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Basic parameters of sound adapters
Functional diagram of the sound adapter
When considering the architecture, there is no need to distinguish between the software and hardware parts of the adapter and consider them as one whole.
Block diagram
Let's look at each block in detail.
Analog ports
The Turtle Beach Santa Cruz map is shown as an example. Note that a port is a logical concept, and a connector is a physical concept, since one connector can combine several ports (this is practiced due to the lack of space on the card plate; switching modes is done through the management utility).
External connectors– cards placed on the bracket (bracket) or the outer edge of the motherboard in the case of an integrated adapter. Executed in the format stereo mini jack.
Internal connectors– located on the map itself. Usually executed in MPC format.
There is an AC"97 specification indicating whether a port is optional or mandatory (which is usually followed). The color of the connectors is determined by the PC"99 specification (and is strictly followed).
Line inputs
Linear stereo input.
Required. Designed for playing or recording an analog signal (stereo or mono) from the linear output of other, usually external analog devices, for example, an audio player, radio, VCR, etc.
Painted blue.
In addition to it, the sound card is required (and optional in the case of an integrated adapter) two additional internal line inputs.
Stereo CD Audio input.
Required. Designed to connect a CD drive with an audio cable. Allows you to play audio CDs. This uses the internal DAC of the drive and the sound card mixer.
Note that on modern CDs/ DVD drive x with the IDE interface under Windows OS (starting from version 98), digital reading of CD-DA audio tracks is possible, which is preferable due to the absence of interference. The Digital CD audio option is enabled in the drive properties. It is not necessary to disconnect the analog audio cable. Made in MPC format, 4-pin. Painted white.
Additional linear input (AUX-In).
Required. It transmits analog audio from FM or TV tuner cards, or other internal devices, for example, a second CD drive, DVD drive or MPEG2 decoder card. Painted blue.
PC-Beep.
Optional, monophonic. Connects to system board and allows you to redirect signals for the system speaker to external speakers via the line output. The connector consists of two needle contacts.
Note that signals can be sent to the line inputs simultaneously - the analog mixer will combine them.
Do not think that recording is necessarily carried out through the input ports. For example, you can feed a signal from a cassette player to the line input and listen to it immediately (and without launching the software player!). The audio stream not only will not “cross” the adapter, but will not even reach the codec, being limited to the mixer.
Note that if you can play many streams ( sound files), then record – a maximum of two (usually stereo sound).
Microphone input
Mandatory, outdoor. Monaural, with automatic adjustment amplification and with support for both electret and electrodynamic microphones. Support for electrodynamic microphones, which are characterized by a weak signal, is provided by the additional gain activation mode (20 dB Boost).
Used, for example, to insert speech comments into a clip in a photo album or an Internet conversation. Painted red or pink.
The input is of low quality and is only suitable for speech recording. The fact is that a normal microphone preamplifier costs $50 or more, so it would sharply raise the price of the adapter. Well-known sound engineer E. Petrov even recommends removing the built-in amplifier (after working with a soldering iron), which leaves only 2 modes: an additional linear input and a 20 dB amplifier for an electrodynamic microphone.
Fans of karaoke or those interested in recording vocals need to purchase a microphone preamplifier that connects to the line input of the adapter. Such devices have different designs:
in the form of an external box with batteries;
in the form of an internal board on a bracket in the rear of the case;
in the form of a port module included with the sound card.
Line outputs
Such outputs are necessarily present, are external and are designed to output sound to active speakers, an amplifier or the line input of any external device (for example, a recorder). The number of output (analog) channels determines the channel of the adapter and can be equal to 2, 4, 6.
Linear stereo output front.
Required. Designed to connect frontal (i.e. located in front) acoustic speakers. Made in stereo mini-phone jack format. Painted green.
Linear stereo rear output.
Present in 4- and 6-channel adapters. Used to connect rear speakers.
Usually made in stereo mini-phone jack format and then painted black. To save space, it can be output to a compact integrated G9 connector.
Center speaker and subwoofer output(“theater port”).
Present only in 6-channel adapters. Dual channel.
It can be made in stereo mini-phone jack format, but to save space it can be output to a compact integrated G9 connector (or combined, see below).
