What's included in a pc sound system. PC sound system device. Audio information processing and playback systems
1.3 Equipment of the workplace ………………………….
1.4 Safety regulations when working with SVT and computer network ………………………………….
2 Fulfillment of an individual assignment ………… ....
2.2 Description and technical characteristics of the audio system …………………………………………………
2.3 Operating principle of the PC Sound System ……… ..
2.4 Steps for setting up and configuring
sound system PC………………………………………….
2.5 Tools for diagnostics and repair of the Sound system ………………………………………………
2.6 Types of PC sound system malfunctions 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 the choice of equipment …………………………………………………… ..
3.3 Stages of installation and configuration of a computer network …………………………………………………………………………
3.4 Methods and tools for network testing
Bibliography……………………………………..
Appendix A The device and the principle of operation of the PC sound system ………………………………………………
Appendix B Analysis of financial costs for
repair of the audio system ………………… .. …………………….
Appendix B Computer network project in
compass program ………………………………………………
Appendix 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 Listing of Configuration Commands
active network equipment in the CiscoPack-etTracer program …………………………………………………… ..
Appendix G Diagram screenshot virtual networks and listing of VLAN configuration commands in the CiscoPack-etTracer program. …………………………………………….
Introduction
The industrial practice takes place according to the PM 03 module "Maintenance and repair computer systems and complexes ".
The place of the internship is the LLC TelKom enterprise. 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, servicing PCs and peripheral devices.
Practice objectives:
study of the structure of the enterprise and the official instructions according to the rules of the order;
familiarization with the equipment of the workplace and safety rules when working with SVT and a computer network;
fulfillment of an individual task;
development of practical skills for laying networks and maintaining PCs.
The topic of the individual task is the PC sound system.
This topic is relevant because this device is used on all PCs, and serves to reproduce sound. Like all other devices, the sound system can fail. At the TELCOM LLC enterprise, it will be necessary to correct any faults in the sound system, if any.
Practice methods:
monitoring the process of work of the PC sound system;
analysis of malfunctions of the PC sound system;
forecasting 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 should know:
Work programs, instructions, layouts and other guidance materials that determine the sequence and technique for performing computational operations;
- technology of mechanized and automated information processing;
- methods of designing mechanized and automated information processing;
- facilities computing technology, collection, transmission and processing of information and the rules for their operation;
- types of technical data carriers, rules for their storage and operation;
- operating systems of numbers, ciphers and codes;
- methods of calculations and computational work, as well as the calculation of the work performed;
- rules and norms of labor protection;
- internal labor regulations;
- the main formalized programming languages;
- basics of programming.
1.2.2 The software technician fulfills the following duties:
Performing preparatory operations related to the implementation of the computing process, monitoring the operation of machines;
-performing work on the preparation of technical information carriers that provide automatic data entry into a computer, on the accumulation and systematization of indicators of the normative and reference fund, development of outgoing document forms, making the necessary changes and timely correction of work programs ;
- keeping records of the use of computer time, the volume of work performed;
-performing individual service assignments of his direct leader;
-participation in the design of data processing systems and computer software systems;
-participation in the implementation various operations technological process of information processing (reception and control of input information, preparation of initial data, information processing, release of outgoing documentation and its transfer to the customer);
- drawing up simple diagrams of the technological process of information processing, an algorithm for solving problems, switching diagrams, layouts, work instructions and the necessary explanations to them;
-development of programs for solving simple problems, carrying out 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 for assistance in the performance of their duties and rights;
- with suggestions for improving the work associated with the responsibilities provided for by this instruction;
- with messages within their competence about all the shortcomings in the activities of the center (its structural divisions) revealed in the process of fulfilling their official duties and make proposals for their elimination.
To request, personally or on behalf of the immediate supervisor, from the heads of the center's divisions and specialists, information and documents necessary for the performance of their official duties.
To involve specialists of all (individual) structural divisions in solving the tasks assigned to him (if this is provided for by the provisions on structural divisions, if not - with the permission of the head of the computer center (ITC).
1.2.4 Working and rest hours
Normal working hours of workers and employees cannot 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 working week with two days off is established. With a five-day working week, the duration of daily work is determined by the internal work regulations. 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. The break is not included in working hours.
On the eve of holidays, the duration of the work of workers, employees is reduced by one hour. Overtime work is generally not allowed.
1.3 Workstation software
The company "TelKom" LLC provides each student in the production practice with its own car, behind which a kind of work is performed for each person.
A number of programs are used at this enterprise, TelCom LLC, for example, such as:
Figure 1 - General view of the program of LLC TelCom
(for subscribers of the city of Korkino)
In this program, we see that for every day the operator composes "open orders", which the workers must complete during the day.
Each window contains general information:
- the time at which it was agreed to arrive at the connection point;
- full name of the connected subscriber;
- place of residence;
- cell number.
With this information, the installers must complete the connection at the agreed time.
The next program, which is used at the enterprise, is connected with the main server of TELCOM LLC, which continuously monitors the availability and performance of servers. In case of errors and server malfunctions, HostMonitor warns the administrator (or tries to fix the problem on its own). The program uses 60 testing methods, there is a large number of settings. In addition, HostMonitor allows you to create detailed logs in various formats (Text, HTML, DBF and ODBC), there is a built-in report editor, a convenient and intuitive interface, etc. V 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 continuously monitor the availability and performance of servers, it is necessary to create a base for each region, 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 next 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 completely all information about connected subscribers, such as: login, full name, personal account number, personal IP address, balance, etc.
1.4 Safety regulations when working with SVT and computer network
1.4.1 Safety requirements before starting work
Before starting work, you should make sure that the electrical wiring, switches, sockets, with which the equipment is connected to the network, are in good working order, that the computer is grounded, and that it is working properly. 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 norms. To avoid damage to the insulation of the wires and the occurrence of short circuits, it is not allowed to: hang anything on the wires, paint and whitewash the cords and wires, lay wires and cords behind gas and water pipes, for the heating system batteries, pull the plug out of the socket by the cord, force must be attached to the body of the plug.
To exclude defeat electric shock it is forbidden to: often turn on and off the computer unnecessarily, touch the screen and the back of the computer blocks, work on computer technology and peripheral equipment with wet hands, work on computer technology and peripheral equipment that have compromised the integrity of the case , wire insulation faults, faulty power-on indication, with signs of electrical voltage on the case, put foreign objects on computer equipment and peripheral equipment.
Do not clean electrical equipment from dust and dirt while energized.
It is forbidden to check the performance of electrical equipment in rooms unsuitable for operation with conductive floors, damp, not allowing accessible metal parts to be earthed.
It is unacceptable to carry out repairs of computer technology and peripheral equipment under voltage. Repair of electrical equipment is carried out only by specialist technicians in compliance with the necessary technical requirements.
In order to avoid electric shock, when using electrical appliances, you must not touch any pipelines, radiators, metal structures connected to the ground at the same time.
Take special care when using electricity in damp areas.
1.4.3 Safety requirements in emergency situations
If a malfunction is detected, immediately de-energize the electrical equipment, notify the administration. Continuation of work is possible only after elimination of the malfunction.
If a broken wire is found, it is necessary to immediately inform the administration about this, to take measures to exclude people from contact with it. Touching the wire is life-threatening.
In all cases of electric shock to a person, immediately call a doctor. Before the arrival of the doctor, it is necessary, without wasting time, to begin providing first aid to the suffering person.
Artificial respiration for an electric shock is performed until the arrival of a doctor.
It is forbidden to have flammable substances in the workplace.
It is prohibited in the premises:
light a fire;
turn on electrical equipment if the room smells like no gas;
smoking;
dry anything on heating devices;
cover the ventilation openings in the electrical equipment.
Sources of ignition are:
spark during discharge of 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, and at the same time notify the administration of the fire.
1.4.4 Safety requirements at the end of work
After completing the work, it is necessary to de-energize all computer facilities and peripheral equipment. In the case of a continuous production process, it is necessary to leave only the necessary equipment on.
2Performing an individual
tasks
2.1 Concept and components of the PC sound system
The sound system of a PC is constructively sound cards, either installed in the motherboard slot, or integrated on motherboard or an expansion card for another PC subsystem. Separate functional modules of the sound system can be performed in the form of daughter cards installed in the corresponding connectors sound card.
Sound system personal computer serves to reproduce sound effects and speech accompanying the reproduced video information.
Includes:
recording / playback module;
synthesizer;
interface module;
mixer;
speaker system.
Figure 4 - The structure of the PC sound system
The components of the sound system (excluding the speaker system) are structurally designed as a separate sound card or are partially implemented as microcircuits on the computer's motherboard.
1. The module for recording and reproducing the sound system performs analog-to-digital and digital-to-analog conversions in the mode of software transmission of audio data or their transmission via DMA channels (DirectMemoryAccess - direct memory access channel).
2. Electromusical 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 pass-through, as well as two connectors for Joysticks.
4. The sound card mixer module performs:
switching (connecting / disconnecting) sources and receivers of sound signals, as well as regulating their level;
mixing (mixing) of several sound signals and adjusting the level of the resulting signal.
