Rule 2. Before turning on the device in the network, look at what is written on the back of the device.
Check the voltage at the output of the autotransformer at idle before connecting the device to it.
Check the value of the voltage supplying the device during the production of copies.
When finished, unplug the autotransformer from the mains. Do not leave the autotransformer energized!
Rule 3. It is very important to consider the requirements for installing a copier. The device must be installed on a flat horizontal surface. Deviation from the horizontal position leads to the redistribution of toner and media in the cartridge of the machine in the direction of the slope. Accordingly, their mixing is difficult and the uniformity of the coating of the magnetic roller with toner is disturbed.
Laboratory work. Studying the principle of operation of sound processing devices
Goal of the work
To study the block diagram of the PC sound system, the components of the sound system.
7.2 Work progress:
1) Familiarize yourself with the block diagram of the PC sound system.
2) To study the main components (modules) of the sound system.
3) Familiarize yourself with the principle of operation of the synthesizer module.
4) Familiarize yourself with the principle of operation of the interface module.
5) Familiarize yourself with the principle of operation of the mixer module.
1) Theme, goal, course of work;
2) Formulation and description of the individual task;
1) Name the main modules of a classic sound system.
2) What is the essence of synthesizing.
3) Name the phases of the audio signal.
4) What methods of sound synthesis do you know?
5) List modern interfaces of sound devices.
Methodical instructions.
PC sound system structure
Sound system A PC is structurally a sound card, either installed in a slot on the motherboard, or integrated on the motherboard or expansion card of another PC subsystem.
A classic sound system, as shown in Figure 23, contains:
1. sound recording and playback module;
2. synthesizer module;
3. interface module;
4. mixer module;
5. acoustic system.
Figure 23 - Structure of the PC sound system
synthesizer module
The electronic musical digital synthesizer of the sound system allows you to generate almost any sounds, including the sound of real musical instruments. The principle of operation of the synthesizer is illustrated in Figure 24.
Synthesizing is the process of recreating the structure of a musical tone (note). The sound signal of any musical instrument has several time phases. In Figure 24, A shows the phases of the sound signal that occurs when you press ml vichy piano. For each musical instrument, the type of signal will be peculiar, but three phases can be distinguished in it: attack, support and decay. The combination of these phases is called amplitude envelope, the shape 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 hundreds of milliseconds. In the phase called support, the amplitude of the signal does not change much, and the pitch of the musical tone is formed during the support. The last phase, attenuation, corresponds to a section of a fairly rapid decrease in the signal amplitude.
In modern synthesizers, the sound is created as follows. A digital device using one of the synthesis methods generates a so-called excitation signal with given height sound (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 24, b. Next, the excitation signal is fed to a filter that simulates the frequency response of a real musical instrument. The other input of the filter is fed with the amplitude envelope signal of the same instrument. Further, the set of signals is processed in order to obtain special sound effects, for example, echo (reverberation), choral performance. Next, a digital-to-analogue conversion and signal filtering are performed using a filter low frequencies(LPF).
The main characteristics of the synthesizer module:
sound synthesis method;
Memory size;
Possibility of hardware signal processing to create sound effects;
Polyphony - the maximum number of simultaneously reproduced elements of sounds.
sound synthesis method, used in a PC sound system determines not only the sound quality, but also the composition of the system. In practice, synthesizers are installed on sound cards that generate sound using the following methods.
Figure 24 - The principle of operation of a modern synthesizer: a - phases of an audio signal; b - synthesizer circuit
Synthesis method based on frequency modulation ( Frequency Modulation Synthesis - FM synthesis) involves the use of at least two signal generators of complex shape 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, overtones that determine the timbre of the sound of a particular instrument. Envelope generator controls the amplitude of the resulting signal FM generator provides acceptable sound quality, is inexpensive, but does not implement sound effects. As such, sound cards using this method are not recommended under the PC99 standard.
Sound synthesis based on the wave table (Wave Table Synthesis - WT-synthesis) is produced 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 a WT generator integrated into the memory chip. WT synthesizer provides sound generation with high quality. This synthesis method is implemented in modern sound cards.
Memory on sound cards with a WT synthesizer, it can be increased by installing additional memory elements (ROM) for storing banks with instruments.
Sound effects are formed using a special effect processor, which can be either 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.
Synthesis of sound based on physical modeling involves the use of mathematical models of sound production of real musical instruments for generation in digital form and for further conversion into an audio signal using a DAC. Sound cards using the physical modeling method have not yet become widespread, since they require a powerful PC to work.
Interface module
The interface module provides data exchange between the sound system and other external and internal devices.
ISA interface in 1998 it was superseded in sound cards by the PCI interface.
PCI interface provides a wide bandwidth (for example, version 2.1 - more than 260 Mbps), which allows you to transmit streams of audio data in parallel. Using the PCI bus allows you to improve the sound quality, providing a signal-to-noise ratio of over 90 dB. In addition, the PCI bus enables cooperative audio data processing, where data processing and transmission tasks are shared between the audio system and the CPU.
MIDI (Musical Instrument Digital Interface)- digital interface musical instruments) is regulated by a special standard containing specifications for the hardware interface: types of channels, cables, ports with which MIDI devices are connected to one another, as well as a description of the data exchange procedure - the protocol for exchanging information between MIDI devices. In particular, using MIDI commands, you can control lighting equipment, 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 that 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. There is no limit to the total length of the MIDI chain, but the maximum cable length between two MIDI devices must not exceed 15 meters.
Connecting a PC to a MIDI network is carried out using a special MIDI adapter, which has three MIDI ports: input, output and data through, as well as two connectors for connecting joysticks.
The sound card includes an interface for connecting CD-ROM drives.
7.5.4 Mixer module
The sound card mixer module performs:
Switching (connection / disconnection) of sources and receivers of sound signals, as well as regulation of their level;
Mixing (mixing) several audio signals and adjusting the level of the resulting signal.
The main features of the mixer module include:
The number of mixed signals on the playback channel;
Adjustment of the signal level in each mixed signal;
Adjustment of the level of the total signal;
Amplifier output power;
Existence of sockets for connection of external and internal receivers/sources of sound signals.
Audio sources and receivers are connected by the mixer module via external or internal connectors. External audio connectors are usually located on the back of the case system block: Joystick/MIDI- for connecting a joystick or a MIDI adapter; Mic In- to connect a microphone; Line In- line input for connection of any sources of sound signals; line out- line output for connection of any receivers of sound signals; speaker for connecting headphones (headphones) or a passive speaker system.
The software control of the mixer is carried out either Windows tools, or using the mixer program supplied with software sound card
Sound system compatibility with one of the sound card standards means that the sound system will provide high-quality reproduction of audio 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.
Sound Blaster standard support applications in the form of games for DOS, in which the soundtrack is programmed with a focus on sound cards of the Sound Blaster family.
Windows Sound System Standard (WSS) Microsoft includes a sound card and a software package focused primarily on business applications.
Examples of performing individual tasks
Model 1 - SB PCI CMI 8738 Sound Card
Figure 25 - Appearance sound card SB PCI CMI 8738
Description: 5.1 sound card
Hardware type: Multimedia sound card
Chip: C-Media 8738
Analog inputs: 2
Analog outputs: 3
Connectors: External: line in, microphone in, front speaker out, rear speaker out, center/subwoofer out; internal: line-in, CD-in
Ability to connect 4 speakers: Yes
Dolby Digital 5.1 Support: Yes
EAX support: EAX 1.0 and 2.0
Interface: PCI
Ability to connect 6 speakers: Yes
Model 2 - SB PCI Terratec Aureon 5.1 PCI Sound Card
Figure 26 - External view of the SB PCI Terratec Aureon 5.1 PCI sound card
Description: 6-channel sound card.
3D Sound: EAX 1.0, EAX 2.0, Sensaura, Aureal A3D 1.0, Environment FX, Multi Drive, Zoom FX, I3DL2, DirectSound 3D
Chip: C-media CMI8738/PCI-6ch-MX
DAC: 16bit/48kHz
ADC: 16bit/48kHz
Number of columns: 5.1
Analog inputs: 1x unbalanced miniJack connector, miniJack microphone input, internal connectors: AUX, CD-in.
Analog outputs: MiniJack audio outputs for connecting 5.1 acoustics (front-out, rear-out, sub/senter-out).
S/PDIF: 16bit/48kHz
Digital inputs/outputs: Optical (TOSLINK) output, Optical (TOSLINK) input.
Sampling frequency: 44.1, 48 kHz
System requirements (minimum): Intel PentiumIII, AMD K6-III 500 MHz 64 MB memory
Interface: PCI 2.1, 2.2
know:
PC sound system. Composition of the PC sound system. working principle and specifications sound cards. Directions for improving the sound system. The principle of sound information processing. Specification of sound systems.
Guidelines
PC sound system- a set of software and hardware tools that perform the following functions:
Figure 10 - Structure of the PC sound system
The classic sound system, as shown in fig. 5.1 contains:
The designs of the PC sound system are undergoing significant changes; meet motherboards with Chipset installed on them for sound 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:
Topic 6.2 Audio Information Processing Interface Module
The student must:
have an idea:
know:
Composition of the sound subsystem of the PC. Recording and playback module. synthesizer module. Interface module. Mixer module. The principle of operation and technical characteristics of acoustic systems. Software. Sound file formats. Speech recognition tools.
Guidelines
Sound System Recording and Playback Module performs analog-to-digital and digital-to-analog conversions in the mode of program transmission of audio data or their 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 represented in analog or digital form.
In most cases, the audio signal is fed to the input of a 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, inverse conversion is necessary. digital signal to analog.
Analog-to-digital conversion is the conversion of an analog signal into a digital one and consists of the following main steps: sampling, quantization and encoding.
^ The pre-analogue audio signal is fed to an analog filter that limits the signal's bandwidth.
Sampling of the signal consists in sampling samples of the analog signal with a given periodicity and is determined by the sampling frequency. Moreover, the sampling frequency must be at least twice the frequency of the highest harmonic (frequency component) of the original audio signal.
Amplitude quantization is a measurement of the instantaneous values of the amplitude of a time-discrete signal and its transformation into a signal discrete in time and amplitude. Figure 11 shows the analog signal level quantization process, with the instantaneous amplitude values encoded as 3-bit numbers.
^
Figure 11 - Scheme of analog-to-digital conversion of an audio signal
Coding consists in 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 - Discretization in time and quantization in terms of the level of the analog signal of the quantization of the amplitude of the reading.
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 received digital data stream, i.e. the signal includes both useful and unwanted high-frequency interference, for filtering which the received digital data is passed through a digital filter.
Digital-to-analogue 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 (compression) is used, which consists in reducing the number of samples and quantization levels or the number of bits per sample.
^
Figure 13 - Scheme of digital-to-analogue conversion
Such methods of encoding audio data using special encoders can reduce the amount of information flow to almost 20% of the original. The choice of encoding method for 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 conversions, the digital audio recording and playback module contains an ADC, a DAC and a control unit, which are usually integrated into one chip, also called a codec. The main characteristics of this module are: sampling rate; type and capacity of ADC and DAC; a method for encoding audio data; the ability to work in Full Duplex mode.
The sampling rate determines the maximum frequency of the signal being recorded or played back. To record and reproduce human speech, 6 - 8 kHz is sufficient; music with low quality - 20 - 25 kHz; For 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 44.1 kHz or 48 kHz sampling rates.
^ The bit depth of the ADC and DAC determines the bit depth of the digital signal representation (8, 16 or 18 bits).