A small number of channels can be compensated by a digital output port (through which several channels are transmitted), but this increases the cost of both the adapter itself and especially the acoustics, and, in addition, it only has an effect on expensive acoustics.
Modern adapters allow you to do more sophisticated upmixing and downmixing, for example, dividing 6-channel sound into 4 speakers and vice versa.
Adapter channel
2 channel adapters in the form of sound cards are now represented by only a small number of models. The main part of them is represented by adapters integrated into motherboards. These adapters work with a 2-3 component speaker system (the third component is a subwoofer). This solution is addressed to those who do not want or do not have the opportunity to clutter their workplace with many speakers and become entangled in wires.
4 channel adapters designed for a 4-5 component speaker system (the fifth is a subwoofer) and already provide full sound in games.
6 channel adapters They provide full-fledged sound not only in games, but also in the home theater (movies on DVD). The acoustics are 6-component and differ from 4-channel by adding only one (central) speaker, but it provides many more possibilities. Therefore, preference should be given to 6-channel adapters over 4-channel ones, especially since the price difference between the adapters is also minimal.
Headphone output
Usually combined with a linear output to the front speakers. According to AC"97, the output has an impedance of 32 Ohms, so headphones must also be selected with the same impedance. A stereo mini-phone jack is used.
Telephone port
This is an optional bidirectional internal port (MPC3 format) called TAD (Telephone Answering Device). It is connected by a cable to the internal voice modem and makes it possible not to switch the microphone from the sound card to the modem again), and also to play sound from the modem. Painted red. Note that at present, unfortunately, there are no hardware PCI modems costing up to $100 that are adapted for the domestic telephone network.
Connector colors
According to the PC"99 specification, audio connectors have the following colors.
Connector colors
| Connector | Color |
|---|---|
| External on the map | |
| Line input | Blue |
| Line output front / headphones | Green |
| Rear line output | Black |
| Microphone | Red |
| MIDI/Game | Golden |
| Domestic on the map | |
| CD Audio-In | White |
| AUX-In | Blue |
| TAD | Red |
| Columns | |
| Columns | Brown |
| Subwoofer | Orange |
Analog mixer
The adapter contains analog input and output mixers, volume controls, complemented complete shutdown each channel (Mute). Occasionally there are mixers with tone control. An explanatory diagram is shown for the case of a 2-channel card.
Mixer features include integral part standard application Volume Control – volume control. According to AC"97, it is possible to close any channel and adjust the volume of not only analogue but also digital sources.
Instead of separate controls on stereo channels, this application uses master volume and channel balance. Here:
Volume Control– linear output. Located in the mixer. Another common name is Master Volume.
Wave – digital channel for Wave files.
ADC and DAC
ADC (A tax ts digital P converter, Analog to Digital Convertor, ADC) is used for digitization analog sound(usually writing to a file).
DAC (C ifro- A tax P converter, Digital to Analog Convertor, DAC) carries out inverse conversion digital audio to analog (dedigitization).
The main parameters of the ADC are the sampling frequency and the quantization bit depth of the ADC when digitizing using the PCM (Pulse Code Modulation) method. The essence of the method is to approximate the signal amplitude as a function of time by a step function. The sampling frequency determines the frequency of the “columns”, and the bit depth determines maximum number levels in columns (in PCM this is a power of two). The more these characteristics, the better.
The DAC has similar parameters. It is clear that these parameters must be no less than those of the reproduced PCM file.
Note that “from the inside” modern mainstream ADCs and DACs are so-called single-bit, but “from the outside” they look exactly like PCM, and therefore we can talk about bit depth in the usual sense. Details are in the corresponding appendix.
For reference: CD Audio is a digital audio format on music compact discs (CDs), which is a recognized standard for Hi-Fi quality, uses 44.1 kHz and 16 bits.
Mainstream adapters record and play back no more than 16-bit PCM files. The frequency is usually 44.1, 48 kHz, but for low-quality sound, even lower multiples of their derivatives (22.5 kHz, etc.) are used. 48 kHz is standard for computer audio, digital interfaces, DVD audio, and should be preferred (unless the file is a copy of an audio track).
When passing through the digital part of the adapter, the audio may undergo some processing, and the results of intermediate calculations are outside the 16-bit grid. Therefore, it is highly desirable for the DAC/ADC to have a higher bit depth in order to really “convey” 16 bits. Therefore, almost all mainstream audio adapters use 18- and 20-bit DACs/ADCs.
