in the formula and carry out the calculation. In case you get an excess of the power drop, or vice versa, the cross section will be redundant, you will have to select a cable of another section and return to the initial point of the calculation. I usually recommend to start the calculation from the section 2x2.5 square meters (7.5-8 ohm / km) for low-level lines and 2x1.5 square meters (about 13 ohm / km) for transformer lines. Of course, it will make you spend some time calculating, but for convenience you can create a calculator in Excel, making the formula and the values \u200b\u200bof the resistance of the cables of different sections - it will take some time at once, but will save from the need for manual calculation in the future.

Thank you Digis for the materials provided

1. Viscovic system PC

The audio system of the PC in the form of a sound card appeared in 1989, significantly expanding the possibilities of the PC as a technical means of informatization.

PC sound system -software and hardware complex performing the following functions:

recording audio signals from external sources, such as a microphone or a tape recorder, by converting input analog audio signals into digital and subsequent storage on the hard disk;

play recorded audio data using an external speaker system or headphones (headphones);

playing audio CDs;

mixing (mixing) when recording or playing signals from multiple sources;

simultaneous recording and playback of audio signals (mode FULLDuplex);

processing of sound signals: editing, combining or separating signal fragments, filtering, change its level;

processing of the sound signal in accordance with the algorithms of volumetric (three-dimensional - 3 D.- Sound.) sound;

generation using musical instruments synthesizer, as well as human speech and other sounds;

managing the work of external electronic musical instruments through a special MIDI interface.

The PC sound system is structurally sound cards, or installed in the motherboard slot, or the other subsystem of the PC installed on the motherboard or the extension card. Separate sound system functional modules can be performed as childboards installed in the appropriate sound card connectors.

The classic sound system, as shown in Fig. 5.1, contains:

Recording and sound recording module;

synthesizer module;
interface module;
module Mixer;
acoustic system.

The first four modules are usually installed on the sound card. Moreover, there are sound cards without a synthesizer module or a digital sound recording / playback module. Each of the modules can be performed either as a separate chip, or to enter a multifunctional chip. Thus, the chipset of the audio system may contain both several and one microcircuit.

Constructive performances of the PC sound system undergo significant changes; There are motherboards with chipset installed on them for sound processing.

However, the purpose and function of the modules of the modern sound system (regardless of its design) do not change. When considering the sound card functional modules, it is customary to use the terms "PC Sound System" or "Sound Card".

2. Record and Play Module

The recording and playback module of the audio system performs analog-digital and digital conversion in software transmission mode or transmission to DMA channels (Direct.Memory.Access- Channel direct memory access).

The sound is known is longitudinal waves freely distributing in air or other medium, so the beep sounds continuously in time and in space.

Sound recording is the saving of information about sound pressure fluctuations at the time of the recording. Currently, analog and digital signals are used to record and transmit information about sound. In other words, the beep can be represented in analog or digital form.

If when recording sounds, use a microphone that converts the electrical signal continuous in time into time in time, the electrical signal is obtained in analog form. Since the amplitude of the sound wave determines the volume of the sound, and its frequency is the height of the audio tone, the electrical signal should be proportional to the height of the sound, and its frequency must match the frequency of the oscillations of the sound pressure.

At the entry of the PC sound card in most cases, the beep is applied in analog form. Due to the fact that the PC operates only by digital signals, the analog signal must be transformed into digital. At the same time, the acoustic system installed at the output of the PC sound card perceives only analog electrical signals, so after processing the signal using the PC, the inverse conversion of the digital signal to the analog one is necessary.

Analog-Digital Conversionit is the conversion of an analog signal into digital and consists of the following main steps: sampling, quantization and coding. The diagram of analog-digital conversion of the beep is presented in Fig. 5.2.

The pre-analog beep enters an analog filter that limits the signal frequency band.

Signal sampling is to select an analog signal sample with a given frequency and is determined by the sampling rate. Moreover, the discretization frequency should be at least twice the highest harmonic frequency (frequency component) of the source audio signal. Since a person is able to hear sounds in the frequency range from 20 Hz to 20 kHz, the maximum frequency of sampling of the source sound signal should be at least 40 kHz, i.e., the counts are required to carry out 40,000 times per second. In this regard, in most modern sound systems PC, the maximum frequency of sampling of the sound signal is 44.1 or 48 kHz.

The amplitude quantization is the measurement of the instantaneous values \u200b\u200bof the amplitude of the discrete signal in time and the transformation of it into the discrete time and amplitude. In fig. 5.3 shows the process of quantization by an analog signal level, and instantaneous amplitude values \u200b\u200bare encoded by 3-bit numbers.

Coding is to convert to the digital code of the quantized signal. In this case, the accuracy of measuring during quantization depends on the number of category discharges. If the values \u200b\u200bof amplitudes are recorded using binary numbers and set the length of the code word N.discharges, the number of possible values \u200b\u200bof the code words will be equal 2 N. . There can be the levels of quantization of the amplitude of the countdown. For example, if the value of the countdown amplitude is represented by a 16-bit code word, the maximum number of amplitude gradations (quantization levels) will be 2 16 \u003d 65 536. For an 8-bit view, respectively, we obtain 2 8 \u003d 256 amplitude graduations.

Analog-to-digital conversion is carried out by a special electronic device - analog-digital converttelem.(ADC), in which discrete signal counts are converted into a sequence of numbers. The resulting flow of digital data, i.e. The signal includes both useful and unwanted high-frequency interference, to filter which the digital data obtained is passed through a digital filter.

Digid transformationin general, occurs in two stages, as shown in Fig. 5.4. At the first stage, from a digital data stream with a digital-to-analog converter (DAC), the signal counts are isolated from the sampling frequency. At the second stage, a continuous analog signal is generated from the discrete samples by smoothing (interpolation) using a low frequency filter, which suppresses the periodic components of the spectrum of the discrete signal.