Full Duplex (full duplex) - data transmission mode over the channel, according to which the sound system can simultaneously receive (record) and transmit (play back) audio data. However, not all sound cards fully support this mode, since they do not provide high sound quality with intensive data exchange. Such cards can be used to work with voice data on the Internet, for example, when conducting teleconferencing, when high sound quality is not required.
synthesizer module
The electronic musical digital synthesizer of the sound system allows you to generate almost any sounds, including the sound of real musical instruments. The principle of operation of the synthesizer is illustrated in Figure 14.
Synthesizing 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 occurs when a piano key is pressed. For each musical instrument, the type of signal will be peculiar, but three phases can be distinguished in it: attack, support and decay. The combination of these phases is called the amplitude envelope, the shape 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 hundreds of milliseconds. In the phase called support, the amplitude of the signal does not change much, and the pitch of the musical tone is formed during the support. The last phase, attenuation, corresponds to a section of a fairly rapid decrease in the signal amplitude.
In modern synthesizers, the 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 that are 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 frequency response of a real musical instrument. The other input of the filter is fed with the amplitude envelope signal of the same instrument. Further, the set of signals is processed in order to obtain special sound effects, for example, echo (reverberation), choral performance (chorus). Further, digital-to-analogue conversion and signal filtering are performed using a low-pass filter (LPF).
Figure 15 - The principle of operation of a modern synthesizer: a - phases of an audio signal; 6 - synthesizer circuit
The main characteristics of the synthesizer module:
The synthesis method based on frequency modulation (Frequency Modulation Synthesis - FM-synthesis) involves the use of at least two complex-shaped signal generators 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, overtones that determine the timbre of the sound of a particular instrument. The envelope generator controls the amplitude of the resulting signal. The FM generator provides acceptable sound quality, is not expensive, but does not implement sound effects. As such, sound cards using this method are not recommended under the PC99 standard.
Sound synthesis based on the wave table (Wave Table Synthesis - WT-synthesis) 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 a WT generator integrated into the memory chip. WT synthesizer provides high quality sound generation. This synthesis method is implemented in modern sound cards.
^ The amount of memory on sound cards with WT-synthesizer can be increased by installing additional memory elements (ROM) to store instrument banks.
Sound effects are formed using a special effect processor, which can be either 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. Synthesis of sound based on physical modeling involves the use of mathematical models of sound production of real musical instruments for generation in digital form and for further conversion into an audio signal using a DAC. Sound cards using the physical modeling method have not yet become widespread, since they require a powerful PC to work.
Interface module provides communication between the sound system and other external and internal devices.
The PCI interface provides a wide bandwidth (for example, version 2.1 - 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 a signal-to-noise ratio of over 90 dB. In addition, the PCI bus enables cooperative audio data processing, where data processing and transmission tasks are shared between the audio system and the CPU.
MIDI (Musical Instrument Digital Interface - digital interface of musical instruments) is regulated by a special standard containing specifications for a hardware interface: types of channels, cables, ports with which MIDI devices are connected one to another, as well as a description of the data exchange procedure - an information exchange protocol between MIDI devices. In particular, using MIDI commands, you can control lighting equipment, 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. There is no limit to the total length of the MIDI chain, but the maximum cable length between two MIDI devices must not exceed 15 meters.
Connecting a PC to a MIDI network is carried out using a special MIDI adapter, which has three MIDI ports: input, output and data 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 performs:
The software control of the mixer is carried out either by means of Windows or with the help of the mixer program supplied with the sound card software.
Sound system compatibility with one of the sound card standards means that the sound system will provide high-quality reproduction of audio 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 games for DOS, in which the soundtrack 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 a software package focused primarily on business applications.
Acoustic system (AC) directly converts the sound electrical signal into acoustic vibrations and is the last link in the sound reproducing path. The composition of the speakers, as a rule, includes several speakers, each of which can have one or more speakers. The number of speakers in the speakers depends on the number of components that make up the audio signal and form separate audio channels.
As a rule, the principle of operation and the internal structure of sound speakers for domestic use and those used in technical means of informatization as part of a PC speaker system practically do not differ.
Basically, a PC speaker consists of two 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 allow you to reproduce sound in almost everything audible. frequency range thanks to the use of a special design of the speaker cabinet or loudspeakers.
To reproduce low and ultra-low frequencies with high quality, in addition to two speakers, a third sound unit is used in the speakers - a subwoofer (Subwoofer), installed under the desktop. This 3-way PC speaker consists of two so-called satellite speakers that reproduce mid and high frequencies (approximately 150 Hz to 20 kHz) and a subwoofer that reproduces frequencies below 150 Hz.
A distinctive feature of speakers for PC 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 an active speaker is the ability to connect to the line-out of a sound card. The active speaker is powered either from batteries (accumulators) or from the mains through a special adapter made in the form of a separate external unit or power module installed in the case of one of the speakers.
The output power of PC speakers can vary widely and depends on the specifications of the amplifier and speakers. If the system is intended for sound computer games, enough power 15 - 20 watts per speaker 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 watts per channel. With an increase in the power of the AU, its overall dimensions increase and the cost increases.
^ The main characteristics of the speakers: the band of reproducible frequencies, sensitivity, harmonics, power.
The band of reproducible frequencies (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 use of a subwoofer can solve the problem of low frequency reproduction.
Sound column sensitivity (Sensitivity) is characterized by the sound pressure that it creates at a distance of 1 m when applied to its input electrical signal power of 1 W. 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 musical program. The difference between the "quietest" and the "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 the non-linear distortion associated with the appearance of new spectral components in the output signal. The harmonic coefficient is normalized in several frequency ranges. For example, for high-quality Hi-Fi speakers, this coefficient should not exceed: 1.5% in the frequency range of 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 value of the harmonic coefficient, the better the speaker.
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 the sensitivity, and the power of the AC- mainly determines its reliability.
Often, on the packaging of speakers for a PC, the value of the peak power of the speaker system is indicated, which does not always reflect the real power of the system, since it can exceed the nominal power by 10 times. Due to the significant difference in the physical processes occurring during the tests of the AU, the values of electrical powers may differ by several times. To compare the power of different speakers, you need to know exactly what power the product manufacturer indicates and what test methods it is determined by.
Some Microsoft speaker models do not connect to a sound card, but to USB port. In this case, the sound enters the speakers in digital form, and its decoding is performed by a small Chipset installed in the speakers.
Questions for self-control:
Practice 8. PC sound system
The student must:
have an idea:
know:
be able to:
Section 7 Printing Devices
Topic 7.1 Printer
The student must:
have an idea:
know:
General characteristics of printing output devices. Classification of printing devices. Impact printers: principle of operation, mechanical components, features of operation, technical characteristics, operating rules. Basic modern models.
^ Inkjet printers: principle of operation, mechanical components, features of work, technical characteristics, operating rules. Basic modern models.
Laser printers: principle of operation, mechanical components, features of work, specifications, operating rules. Basic modern models.
Guidelines
Printers- devices for outputting data from a computer that convert information ASCII codes into their corresponding graphic characters and fix these characters on paper.
Printers can be classified according to a number of characteristics:
When working in text mode The printer receives from the computer character codes that need to 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 stored in the printer's ROM and can only be read from there.
To print text information, there are print modes that provide different quality:
According to the method of applying an image to paper, printers are divided into impact printers, inkjet printers, photoelectric printers, and thermal printers.
Sound Systems for the IBM PCINTRODUCTION
The interaction of a person with a computer should be primarily mutual (that's why it is communication). Reciprocity, in turn, provides for the possibility of communication both between a person and a computer, and a computer with a person. It is an undeniable fact that visual information, supplemented by sound, is much more effective than a simple visual impact. Try, plugging your ears, chat with someone for at least a minute, I doubt that you will get great pleasure, as well as your interlocutor. However, while many orthodox programmers/designers still do not want to admit that the sound effect can play the role of not only a signaling device, but an information channel, and, accordingly, due to inability and / or unwillingness, they do not use the possibility of non-visual communication between a person and a computer in their projects, but even they never watch TV without sound. At present, any major project that is not equipped with multimedia tools (hereinafter, under the word "media tools" we will primarily mean 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's memory in digital form (in the form of samples) and with the help of the DAC is transformed into an analog signal, which is fed to amplifying equipment, and then to headphones, speakers, etc.
2. Synthesis. The computer sends note 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 with the mathematical representation of the 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 perform very well at playing "computer" music, but the attempt to simulate the sound of live instruments does not work very well. The disadvantage of FM synthesis is that it is very difficult (almost impossible) to create really realistic instrumental music with a lot 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 types of chips, such as the Yamaha YMF262 (OPL-3) FM.
b) synthesis according to the wavetable synthesis, with this method of synthesis, the given sound is "collected" not from the sines of mathematical waves, but from a set of really sounded instruments - samples. Samples are stored in the RAM or ROM of the sound card. A special sound processor performs operations on sounds (with the help of various kinds of mathematical transformations, the pitch and timbre are changed, the sound is supplemented with special effects).
Since the 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 more and more popular now. An example of a popular map using WS Gravis Ultra Sound (GUS).
3. MIDI. Computer sends to MIDI interface special codes, each of which indicates an action that the MIDI device (usually a synthesizer) should perform. (General) MIDI is the main standard for most sound cards. Sound card, independently interprets the sent codes and matches them with sound samples (or patches) stored in the card's memory. The number of these patches in the GM standard is 128. On PC - compatible computers, two MIDI interfaces have historically developed: UART MIDI and MPU-401. The first is implemented in SoundBlaster's cards, the second was used in early Roland models.
SOUND FEATURES OF THE IBM PC FAMILY
Even the very first IBM PC models had a built-in speaker, which, however, was not designed for accurate sound reproduction: it did not reproduce all frequencies of 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, this is more a tribute to tradition than a vital necessity, because the speaker has never played any serious role in human communication with 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 a completely 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 volume and frequency control. However, due to marketing errors, the PCjr model did not become widespread, was declared unpromising, discontinued and its support was discontinued. From that moment on, IBM no longer equipped its computers sound means own development. And from that moment on, sound cards have firmly taken their place on the market.
OVERVIEW OF SOUND CARDS
A kind of "illegitimate son" of the PC and the desire of a person to hear a decent sound with a minimum of financial costs. It is not for nothing that Covox is called the "SoundBlaster for the poor" because its cost is an order of magnitude lower than the cheapest sound card. The essence of Covox "a 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 quite 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 it with 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 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 a Yamaha YM3812 chip. 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).SB-pro is capable of digitizing / Real sound playback up to 44.1 Hz (CD player frequency) in stereo mode This card also supports General MIDI interface with external drivers.
External line in.
SB compatible MIDI,
SB CD-ROM interface.
The SB-pro was fully compatible with the Adlib card, which made it an amazing success in the low cost home sound market (primarily for games). And although professionals were dissatisfied with the unnatural "metal" sound, and the MIDI simulation left much to be desired, but this card appealed to numerous fans of computer games, who encouraged developers to insert support for SundBlaster cards into their games, which finally consolidated 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 the de-facto SB standard. Otherwise, she risks being simply not noticed.
The SoundBlaster 16 (SB 16) is an improved version of the SB-pro that is capable of recording and playing back 16-bit stereo audio. And of course SB16 is fully compatible with Adkib & SB. The SB-16 is capable of playing 8-bit and 16-bit stereo samples up to 44.1 KHz with dynamic sound filtering (this card allows you to suppress unwanted frequency range during playback). The SB16 can also be equipped with a dedicated ASP (Advanced (Digital) Signal Processor) chip that can perform audio compression/decompression on the fly, thus offloading 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 optionally install a special WaveBlaster expansion board, which provides better sound quality in General MIDI mode.