The vast majority of the time, sound adapters are used for playback, and only occasionally and by very few users - for digitization. Therefore, the quality of the DAC is usually higher than that of the ADC (by about 2-10 dB), and the bit depth of the DAC (playback) is always no less than the ADC (recording). For example, often the ADC is 18-bit and the DAC is 20-bit.
In this regard, we note that the quality of mainstream recording adapters is low. Therefore, to record, for example, old vinyl records, it is recommended to use professional so-called 24/96 adapters, where 24 is the bit depth and 96 is the sampling frequency in kHz.
Let us also note that the same manufacturer has ADCs with the same bit depth, but different quality, so a high bit capacity is not a guarantee of quality, but only a necessary condition. There are more objective parameters, such as signal-to-noise ratio. These are discussed below in the codec section, as DAC/ADC are now implemented integrated into the codec.
It is important to note that it is the DACs that mainly determine the purity of the adapter’s sound. At the same time, the relative cost of the DAC/ADC in the adapter is low.
Digital equalizer
Not present on all adapters. The main purpose is to compensate for the shortcomings of acoustics. Affects all output streams equally.
Such advanced software players as WinAmp, WMP, Apollo have built-in digital equalizers (you only need a sufficiently powerful processor). However, a hardware equalizer distorts the sound less. The equalizer is characterized by the number of bands (usually 10), the number of presets and the ability to remember user settings.
Thread processor
As can be seen from the diagram, this is a central block, the services of which are used by other blocks. Therefore, all modern adapters (except perhaps those integrated into the chipset) have hardware acceleration for this block (other blocks can be implemented in software). Note for advanced users that since we are talking exclusively about the Windows + ActiveX combination, this is equivalent to hardware acceleration for the MS DirectSound API.
The block controls digital processing of audio streams. Processing functions include: mixing/splitting streams, adjusting their volume, stereo balance, stream routing, i.e. sending them to additional processing units and receiving processed streams.
Note that hardware digital mixing is performed so quickly that the “gaps” between the blocks of mixed streams are invisible to the ear (unlike software mixing).
An example of routing is directing a violin audio stream from MIDI to a 3D block. The output flow describes a violin circling overhead.
The parameter of the thread processor is the number of simultaneously hardware-accelerated threads, but the number itself is not so important for the choice, the main thing is the very fact of such acceleration. Note that if the application does not have enough hardware threads, then DirectSound provides for adding an unlimited number of software-processed threads (the CPU power would be enough). “Multi-threading” is used, for example, in games, as well as when editing multiple recordings.
MIDI block
Only the information necessary for understanding is provided here. Details are in the corresponding appendix.
Introduction
The MIDI block allows you to synthesize the sound of musical instruments, turning musical notation into audio streams. Usually reproduced MIDI file s in which the score (musical notation) of an orchestral work is recorded.
MIDI files are a thousand times smaller than regular PCM files (for stereo good quality that's 10 KB/min compared to 10 MB/min). This compactness is used in games, for network applications and karaoke (see, for example, karaoke server http://www.fcenter.ru/www.karaoke.ru). In karaoke, the ability to play a MIDI file with a randomly selected tempo and key is valuable.
The MIDI block parameters are the number of hardware and software synthesized instrument voices (polyphony). The MIDI block of modern adapters can have, for example, 64 hardware voices and 512 software voices. In principle, 64-voice polyphony is sufficient, since only selected listeners can distinguish a larger number of voices.
To apply effects (changing the sound of instruments to give greater expressiveness), the MIDI block uses an effect processor. Although effects can be applied to any stream, they first appeared in MIDI. Therefore, the possibilities for applying effects are often given as MIDI parameters.
MIDI Standards
The process of MIDI development is reflected in standards that reflect: the number of instruments, polyphony, a list of effects, how many effects can be applied simultaneously to all instruments and how many to individual ones. Eat international standard GM1 (General MIDI Level 1), its further development GM2 (General MIDI Level 2) and proprietary extensions GM1: GS (General Synth), XG (Extended General). A MIDI block must support one of these standards.