To write and store the audio signal in digital form requires a large amount of disk space. For example, a stereo sound signal with a duration of 60 s, digitized with a sampling frequency of 44.1 kHz with a 16-bit quantization for storage requires about 10 MB on the hard drive.

To reduce the amount of digital data required to represent a sound signal with a given quality, use compression (compression), which consists in decreasing (the number of samples and quantization levels or the number of bits, I.holy on one countdown.

Such methods for encoding audio data using special coding devices allow you to reduce the amount of information flow to almost 20% of the initial one. The selection of the encoding method when recording audio information depends on the set of compression programs - codecs (coding-decoding) supplied with the sound card software or part of the operating system.

Performing the functions of analog-digital and digital signal transformations, the recording module and digital sound reproduction contains the ADC, DAC and control unit, which are usually integrated into one chip, also called the codec. The main characteristics of this module are: sampling frequency; type and discharge of ADC and DAC; method of encoding audio data; opportunity to work in mode FULLDuplex.

The sampling frequency determines the maximum frequency of the recorded or playable signal. To record and reproduce human speech, 6 is 8 kHz; music with low quality - 20 - 25 kHz; To ensure high-quality sound (audio drive), the discretization frequency should be at least 44 kHz. Almost all sound cards support recording and playing a stereo sound signal with a sampling frequency of 44.1 or 48 kHz.

The discharge of the ADC and the DAC determines the discharge of the digital signal representation (8, 16 or 18 bits). The overwhelming majority of sound cards are equipped with 16-bit ADCs and DACs. Such sound maps are theoretically attributed to the Hi-Fi class, which must provide studio sound quality. Some sound cards are equipped with 20- and even 24-bit ADCs and dads, which significantly improves the quality of recording / playing sound.

FULLDuplex(Full duplex) - data transmission mode on a channel, according to which the sound system can simultaneously receive (write) and transmit (reproduce) audio data. However, not all sound cards support this mode in full, as they do not provide high sound quality with intensive data exchange. Such cards can be used to work with voice data in the Internet, for example, when conducting teleconferences, when high sound quality is required.

3. Module of synthesizer

Effective sound system synthesizer allows you to generate almost any sounds, including the sound of real musical instruments. The principle of the synthesizer is illustrated in Fig. 5.5.

Synthesis is the process of recreating the structure of the musical tone (notes). A sound signal of any musical instrument has several time phases. In fig. 5.5, and show the phases of the sound signal arising when the piano key is pressed. For each musical instrument, the view of the signal will be peculiar, but three phases can be distinguished: attack, support and attenuation. The combination of these phases is called an amplitude envelope, the form of which depends on the type of musical instrument. The duration of the attack for different musical instruments varies from units to several tens or even up to hundreds of milliseconds. In the phase, called support, the amplitude of the signal almost does not change, and the height of the musical tone is formed during support. The last phase, attenuation, corresponds to a plot of a fairly rapid decrease in the amplitude of the signal.

In modern synthesizers, the sound is created as follows. A digital device using one of the synthesis methods generates the so-called excitation signal with a given sound height (note), which must have spectral characteristics, as close as possible to the characteristics of the imitated musical instrument in the support phase, as shown in Fig. 5.5, b. Next, the excitation signal is fed to the filter, simulating the amplitude-frequency response of the real musical instrument. An amplitude envelope of the same tool is applied to another filter input. Next, the set of signals is processed in order to obtain special sound effects, for example, echo (reverb), choral performance (Ho-Rus). Next, a digitalog conversion and signal filtering are made using a low frequency filter (FNH). The main characteristics of the synthesizer module:

Sound synthesis method;

Memory size;

The ability to hardware signal processing to create sound effects;

Sound synthesis method,used in the PC sound system, it determines not only the sound quality, but also the composition of the system. In practice, synthesizers generating sound using the following methods are installed on the sound cards.

Freewood Synthesis Method (FrequencyModulationSynthesis- FM synthesis) implies use to generate a voice of a musical instrument at least two generators of the challenges of complex shape. The carrier generator generates the main tone signal, the frequency-modulated signal of additional harmonics, overtones that determine the timbre of the sound tool. The envelope generator manages the amplitude of the resulting signal. The FM generator provides acceptable sound quality, has a low cost, but does not implement sound effects. In this regard, the audio cards using this method are not recommended in accordance with the RS99 standard.

Sound synthesis based on wave table (WaveTableSynthesis - WT-synthesis) is made by using pre-digitized samples of the sound of real musical instruments and other sounds stored in a special ROM, made in the form of a memory chip or integrated in the WT generator's memory microcircuit. WT synthesizer provides high quality sound generation. This synthesis method is implemented in modern audio cards.

Memory sizeon the sound cards with the WT synthesizer, it may increase due to the installation of additional memory elements (ROM) for storing banks with tools.

Sound effectsformulate with a special processor effect, which can be either an independent element (chip), or integrate into the WT synthesizer. For the overwhelming majority of cards with WT-synthesis, the effects of reverb and chorus have become standard. Synthesis of sound based on physical modeling provides for the use of mathematical models of sound formation of real musical instruments for generating digital form and for further conversion to a beep with a DAC. Sound cards using the physical modeling method have not yet been widespread, since there is a powerful PC for their operation.

4. Interface module

The interface module provides data exchange between the sound system and other external and internal devices.

InterfaceISA.in 1998, the PCI interface was displaced in the audio cards.

InterfacePCIprovides a wide bandwidth (for example, version 2.1 is more than 260 Mbps), which allows you to transmit audio data streams in parallel. Using the PCI bus allows you to improve the sound quality, providing signal-to-noise ratio over 90 dB. In addition, the PCI bus ensures the possibility of cooperative sound data processing, when the processing and data transmission tasks are distributed between the sound system and the CPU.