Pro Audio Spectrum Plus and Pro Audio Spectrum 16 The Media Vision "s
The Pro Audio Spectrum Plus and -16 (PAS+ and PAS-16) are one of many attempts to add to the family of SB-like cards. Both cards are almost identical, except that the PAS-16 supports 16-bit sampling. Both cards are able to bring the playback frequency up to 44.1 KHz, dynamically filter the audio stream. Like SB-pro and SB-16, PAS performs FM synthesis via Yamaha YMF262 chip (OPL-3)
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 compliant, this is not entirely true and many people have had bad experiences with this card when trying to use it as SB. However, this is somewhat offset by the excellent stereo sound and very low noise levels.
The Gravis UltraSound
The Advanced Gravis"
Gravis UltraSound (GUS) is the undisputed leader in the field of WS-synthesis. The standard GUS has 256 or 512 kilobytes of memory "on board" for storing samples (also called patches), by playing which the GUS generates all sound effects and music. The GUS can operate at sampling rates up to 44.1 KHz and can produce 16-bit stereo sound. Recording is somewhat more complicated - initially standard GUS models only recorded 8-bit sound, but newer models (GUS MAX) are capable of 16-bit recording as well. In general, the sound played by the GUS is more realistic (due to the use of WS-synthesis, instead of FM), and of course the GUS provides excellent support for General MIDI due to the fact that it does not need to "design" 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).
The 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 the GUS "it. However, in practice, satisfactory operation of SBOS is more accidental than natural. In addition, SBOS significantly slows down the processor , which makes GUS practically unusable for multimedia applications written exclusively for SB. sound qualities GUS "I 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 after SB is the card GUS.
The problems of GUS advancement to the modern gaming market are complicated by the fact that currently 45% of games are written on Miles Design AIL 2.0 - 3.15, 50% on HMI SOS 3.0 - 4.0, the remaining 5% on self-made sound libraries. Only AIL 3.15 learned how to support GUS, and then only almost. Prior to 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 in the standard 512 -and kilobyte GUS memory "I fit 30-50 timbres), AIL 3.15 is a little more humane - timbres are loaded as needed (almost) but not unloaded (!!), thus the situation is reduced to the previous one. I'm silent that the original timbres use rare units of manufacturers and I understand the rest very well - for the sake of one GUS "and it makes no sense to buy timbres and" drag "music. Not to mention the manufacturers' problems with creating music for standard timbres and inventing how to stuff 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. The LAPC-1 uses a combined way 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, so only 8 instruments can play at the same time, 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 the 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) sewn in internal memory ROM. A patch can be up to 24 partials, but most patches are one partials. 15 instruments and one percussion can be played at the same time. Although there is no possibility of changing the internal samples, this is to some extent compensated by the presence of two sound effects: hall and echo. One of the most serious shortcomings of the Roland family of cards is that none of them are equipped with DAC/ADC, and do not contain a CD-ROM controller, which makes it impossible to use them in multimedia systems that comply with the MPC standard.
The sound quality of the LAPC-1 is very high. Some patches (like a piano or flute) are superior in quality to similar GUS "i 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 be recognized as one of the best for creating professional instrumental music, however they are completely unsuitable for operation in multimedia systems.Moreover, Roland cards are not compatible with any modern sound standard.
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 playing up to 44.1 KHz.
Similar to Adlib Gold 1000 but does 16 bit sampling.
Based on the Yamaha YMF3812 FM chip. 11 channels.
8-bit mono sound up to 22 KHz. Compatible with SB standard. Contains a MIDI interface.
Adlib and SB compatible card based on Yamaha YM3812FM chip. 11 channels. 8-bit stereo sound up to 44.1 KHz. Contains a MIDI interface.
Turtle Beach
Based on the 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 up to 44.1 KHz.
TXX SOUND BOARDS: BASIC CONCEPTS
Before proceeding to the next section, which touches upon the practical issues of purchasing a sound card, it is necessary to clarify a number of terms:
Frequency response (FrequencyResponse)
Indicates 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 a decrease in the characteristic by -2dB can be observed due to the presence of a high / low pass filter, however, it is considered that a deviation below -3dB is unacceptable.
Signal to noise ratio (S/N Ratio)
It is the ratio (in dB) of the board's undistorted maximum signal to the level of electronic noise generated by the board's own circuitry. Since humans perceive noise at different frequencies differently, a standard A-weighting grid has been developed that takes into account annoying noise levels. This number is usually what is meant when talking about the S/N Ratio. The higher this ratio, the better the sound system. Reducing this parameter to 75 dB is unacceptable.
Quantization noise
Residual noise, characteristic of digital devices, which occurs due to imperfect conversion of a signal 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 allowed output signal. The lower this level, the higher the sound quality.
Total harmonic distortion + noise Reflects the effect of distortion introduced by the audio 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
Just a number indicating to what extent the left and right channels remain mutually independent. Ideally, the separation of channels should be complete (absolute stereo effect), however, in practice, there is a 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 is allowed to pass. Typically the dynamic range is measured at 1Khz. In an ideal digital audio system, the dynamic range should be close to 98dB.
Intermodulation distortion
Potential Gain
The maximum gain provided by the preamp of the sound card. It is desirable to have high potential gain at low input voltage. Low voltage is considered to be 0.2V, which corresponds to the typical output signal of a household tape recorder.
WHICH BOARD TO CHOOSE?
As can be seen above in this moment Just a huge number of sound systems for personal computers have been thrown into the market. Therefore, choosing a sound card becomes a difficult 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's take the liberty, in conclusion, to give some advice to those who are going 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 desirable that the sound card you choose already carries a CD-ROM controller "a of your chosen design.
1.3 Equipment of the workplace………………………….
1.4 Safety regulations when working with SVT and computer network…………………………………….
2 Performing an individual task…………....
2.2 Description and technical characteristics of the audio system……………………………………………………
2.3 Working Principle of PC Sound System………..
2.4 Setup and configuration steps
PC sound system………………………………………….
2.5 Tools for diagnostics and repair of the Sound system…………………………………………………
2.6 Types of malfunctions of the PC sound system and their elimination………………………………………………………
3 Working with a computer network…………………….
3.1 Description of the location of the network and
available equipment…………………………………….
3.2 Design computer network and equipment selection……………………………………………………..
3.3 Stages of installation and configuration of a computer network……………………………………………………………………………
3.4 Methods and tools for network testing
Bibliography……………………………………..
Appendix A Structure and principle of operation of the PC sound system…………………………………………………
Annex B Analysis of financial costs for
audio system repair…………………..…………………….
Appendix B Project of a computer network in
compass program…………………………………………………
Annex D Analysis of financial costs for
creation of a computer network………………………………..
Appendix E Screenshot of the network diagram and listing of commands for setting up workstations in the CiscoPacket-Tracer program………………………………………………………..
Appendix E Configuration command listing
active network equipment in the CiscoPack-etTracer program……………………………………………………..
Appendix G Schematic screenshot virtual networks and a listing of the VLAN configuration commands in CiscoPack-etTracer. …………………………………………….
Introduction
The production practice takes place according to the module PM 03 "Maintenance and repair computer systems and complexes.
The place of internship is the enterprise "OOOTelCom". The practice takes place in the division of installation and laying of fiber optic networks.
The purpose of the internship is to acquire practical skills in the process of laying a network, maintaining a PC and peripheral devices.
Practice objectives:
study of the structure of the enterprise and the job description according to the rules of the order;
Familiarization with the equipment of the workplace and the safety regulations for working with computer and computer networks;
- fulfillment of an individual task;
development of practical skills in networking and PC maintenance.
The topic of the individual task is PC sound system.
This topic is relevant because this device is used on all PCs, and is used to play sound. Like all other devices, the sound system can fail. At the enterprise "OOO TELCOM" it will be necessary to correct the malfunctions of the sound system, if any.
Practice methods:
monitoring the process of PC sound system operation;
analysis of malfunctions of the PC sound system;
prediction of possible malfunctions;
practical work Troubleshooting the PC sound system;
network design;
experiment on laying fiber optic networks.
1 General
1.1 Enterprise structure
1.2 Job description and the rules of the technician-programmer.
1.2.1 The programming technician must know:
Work programs, instructions, layouts and other guidance materials that determine the sequence and technique for performing settlement operations;
- technology of mechanized and automated processing information;
- design methods for mechanized and automated information processing;
- facilities computer science, collection, transmission and processing of information and rules for their operation;
- types of technical information carriers, rules for their storage and operation;
- operating systems calculus, ciphers and codes;
- methods for carrying out calculations and computational work, as well as calculating the work performed;
- rules and norms of labor protection;
- internal labor regulations;
- the main formalized programming languages;
- basics of programming.
1.2.2 The programmer-technician performs the following duties:
Performing preparatory operations related to the implementation of the computing process, monitoring the operation of machines;
- performance of work on the preparation of technical storage media that provide automatic data entry into a computer, on the accumulation and systematization of indicators of the normative and reference fund, the development of outgoing document forms, the introduction of necessary changes and the timely adjustment of work programs ;
- keeping records of the use of machine time, volumes of work performed;
- fulfillment of individual official assignments of his non-mediocre leader;
-participation in the design of data processing systems and software systems for the machine;
-participation in the performance of various operations of the technological process of information processing (reception and control of input information, preparation of initial data, processing of information, release of outgoing documentation and its transfer to the customer);
- drafting simple circuits the technological process of information processing, the algorithm for solving problems, switching schemes, layouts, work instructions and the necessary explanations for them;
-development of programs for solving simple problems, conducting their debugging and experimental verification of individual stages of work.
1.2.3 The network technician has the right to contact the management of the enterprise:
With the requirements of assistance in the performance of their official duties and rights;
- with proposals for improving the work related to the duties stipulated by this instruction;
- with reports within their competence about all shortcomings in the activities of the center (its structural divisions) identified in the process of exercising their official duties and make proposals for their elimination.
Request personally or on behalf of the immediate supervisor from the heads of departments of the center and specialists information and documents necessary for the performance of their duties.
Involve specialists from all (individual) structural divisions in solving the tasks assigned to it (if it is provided for by the regulations on structural divisions, if not, with the permission of the head of the Computing Center (EC).
1.2.4 Work time and rest time
The normal working hours of workers and employees may not exceed 40 hours per week. As economic and other necessary conditions there will be a transition to a shorter working week.
For workers and employees, a five-day work week with two days off is established. With a five-day working week, the duration of daily work is determined by the rules of the internal labor schedule. At our enterprise, the working day is from 8-00 to 17-00 - for employees and engineers.
Workers and employees are provided with a lunch break for rest and meals lasting at least 1 hour. Breaks are not included in working hours.
On the eve of holidays, the duration of work of workers and employees is reduced by one hour. Overtime work is generally not allowed.
1.3 Software equipment of the workplace
The TelCom LLC company provides each student with his own car during the production practice, behind which a kind of work is performed for each person.
At this enterprise LLC "TelCom" a number of programs are used, for example, such as:
Picture 1 - General form programs of TelCom LLC
(for subscribers of the city of Korkino)
In this program, we see that for each day the operator draws up “open orders” that the workers must complete during the day.
Each box contains general information:
- the time at which it was agreed to arrive at the place of connection;
- full name of the connected subscriber;
- location;
- cell number.
With the help of this information, the installers must complete the connection at the agreed time.
The following program, which is used at the enterprise, is connected to the main server. TELCOM LLC constantly monitors the availability and performance of servers. In case of server errors and failures, HostMonitor warns the administrator (or tries to fix the problem itself). The program uses 60 testing methods, there are a large number of settings. In addition, HostMonitor allows you to create detailed logs in various formats (Text, HTML, DBF and ODBC), has a built-in report editor, a convenient and intuitive interface, etc. IN new version improved performance of HostMonitor, LogAnalyzer, RemoteControlConsole, RMA Manager, WebService and MIB Browser
Figure 2 - General view of the KS-HostMonitor program
In the KS-HostMonitor program, in order to constantly monitor the availability and performance of servers, you need to create a database for each area, and use the IP address to enter access to each switch, which will be referred to as the address of its location (for example, "Tereshkova 12 ”, “Kalinina 14”, etc.).