Mixer software control is carried out either Windows tools, or using 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 of the sound reproducing path. As a rule, the speaker system includes several sound speakers, each of which can have one or more dynamic miks.
The number of speakers in the speaker depends on the number of components that make up the sound signal and form separate sound channels.
2.2 Description and technical characteristics of the PC sound system
Figure 5 - Sound card Creative SB 5.1 VX
Sound card specifications:
General characteristics.
Type - internal;
Connection type - PCI;
Necessity additional food- No;
Ability to output multichannel sound - yes;
Sound characteristics.
DAC capacity - 24 bits;
Maximum DAC frequency (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
Specifications.
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;
Power supply - 220 V network;
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 consists in the following stages.
1. Module for recording and reproducing sound.
The audio signal can be presented in analog or digital form.
If, when recording sound, a microphone is used that converts a time-continuous sound signal into a time-continuous electrical signal, an analogue 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 maintain 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.
In most cases, a sound signal is sent 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 using a PC, it is necessary to convert the digital signal back to 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 received stream of digital data, 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.
In general, digital-to-analog conversion takes place in two stages. At the first stage, the signal samples are extracted from the digital data stream using a digital-to-analog converter (DAC), following with 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 spectrum of the discrete signal.
2. Synthesis - the computer sends musical information to the sound card, and the card converts it into an analog signal (music). There are two ways to synthesize:
a) FrequencyModulation (FM) synthesis, in which the sound is reproduced by a special synthesizer, which operates with a mathematical representation of a sound wave (frequency, amplitude, etc) and from the totality of such artificial sounds, almost any necessary sound is created.
Most systems equipped with FM synthesis show very good results in playing "computer" music, but the attempt to simulate the sound of live instruments is not very good. The disadvantage of FM synthesis is that it is very difficult (almost impossible) to create truly realistic instrumental music with a large presence of high tones (flute, guitar, etc.). The first sound card to use this technology was the legendary Adlib, which used a chip from the Yamaha YM3812FM synthesis 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 (Wavetablesynthesis), with this method of synthesis a given sound is "typed" not from the sines of math waves, but from a set of really sounded instruments - samples. Samples are saved in RAM or ROM of the sound card. A special sound processor performs operations on the samals (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 digitization of real instruments, they make the sound extremely realistic. Until recently, a similar technique was used only in hi-end instruments, but it is becoming more and more popular now. An example of a popular card 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 basic standard of most sound cards. The sound card independently interprets the codes being sent and matches them with the sound samples (or patches) stored in the memory of the card. The number of these patches in the GM standard is 128. On PC - compatible computers, historically, two MIDI interfaces have developed: UART MIDI and MPU-401. The first is ralized in SoundBlasters cards, the second was used in early Roland models.
4. ISA or PCI interface block
The ISA interface was supplanted in sound cards by the PCI interface in 1998.
The PCI interface provides a wide bandwidth (for example, version 2.1 - more than 260 Mbps), which allows you to transfer audio data streams in parallel. The use of the PCI bus allows you to improve the sound quality, providing a signal-to-noise ratio of more than 90 dB. In addition, the PCI bus allows for cooperative processing of audio data, when processing and data transfer 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. The sound card will be configured in 2 stages.
1. Software installation.
First of all, you need to install the drivers. Of course, most likely Windows has already found and installed the drivers for the sound device itself, however, 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 specified here were checked on the R2.67 driver version. , we perform a simple installation procedure (by running HD_Audio / Setup.exe), restart the computer. After loading the OS, a brown speaker icon should appear in the system tray.
2. Driver configuration
Windows Control Panel-> Hardware and Sound-> Sounds, after making sure that our headphones or speakers are connected to the green jack of the sound card, turn off 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, break down over time. To diagnose and repair the PC Sound System, the following tools are required:
electric soldering iron;
Figure 8 - Electric Soldering Iron
A soldering iron is a hand tool used for puddling and soldering to heat parts, flux, melt the solder and add it to the contact point of the parts to be soldered. The working part of the soldering iron, usually called the tip, is heated by a flame (for example, from a blowtorch) or electric current.
Using a soldering iron, you can solder faulty components on the sound card or solder cable wiring to the plug.
Insulating tape - designed for electrical insulation of live parts.
Electrical tape is wrapped around the cable where the soldering took place.
Figure 9 - electrical tape
Screwdrivers - are used to dismantle and install the sound card and speaker system.
Figure 10 - screwdrivers
BIOS - Basic Input / Output System. You can configure the connection of the sound card.
wires and plugs - serve 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. Malfunctions of a sound card are a very common phenomenon, this breakdown occurs very easily, but it is very difficult to eliminate it, since the cause of the occurrence in the absence of sound can be hidden in the most unexpected places of the computer.
a) When turned on system unit, there are no sound signals. Causes of breakdown and how to eliminate:
check the correctness of the connection of the speakers to the sound card connector and the connection to the power supply of the speakers themselves.
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 hardware in the system device manager and, if necessary, remove conflicting programs and install the necessary drivers ...
a sound card malfunction may be accompanied by faulty elements and parts of the sound card, for example, the sound card output itself or the soldering on the track itself, which must be soldered.
a sound card, especially if it is built-in, can simply be disabled in 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 connecting them, 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 system unit.
b) There is a hum and an incomprehensible background from the speakers - the connection plugs are faulty, which must be soldered or replaced; over time, the capacitance of the capacitors on the sound card and in the pre-amplification unit of the speakers themselves is lost.
c) There are incomprehensible intermittent sounds and extraneous noises coming from the speakers - in this case, there are no necessary audio codecs. Which must be replaced or updated through the necessary software.
2. Malfunction of speaker systems.
Speaker systems, especially inexpensive and from unknown manufacturers, cannot withstand long-term operation at maximum power, since their built-in power supply is designed for the nominal 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 unit experiences increased loads, and this causes overheating of the circuit elements, and, as a result, their damage.
Mechanical volume controls are a common cause of sound distortion. It is easy to "calculate" such a regulator, it is enough to add or subtract the volume of the sound, the wheezing and crackling that occurs at this time will indicate 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 coil winding of the speaker may burn out, such a loudspeaker will have to be replaced, speakers with significant damage to the diffuser must 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 Whatman paper.
When operating the speakers at high power, a break occurs in the conductor connecting the external terminal of the loudspeaker with the terminal of its diffuser, in this case everything is repaired by ordinary soldering.
Often the cause of a malfunction is wire breaks near the connection plugs, and the insulation of these wires in most cases remains intact, which complicates the diagnosis. You can "call" such damage using a multimeter, if you do not have one, you can use a battery and an ordinary 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, but then everything is clear - the light bulb lit up 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 too. To do this, you need to clean the plug from the plastic covering it, solder the cable wiring to it again, then carefully wrap it all with electrical tape.
3 Working with a computer network
3.1 Description of the place of laying the computer network and the available equipment
Place of laying the network 2nd floor of the building sushi bar "Samu-rai" on the street. Zwilinga 21.
This floor contains one room, room size: room length 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 batteries, one chair, wardrobe, sofa, refrigerator, computer desk.
Available equipment:
- D-LinkDES-1210-28 / ME switch;
- cable NETLANEC-UU002-5-PVC-GY, 2 pairs, Cat.5, internal;
- network sockets for connecting an 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 the central node (switch), forming a physical segment of the network.
The type of cable used in the design of the network - shielded twisted pair of category 5, provides a throughput of 100 Mbit / s, is intended for indoor installation. The advantages of this cable include its inexpensive cost, but at the same time it fully complies with 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 walls 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 latch lock allows you to fix the box door in the open position;
- all screws included in the box have a universal head. It fits both a Phillips screwdriver and a flat head screwdriver.
The switch used in the design of the network was chosen by D-LinkDES-1210-28 / ME. Since this switch has advanced functionality, and, moreover, are inexpensive solution This switch features a high port density, 24 FastEthernet ports, and 4 GigabitEthernet ports, including 2 combo 1000Base-T / SFP ports that support both SFP Gigabit transceivers and and 100BASE-FX.
The benefits include: broadcast storm control, which minimizes the likelihood of virus attacks on the network, as well as port mirroring, which simplifies traffic diagnostics, and also helps administrators monitor the performance of the switch and change it if necessary. bridges.
Appendix B
3.3 Stages of installation and configuration of a computer network
During the installation, 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 gasket... The rigid polyethylene shell is not afraid of ultraviolet radiation, is resistant to cold up to minus 60 degrees, and external influences.
When fastening the cable, it is necessary to remove the braiding layer with a clerical knife, under which there is a stranded steel cable. Next, using a screwdriver and a hex head bolt, we wind a steel cable onto the bolt, which, when tightened, will tighten the cable base, this completes the installation.
Next, you need to crimp the shielded twisted pair according to the standard used in the enterprise. It looks like this:
1 –white orange;
2 - orange;
3 - white blue;
4 - green;
5 - white-green;
6 - blue;
7 - white-brown;
8 - brown.
Before crimping the cable, you must prepare it. First, remove the braid by carefully cutting the cable. Then we remove the shielded film. AND the final stage will straighten each strand to make it look like a string, and insert it into an RJ-45 connector as standard, and crimp with a crimping tool.