Midi. (Musical.InstrumentDigital.Interface.- The digital interface of musical instruments) is governed by a special standard containing specifications on the hardware interface: channel types, cables, ports, with which MIDI devices are connected one to another, as well as a description of the data exchange of information - information exchange protocol between MIDI devices. In particular, using MIDI commands can be controlled by lighting equipment, video equipment in the process of performing a musical group on the scene. Devices with MIDI interface are connected sequentially by forming a kind of MIDI network that includes a controller - a control device, which can be used as a PC and a music key synthesizer, as well as driven devices (receivers), transmitting information to the controller for 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 the MIDI network is carried out using a special MIDI adapter, which has three MIDI ports: input, output and pass-through data transmission, as well as two connections for connecting the joysticks.

The audio card includes an interface for connecting the CD-ROM drives.
5. Module Mixer

The sound card mixer module performs:

switching (connection / disconnection) of sources and sound signals, as well as regulation of their level;

mixing (mixing) multiple audio signals and adjust the level of the result.

The main characteristics of the mixer module include:

the number of mixed signals on the playback channel;
control signal level in each mixed channel;
regulation of the level of the total signal;
output power amplifier;
the presence of connectors to connect external and internal receivers / sources of sound signals.

Sources and sound signal receivers are connected to the mixer module through external or internal connectors. External sound system connectors are usually located on the rear panel of the system unit housing: JoyStick./ Midi. - to connect a joystick or MIDI adapter; MICIN.- for connecting the microphone; LineIN.- linear input to connect any sources of sound signals; LineOut.- linear output to connect any audio receivers; SPEAKER- To connect headphones (headphones) or passive acoustic system.

Software management mixer is carried out either by Windows tools or using a mixer program supplied with a sound card software.

Compatibility of the audio system with one of the standards of sound cards means that the audio system will provide high-quality sound signals. Compatibility issues are especially important for DOS applications. Each of them contains a list of audio cards, to work with which the DOS application is oriented.

StandardSound.Blaster.support applications in the form of games for DOS, in which sound support is programmed with the orientation of the Sound Blaster Sound Card.

StandardWindowsSound.System.(WSS.) microsoft includes a sound card and a software package focused mainly on a business application.

6. Acoustic system

The acoustic system (AC) directly converts the sound electrical signal into acoustic oscillations and is the last link of the sound-reproducing path.

The AC, as a rule, includes several audio speakers, each of which can have one or more speakers. The number of speakers in speakers depends on the number of components that make up the beep and forming separate audio channels.

For example, a stereo signal contains two components - signals of the left and right stereochanals, which requires at least two columns as part of a stereo acoustic system. A sound signal in Dolby Digital contains information for six audio channels: two front stereo channels, a central channel (dialog channel), two rear channels and an ultra-low channel canal. Therefore, to play the Dolby Digital signal, the acoustic system should have six sound columns.

As a rule, the principle of operation and the internal device of the sound columns of domestic and used in technical means of informatization in the composition of the acoustic system PC is practically not vary.

Basically, the AC for PC consists of two sound columns that provide a stereo signal playback. Usually, each column in the AC for PC has one speaker, however, two are used in expensive models: for high and low frequencies. At the same time, modern models of acoustic systems make it possible to reproduce the sound in almost a whole hearing frequency range due to the use of the special design of the column or loudspeakers.

To reproduce low and ultra-low frequencies with high quality ACs, in addition to two columns, use the third sound unit - subwoofer (Subwoofer.), installed under the desktop. Such a three-component AC for PC consists of two so-called satellite speakers reproducing medium and high frequencies (from about 150 Hz to 20 kHz), and a subwoofer, which reproduces the frequency below 150 Hz.

A distinctive feature of the AC for PC is the possibility of presence of its own built-in power amplifier. The speaker with the built-in amplifier is called active. PassiveAC amplifier has no.

The main advantage of the active speaker is to connect to the linear output of the sound card. The active AC power is carried out either from batteries (batteries), or from an electrical network via a special adapter, made in the form of a separate external unit or power module installed in the body of one of the columns.

The output power of the acoustic systems for the PC may vary in a wide range and depends on the technical characteristics of the amplifier and speakers. If the system is designed for

sounding computer games, sufficient power of 15 -20 W per column for medium-sized space. If you need to ensure good audibility during a lecture or presentation in a large audience, it is possible to use one AU, having a power of up to 30 W per canal. With an increase in the power of the AU, its overall dimensions increase and cost increases.

Modern models of acoustic systems have a nest for headphones, when connecting which sound playback through the speakers is automatically terminated.

The main characteristics of the AC:band reproducible frequency, sensitivity, harmonic coefficient, power.

Band reproducible frequency (FrequencyRespon. sE) - This is an amplitude-frequency dependence of sound pressure, or the dependence of the sound pressure (sound force) from the frequency of alternating voltage, summing up to the coil of the speaker. The frequency band perceived by the Human Eh is ranging from 20 to 20,000 Hz. Columns, as a rule, have a range limited in the low frequency range of 40 - 60 Hz. Solve the reproduction problem of low frequencies allows the use of a subwoofer.

Sensitivity of the sound column (SENSITITIY.) it is characterized by sound pressure, which it creates at a distance of 1 m when the electrical signal is applied to its input with a power of 1 W. In accordance with the requirements of standards, the sensitivity is defined as the average sound pressure in a certain frequency band.

The higher the value of this characteristic, the better the speakers transmits the dynamic range of the music program. The difference between the most "quiet" and the most "loud" sounds of modern phonograms 90-95 dB and more. The speakers with high sensitivity are well reproduced as quiet and loud sounds.

The coefficient of harmonic (TotalHarmonicDistortion.- THD) it estimates nonlinear distortions associated with the appearance of new spectral components in the output signal. The harmonic coefficient is normalized in several frequency bands. 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 of 1000-2000 Hz and 1.0% in the frequency range of 2000 - 6300 Hz. The smaller the value of the harmonic coefficient, the better the AU.

Electric power (PowerHandling), which is withstanding the AU, is one of the main characteristics. However, there is no direct relationship between the power and quality of sound playback. The maximum sound pressure depends

rather, the sensitivity, and the power of the AC basically determines its reliability.