Synthesis by wave table
Currently, the main type of MIDI synthesis is Wave Table synthesis. Its essence is to use samples for synthesis - digitized samples of the sound of real instruments. The samples are collected in a file called an instrument bank. This bank is loaded into memory during synthesis (possibly in parts). Banks, of course, also comply with a certain MIDI standard.
One or more banks of various sizes may also be supplied with the adapter; smaller banks are used when the computer memory size is small.
Banks come in a wide variety of qualities and differ in sample digitization parameters (for example, there are 14-, 16- and 18-bit samples), the completeness of samples (whether the full note is included or just the attack and a piece of a constant level, which is then played back cyclically to obtain a large length) , the presence of sample variations (sharp and soft sound extraction), the fullness of the pitches of the voice (the number of basic notes from which the rest are obtained). An interesting parameter of both the bank and the synthesizer itself is the number of phases of the synthesizer envelope. The higher the number of phases, the more realistic the sound. 4 is low, 5 is medium, 6-8 is high.
Almost all of these parameters are unavailable, but you can focus on the overall size of the bank: the larger it is, the better. The count is in megabytes. For reference: a good quality bank from Yamaha of the XG standard with 3 times compressed 18-bit samples takes up 4 MB.
Downloadable Sounds (DLS) technology
This modern technology eliminates such disadvantages of MIDI standards as fixed sets of instruments and uneven sound on different adapters. Numbers of 128 instruments can be dynamically (by loading into memory, where the name comes from) assigned to any instrument (called “loadable”). An analogue of MIDI banks are DLS files containing samples and “articular” information (how to play). The DLS message contains much more complete attributes about the note, including volume, expression, envelope parameters, etc., which eliminates ambiguity.
The known specifications are DLS Level 1 (DLS1) from 1997 and DLS Level 2 (DLS2) from 1998. MS DirectX 8.0 already supported the DLS2 synthesizer and contained its software implementation. The DLS2 bank was 3.3 M in size with 16-bit samples, 226 melodic and 9 percussion instruments.
A modern adapter should support at least DLS1, with support for DLS appearing to be more important than support for MIDI standards. The PC"99 Audio specification recommends hardware support for DLS.
Soft WT-synthesis
Pure soft synthesis using a wave table does not require many resources: a 300 MHz CPU power with MMX support is quite enough for this. Therefore, parameters such as the number of hardware-accelerated voices have now become irrelevant (and only relatively recently, the number of software(!) voices was included in the name of a sound card, for example, SB PCI-128, SB PCI-1024, etc.).
As already mentioned, in Windows, after installing DirectX, a 6-phase soft synthesizer that supports DLS2 is available. By default, the Roland GM/GS Sound Set bank of 3.3 MB is used.
With sound adapters based primarily on software processing (HSP adapters), more advanced Yamaha S-YXGxxx XG software synthesizers are also included in the package (they can also be found even on the driver disk included with system board; I came across one nice player with video accompaniment Yamaha XGStudio Mixer).
There are also professional soft synthesizers with huge gigabyte banks on the hard drive, for example, Nemesys GigaXXX (see the corresponding application for more details), but the card requires certain support.
You can view CPU load in Windows 2000 by turning on the player and looking at the CPU column in Task Manager (Ctrl+Alt+Del \ Task Manager).
Advice: If you have a decent processor, then do not attach importance to the hardware MIDI capabilities of the adapter and use soft synthesizers.
Compatible with bank format
The MIDI block is compatible with a specific bank format. Thus, cards on the Creative CT5880 audio controller use banks of a specific format from Ensoniq. The most attractive is the DLS compatibility. In this case, you can usually convert other jar formats to DLS.
To use Yamaha S-YXGxxx soft synthesizers, the adapter only requires trivial support for PCM 16-bit 44.1 kHz playback.
However, Nemesys GigaXXX soft synthesizers that use hard drive banks require support from the GSIF fast interface card (see Glossary).
MS DirectMusic
If an audio adapter supports DirectMusic hardware acceleration, then this increases its appeal among gamers. DirectMusic is a DirectX component that allows you to apply dynamic effects to MIDI fragments. DirectMusic supports DLS banks.
Note that hardware acceleration of the DirectMusic API is only possible when using Windows starting from 98SE.