After all the manipulations with the cable, you need to configure it on the computer. In order to configure it, you need to set your personal IP address, subnet mask, main gateway, preferred gateway, and an alternative gateway, which for each subscriber differs from another subscriber, since each subscriber is issued his own personal contract, which contains the entire required setup information.
After completing the steps, we measure the speed and ping using the site www.speedtest.net, 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 in a simple way testing of all features local network is the use of different kinds of testers. They allow you to automate and simplify the testing process as much as possible, therefore, if there is such an opportunity, it is advisable to use this particular method.
There are different types of testers, differing in testing methods, the number of different tests, and the way the results are presented. The cost of testing equipment directly depends on these functions. There are 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 that provides professional services for SCS installation 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 in 90% of cases is quite enough.
There are two main types of testers: for testing physical lines and network analyzers.
Testers for testing physical lines are most prevalent due to their price. Such a tester is capable of detecting a malfunction of the cable segment on physical level, up to determining the location of the breakage of the conductors. In addition, it can, for example, test the impedance of a line or measure the baud rate, which allows you to determine the used network standard or compliance with a specific standard. Even a small company can afford the purchase of such a tester, which will make it possible to quickly identify and eliminate 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, you can not only investigate the characteristics of the cable structure, but also obtain complete information about the process that occurs when a signal passes 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 you to assess 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 the tester often comes with various adapters and auxiliary tools. For example, using the appropriate adapters, you can test both coaxial segments and twisted pair cable segments. As for fiber-optic lines, the equipment for testing them has a more complex design and is often focused only on fiber-optic testing.
Cable segment testing is in progress different ways that depend on the availability of cable access. 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 at the other end. As a result, the tester can check the resistance of each conductor, as well as the compliance of their connection with one of the standards. Using the 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 programmatic method
When there is no opportunity 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 system utility ping.
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 test 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 enter the following command: ping 192.168.2.2
If as a result of execution of this command 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 the command execution, the message "The waiting interval for request has been exceeded" appears on the screen, this will indicate that the cable has a break or the connectors are crimped incorrectly.
Similarly, you can test any cable, including twisted pair cable. In the case of a twisted pair cable, this kind of connection is possible only for the crossover version. If it is necessary to test the functionality of a patch cord cable, it must be connected to a central unit, for example, a switch, and paired with it, use a working cable that is connected to the second computer.
Conclusion
In the process of passing the industrial practice according to PM 03. "Maintenance and repair of computer systems and complexes" the following tasks were completed:
the structure of the enterprise LLC "TELKOM" and the main types of its activities were studied;
the operation of the PC sound system has been studied, it consists of 4 stages. (Appendix A)
Considered the stages of setting up and configuring the sound system of a PC, which consist of two stages;
the tools necessary for diagnostics and repair of the PC Sound system are listed, 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. Standards for local area networks: Handbook / VK Shcherbo, VM Kireichev, SI Samoilenko; ed. S. I. Samoilenko. - M .: Radio and communication, 2005.
2. Practical data transfer: Modems, networks and protocols / F. Jennings; per. from English - M .: Mir, 2000.
3. Computer networks: protocols, standards, interfaces / Yu. Black; per. from English - M .: Mir, 1999.
4. Fast Ethernet / L. Quinn, R. Russell. - BHV-Kiev, 2007.
5. Switching and routing of IP / IPX traffic / M. V. Kulgin, IT Co. - M .: Computer-press, 2001.
6. Fiber optics in local and corporate communication networks / AB Semenov, IT Co. - M .: Computer press, 1998.
7. Internet Protocols... S. Zolotov. - SPb .: BHV - Saint Petersburg, 2002.
8. Personal computers in TCP / IP networks. Craig Hunt; per. from English - BHV-Kiev, 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 - Kiev: Dia-Soft, 2006.
Appendix A
The device and principle of operation of the PC sound system
Appendix B
Analysis of financial costs for the repair of the audio system
Appendix B
Computer network project
Appendix D
Analysis of financial costs for creation
computer network
Appendix D
Screenshot of the network diagram and listing of commands for setting up workstations in the CiscoPacketTracer program
Appendix E
Listing of configuration commands
active network equipment in the Cisco-PacketTracer program
Appendix G
Listing of commands for configuring active network equipment in CPT
1.Pc sound system
The sound system of the PC in the form of a sound card appeared in 1989, significantly expanding the capabilities of the PC as a technical means of informatization.
PC Sound System - a complex of software and hardware that performs 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 the 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 sound signals (mode FullDuplex);
processing of sound signals: editing, combining or dividing signal fragments, filtering, changing its level;
processing of a sound signal in accordance with the algorithms of surround (three-dimensional - 3 D- Sound) sound;
generating with the help of 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 PC sound system is constructively sound cards, either installed in the motherboard slot, or integrated on the motherboard or expansion card of another PC subsystem. Separate functional modules of the sound system can be implemented in the form of daughter cards installed in the corresponding connectors of the sound card.
A classic sound system, as shown in fig. 5.1, contains:
Sound recording and playback module;
synthesizer module;
interface module;
mixer module;
speaker system.
The design of the PC sound system is 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 functional modules of a sound card, it is customary to use the terms "PC sound system" or "sound card".
2. Recording and playback module
The module for recording and reproducing the sound system carries out analog-to-digital and digital-to-analog conversions in the mode of software transmission of audio data or their transmission via DMA channels (DirectMemoryAccess- direct memory access channel).
Sound, as you know, is longitudinal waves that freely propagate in air or other medium, therefore the sound signal is continuously changing in time and space.
Sound recording is storing information about sound pressure fluctuations at the time of recording. Currently, analog and digital signals are used to record and transmit information about sound. In other words, the audio signal can be analog or digital.
If, when recording sound, a microphone is used that converts a time-continuous sound signal into a time-continuous electrical signal, the sound signal is obtained in analog form. Since the amplitude of the sound wave determines the loudness of the sound, and its frequency determines the pitch of the sound tone, so far as to preserve reliable information On 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 sent to the input of the PC sound card in analog form. Due to the fact that the PC operates only with digital signals, the analog signal must be converted to digital. At the same time, the acoustic system installed at the output of the PC sound card perceives only analog electrical signals, therefore, after processing the signal using a PC, it is necessary to convert the digital signal back to analog.
Analog to digital conversion is the conversion of an analog signal to digital and consists of the following main stages: sampling, quantization and encoding. A diagram of analog-to-digital conversion of an audio signal is shown in Fig. 5.2.
The analog audio signal is preliminarily fed to an analog filter, which limits the signal bandwidth.
Signal sampling consists in sampling samples of an analog signal with a specified frequency 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 frequency of the original sound signal should be at least 40 kHz, i.e. 40,000 samples per second are required. Therefore, in most modern PC sound systems, the maximum audio sampling rate is 44.1 or 48 kHz.
Amplitude quantization is a measurement of instantaneous values of the amplitude of a time-discrete signal and its transformation into a discrete one in time and amplitude. In fig. 5.3 shows the process of quantizing the level of an analog signal, and the instantaneous values of the amplitude are encoded with 3-bit numbers.
Encoding consists in converting the quantized signal into a digital code. In this case, the measurement accuracy during quantization depends on the number of codeword bits. If the amplitude values are written using binary numbers and the length of the code word is set N bits, the number of possible values of codewords will be equal to 2
N .
The same number of levels of quantization of the reference amplitude can be. For example, if the value of the sample amplitude is represented by a 16-bit codeword, the maximum number of amplitude grades (quantization levels) will be 2 16 = 65 536. For an 8-bit representation, respectively, we get 2 8 = 256 amplitude grades. 
Analog-to-digital conversion is carried out by a special electronic device - analog-to-digital conversiontelem(ADC), in which discrete signal samples are converted into a sequence of numbers. The received stream of digital data, 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 stages, as shown in Fig. 5.4. At the first stage, samples of the signal with the sampling frequency are extracted from the digital data stream using a digital-to-analog converter (DAC). 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.
Recording and storing audio in digital form requires a large amount of disk space. For example, a stereo audio signal with a duration of 60 seconds, digitized at a sampling rate of 44.1 kHz with 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, when I groomed for one count.
Such methods of encoding audio data using special encoding devices can reduce the volume of information flow up to almost 20% of the original. The choice of the 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 reproduction module contains an ADC, DAC and a control unit, which are usually integrated into a single microcircuit, also called a codec. The main characteristics of this module are: sampling rate; type and capacity of ADC and DAC; method of encoding audio data; the ability to work in the mode FullDuplex.
The sampling rate determines the maximum frequency of the signal being recorded or played back. For recording and reproducing human speech, 6 - 8 kHz is sufficient; music with not high quality- 20 - 25 kHz; for high quality sound (Audio CD), the sampling rate must be at least 44 kHz. Almost all sound cards support the recording and playback of a stereo audio signal with a sampling rate of 44.1 or 48 kHz.
The bit width 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 classified as hi-fi, which should provide studio sound quality. Some sound cards are equipped with 20- and even 24-bit ADCs and PAPs, which significantly improves the quality of sound recording / playback.