Often, the PC packaging for PC indicates the peak power of an acoustic system, which does not always reflect the real power of the system, because it may exceed a nominal 10 times. Due to the significant difference in the physical processes occurring during the tests of the AC, the values \u200b\u200bof electrical capacities may differ several times. To compare the capacity of various speavers, it is necessary to know which power indicates the manufacturer of products and what kind of test methods it is defined.

Among the manufacturers of high-quality and expensive AC - firms Creative, Yamaha, Sony, Aiwa. AC lower class produce Genius, Altec, Jazz.Hipster.

Some Microsoft column models are connected to a sound card, but to the USB port. In this case, the sound comes on the speakers in digital form, and its decoding produces a small chipset installed in the columns.
7. Directions of improving the sound system

Currently, Intel, Compaq and Microsoft offered a new PC sound system architecture. According to this architecture, the sound signal processing modules are taken out of the PC housing, in which they have electrical interference, and are placed, for example, in the speakers of the acoustic system. In this case, the sound signals are transmitted in digital form, which significantly increases their noise immunity and the quality of sound playback. To transmit digital data in digital form, the use of high-speed USB tires and neck 1394 is provided.

Another direction of improving the sound system is the creation of a bulk (spatial) sound, called three-dimensional, or 3D-SOUND (Three.DimentionalSound.). To obtain a surround sound, a special signal phase processing is performed: the phases of the output signals of the left and right channels are shifted relative to the source. In this case, the human brain property is used to determine the position of the sound source by analyzing the amplitude ratio and the phases of the sound signal perceived by each ear. A sound system user equipped with a special 3D sound processing module, feels the effect of "movement" of the sound source.

The new direction of the use of multimedia technologies is the creation of a home theater based on PC (PC.- Theater), those. variant of the multimedia PC, intended to simultaneously multiply users to monitor the game,

view of the educational program or film in the DVD standard. PC-Theater in its composition has a special multichannel acoustic system forming a surround sound (SurroundSound.). Surround Sound systems create various sound effects in the room, and the user feels that it is located in the center of the sound field, and the sound sources around it. Surround Sound multichannel sound systems are used in cinemas and are already starting to appear in the form of domestic devices.

In multi-channel domestic systems, the sound is recorded on two laser video discs or video cassettes using Dolby Surround developed by Dolby Laboratories. The most famous developments in this direction include:

Dolby (Surround) Pro.Logic.- four-channel sound system containing left and right stereokanlas, central channel for dialogs and rear channel for effects.

DolbySurroundDigital.- Sound system consisting of 5 + 1 channels: left, right, central, left and right channels of rear effects and ultra-low frequency channel. Recording signals for the system is performed as a digital optical phonogram on a film.

In separate models of acoustic speakers, in addition to standard high / low frequency regulators, volume and balance, there are buttons to include special effects, such as ZD-sound, Dolby Surround, etc.

Control questions

What are the main functions of the PC sound system?

What are the main components of the PC sound system?

Based on which reasons, the signal sampling frequency is distinguished during analog-digital conversion?

List the main steps of analog-digital and digital-based transformation.

What basic parameters characterize the recording and sound playback module?

What are the methods of sound synthesis?

What functions perform the module of the mixer and what applies to the number of its main characteristics?

What is the difference between the passive acoustic system from active?

1.SvukovayasystemPC

The audio system of the PC in the form of a sound card appeared in 1989, significantly expanding the possibilities of the PC as a technical means of informatization.

Sound PC System - A complex of software and hardware performing the following functions:

recording audio signals from external sources, such as a microphone or a tape recorder, by converting input analog audio signals into digital and subsequent storage on the hard disk;
play recorded audio data using an external speaker system or headphones (headphones);
playing audio CDs;
mixing (mixing) when recording or playing signals from multiple sources;
simultaneous recording and playback of audio signals (mode Full duplex);
processing of sound signals: editing, combining or separating signal fragments, filtering, change its level;
processing of the sound signal in accordance with the algorithms of volumetric (three-dimensional - 3D-Sound.) sound;
generation using musical instruments synthesizer, as well as human speech and other sounds;
managing the work of external electronic musical instruments through a special MIDI interface.
Download the lecture "Processing and Reproduction Systems of Audio Information"

The classic sound system, as shown in Figure 1, contains:

PC Sound System Structure

record and sound recording module:
synthesizer module;
interface module;
module Mixer;
acoustic system.

The first four modules are usually installed on the sound card. Moreover, there are sound cards without a synthesizer module or a digital sound playback recording module. Each of the modules can be performed either as a separate chip, or to enter a multifunctional chip. Thus, the chipset of the audio system may contain both several and one microcircuit.

Constructive performances of the PC sound system undergo significant changes; There are motherboards with chipset installed on them for sound processing.

2. Moduleentriesandplay

The recording and playback module of the audio system performs analog-to-digital and digital conversion in software transmission mode or transmission to channels DMA (Direct Memory Access — direct memory access channel).

Sound As you know, is longitudinal waves, freely extending in the air or other environment, Therefore, the beep sounds continuously in time and in space.

Sound recording - It is saving information about sound pressure fluctuations at the time of recording. Currently, analog and digital signals are used to record and transmit information about sound. In other words, the beep can be presented in analog or digital form .

it is the conversion of an analog signal into digital and consists of the following main steps: sampling, quantization and coding. The diagram of analog-digital conversion of the beep is presented in Fig. 2.

The pre-analog beep enters an analog filter that limits the signal frequency band.

Signal sampling

Signal sampling It is to select an analog signal sample with a given frequency and define the sampling frequency. Moreover, the discretization frequency should be at least twice the highest harmonic frequency (frequency component) of the source audio signal. Since a person is able to hear sounds in the frequency range from 20 Hz to 20 kHz, the maximum frequency of sampling of the source sound signal should be at least 40 kHz, i.e., the counts are required to carry out 40,000 times per second. In this regard, in most modern sound systems PC, the maximum frequency of sampling of the sound signal is 44.1 or 48 kHz.