The following program, connects to the main database by IP address, and contains information about the subscribers who are connected.
Figure 3 - General view of the program "Korkino2"
The program contains all the information about connected subscribers, such as: login, full name, number personal account, personal IP address, balance, etc.
1.4 Safety precautions for working with SVT and computer network
1.4.1 Safety requirements before starting work
Before starting work, you should make sure that the wiring, switches, sockets, with which the equipment is connected to the network, are in good condition, that the computer is grounded, and that it is working. In case of malfunctions, inform the head of the organization.
1.4.2 Safety requirements during work
To reduce or prevent the influence of hazardous and harmful factors, it is necessary to observe sanitary rules and regulations. To avoid damaging the wire insulation and causing short circuits it is not allowed: to hang anything on wires, paint over and whitewash cords and wires, lay wires and cords behind gas and water pipes, for radiators of the heating system, pull out the plug from the socket by the cord, force must be applied to the body of the plug.
To avoid electric shock, it is forbidden to: frequently turn on and off the computer unnecessarily, touch the screen and the back of the computer blocks, work on computer equipment and peripheral equipment with wet hands, work on computer equipment and peripheral equipment , having violations of the integrity of the housing, violations of the insulation of wires, faulty indication of power on, with signs of electrical voltage on the housing, put foreign objects on computer equipment and peripheral equipment.
It is forbidden to clean electrical equipment from dust and dirt under voltage.
It is forbidden to check the operability of electrical equipment in rooms unsuitable for operation with conductive floors, damp, which do not allow accessible metal parts to be grounded.
It is unacceptable to carry out repairs of computer equipment and peripheral equipment under voltage. Repair of electrical equipment is carried out only by specialist technicians in compliance with the necessary technical requirements.
To avoid electric shock, when using electrical appliances, do not simultaneously touch any pipelines, radiators, metal structures connected to the ground.
Take special care when using electricity in damp rooms.
1.4.3 Safety requirements in emergency situations
If a malfunction is detected, immediately turn off the power to the electrical equipment, notify the administration. Continuation of work is possible only after the malfunction has been eliminated.
If a broken wire is found, it is necessary to immediately inform the administration about this, take measures to exclude people from contact with it. Touching the wire is life-threatening.
In all cases of electric shock to a person, a doctor is immediately called. Before the arrival of the doctor, it is necessary, without wasting time, to start providing first aid to the victim.
Artificial respiration to the affected person is carried out until the doctor arrives.
It is forbidden to have flammable substances in the workplace
In the premises it is prohibited:
- kindle a fire;
turn on electrical equipment if the room smells or does not smell of gas;
- smoking;
dry something on heaters;
close ventilation openings in electrical equipment.
Sources of ignition are:
spark when discharging static electricity;
- sparks from electrical equipment;
- sparks from impact and friction;
- open flame.
In the event of a fire hazard or fire, personnel must immediately take the necessary measures to eliminate it, at the same time notify the administration about the fire.
1.4.4 Safety requirements at the end of work
After finishing work, it is necessary to de-energize all computer equipment and peripheral equipment. In the case of a continuous production process, it is necessary to leave only the necessary equipment switched on.
2Performing an individual
tasks
2.1 The concept and components of a PC sound system
The sound system of a PC is structurally a sound card, either installed in a slot on the motherboard, or integrated on the motherboard or expansion card of another PC subsystem. Separate functional modules of the sound system can be implemented as daughter boards installed in the corresponding slots of the sound card.
The sound system of a personal computer is used to reproduce sound effects and speech that accompanies the reproduced video information.
Includes:
recording/playback module;
- synthesizer;
- interface module;
- mixer;
- acoustic system.
Figure 4 - The structure of the PC sound system
The components of the sound system (excluding the speaker system) are designed in the form of a separate sound board or are partially implemented as microcircuits on the computer motherboard.
1. The recording and playback module of the sound system performs analog-to-digital and digital-to-analogue conversions in the mode of program transmission of audio data or their transmission via DMA channels (DirectMemoryAccess - direct memory access channel).
2. The electronic musical digital synthesizer of the sound system allows you to generate almost any sound, including the sound of real musical instruments.
3. The interface module provides data exchange between the sound system and other external and internal devices.
Connecting a PC to a MIDI network is carried out using a special MIDI adapter, which has three MIDI ports: input, output and data through, as well as two connectors for connecting joysticks.
4. The sound card mixer module performs:
- switching (connection / disconnection) of sources and receivers of sound signals, as well as regulation of their level;
mixing (mixing) of several audio signals and adjusting the level of the resulting signal.
The software control of the mixer is carried out either by means of Windows or with the help of the mixer program supplied with the sound card software.
5. The speaker system (AC) directly converts the sound electrical signal into acoustic vibrations and is the last link in the sound reproducing path. mikov.
The number of speakers in the speakers depends on the number of components that make up the audio signal and form separate sound channels.
2.2 Description and specifications of the PC sound system
Figure 5 - Sound card Creative SB 5.1 VX
Sound card specifications:
General characteristics.
Type – internal;
Type of connection - PCI;
Necessity additional food- No;
Possibility to output multi-channel sound – yes;
Sound characteristics.
DAC bit depth – 24 bits;
The maximum frequency of the DAC (stereo) - 96 kHz;
analog inputs.
- Input analog channels – 2;
- Input connectors jack 3.5 mm - 1;
- Microphone inputs - 1;
analog outputs.
Output analog channels - 6;
- Output analog connectors - 3;
Standards support.
Support for EAX - v. 2;
- ASIO support - no.
Figure 6 - Acoustic system
Ritmix SP-2025
Characteristics.
Management - volume control, on / off button. nutrition;
Range of reproducible frequencies - 210 - 20 000 Hz;
Sound power (speakers) - 5 W (RMS);
Emitter diameter - 51 x 102 mm;
Food - network 220 V;
- Outputs - 3.5 mm (for headphones);
- Dimensions - 79 x 86 x 210 mm;
Weight - 673 g
2.3 How the PC sound system works
The principle of operation of the PC sound system is as follows.
1. Sound recording and playback module.
The audio signal can be presented in analog or digital form.
If a microphone is used to record sound, which converts a time-continuous sound signal into a time-continuous electrical signal, an analog sound signal is obtained. Since the amplitude of the sound wave determines the loudness of the sound, and its frequency determines the pitch of the sound tone, in order to preserve reliable information about the sound, the voltage of the electrical signal must be proportional to the sound pressure, and its frequency must correspond to the frequency of sound pressure fluctuations.
In most cases, the audio signal is fed to the input of a 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 acoustic system installed at the output of the PC sound card perceives only analog electrical signals, therefore, after processing the signal with the help of a PC, it is necessary to convert the digital signal into analog.
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. the signal includes both useful and unwanted high-frequency noise, to filter which the received digital data is passed through a digital filter.
Digital-to-analogue conversion generally occurs in two stages. At the first stage, signal samples are separated 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.
2. Synthesis - the computer sends note information to the sound card, and the card converts it into an analog signal (music). There are two synthesis methods:
a) FrequencyModulation (FM) synthesis, in which the sound is reproduced by a special synthesizer that operates with the mathematical representation of the 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 on playing "computer" music, but the attempt 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 really 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, ProAudioSpectrum) also use this technology, only on other more modern types of microcircuits, such as the Yamaha YMF262 (OPL-3) FM.
b) synthesis according to the wave table (Wavetable synthesis), with this method of synthesis, the given sound is "collected" not from the sines of mathematical waves, but from a set of actually sounded instruments - samples. Samples are stored in the RAM or ROM of the sound card. A special sound processor performs operations on the sounds (with the help of various kinds of mathematical transformations, the pitch and timbre are changed, the sound is supplemented with special effects).
Since the 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 more and more popular now. An example of a popular map using WS GravisUltraSound(GUS).
3. MIDI. The computer sends special codes to the MIDI interface, each of which indicates the action that the MIDI device (usually a synthesizer) should perform. (General) MIDI is the main standard for most sound cards. The sound card independently interprets the sent codes and matches them with sound samples (or patches) stored in the card's memory. The number of these patches in the GM standard is 128. On PC - compatible computers, two MIDI interfaces have historically developed: UART MIDI and MPU-401. The first one is implemented in SoundBlasters cards, the second one was used in early Roland models.
4. ISA or PCI interface box
The ISA interface in 1998 was supplanted in sound cards by the PCI interface.
The PCI interface provides a wide bandwidth (for example, version 2.1 - more than 260 Mbps), which allows you to transmit streams of audio data in parallel. The use of the PCI bus allows you to improve the sound quality, providing a signal-to-noise ratio of over 90 dB. In addition, the PCI bus provides the possibility of cooperative processing of audio data, when processing and data transmission tasks are distributed between the sound system and the CPU.
Figure 7 - Device and principle of operation.
2.4 Steps for setting up and configuring the PC Sound System
The sound card can be built into the motherboard or installed separately in a separate slot on the MP. Sound card setup will be done in 2 stages.
1. Software installation.
First of all, you need to install the drivers. Of course, most likely OS Windows already I found and installed the drivers for the sound device myself, however, in order to gain access to all the functionality, as well as for peace of mind, we will install the driver package directly from Realtek. The settings indicated here were checked on the R2.67 driver version. -tiv HD_Audio/Setup.exe), restart the computer. After loading the OS, a brown speaker icon should appear in the system tray.
2. Driver settings
Windows Control Panel->Hardware and Sound->Sounds, making sure that our headphones or speakers are connected to the green sound card jack, disable all unnecessary devices, and make our connected device the default device.
When the sound card setup is completed, you can connect the speaker system.
2.5 Diagnostic and repair tools
PC sound system
The sound system of a PC, like all other components of a computer, fails over time. The following tools are needed to diagnose and repair the PC Sound System:
- electric soldering iron;
Figure 8 - electric soldering iron
A soldering iron is a hand tool used in tinning and soldering to heat parts, flux, melt solder and bring it into the place of contact of the soldered parts. The working part of the soldering iron, usually called the tip, is heated by a flame (for example, from a blowtorch) or an electric current.
Using a soldering iron, you can solder faulty components on the sound card or solder cable wires to the plug.
Insulating tape - designed for electrical insulation of current-carrying parts.
Tape is wrapped around the cable where the soldering took place.
Figure 9 - electrical tape
Screwdrivers - used for dismantling and mounting the sound card and speaker system.
Figure 10 - screwdrivers
BIOS - basic input/output system. You can configure the sound card connection.
wires and plugs - used to replace faulty wires and plugs.
Figure 11 - wires and plugs
- multimeter;
Figure 9 - Multimeter
The multimeter is used to measure control parameters.
2.6 Types of sound system malfunctions and their elimination
1. Sound card malfunctions are a very common phenomenon, this failure occurs very easily, but it is very difficult to fix it, since the cause of the absence of sound can be hidden in the most unexpected places on the computer.
a) When you turn on the system unit, there are no beeps. Causes of failure and how to fix:
check the correct connection of the speakers to the sound card connector and the connection of the speakers themselves to the power supply.
the lack of drivers and hardware incompatibility of programs can lead to a software error or a malfunction of the sound card, here you need to check the software and hardware compatibility of the sound card with the rest of the equipment in the device manager of the system and, if necessary, remove conflicting programs and install the necessary drivers .
- a sound card malfunction may be accompanied by failed elements and parts of the sound card, for example, the sound card output itself or the soldering on the track itself has come off, which need to be soldered.