FullDuplex(full duplex) is a channel data transmission mode, according to which a sound system can simultaneously receive (record) and transmit (play) audio data. However, not all sound cards fully support this mode, since they do not provide high quality sound during intensive data exchange. Such cards can be used to work with voice data on the Internet, for example, during teleconferencing, when high quality sound is not required.
3. Synthesizer module
The electronic music digital synthesizer of the sound system allows you to generate almost any sound, including the sound of real musical instruments. The principle of operation of the synthesizer is illustrated in Fig. 5.5.
Synthesis is the process of recreating the structure of a musical tone (note). The sound signal of any musical instrument has several time phases. In fig. 5.5, 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 unique, 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 an attack for different musical instruments varies from units to several tens or even hundreds of milliseconds. In a phase called support, the amplitude of the signal remains almost unchanged, 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, sound is created in the following way. 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. 5.5, b. Next, the excitation signal is fed to a filter that simulates the frequency response of a real musical instrument. The signal of the amplitude envelope of the same instrument is fed to the other input of the filter. Further, the set of signals is processed in order to obtain special sound effects, for example, echo (reverberation), choral performance (ho-rus). Next, digital-to-analog conversion and signal filtering are performed using a low-pass filter (LPF). Key features of the synthesizer module:
Sound synthesis method;
Memory;
Possibility of hardware signal processing to create sound effects; 
Sound synthesis method, used in the 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.
Frequency Modulation Synthesis Method (FrequencyModulationSynthesis- FM synthesis) involves the use of at least two signal generators of complex shapes to generate the voice of a musical instrument. The carrier frequency generator generates a fundamental tone signal, frequency modulated by a signal of additional harmonics, overtones that determine the timbre of a particular instrument. The envelope generator controls the amplitude of the resulting signal. The FM generator provides acceptable sound quality, is inexpensive, but does not provide sound effects. Therefore, sound cards using this method are not recommended according to the PC99 standard.
Sound synthesis based on the wave table (WaveTableSynthesis - 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 integrated into a WT-generator memory chip. WT synthesizer provides high quality sound generation. 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 either be an independent element (microcircuit), or be integrated into the WT synthesizer. For the vast majority of cards with WT synthesis, reverb and chorus effects have become standard. Sound synthesis based on physical modeling involves the use of mathematical models of sound production of real musical instruments for digital generation and for further conversion into an audio signal using a DAC. Sound cards that use physical modeling are not yet widely used because they require a powerful PC to run.
4. Interface module
The interface module provides data exchange between the sound system and other external and internal devices.
InterfaceISA in 1998 it was supplanted in sound cards by the PCI interface.
InterfacePCI provides a wide bandwidth (for example, version 2.1 - more than 260 Mbps), which allows the transmission of audio streams in parallel. Using the PCI bus allows you to improve the sound quality, providing a signal-to-noise ratio of more than 90 dB. In addition, the PCI bus allows for cooperative processing of audio data, where processing and transmission tasks are shared between the audio system and the CPU.
MIDI (MusicalInstrumentDigitalInterface- musical instrument digital interface) is regulated by a special standard containing specifications for the hardware interface: channel types, cables, ports through 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, which includes a controller - control device, which can be used as a PC, and a musical keyboard synthesizer, as well as slave devices (receivers) that transmit information to the controller at its request. The total length of the MIDI chain is not limited, but the maximum cable length between two MIDI devices should not exceed 15 meters.
Connecting a PC to a MIDI network is carried out using a special MIDI adapter, which has three MIDI ports: input, output and pass-through, as well as two connectors for connecting joysticks.
The sound card includes an interface for connecting CD-ROM drives.
5. Mixer module
The sound card mixer module performs:
switching (connecting / disconnecting) sources and receivers of sound signals, as well as regulating their level;
mixing (mixing) several audio signals and adjusting the level of the resulting signal.
The main features of the mixer module are:
the number of mixed signals on the playback channel;
regulation of the signal level in each mixing channel;
regulation of the total signal level;
amplifier output power;
the presence of connectors for connecting external and internal receivers / sources of audio signals.
The software control of the mixer is carried out either by means of Windows or using 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 sound signals. Compatibility issues are especially important for DOS applications. Each of them contains a list of sound cards that the DOS application is designed to work with.
StandardSoundBlaster support DOS game applications in which the soundtrack is programmed for the Sound Blaster family of sound cards.
StandardWindowsSoundSystem(WSS) Microsoft includes a sound card and software package focused primarily on business applications.
6. Acoustic system
The speaker system (AC) directly converts the sound electrical signal into acoustic vibrations and is the last link in the sound reproducing path.
The structure of the speaker, as a rule, includes several speakers, each of which can have one or more speakers. The number of speakers in a speaker 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, the left and right stereo channels, which requires at least two speakers in a stereo speaker system. Dolby Digital audio contains information for six audio channels: two front stereo channels, a center channel (dialogue channel), two rear channels, and a subwoofer channel. Therefore, a speaker system must have six speakers in order to reproduce a Dolby Digital signal.
As a rule, the principle of operation and the internal structure of audio speakers for household 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, which provide a stereo signal reproduction. Typically, each speaker in a PC speaker has one speaker, however, expensive models two are used: for high and low frequencies. At the same time, modern models of acoustic systems allow reproducing sound in almost the entire audible frequency range due to the use of a special design of the speaker housing or loudspeakers.
To reproduce low and ultra-low frequencies with high quality in the speakers, in addition to two speakers, a third sound unit is used - a subwoofer (Subwoofer), installed under the desktop. This three-piece 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 PC speakers is the ability to have your own built-in power amplifier. A speaker with a built-in amplifier is called active. Passive The speaker does 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 electrical network across special adapter, made in the form of a separate external block or power module, installed in the case of one of the speakers.
PC speaker output power can vary widely depending on amplifier and speaker specifications. If the system is designed for
scoring computer games, enough power 15-20W 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 W per channel. With an increase in the power of the speaker, its dimensions and the cost rises.
Modern models of speaker systems have a jack for headphones, when connected, the sound reproduction through the speakers automatically stops.
The main characteristics of the speaker: frequency band, sensitivity, harmonic distortion, power.
Frequency Response Band (FrequencyRespon se) is the amplitude-frequency dependence of the sound pressure, or the dependence of the sound pressure (sound force) 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. Loudspeakers, as a rule, have a range limited in the low frequency range of 40 - 60 Hz. To solve the problem of reproducing low frequencies, use a subwoofer.
Speaker sensitivity (Sensitivity) characterized by the sound pressure that it creates at a distance of 1 m when an electrical signal with a power of 1 W is applied to its input. In accordance with the requirements of the standards, sensitivity is defined as the average sound pressure in a certain frequency band.
The higher the value of this characteristic, the better the speaker transmits dynamic range music program. The difference between the "quietest" and "loudest" sounds of modern phonograms is 90-95 dB or more. Speakers with high sensitivity reproduce both quiet and loud sounds quite well.
Harmonic coefficient (TotalHarmonicDistortion- THD) evaluates nonlinear distortions associated with the appearance of new spectral components in the output signal. Harmonic distortion is standardized in several frequency ranges. For example, for high-quality Hi-Fi speakers, this ratio should not exceed: 1.5% in the frequency range 250-1000 Hz; 1.5% in the frequency range 1000-2000 Hz and 1.0% in the frequency range 2000 - 6300 Hz. The lower the value of the harmonic distortion, the better the speaker.
Electric power (PowerHandling), which 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 on
rather, on sensitivity, and the power of the speaker mainly determines its reliability.
Often on the packaging of PC speakers, 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 testing of the AU, the values of electrical powers may differ several times. To compare the power of different speakers, you need to know what kind of power is indicated by the manufacturer of the product and by what test methods it is determined.
Among the manufacturers of high-quality and expensive speakers are Creative, Yamaha, Sony, Aiwa. AC over low class produced by Genius, Altec, JAZZ Hipster.
Some Microsoft speaker models are not connected 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.
7. Directions for improving the sound system
Intel, Compaq and Microsoft have now proposed a new architecture for the PC sound system. According to this architecture, the audio signal processing modules are removed from the PC case, in which they are exposed to electrical noise, and are placed, for example, in the speakers of the loudspeaker 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 transmission of digital data in digital form, the use of high-speed USB bus and NECK 1394.
Another area of improvement of the sound system is the creation of surround (spatial) sound, called three-dimensional, or 3D-Sound (ThreeDimentionalSound). To achieve 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. In this case, the property of the human brain is used 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 feels the effect of "moving" the sound source.
A new direction for the use of multimedia technologies is the creation of a home theater based on a PC. (PC- Theater), those. version of a multimedia PC intended for several users at the same time to watch the game,
review educational program or a movie in the DVD standard. PC-Theater includes a special multi-channel speaker system that creates surround sound (SurroundSound). Surround Sound systems create a variety of sound effects in a room, with the user feeling like they are in the center of the sound field and the sound sources around it. Multichannel surround sound systems are used in movie theaters and are already beginning to emerge as consumer devices.
In multi-channel consumer systems, sound is recorded on two tracks of laser video discs or videotapes over Dolby technology Surround developed by Dolby Laboratories. The most famous developments in this area include:
Dolby (Surround) ProLogic- a four-channel sound system containing left and right stereo channels, a center channel for dialogue and a rear channel for effects.