Quantization

Quantizationthe amplitude is the measurement of the instantaneous values \u200b\u200bof the amplitude of the discrete signal in time and the transformation into discrete time and amplitude. In fig. 3 shows the quantization process by an analog signal level, and the instantaneous amplitude values \u200b\u200bare encoded by 3-bit numbers.

Coding

Coding It is to convert to the digital code of the quantized signal. In this case, the accuracy of measuring during quantization depends on the number of category discharges. If the amplitude values \u200b\u200bare recorded using binary numbers and set the length of the category N of discharges, the number of possible values \u200b\u200bof the code words will be 2 n. There can be the levels of quantization of the amplitude of the countdown. For example, if the value of the countdown amplitude is represented by a 16-bit code word, the maximum number of amplitude gradations (quantization levels) will be 2 1b \u003d 65 536. For an 8-bit view, respectively, we obtain 2 8 \u003d 256 amplitude gradations.

Analog-digital and digital-making conversion

Analog-Digital Conversion carried out by a special electronic device - analog-to-digital converter(ADC)in which the discrete signal counts are converted into a sequence of numbers. The resulting flow of digital data, i.e. The signal includes both useful and unwanted high-frequency interference, to filter which the digital data obtained is passed through a digital filter.

Digid transformation in general, occurs in two stages, as shown in Fig.4. At the first stage, from a digital data stream with a digital-to-analog converter (DAC), the signal counts are isolated from the sampling frequency. At the second stage, a continuous analog signal is generated from the discrete samples by smoothing (interpolation) using a low frequency filter, which suppresses the periodic components of the spectrum of the discrete signal.

To reduce the amount of digital data required to represent a sound signal with a given quality, use compression (compression) which consists in reducing the number of samples and levels of quantization or the number of bits occurring on one countdown.

Such methods for encoding audio data using special coding devices allow you to reduce the amount of information flow to almost 20% of the initial one. The selection of the encoding method when recording audio information depends on the set of compression software - codecs (Coding-decoding)Supplied with the Software Card Software or the operating system.

Performing analog-digital and digital signal conversion functions, a digital sound recording and playback module contains ADC, DAC and control unit, which are usually integrated into one chip, also called codec .

The main characteristics of this module are: sampling frequency; type and discharge of ADC and DAC; method of encoding audio data; opportunity to work in mode FULLDuplex.

Sampling frequency Defines the maximum frequency of the recorded or playable signal. To record and reproduce human speech, 6 is 8 kHz; music with low quality - 20 - 25 kHz; To ensure high-quality sound (audio drive), the discretization frequency should be at least 44 kHz. Almost all sound cards support recording and playing a stereo sound signal with a sampling frequency of 44.1 or 48 kHz.

The discharge of the ADC and DAC determines the digital signal presence) (8, 16 or 18 bits). The overwhelming majority of sound cards are equipped with 16-bit ADCs and DACs. Such sound maps are theoretically attributed to the Hi-Fi class, which must provide studio sound quality. Some sound cards are equipped with 20- and even 24-bit ADCs and DACs, which significantly improves the quality of recording / playing sound.

Full duplex(full duplex) - Data transfer mode by channel, according to which the audio system can simultaneously receive (write) and transmit (reproduced) audio data. However, not all sound cards support this mode in full, as they do not provide high sound quality with intensive data exchange. Such cards can be used to work with voice data in the Internet, for example, when conducting teleconferences, when high sound quality is required.

3. Modulesynthesizator

Electrical Sound System Synthesizer allows you to generate almost any sounds, including the sound of real musical instruments. The principle of the synthesizer is illustrated in Fig. five

Synthesizing it is the process of recreating the structure of musical tone (notes). A sound signal of any musical instrument has several time phases. In fig. five butthe phases of the audio signal appear when pressing the piano key are shown. For each musical instrument, the view of the signal will be peculiar, but it can be allocated in it three phases: attack, support and attenuation. The combination of these phases is called amplitude envelope , the form of which depends on the type of musical instrument. Duration ataaki. For different musical instruments, it changes from units to several tens or even to hundreds of milliseconds. In phase called Support, the amplitude of the signal almost does not change, and the height of the musical tone is formed during support. Last phase atochingThe site corresponds to a sufficiently rapid decrease in the amplitude of the signal.

In modern synthesizers, the sound is created as follows. A digital device using one of the synthesis methods generates the so-called excitation signal with a given sound height (note), which must have spectral characteristics, as close as possible to the characteristics of the imitated musical instrument in the support phase, as shown in Fig. 5 B. Next, the excitation signal is fed to the filter, simulating the amplitude-frequency response of the real musical instrument. An amplitude envelope of the same tool is applied to another filter input. Next, the set of signals is processed in order to obtain special sound effects, for example, echo (reverb), choral performance (Ho-Rus). Next, a digitalog conversion and signal filtering are made using a low frequency filter (FNH).

The main characteristics of the synthesizer module:

sound synthesis method;
memory size;
the ability to hardware signal processing to create sound effects;
polyphony - the maximum number of simultaneously reproducible sound elements.

Sound synthesis method

Sound synthesis method , used in the PC sound system, it determines not only the sound quality, but also the composition of the system.

In practice, synthesizers generating sound using the following methods are installed on the sound cards.

1. Synthesis method based on frequency modulation (Frequency Modulation Synthesis - FM synthesis) Ensures use to generate a voice of a musical instrument for at least two generators of signals of complex shape. The carrier generator generates the main tone signal, the frequency-modulated signal of additional harmonics, overtones that determine the timbre of the sound tool. The envelope generator manages the amplitude of the resulting signal. The FM generator provides acceptable sound quality, has a low cost, but does not implement sound effects. In this regard, the audio cards using this method are not recommended in accordance with the RS99 standard.

2. Sound synthesis based on wave table (Wave Table Synthesis - WT-synthesis) It is performed by using pre-digitized samples of the sound of real musical instruments and other sounds stored in a special ROM made in the form of a memory chip or integrated in the WT generator memory chip. WT synthesizer provides high quality sound generation. This synthesis method is implemented in modern audio cards.