Effect processor
Sound effects first appeared in MIDI and consisted of changing the sound of instruments to give greater expressiveness. The main ones are chorus And reverberation. The total number of effects is quite large, e.g. echo, flanger, sustain, distortion, portamento, and the number of variations of each is in the tens.
With the release of DirectX 8 and DLS2 support, most sound effects can also be applied to PCM streams (an example of a MIDI-specific effect is breath - “aspiration” for wind instruments). The effect processor does all this. Its hardware implementation is included only in adapters with DSP on board (see below). In the absence of DSP, several simple effects are applied using software.
The adapter's hardware ability to apply effects is expressed in the following:
what are these effects?
how many of them can be applied simultaneously to all flows (tools);
How many channels can you apply effects to individually?
Modern adapters support a sufficient number of effects, so as a first approximation, you can ignore these parameters.
3D block
What does a 3D block do?
This block handles support for positioned 3D audio (just 3D audio for short). Ideally, this means that you can hear a point source such as the squeak of a mosquito. Of course, the possibilities of 3D sound are not limitless; for example, it is impossible to reproduce sound coming from below. Different types of acoustics (headphones, 2 and 4 speakers) have different capabilities for reproducing 3D sound. You can read more about this in the corresponding appendix.
The implementation of 3D sound is based on two elements: creating a point source of sound in an infinite space and superimposing environmental effects on the sound stream created by it ( room reverberation, occlusions And obstructions), since games mostly take place not in open areas, but indoors. This could be the echo of footsteps in the hall, the echo of rocks, etc.
Note that applying environmental effects (such as a concert hall, a room with carpets) allows you to “revive” even ordinary stereo audio files.
Currently, adapters use a small number of 3D technologies. The bulk of the API of these technologies is standard (open). Some technologies have small API extensions to this standard part. Standard part The API consists of DS3D, I3DL2, EAX2 components, which are supported (by libraries) MS DirectX. DirectX has a remarkable property: if the adapter has support for the called function (software or hardware), then the execution is transferred to the adapter, otherwise the function is performed using DirectX library functions. The audio component of DirectX (DirectX Audio) has minimal capabilities, operates with low-quality sound (with parameters of 8 bits and 22 kHz) and has a slow software mixer for 3D streams.
Despite the fact that most of the APIs are common to all technologies, they differ in implementation algorithms. Moreover, each technology also has different implementations for headphones, 2, 4 or more speakers (3D sound must be reproduced on at least 2 channels). The possibilities of different acoustic configurations for 3D sound are presented in the following table.
Another disadvantage of headphones is the feeling that the sound source is closer than it actually is.
Technologies used
Here is a list of the main technologies:
technology from Creative (has no special name);
Sensaura 3D from Sensaura;
Q3D (QSound3D) from QSound Labs.
Technology from Creative is used only in adapters produced by the company itself. Sensaura and QSound Labs, on the other hand, do not produce adapters, but license their technology to adapter manufacturers.
Creative's technology is proprietary and little is known about it. However, its EAX Environment Accounting API has become a de facto standard.
Q3D technology from Qsound is based on average listening results and does not require labor-intensive calculations.
Sensaura's technologies are the most advanced and are currently used by most adapter manufacturers. There are several technologies whose APIs are proprietary extensions of standard ones. These are the following components of Sensaura 3D:
MacroFX– reproduction of close sounds, for example, the squeak of a mosquito, the flight of a bullet at the temple.
ZoomFX– reproduction of sounds from non-point sources, for example, from a large locomotive rushing past. It is modeled by a variety of sources, “smeared” over the volume of the body.
Virtual Ear– adjusting the sound to the user’s ear. The fact is that a person’s perception greatly depends on the shape and size of his ears and head, as well as individual hearing (especially for high frequencies). All technologies are designed for some average ear. By adjusting the parameters to suit you, you can get more realistic effects without additional resources. Virtual Ear technology is implemented by an adjustment utility, where individual ear sizes and other parameters are entered.
The component implementing environmental effects is called Sensaura EnvironmentFX. It is API compatible with EAX2 (the de facto standard), but sounds slightly different (which does not mean worse).