DolbySurroundDigital- a sound system consisting of 5 + 1 channels: left, right, center, left and right channels of rear effects and an ultra-low frequency channel. Signals for the system are recorded in the form of a digital optical phonogram on film.
In some speaker models, in addition to the standard controls for high / low frequencies, volume and balance, there are buttons for turning on special effects, for example, 3D sound, Dolby Surround, etc.
Control questions
List the main steps for analog-to-digital and digital-to-analog conversion.
What are the main functions of a PC sound system?
What are the main components of a PC sound system?
Based on what considerations is the sampling frequency of the signal in the process of analog-to-digital conversion allocated?
What are the main parameters of the sound recording and playback module?
What methods of sound synthesis are used?
What are the functions of the mixer module, and what are its main features?
What is the difference between a passive speaker system and an active one?
CONTENT
Introduction 3
1 ESSENCE OF PC SPEAKERS …………………………… .4
1.1 Sound input / output system - audio adapter ……………………………… ..4
1.2 Sound Reproduction - Acoustic Stereo System …… …………… ... 5
2 PARAMETERS AND FUNCTIONS OF PC SPEAKERS ……. ..9
2.1 Purpose …………………………………………………………………… 9
2.2 Classification …………………………………………………………… .... 9
2.3. Basic principles of work …………………………………………… 12
2.4 Basic characteristics ………………………………………………………………………………………………………………………………………………………………………………………………………………………………………………………
2.5 Major manufacturers ……………………………………………… .14
Conclusion ……………………………………………………………………… ... 16
Bibliography............. .............................. ..... ......................... ....................... 17
INTRODUCTION
At present, our life is absolutely inconceivable without the everyday use of technology, in particular, computer technology. Computer technology combines hundreds of different functions, providing an example of unlimited performance, focus and, of course, practicality.
A modern multimedia PC in full "armament" resembles a home stereo Hi-Fi complex, combined with a display-TV. It is equipped with powered stereo speakers, microphone and optical CD drive. In addition, a new device for a PC is hidden inside the computer - an audio adapter, which made it possible to switch to listening to pure stereo sounds through speakers with built-in amplifiers.
The emergence of multimedia systems undoubtedly brings about revolutionary changes in such areas as education, computer training, in many areas of professional activity, science, art, computer games, etc.
Any user needs high-quality hardware and, of course, a good speaker system for a PC. There are a lot of manufacturers of acoustics at the moment. Each firm has both advantages and disadvantages. Therefore, choosing a good speaker system for your computer is often difficult. If you need good sound quality when listening to music, watching movies, or when playing a three-dimensional game, then you should take the purchase of acoustics more seriously. You will have to tinker a little with the purchase of high-quality acoustics for music, games and films! This is explained by the fact that the sound quality depends on many factors, which will be discussed below.
Modern speaker systems are a ready-made convenient solution for creating a home theater. Ideal for small spaces where it is important to efficiently use the available space. Distinctive advantages - high quality sound and ease of use.
1 ESSENCE OF PC SPEAKERS.
A PC speaker system is a device designed to output sound information processed on a computer. Under the acoustic system in the broad sense of the word we mean an electromechanical converter of electrical sound signals into acoustic ones.
We are all already accustomed to the fact that a modern personal computer can produce a wide variety of sounds. At first, they could only hum and squeak in different ways, then programs appeared, uttering quite distinct words and playing a distant semblance of music heard through a drainpipe; computer games quite quickly learned to emit something like shots and explosions even with the help of a built-in loudspeaker. And now the ubiquity of inexpensive sound cards has made it possible to reproduce any theoretically possible sound with their help. However, in most cases, we hear only those sounds that were laid down in the development of a particular program, and yet many want much more. All this is quite possible - if you have the required hardware and / or programs, and most importantly - knowledge about how to extract the desired sounds from such a seemingly non-musical device like a computer, since a computer, by its initial definition, is a device for storing, processing and transmitting information.
Over the years, the list of tasks performed on a PC has gone beyond just using spreadsheets or text editors. The personal computer is becoming a multimedia complex.
Multimedia is the sum of technologies that enable a computer to input, process, store, transmit and display (output) types of data such as text, graphics, animation, digitized still images, video, sound and speech.
Audio CDs, multimedia presentation preparation, video conferencing and telephony, as well as playing and listening to audio CDs all require sound to become an integral part of the PC. This requires a sound card and speaker system.
1.1 Sound I / O System - Audio Adapter
A microphone is used to input sound into a computer. Continuous electrical vibrations from the microphone are converted into a numerical sequence. This work is done by a device connected to your computer called an audio adapter, or sound card. Reproduction of sound recorded in computer memory also occurs with the help of an audio adapter that converts the digitized sound into an analog electrical audio signal of audio frequency, which is fed to speakers or stereo headphones.
The audio adapter has an analog-to-digital converter (ADC), which periodically detects the audio signal level and converts this reading into a digital code. He writes on external media already as a digital signal.
Digital samples of the real sound signal are stored in the computer memory (for example, in the form of WAV files). The digital signal read from the disk is fed to a digital-to-analog converter (DAC), which converts the digital signals into analog ones. After filtering, they can be amplified and fed to the speakers for reproduction. The important parameters of the audio adapter are the sampling rate of audio signals and the bit quantization.
It follows from the above that the sound card combines the functions of a DAC and an ADC (Figure 1).
Figure 1 - Converting audio for input and output
An audio adapter is a rather complex technical device built on the basis of using the latest advances in analog and digital audio technology.
1.2 Sound reproduction - acoustic stereo system.
No matter how modern it is electronic system sound recording and playback, no matter how many recording formats it serves, combined in one unit, at the end of it, there will be a "speaker" at the output - as it was called before. And at first he was one, well, two - for reproducing high and low sound frequencies in one box-box. With the advent of stereo phonograph records in the 1950s, there were two boxes - separate for the right and left audio channels.
The famous long-standing experience of broadcasting sound transmission was undertaken by the Frenchman Clement Adler as early as 1881 at the Paris Electric Exhibition. Eighty pairs of telephone wires were run from the stage of the Paris Opera into four rooms of a nearby hotel. In this way, visitors to the exhibition were shown the opportunity to listen to an opera performance from a distance. Musical images influenced the listener with the help of two free-standing microphones located on the stage.
After 50 years in the research units of BELL Labs, Harvey Fletcher, the famous American theoretical and practical scientist, founder and leader of the Acoustic Society and President of the Physics Society of the USA, in collaboration with Arthur C. Keller and in collaboration with the eminent the conductor of the symphony orchestra Leopold Stokowski conducted the first experiments on mono- and binaural sound recording. In England, at the same time, similar research was carried out by the engineer of the record company EMI, Alan D. Blumlein, who on December 14, 1931 filed documents for the patent of a spatial-perceived sound recording, also called binaural.
In the development and production of modern, widely used electrodynamic loudspeakers, innovations known since the mid-1920s are still repeated. The ideas and technical solutions that implement them, which form the basis of an acoustic device that converts electrical vibrations into sound ones, were presented by the engineers of the American company GENERAL ELECTRIC Chester W. Rice and Edward W. Kellog in the proceedings of the American Institute of Electrical Engineers. in 1925. Engaged in electroacoustics in parallel with them and independently of them in the same year, engineer Edward Wente from the American company BELL Laboratories also applied for a patent for a similar emitter of sound vibrations.
However, C. Rice and E. Kellogg cited in the article a description of a 1 W amplifier for their loudspeaker. And already in 1926, at their suggestion, the American firm RCA (Radio Corporation of America) developed and made a loud-sounding radio receiver in one case. In addition to the acoustic head, it contained input tuning circuits, tube amplifier and a power supply rectifier. The radio receiver received the popular name "radio", and the dynamic type loudspeaker began to be called simply: "speaker".
A loudspeaker - a device for converting electrical vibrations into acoustic vibrations of the air environment, is the last and one of the most important links in any acoustic path, since its properties have an extremely large impact on the quality of this path as a whole.
According to the method of converting oscillations, loudspeakers are divided into electrodynamic reel-to-reel (the overwhelming number of modern types of loudspeakers), electromagnetic, electrostatic, piezoelectric and some others; by type of radiation - to loudspeakers of direct radiation, diffuser and horn; according to the reproducible range - into broadband, low, medium and high frequency; in terms of consumed electric power - into powerful and low-power.
In the overwhelming majority of modern acoustic systems (more than 90%), the conversion of electrical sound signals into acoustic signals is carried out using electrodynamic heads, the principle of which is based on the interaction of the magnetic field of a permanent magnet with a voice coil wire. When audio-frequency currents flow through the wire under the influence of the electrodynamic force, the loudspeaker coil is alternately drawn in and out of the annular gap of the magnet, depending on the direction of the electric current. Well, then everything is simple: the voice coil is mechanically connected to the emitter - the diffuser, which, in fact, creates a thickening and rarefaction of air in the space, i.e. acoustic waves. Since the sound wave emitted by the front (front) surface of the diffuser is in antiphase with the acoustic wave emitted by the back of the diffuser, both of these waves, when the dynamic head is operating in an open space, can extinguish each other, which is called an "acoustic short circuit" (according to analogy with a short circuit in electrical networks). To avoid this nuisance, the heads are placed in a housing, the main purpose of which is to exclude this very interaction of sound waves from the front and rear surfaces of the diffuser. Loudspeakers installed in the cabinet together with cross-over filters form a speaker system, sometimes called a sound column or simply a loudspeaker.