Memory size on the sound cards with the WT synthesizer, it may increase due to the installation of additional memory elements (ROM) for storing banks with tools.

Sound effectsform with the help of special effect processor which can be either an independent element (microcircuit), or integrate into the WT synthesizer. For the overwhelming majority of cards with WT-synthesis, the effects of reverb and chorus have become standard.

Sound synthesis based on physical modeling . Provides for the use of mathematical models of sound formation of real musical instruments for generating digital form and for further conversion to a beep with a DAC. Sound cards using the physical modeling method have not yet been widespread, since there is a powerful PC for their operation.

Polyphony - The maximum number of simultaneously reproducible elementary sounds. For each type of sound card, the value of the polyphony can be its own. (from 20 and more votes).

4. Moduleinterfaces

Interface module Provides data exchange between the sound system and other external and internal devices.

InterfaceISA. in 1998, the PCI interface was displaced in the audio cards.

RSI interface provides a wide bandwidth (for example, version 2.1 is more than 260 Mbps), which allows you to transmit audio data streams in parallel. Using the PCI bus allows you to improve the sound quality, providing signal-to-noise ratio over 90 dB. In addition, the PCI bus ensures the possibility of cooperative sound data processing, when the processing and data transmission tasks are distributed between the sound system and the CPU.

Midi. MUSICAL INSTRUMENT DIGITAL INTERFACE) - the digital interface of musical instruments) is governed by a special standard containing specifications on the hardware interface: channel types, cables, ports, with which MIDI devices are connected one to another, as well as a description of the procedure for exchanging data - information exchange protocol between MIDI devices. In particular, using MIDI commands can be controlled by lighting equipment, video equipment in the process of performing a musical group on the scene. Devices with MIDI interface are connected sequentially by forming a kind of MIDI network that includes a controller - a control device, which can be used as a PC and a music key synthesizer, as well as driven devices (receivers), transmitting information to the controller for 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.

The audio card includes an interface for connecting the CD-ROM drives.

5. Modulemixer

The sound card mixer module performs:

switching (connection / disconnection) sources and sound signals, as well as regulation of their level;
mixing (mixing) Multiple sound signals and adjust the result of the result.

The main characteristics of the mixer module include:

the number of mixed signals on the playback channel;
control signal level in each mixed channel;
regulation of the level of the total signal;
output power amplifier;
the presence of connectors to connect external and internal receivers / sources of sound signals.

Joystic.k./ Midi. - to connect a joystick or MIDI adapter;
MIC IN. - for connecting the microphone;
Line In. - linear input to connect any sources of sound signals;
Line Out. - linear output to connect any audio receivers;
SPEAKER - To connect headphones (headphones) or passive acoustic system.

Software management mixer is carried out either by Windows tools or using a mixer program supplied with a sound card software.

StandardSound Blaster. support applications in the form of games for DOS, in which sound support is programmed with the orientation of the Sound Blaster Sound Card.

StandardWindows Sound System (WSS) microsoft includes a sound card and a software package focused mainly on a business application.

6. Acousticsystem

Acoustic system (AC) Immediately converts a sound electrical signal into acoustic oscillations and is the last link of the sound-reproducing path.

Acoustic system

The AC, as a rule, includes multiple audio speakersEach of which can have one or more speakers.

The number of speakers in speakers depends on the number of components that make up the beep and forming separate audio channels.

For example, stereo signal contains two components - Signals of the left and right stereokanal, which requires at least two columns as part of a stereo acoustic system.

A beep in Dolby Digita formatl contains information for six audio channels: two front stereo channels, central channel (channel dialogs), two rear channels and an ultra-low channel canal. Therefore, to play the Dolby Digital signal, the acoustic system should have six sound columns.

Mostly AC for PC consists from two audio columnswhich provide playback of the stereo signal. Usually, each column in the AC for PC has one speaker, however, two are used in expensive models: for high and low frequencies. At the same time, modern models of acoustic systems make it possible to reproduce the sound in almost a whole hearing frequency range due to the use of the special design of the column or loudspeakers.

To play low and ultra-low frequencies with high quality in the AU, in addition to two columns, the third sound unit is used - subwoofer (Subwoofer. ) , installed under the desktop. Such a three-component speaker for PC consists of two so-called satellite speakers reproducing medium and high frequencies (from about 150 Hz to 20 kHz), and the subwoofer, the reproducing frequency below 150 Hz.

A distinctive feature of the AC for PC - the ability to have its own Built-in power amplifier. The speaker with the built-in amplifier is called active. PassiveAC amplifier has no.

The main advantage of the active speaker is connection capabilities to linear audio card output. The active AC power is carried out either from batteries (batteries), or from an electrical network via a special adapter, made in the form of a separate external unit or power module installed in the body of one of the columns.

The output power of the acoustic systems for the PC may vary in a wide range and depends on the technical characteristics of the amplifier and speakers. If the system is designed to sound computer games, sufficient power is 15-20 W per column for the mid-size room. If you need to ensure good audibility during a lecture or presentation in a large audience, it is possible to use one AU, having a power of up to 30 W per canal. With an increase in the power of the AU, its overall dimensions increase and cost increases.

Modern models of acoustic systems have a nest for headphones, when connecting which sound playback through the speakers is automatically terminated.

Acoustic system microlab.

The main characteristics of the AC:

reproducible frequency band
sensitivity,
harmonic coefficient
power.

Band reproducible frequency (FrequencyResponse)- this is an amplitude-frequency dependence of the sound pressure, or the dependence of the sound pressure (sound force) from the frequency of the variable voltage, summing up to the coil of the speaker.

The frequency band perceived by the Human Eh is ranging from 20 to 20,000 Hz.

Columns, as a rule, have a range limited in the low frequency range of 40 - 60 Hz. Solve the reproduction problem of low frequencies allows the use of a subwoofer.

Sensitivity of the sound column (Sensitivity) it is characterized by sound pressure, which it creates at a distance of 1 m when the electrical signal is applied to its input with a power of 1 W.