- a sound card, especially if it is built-in, can simply be disabled in the BIOS, which must be enabled.
- very often the built-in sound card simply burns out, and it is replaced with an external or internal one, when they are connected, it is necessary to disable the built-in sound card in the BIOS, this is necessary so that there is no hardware error in the operation of the system unit.
b) There is a hum and an incomprehensible background from the speakers - the connection plugs are faulty, which need to be soldered or replaced, over time, the capacitance of the capacitors on the sound card and in the pre-amplifier of the signal of the speakers themselves is lost.
c) Incomprehensible intermittent sounds come from the speakers and extraneous noise- in this case, the required audio codecs are missing. Which, you need to replace or update through the necessary software.
2. Malfunctions of acoustic systems.
Acoustic systems, especially inexpensive and from unknown manufacturers, cannot withstand long-term operation at maximum power, since their built-in power supply is designed for a rated load, and such a load is created at a sound volume of about 80% of the maximum. Therefore, it is natural that when the system is operated at maximum volume, the power supply experiences increased loads, and this causes overheating of the circuit elements, and, as a result, their damage.
Quite often, mechanical volume controls become the cause of sound distortion. It is easy to "calculate" such a regulator, it is enough to add or decrease the volume of the sound, wheezing and cods that occur at this time will be evidence that the working part of the regulator is worn out, such a regulator must be replaced with a similar one.
When operating at maximum volume, the winding of the speaker coil may burn out, such a speaker will have to be replaced, speakers with significant damage to the diffuser should also be replaced, if the damage is small and the diffuser is made of paper, you can try to glue it with a piece of drawing paper.
When the speakers are operated at high power, a break occurs in the conductor connecting the external terminal of the loudspeaker with the terminal of its diffuser, in which case everything is repaired by ordinary soldering.
Often the cause of the malfunction is wire breaks near the connection plugs, and the insulation of these wires in most cases remains intact, which makes diagnosis difficult. You can “call out” such damage using a multimeter, if you don’t have one, then you can use a battery and a regular light bulb from a flashlight, for this one contact of the light bulb is connected to the battery directly, and the second contact is connected to the battery through the tested cable, well, then everything is clear - the light bulb caught fire; the whole cable did not light up - it was damaged. In case of damage, it is advisable to replace such a cable, because, as we know, damage to cables most often occurs very close to the connector, although you can try to fix this as well. To do this, you will need to clean the plug from the plastic covering it, solder the cable wires to it again, then carefully wrap it all with electrical tape.
3 Working with a computer network
3.1 Description of the location of the computer network and available equipment
Place of laying the network 2nd floor of the building sushi bar "Samu-rai" on the street. Zwillinga 21.
This floor contains one room, the size of the room: the length of the room is 5.10 meters. The width of the room is 3 meters. The height of the room is 3.1 meters. The area of the room is 20 square meters.
The room has: one window, one door, 4 lamps installed in the false ceiling, two radiators, one chair, wardrobe, sofa, refrigerator, computer desk.
Available equipment:
- switchboardD-LinkDES-1210-28/ME;
- NETLANEC-UU002-5-PVC-GY cable, 2 pairs, Cat.5, internal;
- network sockets for connecting RJ-45 cable;
- cable channel.
3.2 Computer network design and equipment selection
When designing a computer network, the network topology was used - a star, since all computers in the network are connected to a central node (switch), forming a physical network segment.
The type of cable used in the design of the network is a shielded twisted pair of category 5, providing a throughput of 100Mbps, designed for indoor installation. The advantages of this cable include its inexpensive cost, but at the same time fully complying with the standards and availability.
To protect the cables, cable channels were used, and TDM boxes for 6 modules were installed. The advantage of these boxes is:
- ease of installation on the side and rear wall of the case, easily removable cable entries are stamped, and the marking with the installation dimensions on the rear wall will make the installation more accurate;
- a special lock-latch allows you to fix the box door in the open position;
- all screws included in the box have a universal head. It fits both Phillips and flathead screwdrivers.
The switch used in the network design was chosen by D-LinkDES-1210-28/ME. Since this switch has advanced functionality, and is also an inexpensive solution for creating a secure and high-performance network. Distinctive features of this switch are high port density, equipped with 24 FastEthernet ports, as well as 4 GigabitEthernet ports, including 2 1000Base combo ports -T/SFP that support both Gigabit and 100BASE-FX SFP transceivers.
Benefits include: broadcast storm management, which minimizes the possibility of virus attacks on the network, and port mirroring, which simplifies traffic diagnostics and helps administrators monitor and modify switch performance if bridges.
Annex B
3.3 Stages of installation and configuration of a computer network
During installation, the NikoLan NKL 4700B-BK cable was used, which is a high-quality shielded 4-pair cable with a solid core and is designed for external installation. Rigid polyethylene shell is not afraid of ultraviolet, resistant to cold down to minus 60 degrees, and external influences.
When attaching the cable, it is necessary to remove the braid layer with a clerical knife, under which there is a stranded steel cable. Next, using a screwdriver and a hexagonal bolt, we wind a steel cable onto the bolt, which, when twisted, will tighten the cable base, this completes the installation.
Next, you need to compress the shielded twisted pair according to the standard used in the enterprise. It looks like this:
1 - white-orange;
2 - orange;
3 - white;
4 - green;
5 - white-green;
6 - blue;
7 - white-brown;
8 - brown.
Before crimping the cable, it is necessary to prepare it. First, remove the braid, carefully cutting the cable. Then remove the shielded film. And the final step will be to straighten each core so that it looks like a string, and insert it into the RJ-45 connector according to the standard, and crimp it with a crimping tool.
After all the manipulations with the cable, you need to configure it on the computer. In order to configure, you need to set your personal IP address, subnet mask, main gateway, preferred gateway, and an alternative gateway which is different for each subscriber from another subscriber, since each subscriber is given his own personal agreement, which contains all necessary information on the setup.
After the steps have been taken, we measure the speed and ping using the www.speedtest.net website, so that approximately everything corresponds to the declared tariff.
3.4 Methods and tools for testing a computer network
3.4. 1 Using testers
The most objective and simple way to test all the features of a local network is to use various kinds of testers. They allow you to automate and simplify the testing process as much as possible, therefore, if possible, it is advisable to use this particular method.
Exist different variants testers who differ in testing methods, the number of various tests, as well as the way results are issued. The cost of testing equipment directly depends on these functions. There are quite a lot of testing equipment on the market from different manufacturers, the cost of which varies in a wide range: from $50 to $20,000. For obvious reasons, only a serious company providing professional SCS installation services can afford to use expensive equipment. In practice, when testing most of the created local networks with 30-50 computers, the simplest testers are used, which only allow you to check the condition of the cable segment, which is quite enough in 90% of cases.
There are two main types of testers: for testing physical lines and network analyzers.
Testers for testing physical lines are most widely used due to their price. Such a tester is able to determine the failure of the cable segment at the physical level, up to determining the location of the broken conductors. In addition, he can, for example, test the impedance of the line or measure the data rate, which allows you to determine the used network standard or meeting a certain standard. Even a small company can afford to buy such a tester, which will make it possible to quickly identify and fix a malfunction during the operation of a local network.
Network analyzers are expensive equipment that only network integrators can afford. With the help of such a network analyzer, one can not only study the characteristics of the cable structure, but also obtain complete information about the process that occurs during the passage of a signal from any node to any node, with the identification of problem segments and bottlenecks. In addition, you can even predict the state of the network in the near future and ways to solve or prevent future problems.
The appearance of the tester, which allows assessing the physical integrity of a cable segment of any length, is shown in Figure 13.
Figure 13 - cable tester with a set of adapters
A good tester allows you to evaluate the maximum number of cable parameters, for which various adapters and auxiliary tools often come with the tester. For example, using the appropriate adapters, you can test both coaxial segments and twisted-pair cable segments. As far as fiber optic lines are concerned, the equipment for testing them is more complex and often focused only on fiber optic testing.
Testing of the cable segment occurs in different ways, which depend on the availability of access to the cable. One way is as follows: the end of the crimped cable is connected to the connector on the tester, and a special plug is installed on the second end. As a result, the tester can check the resistance of each conductor, as well as their connection to one of the standards. Using resistance data allows you to determine the technical characteristics of the cable, as well as find out the distance to the break point.
3.4.2 Using the software method
When it is not possible to purchase a tester, which often happens when installing an office or "home" network, the integrity and quality of the cable segment can also be checked programmatically, using, for example, the ping system utility.
The principle of operation of this method is extremely simple and boils down to trying to transmit any data through the cable.
For example, to check a segment of a coaxial path, you need to connect two computers to them and install terminators on them. Next, you need to configure the IP addressing of each computer, assigning one, for example, the IP address 192.168.2.1, and the second - 192.168.2.2 with a subnet mask of 255.255.255.0. Then, on the computer with the address 192.168.2.1, you should run command line, in which to enter the following command: ping 192.168.2.2
If this command results in the response "Response from 192.168.2.2: number of bytes=32 time< 1мс TTL=64", значит, кабельный сегмент физически цел.
If, as a result of executing the command, the message "Request timeout exceeded" appears on the screen, this will indicate that the cable has a break or the connectors are crimped incorrectly.
In a similar way, you can test any cable, including the twisted pair cable. In the case of a twisted pair cable, this kind of connection is only possible for the crossover option. If it is necessary to test the performance of a patch cord cable, it must be connected to a central node, for example, a switch, and paired with it, use a known working cable that is connected to a second computer.
Conclusion
In the process of passing the production practice according to PM 03. "Maintenance and repair of computer systems and complexes", the following tasks were performed:
studied the structure of the enterprise LLC "TELCOM" and the main types of its activities;
the work of the PC sound system has been studied, it consists of 4 stages. (Appendix A)
The stages of setting up and configuring the sound system of a PC are considered, which consist in two stages;
lists the tools necessary for the diagnostics and repair of the PC Sound System, these include: a set of screwdrivers, electrical tape, a multimeter, a soldering iron.
studied the types of malfunctions of the PC sound system and their elimination.
Thus, the knowledge gained in practice and the skills formed can be applied in future professional activities.
Bibliography
1. Local standards computer networks Reference book / V. K. Shcherbo, V. M. Kireichev, S. I. Samoylenko; ed. S. I. Samoylenko. - M.: Radio and communication, 2005.
2. Practical data transmission: Modems, networks and protocols / F. Jennings; per. from English. - M.: Mir, 2000.
3. Computer networks: protocols, standards, interfaces / Y. Black; per. from English. - M.: Mir, 1999.
4. Fast Ethernet / L. Quinn, R. Russell. - BHV-Kyiv, 2007.
5. Switching and routing of IP / IPX traffic / M. V. Kulgin, IT. - M.: Computer-press, 2001.
6. Fiber optics in local and corporate communication networks / A. B. Semenov, IT. - M.: Computer-press, 1998.
7. Internet protocols. S. Zolotov. - St. Petersburg: BHV - St. Petersburg, 2002.
8. Personal computers in TCP/IP networks. Craig Hunt; per. from English. - BHV-Kyiv, 2003.
9. Computing systems, networks and telecommunications / Pyatibratov et al. - FIS, 2004.
10. High performance networks. User Encyclopedia / A. Mark Sportak et al.; per. from English. - Kyiv: Dia-Soft, 2006.
Annex A
The device and principle of operation of the PC sound system
Annex B
Analysis of the financial costs of repairing an audio system
Annex B
computer network project
Annex D
Analysis of financial costs for creation
computer network
Annex D
Screenshot of the network diagram and listing of commands for setting up workstations in the CiscoPacketTracer program
Appendix E
Configuration command listing
active network equipment in the Cisco-PacketTracer program
Annex G
Listing commands for configuring active network equipment in CPT
Students, graduate students, young scientists who use the knowledge base in their studies and work will be very grateful to you.