A relatively small number of loudspeaker systems use emitters based on other physical principles (electrostatic, piezoelectric, isodynamic, plasma emitters), but these types of “exotic” loudspeakers are practically not used in mass loudspeaker systems.
The sensitivity (radiation efficiency) of the loudspeaker at high frequencies is increased by decreasing the inductance of the voice coil, for example, using Foucault eddy currents; a decrease in inductance reduces its electrical resistance and leads to an increase in current at high frequencies. On the low frequencies the sensitivity of the loudspeaker is increased by using special acoustic designs.
The overwhelming majority of modern sound speakers are a set of two or three electrodynamic loudspeakers, placed inside a rectangular case with a width of 20-30 cm.
An important parameter that characterizes loudspeakers is the directional diagram. With a narrow beam, more sound signals from the acoustic emitter are directed directly towards the listener, and sound images appear more clearly.
As in a real concert hall, at home, performers of works of art are supposed to be in front of the listener. This condition is fully satisfied by two speakers (left and right), installed at a certain distance from the listener and one from the other.
How can speakers be used to reproduce binaural sound (i.e. sound intended for listening with headphones, where part of the signal is for one ear and part for the other ear)? As soon as we connect speakers instead of headphones, our right ear will begin to hear not only the sound intended for it, but also part of the sound intended for the left ear. One solution to this problem is to use the cross-talk-canceled stereo or transaural stereo technique, more often referred to simply as the crosstalk cancellation algorithm (CC for short).
The CC idea is simply expressed in terms of frequencies. In Figure 2, the signals S1 and S2 are reproduced by the speakers. The Y1 signal reaching the left ear is a mixture of S1 and a crosstalk (portion) of the S2 signal.
Figure 2 - Scheme of reproduction of binaural sound by speakers
If we decide to use headphones, then we will clearly know the desired signals Y1 and Y2 perceived by the ears. The problem is that you need to correctly identify the signals S1 and S2 in order to get the desired result.
With the correct use of CC algorithms, very good results are obtained, ensuring the reproduction of sound, the sources of which are located in the vertical and horizontal planes. A phantom sound source can be located far outside the line segment between two speakers.
It has long been known that two audio channels are enough to create convincing 3D sound. The main thing is to recreate the sound pressure on the eardrums in the left and right ears in the same way as if the listener were in a real sound environment.
2 PARAMETERS AND PURPOSES OF SPEAKERS PC.
2.1 Purpose
Designed to play sound and melodies. If your computer is equipped with sound speakers and a sound card, it is called multimedia.
A sound card (also called a sound card or music card) is a card that allows you to work with sound on your computer. Nowadays, sound cards are either built into the motherboard, or as separate expansion cards or external devices.
Sound cards today are a whole class of devices, many of which serve much higher purposes than simply outputting MP3 files to speakers. They become hubs for home theaters, hi-fi systems, home and professional studios.
By the way, boards were called boards actually because they were a printed circuit board inserted into an ISA or PCI slot. Today sound cards are also connected via USB, FireWire, PCMCIA
Active speakers are used as a device for reproducing and amplifying music, speech and sound effects.
2.2 Classification
Built-in sound cards.
Where are they embedded? In motherboards. I / O and codecs are soldered directly to the "mother", and the central processor takes over all the computational processing. Such a sound solution is almost free, therefore it is more than acceptable for unpretentious users - despite the disgusting sound quality.
Multimedia sound cards.
This is the most ancient category of boards: it was they who first appeared and made the computer a means of playing and recording music. These cards, in contrast to the built-in ones, have their own sound processor, which deals with sound processing, calculating three-dimensional sound effects used in games, mixing sound streams, etc., which allows offloading the computer's central processor for processing more important tasks.
As a rule, the sound quality in individual multimedia cards is really higher than that of the built-in ones. You can not hesitate to connect not the worst computer speakers and acoustics sets to them - although there is still a long way to go to the Hi-Fi level. Home theater will sound more or less decent in combination with 5.1-speaker sets made especially for computer use.
Moreover, it is already possible to record sound with the help of multimedia cards: it will be quite capable of being at the level of karaoke. And simple programs for working with sound will function normally.
Several years ago, the market for multimedia cards was very saturated, with fights between manufacturers and their products. The most prominent competitors were Aureal and Creative. The cards of these companies used different algorithms for working with 3D sound - each had its own fans.
With the arrival of motherboards with built-in audio, conflicts were resolved on their own: all manufacturers of cheap sound cards are dead. Only Creative remained afloat with its Sound Blaster Audigy / Audigy2 line, which is considered the top-level in multimedia.
Semi-professional sound cards
Actually, these boards can be called differently - either semi-professional or top-end multimedia. But rather, these are still semi-professional boards. As a rule, they are produced by manufacturers of professional equipment, focusing not on musicians, but on lovers of good sound. In other words - cards for audiophiles.
They differ from multimedia primarily in professional circuitry solutions and high quality sound reproduction. At the same time, they usually do not use serious sound processors, and again, the whole burden of processing 3D sound is borne by the central processor.
But for listening to music, these cards are ideal. With good acoustics, devoid of the shameful definition of "computer", or decent headphones, you can get the sound close to an inexpensive Hi-Fi system. You will finally be able to distinguish MP3-files from normal recordings ... And you will start to fear low-quality "empatresses" like fire.
As a basis for theatrical sound, such cards will also work fine. The sound will be clear, not distorted - in general, very decent.
As a rule, cards from manufacturers of professional equipment are equipped with drivers for professional programs for working with music and sound. So this board will be a great start for a beginner musician. However, many of these cards are unsuitable for professional sound recording and in this respect they are no better than their multimedia counterparts.
Professional sound cards
These cards are designed for professional musicians, arrangers, music producers. Anyone who is involved in the production and recording of music. In accordance with the tasks - and features: the highest quality of sound reproduction and recording, minimum distortion, maximum possibilities for working with professional software and connecting professional equipment.
Professional cards usually lack multimedia drivers and DirectX support, which makes many of them useless in games. They do not even support standard system volume controls - each channel is adjusted in a special control panel that shows the signal level in decibels.
Inputs / outputs instead of the standard "minijack" are made either on "tulips" RCA, or on "big jacks", or in the form of XLR connectors, brought out using special interface cables. Many cards have external blocks where all the connectors are brought out for easy connection. There is simply nowhere to plug computer speakers ... These cards are designed to connect professional studio acoustic monitors, mixing consoles, preamplifiers and other "serious" devices.
However, inexpensive professional cards can be the best choice for a true connoisseur of high-quality sound. Cards with RCA connectors are very convenient for connecting hi-fi equipment and will be a good source of sound for a decent audio system. Cards with stereo jack outputs will allow you to connect expensive headphones without adapters and accompanying distortions. However, as a basis for a home theater, only a few of the professional boards are suitable, the number of outputs of which will allow you to connect all six speakers. After all, the main thing here is not the number of channels, but the sound quality of each of them.
External sound cards
This is a relatively recent trend in the world of sound cards that has developed only over the past year. External sound cards are connected to a computer using USB, USB 2.0 or FireWire.
What are these devices made for?
First, moving the card out of the PC case allows you to easily solve some problems related to pickups and interference from other computer components that affect the sound quality. Manufacturers of expensive boards solve these problems with high-quality elements, special insulation, etc., which increases the cost of the device.
Secondly, barebone systems are gaining more and more popularity - small system units with a large number of interface connectors and, as a rule, no more than one PCI slot, which may have to be occupied by something more necessary for the user than a sound card.
Third, a portable professional sound card that connects on the fly to any computer - it's a complete portable studio!
But there are also problems. The first devices released for USB did not gain due popularity due to the low bandwidth of this interface. Restrictions were introduced on the quantity and quality of transmitted signals. However, there are still plenty of USB multimedia cards on the market that offer decent sound and fewer I / O channels.
Today there is a real boom in professional cards connected via the FireWire bus: due to the high bandwidth of the interface, there are practically no problems with the number of channels and signal quality.
Column classification.
-Active (built-in amplifier, require additional power supplies, volume and tone control);
-Passive (low power).
2.3 Basic principles of operation
How conventional sound cards work
In addition to the usual sound channel to the built-in speaker of a computer, sound cards developed by Creative Technology are the de facto standard for creating sounds on an ordinary computer. All other manufacturers of sound cards try to maintain compatibility with these cards either by hardware or software. Previously, sound cards most often used the 16-bit ISA bus; 8-bit cards have not been produced for several years. Since mid-1996, all new sound card models support Plug & Play. Since the fall of 1998, audio cards with the PCI bus have been actively distributed.
Sound cards consist of two main parts: a synthesizer for processing MIDI commands and a block of analog-to-digital (ADC - Analog Digital Converter - ADC) and digital-to-analog (DAC - Digital Analog Converter - DAC) converter. Besides, the sound card usually contains a joystick controller.