Technologies also differ in different schemes for reproducing 3D sound on 4 speakers (we will call them 4-schemes).
| Technology | 4-scheme |
|---|---|
| Creative 5880 | panoramic |
| Creative EMU | accurate |
| Q3D | variable |
| Sensaura 3D | accurate |
panoramic(panning):
3D sound is calculated and sent to the front speakers, and the rear speakers simply duplicate the front ones. This is the most primitive scheme.
variable(transition):
3D sound is calculated and sent to those speakers that are closest to the sound source. The “remaining” columns simply duplicate the “main” ones. This is a slight improvement on the panoramic scheme.
accurate:
3D sound is calculated and sent separately to the front and rear pairs of speakers.
All technologies also allow you to output 3D sound to 6 speakers (by means of panning).
Expert assessments of technologies
Experts from the authoritative (English-language) site 3D SOUND SURGE, as a result of listening to the best new cards, placed the technologies in the following descending order:
implementation on headphones
1.
Q3D
2.
Sensaura 3D
3.
Creative.
2-column implementation
1.
Sensaura 3D
2.
Q3D
3.
Creative.
implementation for multi-channel acoustics (4 or more speakers)
1.
Sensaura 3D
2.
Creative
3.
Q3D.
As you can see, the “leader in the overall standings” is Sensaura 3D.
3D adapter parameters
The adapter's 3D parameters are:
3D technology
Applied 3D technology.
Number of 3D streams means the number of hardware-accelerated point sources in space. More threads allow you to create more diverse games. Additional threads will be created programmatically. The PC"99 Audio specification recommends hardware support for 8 audio sources. The presence of hardware 3D acceleration is determined by the presence of DSP “on board”.
4-scheme
The technology, in principle, also determines the scheme for using 4 columns, but for convenience this parameter is presented separately.
Home cinema
Soft DVD movie players support playback of 6-channel (or even 7-channel) sound on sound adapters. The image is displayed on a large TV, the sound is displayed on a multi-component speaker system. By placing the speakers not only in front of the listener, but also behind them, you can create Surround Sound, for example, the effect of a passing train.
You could say that cinema surround sound is positioned 2D sound, but not interactive. It is achieved by ordinary panning (i.e. changing the volume of the speakers), without changing the phase and frequency, as in 3D sound.
Home theater sound formats
Format audio track movie could be like this:
AC-3
This format is often associated with Dolby Digital (DD 5.1) technology, which is synonymous with AC-3. This is 6-channel audio, and highly compressed. This is currently the most common format.
DTS
6-channel audio, less compressed than AC-3 and therefore higher quality. Requires large resources when decoding. Newer and less common.
DTS-ES
DTS expansion to 7 channels. The first products appeared only in Q4 2001.
In 6-channel audio, the output channels are distributed as follows:
1.2 – to the front speakers
3.4 – to the rear speakers (just like in games)
5.6 – for the central speaker and subwoofer, respectively.
Dialogues (i.e. speech) are displayed on the central column, which increases their intelligibility. The subwoofer can be placed quite arbitrarily due to the insensitivity of the ear to the location of the bass source.
In 7-channel audio, additional the channel is on to the center rear speaker.
6-channel audio is often recorded as 5.1 channel, 7-channel audio is often recorded as 6.1 channel, where “.1” means subwoofer.
Adapter operating modes
There are the following modes of operation of the adapter, or rather its home theater unit:
through(pass-thru)
the track is transmitted without changes through the adapter’s digital output port to the acoustics, where it is decoded.
The adapter must have a digital output port, and the speaker system must have a corresponding digital input port and a decoder of a certain format. The presence of a decoder sharply (about twice) increases the cost of the speaker system. With the emergence of new formats, the possibility of upgrading the speaker system remains questionable.
However, the decision in question is justified when the corresponding acoustic system has already.
digital
the track is unpacked into channels, the channels are decoded into PCM streams and then packed into 3 streams. These streams are supplied to the corresponding digital output ports of the adapter.
The adapter and the speaker system must each have 3 digital ports. However, an expensive decoder is not required in acoustics.
This mode is only available in Creative SB cards starting with the Live! 5.1 and so far only for the AC-3 format.
Compared to the analog mode, higher quality sound transmission is produced.
analog
the track is completely decompressed, decoded and converted into the appropriate number (6 or 7) of analog output channels, which are fed to the adapter ports.