Hosted at http://www.allbest.ru/
Ministry of Education of the PMR
SEI "Tiraspol Technical School of Informatics and Law"
Graduate work
Subject: Investigation of the PC sound system using a diode plate
Tiraspol
Introduction
Chapter 1. Theoretical part. Studying the sound system of a PC using a diode plate
1.1 Analytical review on the topic
1.2 Practical part
1.2.1 Structural diagram of the transceiver for wireless transmission signal
1.2.2 Selection of the element base for building a device for studying the sound system of a PC
1.2.3 The principle of operation of the device for studying the sound system of a PC
1.2.4 Device use
Chapter 2. Labor protection. Security measures for the maintenance of computer equipment
2.1 Industrial sanitation and occupational health
2.2 Requirements for the organization and equipment of the technician's workplace
2.3 Fire safety requirements
Conclusion
List of used literature
Introduction
In the traditional way, sound transmission from a PC sound card to a speaker amplifier is carried out using cables. In the graduation project, wireless transmission of sound over a laser beam over a distance of up to several meters is considered.
This work is relevant, since the sound system significantly expands the capabilities of the PC as a technical means of informatization. The sound system of a PC is structurally a sound card, either installed in a slot on the motherboard, or integrated on the motherboard or expansion card of another PC subsystem.
The purpose of this thesis is to study circuit solutions for devices for researching the operation of a PC sound system, develop a structural and schematic diagram, and make a layout.
To achieve the goals set, the following tasks need to be solved:
consider literature data on the topic of the diploma, conduct research on this topic (develop circuits, design a device, analyze the performance of the device), provide engineering calculations for this device being developed.
The purpose of labor protection is a scientific analysis of working conditions, technological processes, apparatus and equipment from the point of view of the possibility of occurrence of hazardous factors, the release of harmful production substances. On the basis of such an analysis, hazardous areas of production, possible emergency situations are determined and measures are developed to eliminate them or limit the consequences.
The study and solution of problems associated with ensuring healthy and safe conditions in which human labor takes place is one of the most important tasks in the development of new technologies and production systems.
Exploring and revealing possible causes industrial accidents, occupational diseases, accidents, explosions, fires, and the development of measures and requirements aimed at eliminating these causes make it possible to create safe and favorable conditions for human labor. Comfortable and safe working conditions are one of the main factors affecting the productivity and safety of work, human health.
Chapter 1. Theoretical part. Studying the sound system of a PC using a diode plate
1.1 Analytical review on the topic
The PC sound system in the form of a sound card appeared in 1989, significantly expanding the capabilities of the PC as a technical means of informatization.
The PC sound system is a set of software and hardware tools that perform the following functions:
recording audio signals from external sources, such as a microphone or tape recorder, by converting the input analog audio signals into digital ones and then storing them on a hard disk;
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 multiple sources;
simultaneous recording and playback of audio signals (Full Duplex mode);
audio signal processing: editing, combining or splitting signal fragments, filtering, changing its level;
audio signal processing in accordance with surround (three-dimensional - 3D-Sound) sound algorithms;
generating using a synthesizer the sound of musical instruments, as well as human speech and other sounds;
control of the operation of external electronic musical instruments through a special MIDI interface.
The sound system of a PC is structurally a sound card, either installed in a slot on the motherboard, or integrated on the motherboard or an expansion card of another PC subsystem, as well as devices for recording and playing audio information (acoustic system). Separate functional modules of the sound system can be implemented in the form of daughter cards installed in the corresponding slots of the sound card.
The classic sound system, as shown in fig. 1 contains:
sound recording and playback module;
synthesizer module;
interface module;
mixer module;
acoustic system.
Rice. 1 - PC sound system structure
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 made either in the form of a separate microcircuit, or be part of a multifunctional microcircuit. Thus, the Chipset of a sound system can contain both several and one microcircuit.
The designs of the PC sound system are undergoing significant changes; there are motherboards with a Chipset installed on them for sound 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".
RECORD AND PLAYBACK MODULE
The recording and playback module of the sound system performs analog-to-digital and digital-to-analogue conversions in the mode of program transmission of audio data or their transmission via DMA (Direct Memory Access) channels.
Sound, as you know, is a longitudinal wave that propagates freely in air or other medium, so the sound signal is continuously changing in time and space.
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 represented in analog or digital form.
When sound is recorded using a microphone that converts a time-continuous sound signal into a time-continuous electrical signal, an analog sound signal is obtained. Since the amplitude of the sound wave determines the loudness of the sound, and its frequency determines the pitch of the sound tone, in order to preserve reliable information about the sound, the voltage of the electrical signal must be proportional to the sound pressure, and its frequency must correspond to the frequency of the sound pressure oscillations.
In most cases, the audio signal is fed to the input of a 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 a PC sound card perceives only analog electrical signals, therefore, after signal processing with a PC, it is necessary to convert the digital signal into analog.
Analog-to-digital conversion is the conversion of an analog signal into a digital one and consists of the following main steps: sampling, quantization and encoding. The scheme of analog-to-digital conversion of an audio signal is shown in fig. 2.
Rice. 2 - Scheme of analog-to-digital conversion of an audio signal
The pre-analogue audio signal is fed to an analog filter that limits the signal's bandwidth.
Sampling of the signal consists in sampling samples of the analog signal with a given periodicity and is determined by the sampling frequency. Moreover, the sampling frequency must be at least twice the frequency of the highest harmonic (frequency component) of the original audio signal. Since a person is able to hear sounds in the frequency range from 20 Hz to 20 kHz, the maximum sampling rate of the original audio signal must be at least 40 kHz, i.e., samples must be taken 40,000 times per second. As a result, most modern PC sound systems have a maximum audio sampling rate of 44.1 kHz or 48 kHz.
Rice. 3 - Time sampling and analog signal level quantization
Amplitude quantization is a measurement of the instantaneous values of the amplitude of a time-discrete signal and its transformation into a signal discrete in time and amplitude. On fig. 3 shows the analog signal level quantization process, with the instantaneous amplitude values encoded as 3-bit numbers.
Coding consists in 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. If the amplitude values are written using binary numbers and the codeword length is set to N bits, the number of possible codeword values will be 2N. There can be the same number of levels of quantization of the readout amplitude. For example, if the value of the sample amplitude is represented by a 16-bit code word, the maximum number of amplitude gradations (quantization levels) will be 216 = 65 536. For an 8-bit representation, respectively, we get 28 = 256 amplitude gradations.
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 received digital data stream, i.e. the signal includes both useful and unwanted high-frequency interference, for filtering which the received digital data is passed through a digital filter.
Digital-to-analog conversion generally occurs in two steps, as shown in Fig. 4. At the first stage, from the digital data stream, using a digital-to-analog converter (DAC), signal samples are selected, following with a 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.
Rice. 4 - Scheme of digital-to-analogue conversion
A large amount of space is required to record and store an audio signal in digital form. disk space. For example, a 60-second stereo audio signal, digitized with a sampling frequency of 44.1 kHz at 16-bit quantization, requires about 10 MB on the hard drive for storage.
To reduce the amount of digital data required to represent an audio signal with a given quality, compression (compression) is used, which consists in reducing the number of samples and quantization levels or the number of bits per sample.
Such methods of encoding audio data using special encoders can reduce the amount of information flow to almost 20% of the original. The choice of encoding method when recording audio information depends on the set of compression-codec programs (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 conversions, the digital audio recording and playback module contains an ADC, a DAC and a control unit, which are usually integrated into one chip, also called a codec. The main characteristics of this module are: sampling rate; type and capacity of ADC and DAC; a method for encoding audio data; the ability to work in Full Duplex mode.
The sampling rate determines the maximum frequency of the signal being recorded or played back. To record and reproduce human speech, 6 - 8 kHz is sufficient; music with low quality - 20 - 25 kHz; For 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 44.1 kHz or 48 kHz sampling rates.
The bit depth of the ADC and DAC determines the bit depth of the digital signal representation (8, 16 or 18 bits). The vast majority of sound cards are equipped with 16-bit ADCs and DACs. Such sound cards can theoretically be attributed to the Hi-Fi class, which should provide studio-quality sound. Some sound cards are equipped with 20- and even 24-bit ADCs and DACs, which significantly improves the quality of sound recording / playback.
Full Duplex (full duplex) - data transmission mode over the channel, according to which the sound system can simultaneously receive (record) and transmit (play back) audio data. However, not all sound cards fully support this mode, since they do not provide high sound quality with intensive data exchange. Such cards can be used to work with voice data on the Internet, for example, when conducting teleconferencing, when high sound quality is not required.
SYNTH MODULE
The electronic musical digital synthesizer of the sound system allows you to generate almost any sounds, including the sound of real musical instruments. The principle of operation of the synthesizer is illustrated in Fig. 5.
Rice. 5 - The principle of operation of a modern synthesizer: a - phases of an audio signal; b - synthesizer circuit
Synthesizing is the process of recreating the structure of a musical tone (note). The sound signal of any musical instrument has several time phases. On fig. 5a shows the phases of the sound signal that occurs when a piano key is pressed. For each musical instrument, the type of signal will be peculiar, but three phases can be distinguished in it: attack, support and decay. The combination of these phases is called the amplitude envelope, the shape 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 hundreds of milliseconds. In the phase called support, the amplitude of the signal does not change much, and the pitch of the musical tone is formed during the support. The last phase, attenuation, corresponds to a section of a fairly rapid decrease in the signal amplitude.
In modern synthesizers, the 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 that are as close as possible to the characteristics of the simulated musical instrument in the support phase, as shown in Fig. 5 B. Next, the excitation signal is fed to a filter that simulates the frequency response of a real musical instrument. The other input of the filter is fed with the amplitude envelope signal of the same instrument. Further, the set of signals is processed in order to obtain special sound effects, for example, echo (reverberation), choral performance (chorus). Further, digital-to-analogue conversion and signal filtering are performed using a low-pass filter (LPF). The main characteristics of the synthesizer module:
sound synthesis method;
Memory;
the possibility of hardware signal processing to create sound effects;
polyphony - the maximum number of simultaneously reproduced elements of sounds.
The sound synthesis method used in a PC sound system determines not only the sound quality, but also the composition of the system. In practice, synthesizers are installed on sound cards that generate sound using the following methods.
The synthesis method based on frequency modulation (Frequency Modulation Synthesis - FM-synthesis) involves the use of at least two complex-shaped signal generators 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, overtones that determine the timbre of the sound of a particular instrument. The envelope generator controls the amplitude of the resulting signal. The FM generator provides acceptable sound quality, is not expensive, but does not implement sound effects. As such, sound cards using this method are not recommended under the PC99 standard.
Sound synthesis based on the wave table (Wave Table Synthesis - WT-synthesis) 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 a WT generator integrated into the memory chip. WT synthesizer provides high quality sound generation. This synthesis method is implemented in modern sound cards.
The amount of memory on sound cards with WT-synthesizer can be increased by installing additional memory elements (ROM) to store instrument banks.
Sound effects are formed using a special effect processor, which can be either an independent element (microcircuit) or integrated into a WT synthesizer. For the vast majority of cards with WT synthesis, reverb and chorus effects have become standard.
Synthesis of sound based on physical modeling involves the use of mathematical models of sound production of real musical instruments for generation in digital form and for further conversion into an audio signal using a DAC. Sound cards using the physical modeling method have not yet become widespread, since they require a powerful PC to work.
INTERFACE MODULE
The interface module provides data exchange between the sound system and other external and internal devices.