In the corresponding PC system in the form of a sound card appeared in 1989, significantly expanding the possibilities of the PC as a technical means of informatization. The source and sound signal receivers are connected to the mixer module through external or internal connectors. The external sound system connectors are usually located on the rear panel of the system unit housing: / H3NBSP; Stron / BGBB sensation with standards requirements The sensitivity is defined as the average sound pressure in a specific frequency band.

The higher the value of this characteristic, the better the speakers transmits the dynamic range of the music program. The difference between the "quiet" and the most "loud" sounds of modern phonograms 90 - 95 dB and more. / EMAs with high sensitivity quite well reproduce both quiet and loud sounds.

The coefficient of harmonic

The coefficient of harmonic (Total Harmonic Distortion.- THD) it estimates nonlinear distortions associated with the appearance of new spectral components in the output signal.

The harmonic coefficient is normalized in several frequency bands. 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 of 1000-2000 Hz and 1.0% in the frequency range of 2000 - 6300 Hz.

The smaller the value of the harmonic coefficient, the better the AU.

Electric power

Electric power (Power Handling),which is withstanding the AU, is one of the main characteristics. However, there is no direct relationship between the power and quality of sound playback. The maximum sound pressure depends on the sensitivity, and the power of the AC is mainly determines its reliability.

Among manufacturers of high quality and expensive AC - firms Creative, Yamaha, Sony, AIWA. AC lower class manufactures Genius, Altec, Jazz Hipster.

Some Microsoft column models are connected to a sound card, but to the USB port. In this case, the sound comes on the speakers in digital form, and its decoding produces a small chipulb, each of which can have one or more speakers. SET installed in columns.

7. Directionsimprovingsoundsystems

Currently, Intel, Compaq and Microsoft offered new Architecture of the PC Sound System. According to this architecture sound signal processing modules are taken out of PC housingwhere they act on electrical interference, and are placed, for example, in the speakers of the acoustic system. In this case, the sound signals are transmitted in digital form, which significantly increases their noise immunity and the quality of sound playback. To transmit digital data in digital form, the use of high-speed USB and IEEE 1394 tires is provided.

Another direction of improving the sound system is to create volume (spatial) sound, called three-dimensional, or 3D-SOUND Three Dimient Sound) . To obtain a surround sound, a special signal phase processing is performed: the phases of the output signals of the left and right channels are shifted relative to the source. In this case, the human brain property is used to determine the position of the sound source by analyzing the amplitude ratio and the phases of the sound signal perceived by each ear. A sound system user equipped with a special 3D sound processing module, feels the effect of "movement" of the sound source.

The new direction of application of multimedia technologies is creation of home theater based on PC (PC.— Theater) ,those. A variant of a multimedia PC intended to simultaneously multiple users to monitor the game, view the educational program or film in the DVD standard. PC-Theater in its composition has a special multi-channel speaker system forming sound Sound ( Surround Sound.). Surround Sound systems create various sound effects in the room, and the user feels that it is located in the center of the sound field, and the sound sources around it. Multichannel Surround Sound Systems Used in cinemas and are already starting to appear in the form of domestic devices.

In multichannel domestic systems, the sound is recorded on two laser video discs or video cassettes Dolby Surrounddeveloped by Dolby Laboratories. The most famous developments in this direction include:

Dolby (Surround) Pro. Logic. - a four-channel sound system containing left and right stereokanlas, a central channel for dialogs and a rear channel for effects.

Dolby Surround Digital. - a sound system consisting of 5 + 1 channels: the left, right, central, left and right channels of the rear effects and the ultra-low channel channel. Recording signals for the system is performed as a digital optical phonogram on a film.

Controlsquestions

What are the main functions of the PC sound system?
What are the main components of the PC sound system?
Based on which reasons, the signal sampling frequency is distinguished during analog-digital conversion?
List the main steps of analog-digital and digital-based transformation.
What basic parameters characterize the recording and sound playback module?
What are the methods of sound synthesis?
What functions perform the module of the mixer and what applies to the number of its main characteristics?
What is the difference between the passive acoustic system from active?

Sound system Personal computer is used to play sound effects and speech accompanying the reproducible video information, and includes:

record / Play Module;
synthesizer;
interface module;
mixer;
acoustic system.

The components of the sound system (excluding the speaker system) are constructively drawn up as a separate sound card or partially implemented as chips on the computer's motherboard.

As a rule, signals at the input and output of the recording / playback module have an analog form, but the processing of sound signals is in digital form. Therefore, the main functions of the recording / playback module are reduced to analog-to-digital and digital-analog conversion.

To do this, the input analog signal is subjected to pulse-code modulation (ICM), the essence of which is to discretize the time and representation (measurement) of the analog signal amplitude into the discrete moments of time in the form of binary numbers. It is necessary to choose the sampling frequency and the discharge of binary numbers so that the accuracy of the analog-to-digital conversion corresponds to the requirements for the quality of sound playback.

According to the Kotelnikov Theorem, if the time sampling step separating the adjacent samples (measured amplitudes) does not exceed half the oscillation period of the highest component in the frequency spectrum of the transformed signal, then the discretion of time does not make distortion and does not lead to loss of information. If it is enough for high-quality sound to play a spectrum of 20 kHz wide, the sampling frequency should be not lower than 40 kHz. In sound systems of personal computers (PCs), the frequency of sampling is usually taken equal to 44.1 or 48 kHz.

The limited bittenness of binary numbers representing the amplitudes of the signals causes the discretization of the values \u200b\u200bof the signal. In the audio cards, in most cases, 16-bit binary numbers are used, which corresponds to 216 quantization levels or 96 dB. Sometimes 20- or even 24-bit analog-digital conversion are used.

Obviously, improving the quality of sound by increasing the frequency F of sampling and the number of quantization levels leads to a significant increase in the volume of digital data, since

S \u003d 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 amount of data increases twice. Thus, at a frequency of 44.1 kHz and 216 quantization levels, the amount of information for the presentation of a sound stereo fragment with a duration of 1 min is about 10.6 MB. To reduce the requirements for both the memory capacity for storing audio information and the bandwidth of the data transmission channels, compression (compression) of information is used.