With the help of ADC and DAC, it is possible to record and play audio files in mono or stereo with the quality level from cassette recorder to audio CD. The bit depth of the ADC and DAC (analog-to-digital and digital-to-analog converters) is now, as a rule, 16 bits, the sampling frequency is from 5 to 44, 1 kHz, audio compression is possible (for example, using the ADPCM method), which makes it possible to reduce the volume of created audio files. ISA cards also use 8- and / or 16-bit DMA channel, interrupt and I / O ports. When using two DMA channels, simultaneous recording and playback of audio signals is possible, which is realized only in Full-Duplex cards. The most commonly used interrupt 5 (IRQ 5) and the 1st and 5th DMA channels. The bi-directional capability of many sound cards is now actively used for communication over the Internet, so it is recommended to purchase sound cards that support this mode. PCI audio cards always support full duplex due to the much higher bus speed
The synthesizer provides imitation of the sound of musical instruments and the reproduction of various sounds when executing MIDI commands. The synthesizer can be made both on the basis of FM synthesis, and on the basis of the wave table. With FM synthesis, it is possible to simultaneously sound up to 20 instruments, and using the wave table - up to 512 or more. Very often they confuse the number of simultaneously sounding instruments and the bit depth of the sound card. Once again, we draw your attention to the fact that there are NO 32- and 64-bit classical sound cards. The number 32 or 64 (for example, Sound Blaster 32 or Sound Blaster AWE64) means the maximum number of simultaneously sounding instruments and no more. PCI sound cards generally do not have a built-in wavetable. To reduce their cost, the table (s) are loaded into ordinary computer memory, which allows even the most inexpensive audio cards to use wave tables of a large volume and, accordingly, with a large number of instruments (up to 512) and higher sound quality.
PCI sound cards have a 32-bit bus for data exchange, but digital audio processing and receiving / transmitting processing results can be 64 or more.
The software for the sound card, as a rule, includes a mixer program that allows you to adjust the levels of input and output signals, adjust the tone for low and high frequencies (not in all models). On operating systems such as Windows 95 and Windows NT, a mixer is included in the system, but as a rule, a separate mixer program is included with each sound card.
The sound card has a set of connectors for connecting external analog and digital signals:
- input - microphone, line-in, analog CD-ROM (the connector for its connection is usually located on the card itself for connecting the audio output of the CD-ROM drive), CD-ROM digital input (on some new PCI cards);
output - line out, speaker or headphone out). The built-in amplifier has a power of up to 4 watts per channel, most sound cards since 1999 have an amplifier with an output power sufficient only for headphones.
2.4 Main characteristics
Loudspeaker sensitivity is a value that characterizes the sound pressure created by a loudspeaker when a signal with a certain electrical power is applied to it. The loudspeaker sensitivity is determined by measuring the sound pressure at a distance of 1 m from the head along the main axis with a 1 W signal at the input of the loudspeaker.
Power - nominal, programmed (long-term), or peak (short-term) input power, which the head withstands before its destruction. The head can be destroyed with much less power if the speaker is loaded beyond its mechanical capacity at very low frequencies (eg electronic music with a lot of bass or organ music), and damage can also be caused by overloading (“clipping”) the power amplifier.
Impedance (nominal impedance) - as a rule, dynamic heads have an impedance of 2 Ohm, 4 Ohm, 8 Ohm, 16 Ohm.
Frequency Response - Measured, or declared, output response over a specified range
etc.................
INTRODUCTION
The interaction of a person with a computer should be, first of all, mutual (that is why it is communication). Reciprocity, in turn, provides for the possibility of communication between a person and a computer, and a computer with a person. It is an indisputable fact that visual information, supplemented by sound information, is much more effective than simple visual impact. Try, plugging your ears, chat with someone for at least a minute, I doubt that you will get much pleasure, as well as your interlocutor. However, while many orthodox programmers / designers still do not want to admit that sound effects can play the role of not only a signaling device, but information channel, and, accordingly, from inability and / or unwillingness, they do not use in their projects the possibility of non-visual communication between a person and a computer, but even they never watch TV without sound. Currently any large project that is not equipped with multimedia means (hereinafter under the word "multimedia means" we will first of all mean a set of hardware / software means that complement the traditional visual ways of human interaction with a computer) is doomed to failure.
BASIC SOUND TECHNIQUES
There are many ways to make your computer talk or play.
1. Digital to Analogue (D / A) conversion. Any sound (music or speech) is stored in the computer's memory in digital form (in the form of samples) and with the help of DAC is transformed into an analog signal, which is fed to amplifying equipment, and then to headphones, speakers, etc.
2. Synthesis. The computer sends musical information to the sound card, and the card converts it into an analog signal (music). There are two ways to synthesize:
a) Frequency Modulation (FM) synthesis, in which the sound is reproduced by a special synthesizer, which operates with a 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 truly 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 chip from the Yamaha YM3812FM synthesis 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 wave table (Wavetable synthesis), with this method of synthesis a given sound is "typed" not from the sines of mathematical waves, but from a set of really sounded instruments - samples. Samples are saved in the RAM or ROM of the sound card. A special sound processor performs operations on the saml (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 digitization of real instruments, they make the sound extremely realistic. Until recently, this technique was used only in hi-end instruments, but it is becoming more and more popular now. An example of a popular card using WS Gravis Ultra Sound (GUS).
3. MIDI. The computer sends special codes to the MIDI interface, each of which indicates an action that a MIDI device (usually a synthesizer) should perform (General) MIDI is the basic standard of most sound cards. The sound card independently interprets the codes sent and matches them with the sound samples (or patches) stored in the memory of the card. The number of these patches in the GM standard is 128. On PC - compatible computers, historically, two MIDI interfaces have developed: UART MIDI and MPU-401. The first one was implemented in SoundBlaster's cards, the second one was used in early Roland models.
AUDIO CAPABILITIES OF THE IBM PC FAMILY
Already on the very first models of the IBM PC, there was a built-in speaker, which, however, was not designed for accurate sound reproduction: it did not provide reproduction of all frequencies of the audible range and did not have sound volume controls. And although the PC speaker has survived on all IBM clones to this day, it is more a tribute to tradition than a vital necessity, because the speaker has never played any serious role in human-computer communication.
However, already in the PCjr model, a special TI SN76496A sound generator 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 entirely correct statement - in fact, the TI microcircuit had four independent sound generators, but from the programmer's point of view it was one microcircuit with four independent channels ). All four voices had independent volume and frequency control. However, due to marketing errors, the PCjr model was never widely used, was declared unpromising, was discontinued and support was discontinued. From that moment on, IBM no longer equipped its computers with proprietary sound tools. sound cards have taken a firm hold on the market.
OVERVIEW OF SOUND CARDS
A kind of "illegitimate son" of a PC and a person's desire to hear a decent sound with a minimum of financial costs. It is not for nothing that Covox is called "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 used for a printer). 8-bit codes can be sent to this port, which, after simple mixing at the output, will give quite a satisfactory mono sound.
Unfortunately, due to the fact that the main manufacturers software ignored this simple and ingenious device (collusion with sound card manufacturers), then covox never received any software support. However, it is not difficult to independently write a driver for covox "a 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 port 61 PPI.
The SoundBlaster Pro (SB-pro) The Creative Labs "SoundBlaster (SB) was the first Adlib-compatible sound card that could record and play 8-bit samples, supported FM synthesis using the Yamaha YM3812 chip. The original mono-model SB was equipped with one such chip, and the newer stereo model had two. The most advanced model in this family is SB-pro. 2.0, this card contains the most advanced FM synthesis chip (OPL-3 standard). SB-pro is capable of digitizing / playback of real sound at up to 44.1 Hz (CD-player frequency) in stereo mode Also with the help of external drivers this card supports General MIDI interface Contains built-in 2-watt preamp and CDD controller (usually Matsushita).
External line in.
SB compatible MIDI,
SB CD-ROM interface.
The SB-pro was fully compatible with the Adlib card, which made it a tremendous success in the low-cost home sound system market (primarily in games). And although the professionals were unhappy with the unnatural "metal" sound, and the MIDI simulation left much to be desired, this card was liked by numerous fans of computer games, who encouraged developers to include support for SundBlaster cards in their games, which finally cemented Creative Labs' leadership in the market ... And now any program that claims to produce sound on something other than a PC-speaker is simply obliged to support the de-facto SB standard. Otherwise, she runs the risk of simply not being 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 sound. And of course SB16 is fully compatible with Adkib & SB. The SB-16 is capable of playing 8- and 16-bit stereo samples at a frequency of up to 44.1 KHz with dynamic sound filtering (this card allows you to suppress unwanted frequency ranges during playback). The SB16 can also be equipped with a dedicated ASP (Advanced (Digital) Signal Processor) chip that can compress / decompress audio 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 additionally install a special WaveBlaster expansion card, 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 and dynamically filter the audio stream. Like SB-pro and SB-16, PAS performs FM synthesis via Yamaha YMF262 (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),
Despite the fact that Media Vision claims that its products are fully compatible with the SB standard, this is not entirely true, and many people have received unpleasant surprises from this card when they tried to use it as an SB. However, this is somewhat offset by excellent stereo sound and very low noise levels.