The ISA interface in 1998 was supplanted in sound cards by the PCI interface.
The PCI interface provides a wide bandwidth (for example, version 2.1 - 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 a signal-to-noise ratio of over 90 dB. In addition, the PCI bus enables cooperative audio data processing, where data processing and transmission tasks are shared between the audio system and the CPU.
MIDI (Musical Instrument Digital Interface - digital interface of musical instruments) is regulated by a special standard containing specifications for a hardware interface: types of channels, cables, ports with which MIDI devices are connected one to another, as well as a description of the data exchange procedure - an information exchange protocol between MIDI devices. In particular, using MIDI commands, you can control lighting equipment, 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. There is no limit to the total length of the MIDI chain, but the maximum cable length between two MIDI devices must not exceed 15 meters.
Connecting a PC to a MIDI network is carried out using a special MIDI adapter, which has three MIDI ports: input, output and data 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 performs:
switching (connection / disconnection) of sources and receivers of sound signals, as well as regulation of their level;
mixing (mixing) several audio signals and adjusting the level of the resulting signal.
The main features of the mixer module include:
the 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 sound signals.
Audio sources and receivers are connected to the mixer module via external or internal connectors. External connectors of the sound system are usually located on the rear panel of the system unit case: Joystick/MIDI - for connecting a joystick or MIDI adapter; Mic In - to connect a microphone; Line In - line input for connecting any sources of sound signals; Line Out - line output for connecting any audio signal receivers; Speaker - for connecting headphones (headphones) or a passive speaker system.
The software control of the mixer is carried out either by means of Windows or with the help of the mixer program supplied with the sound card software.
Sound system compatibility with one of the sound card standards means that the sound system will provide high-quality reproduction of audio 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 games for DOS, in which the soundtrack 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 a software package focused primarily on business applications.
ACOUSTIC SYSTEM
The acoustic system (AS) directly converts the sound electrical signal into acoustic vibrations and is the last link in the sound reproducing path.
The composition of the speakers, as a rule, includes several speakers, each of which can have one or more speakers. The number of speakers in the speakers depends on the number of components that make up the audio signal and form separate audio channels.
For example, a stereo signal contains two components - stereo left and right signals, which requires at least two speakers in a stereo speaker system. A Dolby Digital audio signal contains information for six audio channels: two front stereo channels, a center channel (dialog channel), two rear channels, and an ultra-low frequency channel. Therefore, to reproduce a Dolby Digital signal, the speaker system must have six speakers.
As a rule, the principle of operation and the internal structure of sound speakers for domestic use and used in technical means ah informatization in the composition of the PC speaker system practically do not differ.
Basically, a PC speaker consists of two 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 cabinet or loudspeakers.
To reproduce low and ultra-low frequencies with high quality, in addition to two speakers, a third sound unit is used in the speakers - a subwoofer (Subwoofer), installed under the desktop. This 3-way PC speaker consists of two so-called satellite speakers that reproduce mid and high frequencies (approximately 150 Hz to 20 kHz) and a subwoofer that reproduces frequencies below 150 Hz.
A distinctive feature of speakers for PC 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 an active speaker is the ability to connect to the line-out of a sound card. The active speaker is powered either from batteries (accumulators) or from the mains through a special adapter made in the form of a separate external unit or power module installed in the case of one of the speakers.
The output power of PC speakers can vary widely and depends on the specifications of the amplifier and speakers. If the system is intended for scoring computer games, a power of 15 - 20 W per speaker is enough 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 watts per channel. With an increase in the power of the AU, its overall dimensions increase and the cost increases.
Modern models of speaker systems have a jack for headphones, when connected, sound playback through the speakers automatically stops.
The main characteristics of the speakers:
frequency band,
sensitivity,
harmonic coefficient,
power.
The band of reproducible frequencies (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 use of a subwoofer can solve the problem of low frequency reproduction.
The sensitivity of a sound column (Sensitivity) is characterized by the sound pressure that it creates at a distance of 1 m when an electric 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 musical program. The difference between the "quietest" and the "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 the non-linear distortion associated with the appearance of new spectral components in the output signal. The harmonic coefficient is normalized in several frequency ranges. For example, for high-quality Hi-Fi speakers, this coefficient should not exceed: 1.5% in the frequency range of 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 value of the harmonic coefficient, the better the speaker.
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 the sensitivity, and the speaker power mainly determines its reliability.
Often, on the packaging of speakers for a PC, the value of the peak power of the speaker system is indicated, which does not always reflect the real power of the system, since it can exceed the nominal power by 10 times. Due to the significant difference in the physical processes occurring during the tests of the AU, the values of electrical powers may differ by several times. To compare the power of different speakers, you need to know exactly what power the product manufacturer indicates and what test methods it is determined by.
Among the manufacturers of high-quality and expensive speakers are Creative, Yamaha, Sony, Aiwa. Lower class ACs are produced by Genius, Altec, JAZZ Hipster.
Some models of Microsoft speakers do not connect to a sound card, but to a USB port. In this case, the sound enters the speakers in digital form, and its decoding is performed by a small Chipset installed in the speakers.
AUDIO COMPRESSION METHODS
The simplest way to digitally represent signals is called pulse-code modulation (PCM) or PCM (Pulse-Code Modulation). A PCM data stream is a sequence of instantaneous values or samples in binary code. If the converters used have linear characteristic(the instantaneous value of the signal voltage is proportional to the code), then this modulation is called linear (Linear PCM). In the case of PCM, the encoder and decoder do not perform information conversion, but only pack / unpack bits into bytes and data words. The bit rate is defined as the product of the sampling rate (sample rate) by the bit depth and the number of channels. An audio CD gives a stream of 44,100 x16x2 = 1411,200 bps (stereo).
For real audio signals, linear PCM coding is uneconomical. The data stream can be reduced by using a simple compression algorithm used in the Delta PCM (DPCM) system, also known as DPCM (Differential Pulse-Code Modulation). Simplified, this algorithm looks like this: not the instantaneous samples themselves are transmitted in the digital stream, but the scaled difference between the real sample and its value, constructed by the codec according to the data stream previously generated by it. The difference is transmitted with fewer bits than the readings themselves. In ADPCM (Adaptive | DPCM, or ADPCM - Adaptive Differential Pulse-Code Modulation), the scale of the difference is determined by history - if the difference monotonously increases, the scale increases, and vice versa.
Of course, the reconstructed signal with this representation will differ more from the original than with conventional PCM, but a significant reduction in the digital data stream can be achieved. ADPCM has become widely used in the digital storage and transmission of audio information (for example, in voice modems). The ADPCM algorithm from the point of view of the PC processor can be implemented both in software and in hardware using a sound card (modem).
More complex algorithms and a high compression ratio are used in MPEG audio codecs. In an MPEG-1 encoder, the input stream is 16-bit samples at 48 kHz (professional audio), 44.1 kHz (consumer), or 32 kHz (telecom).
The standard defines three "layers" (layer) of compression - Layer I, Layer 2 and Layer 3, working one on top of the other.
The initial compression is carried out on the basis of the psychophysical properties of sound perception. Here, the property of sound masking is played out: if the signal contains two tones with similar frequencies that differ significantly in level, then a more powerful signal will mask the weak one (it will not be heard). The masking thresholds depend on the remoteness of the frequencies.
MPEG full range audio frequencies is divided into 32 subbands (sub-band), in each subband the most powerful spectral components are determined and masking frequency thresholds are calculated for them. The masking effects from several powerful components are cumulative. The masking effect extends not only to signals that are present simultaneously with a strong one, but also to those that precede it for 2-5 ms (premasking) and subsequent ones for up to 100 ms (postmasking). Masked area signals are processed with lower resolution because they have lower signal-to-noise ratio requirements. Due to this "coarseness" compression occurs. Compression on a psychophysical basis is performed by Layer 1.
The next stage (Layer 2) improves the accuracy of the presentation and packs information more efficiently. Here, the encoder has a “window” of 23 ms (1152 samples) in operation.
The last stage (Layer 3) uses complex filter banks and non-linear quantization. The highest compression ratio is provided by Layer 3, for which a compression ratio of 11:1 is achieved with high decoding fidelity.
METHODS FOR PROCESSING AUDIO INFORMATION
Digital storage makes it easy to implement many effects that previously required bulky electromechanical or electroacoustic devices or complex analog electronics.
It is known that in a closed room (for example, a hall), not only direct sound reaches the listener from the source, but also reflected (multiple times) from various surfaces (walls, columns, etc.). Reflected signals arrive relative to the direct signal with different delays and attenuation. This phenomenon is called reverberation. And This phenomenon in digital signal processing can be controlled. Digital storage makes it easy to implement many effects that previously required bulky electromechanical or electroacoustic devices or complex analog electronics.
First of all, it is artificial reverb and echo.
It is known that in a closed room (for example, a hall), not only direct sound reaches the listener from the source, but also reflected (multiple times) from various surfaces (walls, columns, etc.). Reflected signals arrive relative to the direct signal with different delays and attenuation. This phenomenon is called reverberation. And This phenomenon in digital signal processing can be controlled.
More complex effects can be made based on sample bias. In the digital form of representation, the Doppler effect is easily imitated - a change in frequency when the sound source quickly approaches the listener or the source moves away from the listener. Everyone has experienced this effect - the monophonic whistle of an approaching train sounds higher, and the departing train sounds lower than the real tone. In digital playback, accumulation of sample lag will result in a lower tone, while reducing the lag will result in a higher tone.
In addition to tricks with delays, it is possible to use digital filtering- from the implementation of the simplest timbral blocks and equalizers to "cutting out" the voice from the song ("karaoke" effect). Everything is determined by the software and computing resources of the processor.
DIRECTIONS FOR IMPROVING THE SOUND SYSTEM
Currently by Intel, Compaq and Microsoft proposed a new PC sound system architecture. According to this architecture, audio signal processing modules are moved outside the PC case, where they are affected by electrical noise, and placed, for example, in speakers of a speaker system. In this case, audio signals are transmitted in digital form, which significantly increases their noise immunity and the quality of sound reproduction. For the transfer of digital data in digital form, the use of high-speed USB and IEEE 1394 buses is provided.
Another direction for improving the sound system is the creation of surround (spatial) sound, called three-dimensional, or 3D-Sound (Three Dimentional Sound). To obtain surround sound, special signal phase processing is performed: the phases of the output signals of the left and right channels are shifted relative to the original. This uses the ability of the human brain to determine the position of the sound source by analyzing the ratio of the amplitudes and phases of the sound signal perceived by each ear. The user of a sound system equipped with a special 3D sound processing module experiences the effect of "moving" the sound source.
A new direction in the application of multimedia technologies is the creation of a PC-based home theater (PC-Theater), i.e. version of a multimedia PC designed for multiple users simultaneously to watch a game, watch an educational program or a movie in the DVD standard. PC-Theater includes a special multi-channel speaker system that generates surround sound (Surround Sound). Surround Sound systems create various sound effects in the room, whereby the user feels that he is in the center of the sound field, and the sound sources are around him. Multi-channel Surround Sound systems are being used in movie theaters and are already beginning to appear in the form of consumer devices.
In multi-channel consumer systems, sound is recorded on two tracks of laser videodiscs or videocassettes using Dolby Surround technology developed by Dolby Laboratories. The most famous developments in this direction include:
Dolby (Surround) Pro Logic is a four-channel sound system containing left and right stereo channels, a center channel for dialogue, and a rear channel for effects.
Dolby Surround Digital is a sound system consisting of 5 + 1 channels: left, right, center, left and right rear effects channels and an ultra-low frequency channel. Signals for the system are recorded in the form of a digital optical soundtrack on film.