The adapter and the speaker must have the appropriate number of analog ports.
This is the cheapest option. All work can be done by software DVD player and therefore format support depends only on the player used. For example, the DTS format is supported by InterVideo's WinDVD player starting with v3.0.
In all cases downmixing possible for 2- and 4-channel acoustics and headphones. However, this leads to distortions, since transformations are made with compressed sound who “doesn’t like” this.
From the adapter side it is possible optional hardware acceleration on processing (unpacking, decoding) for specific formats.
The theatrical adapter package sometimes includes DVD player software(it is usually included with the DVD drive) and/or remote control(remote control). The latter is more typical for execution with a port module.
High-quality implementation of a home theater depends not so much on acoustics (from $200), a large wide-screen TV (about $2000), but on the presence of a separate room with an area of 20 meters. The room should not be crowded, and the walls should be covered with carpets to absorb sound.
Upmixing
This is the layout of “small-channel” sound into a larger number of channels in order to harness the full power of a multi-channel speaker system. For example, upmixing mono and stereo sound to 4 or 6 speakers, or outputting 4-channel game audio to 6 speakers.
Examples of upmixing technologies are:
CMSS(Creative Multi Speaker Surround)
Upmixing stereo sound into 4 or 6 channels. Implemented through the proprietary Creative PlayCenter player. Allows you to pan any mono or stereo sound in any azimuth on 4 and 6 speakers.
QMSS(QSound Multi-Speaker System)
Upmixing stereo sound into 4 or 6 channels.
MP3 block is optional. Performs hardware acceleration of decoding of compressed MP3 format when playing files. Along with MP3, less common formats can also be supported, for example, WMA, OGG. The block allows you to reduce the processor load by several percent, which is not significant. Only some sound cards have such a block.
Digital ports
Digital transmission is practically unaffected by interference and interference. In the case of a multi-channel speaker system, the cable becomes thinner and the connectors become more compact.
The digital input port (external) allows the use of higher quality external ADCs.
SPDIF has been used as such ports for a long time. It comes with electrical or optical connectors. In multimedia computer acoustics, mainly electric SPDIF ports are used. Therefore, the same format is used in the vast majority of cards. However, in home music equipment there are also optical SPDIF ports. To connect to such equipment, you must select cards with an optical port. Sound cards with an additional port module have a particularly large set of ports.
Input (internal) port allows you to connect internal devices, typically DVD drives.
Note that in the specification PC"99 Audio and AC"97 The universal high-speed two-way IEEE 1394 interface is also recommended (it is also convenient for exchange with digital photo and film cameras).
Some cards also use I2S as an internal bus, complementing the AC-Link bus (see below).
For more information about digital interfaces, see the corresponding appendix.
IEEE 1394
Optional digital bi-directional external port. It has a throughput of up to 400 Mbit/s, PnP, hot pluggability.
CD SPDIF (Digital Audio)
Optional digital internal input port, 2-pin, MPC format. Serves for digital connection CD drive and playback of CD Audio music CDs (such outputs for drives are also optional). The data format is the same as SPDIF, but can use 5V instead of 1V.
SPDIF out
Optional digital output external port. Used for digital communication with an acoustic system (including multi-component ones). The difference compared to analog transmission is noticeable only on a good speaker system. A digital port makes the adapter more expensive by $10 or more.
One port is enough to transmit a stereo or packaged 6-channel stream.
Electrical connector format: stereo mini-phone jack for a single port and G9 (digital DIN) for 3 ports.
I2S input
Optional internal port in MPC format (3 wires used). Used in previous cards from Creative to connect the audio output of an MPEG2 decoder. Better quality than SPDIF.
Ports are anachronisms
MIDI port
used to connect a MIDI musical instrument (usually a 4-6 octave keyboard) via an additional adapter costing about $20.
game port
optional, used to connect game controllers such as joysticks (both analog and digital thanks to the use of different groups of contacts).
Both ports are usually combined in a gold-colored DB-15 connector.
PCM File I/O Block
The PCM file input/output block (Wave in/out) uses the PCI interface to exchange with “ outside world” sound files in PCM format, and can do this simultaneously for several files.
PCI interface
Current version is 2.2. But 2.1 is also used, which is not much different.