The gravis ultrasound
The Advanced Gravis "
Gravis UltraSound (GUS) is the undisputed leader in 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 handle sampling rates up to 44.1 KHz and can handle 16-bit stereo sound. Recording is a little more complicated - initially, the standard GUS models performed only 8-bit audio recording, but the newer models (GUS MAX) are capable of 16-bit recording. In general, the sound reproduced by GUS "is more realistic (due to the use of WS-synthesis, instead of FM), and of course GUS provides excellent support for General MIDI due to the fact that it does not need to" construct "all the variety of sounds from the set sine waves, - he has at his disposal a special library of about 6M in size, from which he can load instruments during playback.
Supported input devices:
Audio Line In.
Supported output devices:
Audio Line Out,
Amplified Audio Out,
Speed compensating joystick (up to 50 Mhz),
General MIDI (requires optional MIDI adapter),
SCSI CD-ROM (requires optional SCSI interface card).
GUS is not SB-compatible card and does not support SB or Adlib standard. Some compatibility, however, can be achieved through software emulation using special SBOS (Sound Board Operating System) drivers supplied with the GUS. However, in practice, satisfactory SBOS performance is more random than natural. In addition, SBOS significantly slows down the processor which makes GUS virtually unusable for multimedia applications written exclusively for SB. Yet the exceptional sound quality of GUS "I made the software makers include drivers for this card in their products. And although support for the GUS standard has not yet become as commonplace as support for the SB standard, there is no doubt that the GUS card is the second most important card after SB.
Problems of promoting GUS to the modern gaming market are complicated by the fact that currently 45% of games are written in Miles Design AIL 2.0 - 3.15, 50% on HMI SOS 3.0 - 4.0, the remaining 5% on homemade sound libraries. How to support GUS learned only AIL 3.15 and then only almost. Before that (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 memory GUS "I fit 30-50 timbres), in AIL 3.15 it is a little more humane - timbres are loaded as needed (almost) but not unloaded (!!), so the situation is reduced to the previous one. original timbres are used by 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 "the music. Not to mention the problems of manufacturers with creating music for standard timbres and inventing how to cram them into 512 / 256K.
The Roland LAPC-1 and SCC-1
The Roland LAPC-1 is a semi-professional sound card based on the Roland MT-32Module. LAPC is identical to the MIDI interface on PC cards. It contains 128 instruments. LAPC-1 uses a combined method of constructing the sound of a note: each note consists of 4 "partials", each of which can be a sample or a simple sound wave. The total number of partials "s is limited to 32", therefore only 8 instruments can be played simultaneously, 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), embedded in the internal ROM memory. A patch can be made up of 24 partials, but most patches are made up of one partials "a. 15 instruments and one percussion can be played simultaneously. 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 biggest drawbacks 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 them impossible to use 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 a flute) are superior in quality to similar instruments GUS "I. The quality of the reproduced sound effects is also very high. The sound quality of the SCC-1 can be considered simply outstanding. Which makes Roland cards one of the best for creating professional instrumental music, however they are completely unsuitable for use in multimedia systems, and Roland cards are not compatible with any current 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 play at up to 44.1 KHz.
Similar to Adlib Gold 1000, but with 16-bit sampling.
Based on the Yamaha YMF3812 FM chip. 11 channels.
8-bit mono sound at a frequency of 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 at a frequency of up to 44.1 KHz. Contains a MIDI interface.
Turtle Beach MultiSound
Based on Motorola 56001 DSP chip. Contains 384 16-bit samples. 15 channels. Special effects. Stereo sound at a frequency of up to 44.1 KHz. Not compatible with any other standard.
AudioBahn 16 from Genoa Systems
Based on the Arial from Sierra semiconductor chip.
Adlib and SB compatible card with SCSI and MIDI interface. Contains 1M samples in ROM. 32 channels. 16-bit stereo sound at a frequency of up to 44.1 KHz.
TXX SOUND BOARDS: BASIC CONCEPTS
Before moving on to the next section, which touches on the practical issues of purchasing a sound card, it is necessary to stipulate a number of terms:
Frequency Response
Shows how well the sound system reproduces sound across the entire frequency range. An ideal device should transmit all frequencies equally from 20 Hz to 20,000 Hz. And although in practice at frequencies above 18000 and below 100, there may be a decrease in the characteristic by an amount of -2dB 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 of the values (in dB) of the undistorted maximum signal of the board to the noise level of the electronics arising from your own electrical diagrams boards. Since humans perceive noise differently at different frequencies, a standard A-weighting grid has been developed that takes into account annoying noise levels. This number is usually what we mean when talking about S / N Ratio. The higher this ratio, the better the sound system. A decrease in this parameter to 75 dB is unacceptable.
Quantization noise
Residual noise, characteristic of digital devices, which occurs due to imperfect signal conversion from analog to digital form. This noise can only be measured in the presence of a signal and is shown as a level (in dB) relative to the maximum allowed output signal. The lower this level, the higher the sound quality.
Total harmonic distortion + noise Reflects the effect of distortion introduced by sound amplification equipment and noise generated by the board itself. It is measured as a percentage of the undistorted output level. A device with an interference level of more than 0.1% cannot be considered good quality.
Channel separation
Simply a number indicating to what extent the left and right channels remain mutually independent. Ideally, the separation of the 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 in dB between the max and min signal that the card can 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 sound card preamplifier. It is desirable to have a high potential gain at a low input voltage. A 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 you can see above in this moment just a huge number of sound systems for personal computers have been thrown onto the market. Therefore, choosing a sound card is not easy, 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 have chosen already carries a CD-ROM controller "a" of your chosen design.
sound card) - additional equipment of a personal computer that allows you to process sound (output to acoustic systems and / or record). At the time of their appearance, sound cards were separate expansion cards installed in the corresponding slot. In modern motherboards, they are presented in the form of a hardware codec integrated into the motherboard (according to the Intel AC'97 or Intel HD Audio specification).
The interaction of a person with a computer should be, first of all, mutual (that is why it is communication). Reciprocity, in turn, provides for the possibility of communication between a person and a computer, and a computer with a person. It is an indisputable fact that visual information, supplemented by sound information, is much more effective than simple visual impact. Try, plugging your ears, chat with someone for at least a minute, I doubt that you will get much pleasure, as well as your interlocutor. However, while many orthodox programmers / designers still do not want to admit that sound impact 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 in their projects the possibility of non-visual communication between a person and a computer, but even they never watch TV without sound. At present, any large project that is not equipped with multimedia means (hereinafter, under the word "multimedia means" we will primarily mean a set of hardware / software means that complement the traditional visual ways of human interaction with a computer) is doomed to failure.
BASIC SOUND TECHNIQUES
There are many ways to make your computer talk or play.
1. Digital to Analogue (D / A) conversion. Any sound (music or speech) is stored in the computer's memory in digital form (in the form of samples) and with the help of DAC is transformed into an analog signal, which is fed to amplifying equipment, and then to headphones, speakers, etc.
2. Synthesis. The computer sends musical information to the sound card, and the card converts it into an analog signal (music). There are two ways to synthesize:
a) Frequency Modulation (FM) synthesis, in which the sound is reproduced by a special synthesizer, which operates with a 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 truly 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 chip from the Yamaha YM3812FM synthesis 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 wave table (Wavetable synthesis), with this method of synthesis a given sound is "typed" not from the sines of mathematical waves, but from a set of really sounded instruments - samples. Samples are saved in the RAM or ROM of the sound card. A special sound processor performs operations on the saml (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 digitization of real instruments, they make the sound extremely realistic. Until recently, this technique was used only in hi-end instruments, but it is becoming more and more popular now. An example of a popular card using WS Gravis Ultra Sound (GUS).
3. MIDI. The computer sends special codes to the MIDI interface, each of which indicates an action that a MIDI device (usually a synthesizer) should perform (General) MIDI is the basic standard of most sound cards. The sound card independently interprets the codes sent and matches them with the sound samples (or patches) stored in the memory of the card. The number of these patches in the GM standard is 128. PC-compatible computers have historically developed two MIDI interfaces: UART MIDI and MPU-401. The first one was implemented in SoundBlaster's cards, the second one was used in early Roland models.
AUDIO CAPABILITIES OF THE IBM PC FAMILY
Already on the very first models of the IBM PC, there was a built-in speaker, which, however, was not designed for accurate sound reproduction: it did not provide reproduction of all frequencies of the audible range and did not have sound volume controls. And although the PC speaker has survived on all IBM clones to this day, it is more a tribute to tradition than a vital necessity, because the speaker has never played any serious role in the communication of a person with a computer.
However, already in the PCjr model, a special TI SN76496A sound generator 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 entirely correct statement - in fact, the TI microcircuit had four independent sound generators, but from the programmer's point of view it was one microcircuit with four independent channels ). All four voices had independent volume and frequency control. However, due to marketing errors, the PCjr model was never widely adopted, it was declared unpromising, was discontinued and support was discontinued. sound cards have taken a firm hold on the market.