In selected models speakers in addition to the standard treble/bass, volume and balance controls, there are buttons for turning on special effects such as 3D sound, Dolby Surround, etc.
1.2 Practical part
1.2.1 Structural diagram of a transceiver for wireless signal transmission
With the rise in popularity wireless technologies the scope of their application is also expanding. IN thesis a solution based on the principle of media data transmission over wireless channels and designed to combine PCs and components of consumer audio equipment into a single multimedia complex is considered.
From time to time, users of personal computers need to connect this device to stationary audio equipment, such as a music center. Of course the most simple option V this case is a cable connection. However, the vast majority of stationary audio components have connectors for connecting signal sources on the rear panel, which is usually not so easy to get to. Second, more serious problem- the absence of many inexpensive radio tape recorders and music centers of inputs for connecting external signal sources.
One of the most universal ways solutions similar problems is the use of low-power radio transmitters that broadcast an audio signal in the VHF band (the ability to receive programs at these frequencies is implemented in almost all modern models of radio tape recorders and music centers). It is also worth noting that a signal broadcast in this way can be received by several nearby radio receivers at once.
In the case of interaction between a digital player and analog equipment (radio recorders, stereos, etc.), analog sound transmission is the only possible option. If we consider the interaction of two digital devices (for example, a computer and a media center), then in this case it is preferable to use the transmission of audio data over a wireless channel in digital form.
The traditional way of transmitting sound from your PC's sound card to your speaker amplifier is through cables. The diploma project considered wireless transmission of sound over a laser beam, over a distance of several meters.
On fig. 6 shows a block diagram of the audio signal receiver:
Rice. 6 - Structural diagram of the audio signal receiver
On fig. 7 shows a block diagram of the audio signal transmitter:
Rice. 7 - Structural diagram of the audio signal transmitter
The primary winding must be directly connected to the audio signal output. The minus of the battery is connected to one of the ends of the secondary winding, the plus of the battery is connected directly to the plus of the laser diode.
We connect the second end of the secondary winding through a 15-47 Ohm resistor to the minus of the laser diode.
1.2.2 Selection of the element base for building a device for studying the sound system of a PC
To assemble a device for wireless signal transmission, you need the following equipment: audio signal source ( Personal Computer, music center or mobile phone), network transformer, power 10-15 W, resistor from 5 to 20 Ohm and battery.
You can use any network transformer, with a power of not more than 20 W, containing a secondary winding of 6 or 12 V., or wind it yourself (primary winding - 15 turns of wire 0.8 mm., Secondary winding - 10 turns of wire 0.8 mm.).
For the receiving device of the sound signal, you will need a photodiode and a low-frequency amplifier.
The LED is used normally. It can be replaced by a laser (significantly increases the transmission distance), which will need to be connected through a 5 ohm, 0.5 W resistor. Also, the source of the light beam can be supplemented with optics from DVD drive, thereby concentrating the beam of light and increasing the transmission distance. The battery used is Li - Ion (lithium - ion) from a mobile phone. Instead, you can use a stabilized power supply for 3.5 - 4 V., with a current strength of not more than 1 A. Solar module parameters: maximum voltage 14 V., with a maximum current of 100 mA. The module can be replaced by any other photodetector.
1.2.3 The principle of operation of the device for studying the sound system of a PC
From a low-power sound source (personal computer, mobile phone), an audio signal is applied to the primary winding of the transformer, leaves the secondary winding, is amplified by the battery and fed to the LED / laser diode. The photodiode, which serves as an audio signal receiver, is directly connected to the input of the power amplifier. Next, turn on the music and direct the beam to the photodetector. The light beam receives the solar module, which is connected to the amplifier, and the power amplifier amplifies weak signal and the result is a fairly high-quality sound. Instead of a laser, you can also use an ordinary LED, but in this case, the transmission range of the sound signal will be no more than 30 centimeters, it is advisable to use white or ultraviolet LEDs from lighters. When using a laser pointer, it is possible to transmit a sound signal up to 15 meters away, and notice the sound quality is quite good. The transmitted sound is quite powerful at a distance of 7 meters, the amplifier at full volume delivered 80 percent of its power to the load.
The quality of the transmitted signal is quite good, there is no sound distortion.
1.2.4 Device use
Such a device has found wide application in science and technology, laser microphones for espionage are based on just such a transmitter and receiver.
Such a device is an excellent accessory for a computer, for example, music is playing on the computer, and the power amplifier is not connected to the computer with a cable, so you can also transmit a conversation, you just need to apply a signal from a microphone (with a preamplifier) to the input of the device and the result is wireless phone or a walkie-talkie, or an excellent bug for short distances.
Chapter 2. Labor protection. Security measures for the maintenance of computer equipment
2.1 Industrial sanitation and occupational health
recording mixer signal transmission
In accordance with GOST 12.0.002 SSBT "Terms and definitions", industrial sanitation is a system of organizational, sanitary and hygienic measures, technical means and methods that prevent or reduce the impact on workers of harmful production factors to values that do not exceed permissible values.
The complex of issues addressed in the framework of industrial sanitation and occupational health includes:
Ensuring sanitary and hygienic requirements for the air of the working area;
Ensuring microclimate parameters at workplaces;
Providing normative natural and artificial illumination;
Noise and vibration protection in workplaces;
Protection against ionizing radiation and electromagnetic fields;
Provision of special nutrition, protective pastes and ointments, overalls and special equipment. shoes, personal protective equipment (gas masks, respirators, etc.);
Provision of sanitary facilities in accordance with the norms, etc.
Occupational hygiene or professional hygiene is a section of hygiene that studies the impact of the labor process and the surrounding production environment on the body of workers in order to develop sanitary and hygienic and therapeutic and preventive standards and measures aimed at creating more favorable working conditions, ensuring health and a high level of human ability to work.
In industrial production, a person is often affected by low and high air temperatures, strong thermal radiation, dust, harmful chemicals, noise, vibration, electromagnetic waves, as well as a wide variety of combinations of these factors that can lead to various health disorders. , to a decrease in performance. To prevent and eliminate these adverse effects and their consequences, the study of the characteristics of production processes, equipment and processed materials (raw materials, auxiliary, intermediate, by-products, production waste) is carried out in terms of their impact on the body of workers; sanitary working conditions (meteorological factors, air pollution with dust and gases, noise, vibration, ultrasound, etc.); the nature and organization of labor processes, changes in physiological functions in the process of work.
Industrial sanitation - a system of organizational, preventive and sanitary-hygienic measures and means aimed at preventing exposure of workers to harmful production factors.
Labor activity can be performed outdoors and indoors.
Industrial premises - closed spaces in any buildings and structures, where during the working hours, labor activity of people is constantly or periodically carried out in various types of production. A person can work in different rooms of one or more buildings and structures. Under such working conditions, it is necessary to talk about the workplace or work area.
The production environment of the workroom is determined by a complex of factors. The presence of these factors (hazards) in the working environment can affect not only the state of the body, but also productivity, quality, labor safety, lead to a decrease in efficiency, cause functional changes in the body and occupational diseases.
In modern conditions of labor automation, a complex of weakly expressed factors acts on the body, the study of the effect of interaction is extremely difficult, therefore, industrial sanitation and labor hygiene solve the following tasks:
taking into account the influence of working environment factors on health and performance;
improvement of methods for assessing working capacity and health status;
development of organizational, technological, engineering, socio-economic measures to rationalize the production environment;
development of preventive and health-improving measures;
improve teaching methods.
The temperature and humidity of the air in the room are the most important parameters that determine the state of comfort inside the room.
The recommended values of indoor air temperature according to various standards are within 20-22Сo and 22-26Сo. Another physical parameter of the internal atmosphere that directly affects the heat exchange of the human body is air humidity, which characterizes its saturation with water vapor. So the lack of humidity, less than 20% relative humidity, leads to drying of the mucous membranes, causes coughing. And exceeding the level of humidity, more than 65%, leads to a deterioration in heat transfer during the evaporation of sweat, there is a feeling of suffocation. Therefore, the temperature must be related to the level of humidity.
The air speed is determined in the working area of the room, i.e. where people are, namely in a space from 0.15 m. from the floor to 1.8 m in height and at a distance of at least 0.15 m from the walls. The air speed in the working area is recommended within 0.13-0.25m/s. At a lower speed - stuffy or even hot, at a higher speed - just a draft, which makes sense only when the temperature rises to the standard values.
Analysis of working conditions
Evaluation of working conditions is carried out according to a special methodology, based on an analysis of the levels of harmful and dangerous factors at a given workplace.
To carry out certification of the workplace, it is also necessary to comprehensively assess working conditions.
Determination of the class of working conditions in the workplace is carried out in order to:
establishing the priority of recreational activities;
creation of a data bank on existing working conditions;
determination of payments and compensations for harmful working conditions.
A harmful production factor is a factor of the environment and the labor process, which can cause a decrease in working capacity, pathology (occupational disease), and lead to a violation of the health of offspring.
Harmful can be:
physical factors: temperature, humidity and air mobility, non-ionizing and ionizing radiation, noise, vibration, insufficient illumination;
chemical factors: gas contamination and dustiness of the air;
biological factors: pathogens;
labor severity factors: physical static and dynamic load; a large number of stereotypical working movements, a large number of body tilts, an uncomfortable working posture;
factors of labor intensity: intellectual, sensory, emotional stress, monotony and duration of work.
A hazardous production factor is a factor of the environment and the labor process that can cause a sharp deterioration in health, injury, or death.
These are: electric current, fire, heated surface, moving parts of equipment, overpressure, sharp edges of objects, height, etc.).
The choice of methods for designing a device for processing and transmitting information. Development of an algorithm of operations for information processing, block diagram of the device. Timing diagram of control signals. Element base for the development of a circuit diagram.
term paper, added 08/16/2012
Devices for recording and reproducing information are an integral part of a computer. The process of restoring information on changes in the characteristics of the medium. Detonation factor. Requirements for the accuracy of manufacturing parts of the transportation mechanism.
abstract, added 11/13/2010
The concept of sound explication. Features of the recording technology used. Layouts of filming equipment on film sets. Rationale for the choice of equipment. Structural diagram of the equipment connection, taking into account the selected synchronization.
term paper, added 12/27/2011
Principles of construction of the radio system "Sagittarius". Wireless data transmission module using ZigBee technology, advantages and disadvantages of its application, principle of operation and assessment of capabilities. Description of the structural and principal electrical circuit of the device.
thesis, added 04/24/2015
The development of information carriers. Sound recording and the process of recording sound information for the purpose of its storage and subsequent reproduction. Musical mechanical instruments. The first two-track tape recorder. Sound and basic recording standards.
abstract, added 05/25/2015
Methods for creating a transmitter for a radio altimeter transceiver module. Feasibility study of the work. Ensuring the safety of personnel working on the project. Classification of production according to fire and explosion hazards.
thesis, added 07/15/2010
Main technical characteristics of the automated receiving-transmitting center. General information and the principle of operation of the device. Automatic 100% redundancy of radio facilities. Methods for outputting transceivers into radiation, device control.
practice report, added 02/12/2016
Algorithms for digital data processing. Scheme of the device of the light and music installation using the example of the ATmega8 microcontroller. Submission, reception and processing of an audio signal. Development of galvanic isolation. A copy of the signal that is applied to the high voltage part.
term paper, added 12/02/2014
Structural diagram of the device for transmitting data and commands. The principle of operation of the temperature sensor. Converting signals from four channels. Data transfer device model. Building code with doubling. Formation of code combinations.
term paper, added 01/28/2015
Scheme for encoding audio information. Analogue and discrete forms of information presentation. Isolation of the number of volume levels in the process of encoding audio information. The quality of the binary audio encoding. Calculation of the information volume.