The interface module is used to transmit digitized audio information to other PC devices (memory, acoustic system) through the computer tires. The ISA bus bandwidth is usually not enough, so other tires are used - PCI, a special MIDI musical instruments interface or some other interfaces.

Using a mixer, you can mix the sound signals, creating a polyphonic sound, impose musical accompaniment for speech, accompanying multimedia fragments, etc.

The synthesizer is designed to generate sound signals, most often to simulate the sound of various musical instruments. For synthesis, frequency modulation, wave tables, mathematical modeling are used. Source data for synthesizers (Music and Toold Codes) are usually presented in MIDI format (MID extension in file name). Thus, when applying the frequency modulation method, control the frequency and amplitude of the summed signals from the main generator and the overtone generator. According to the method of the wavelength, the resulting signal is obtained by combining digitized samples of sounds derived from real musical instruments. In the method of mathematical modeling, mathematical models of sounds are used instead of experimentally obtained samples.

Lecture number 6. Sound-reproducing systems

1. Basic components of the PC sound subsystem.

2. Principles of processing sound information.

The main components of the sound subsystem PC.

The audio system of the PC in the form of a sound card appeared in 1989, significantly expanding the possibilities of the PC as a technical means of informatization.

Sound PC System- A complex of software and hardware performing the following functions:

· Recording sound signals from external sources, for example, microphone or tape recorder, by converting input analog audio signals into digital and subsequent storage on the hard disk;

· Play recorded audio data using an external speaker system or headphones (headphones);

· Play audio CDs;

· Mixing (mixing) when recording or playing signals from several sources;

· Simultaneous recording and playback of audio signals (mode Full Duplex);

· Sound signal processing: editing, combining or separating signal fragments, filtering, change of its level;

· Treatment of sound signal in accordance with volumetric algorithms (three-dimensional - 3D-Sound)sound;

· Generation using musical instruments synthesizer, as well as human speech and other sounds;

· Management of external electronic musical instruments through a special MIDI interface.

The PC sound system is structurally sound cards or installed in the motherboard slot, or the other subsystem of the PC integrated on the motherboard or the expansion card, as well as the recording and playback devices (acoustic system). Separate sound system functional modules can be performed as childboards installed in the appropriate sound card connectors.

The classic sound system, as shown in Fig. 1, contains:

Recording and sound recording module;

Synthesizer module;

Interface module;

Module Mixer (provides data exchange between the sound system and other devices - both external and internal.);

Acoustic system.

Fig. one. Structure of the PC sound system.

Constructive performances of the PC sound system undergo significant changes; There are motherboards with chipset installed on them for sound processing.

Sound equipment and programs.

Special audio adapters are responsible for playing and recording sound adapters. Sound adapter It contains another specialized processor, thereby freeing the main processor from the functions to control the playback of sound. Using the audio adapter, you can record audio information, reproduce speech and music. Also, modern soundboards make it possible to produce sound processing, installation of musical compositions. In addition to the sampling frequency encoded with a given frequency, it is possible to play music created by computer commands. The number of votes is the sound card parameter that determines the maximum number of simultaneously synthesized sounds. The main direction of the development of modern sound boards is the support of surround sound. In this case, the possibility of positioning sound sources in space appears. For reproducing surround sound, at least two acoustic systems are needed. However, to obtain a better effect from surround sound, it is better to use four columns - two in front and two from behind.

The overwhelming majority of modern computers are equipped with a sound card. Good Sound Blaster Audigy Sound Boards Different Versions Releases Creative. At the same time, many motherboards are currently supporting high-quality six-channel sound.

It is extremely important to obtain high-quality sound to have good acoustic systems. Modern soundboards have a digital SPDIF output, which allows connecting to household appliances. However, it is often more convenient to use their own acoustics for the computer. When using a computer to view the videos recorded on the DVD, you must use a modern speaker system from five columns and a subwoofer.

In order to create your own musical works, you may need a special keyboard connected to the MIDI interface. Musical keyboards connected to the sound card differ by the amount of octave (usually from three to seven), as well as the number of keys and their size. The most famous manufacturers are Korg, Roland, Yamaha. Not bad amateur keyboards produces Casio.

For high-quality voice recording, you need to use the corresponding microphones. Simple computer microphones do not provide high sound quality. In addition, the microphone input of most sound boards also do not have good quality. Therefore, it is recommended to use a microphone amplifier that connects to the linear audio input. The microphone amplifier will connect two microphones, which will allow you to record stereo sound.

Recently, miniature digital players that store music in MP3 format were widespread. Music from a computer is recorded in memory of such a device, after which it can be heard anywhere through the headphones.

As an additional sound source, a computer radio can be considered for a computer. It can be implemented as an additional fee, and can be connected to the USB port.

Of course, working with sound on a computer is unthinkable without special programs. The simplest programs for working with sound are included in all versions of Windows. With their help, you can adjust the volume of different sound sources, set the sensitivity of the microphone and linear input. In addition, you can record a small sound fragment, make simple conversion with it and write the result to the file. Also in Windows included CD playback and multimedia files. You can record music on digital players, listen to music from the Internet.

When using a music keyboard, you need to work with sound real time. The most powerful such program is CakeWalk Home Studio, but you can do and easily programs.

For sound processing, use audio editor. The best sound editors are Sound Forge and Wavelab programs. Cool Edit editor is used for multichannel mounting. To create and edit music, as well as to add vocals to music, programs are used, called MIDI sequences and audio. The best programs of this class are CakeWalk Sonar and Cubase VST.

Singing karaoke has recently been quite popular. There are several programs for creating karaoke files and to play them. Quite a convenient program Karaoke Galaxy Maker, which allows you to create karaoke. To play such files, use Karaoke Galaxy Player or Vanbasco's Karaoke Player.