More important is the presence of the adapter in PCI Bus Master mode - a mode for exchanging data via the PCI bus with minimal participation of the processor. This makes it easier for software to implement many audio functions, for example, sound synthesis. Relevant for adapters that do not have certain hardware units, for example, a MIDI synthesizer. PCI Bus Master is currently implemented in almost all adapters.
| Hardware composition of AC"97 sound adapter |
Let's see how the sound adapter architecture is implemented in the Intel AC"97 specification, where AC is an abbreviation for Audio Codec, 97 – the year the first version of the “spec” was adopted – 1997. The current version is 2.2 from 2000.
Close-up
In the diagram, asterisks (*) indicate optional components. The internal codec ports are located on the left side. As you can see, there are the following hardware components:
AC"97 audio codec (Analog codec). This is the analog part of the adapter. Contains an analog mixer, ADCs, DACs, analog ports. The hardware is made in the form of one or two chips.
AC"97 controller (Digital controller). Performs digital processing of audio streams, including mixing, MIDI synthesis, 3D, effects. Performed as a single chip. Tests show that the controller can influence the color of the sound.
The AC-Link bus connects the codec to the controller. A bus branch to a CNR slot is used in integrated solutions.
The AC"97 specification specifically stipulates that the codec and controller are implemented as separate chips. The codec is placed as close as possible to the output connectors, which reduces the noise level.
Harness, operational amplifier
A “strap” is placed between the codec ports and the adapter connectors. This is, firstly, an operational amplifier chip (such as Philips TDA1308), which protects the front linear output from overloads. This allows you to connect headphones with a much lower impedance (standard 32 Ohms) compared to the 1 kOhm impedance for active speakers.
Note that the other outputs (rear, theater) are usually not protected and therefore it is not recommended to connect headphones to them (speakers only).
Input ports can also be protected (by transistors, capacitors, chips, etc.). The importance of this harness is beyond doubt.
A harmful anachronism is the presence of a power amplifier in some cheap adapters (to use cheap passive speakers). Not only is such an amplifier of low quality, it is also an additional source of noise and heating. Therefore, it is recommended to disable it (via a jumper).
Block diagram
In the diagram, the numbers show the connection of the ADC/DAC with the corresponding ports. If the names of internal ports are enclosed in brackets (CD Audio, etc.), this means they are optional. However, all the additional inputs and outputs shown at the top are present on the codec, and it is up to the manufacturer to install the corresponding ports in the adapter or not. 3 stereo line inputs are provided for CD Audio, VIDEO and AUX. 2 mono line inputs are provided for TAD and PC BEEP (see above).
The chip name is an acronym ( co dirating / Dec oding; in English literature - codec, co der / dec oder), derived from the name of its main components - ADC/DAC. The latter perform the transformation sound signal from analogue to digital form (encoding) and back (decoding).
Microphone input has programmable gain, as well as a switchable mode for increasing sensitivity by 20 dB (20 dB Boost). The latter is necessary for microphones dynamic type(unlike electret). The optional second microphone input allows you to simultaneously use one microphone in the headset, used for speech, and a second high-quality desktop one.
Headphone output has an amplifier impedance of 32 Ohms, so you need to choose the same headphones.
SPDIF output. The specification requires the transmission of audio streams with a frequency of 48 kHz, which guarantees compatibility with household appliances. If a PCM file with a sampling frequency other than 48 is output via the interface, then it is transparently converted to 48 kHz before being transmitted via the AC-link bus. Support for other frequencies (i.e. without conversion, “bit exact”) is optional. Note that codecs with an SPDIF port are still rare.
Mixer may have analog tone control. However, a software or hardware digital equalizer allows you to make more subtle adjustments.
ADC/DAC have a bit depth of 16, 18 or 20 (not higher), and the limitation is related to the AC-link bus, see below. However, recording (via the PCI interface) is done to no more than 16-bit PCM files.
The codec is full duplex, i.e. allows you to simultaneously record and playback, and in different modes.
Codec Quality Options
The bit depth of a codec refers to the bit depth of the ADC and DAC included in it. As already mentioned, codec bit depths are far from complete parameters of their audio quality. Thus, the Cirrus Logic CS-4294-KQ and CS-4294-JQ codecs from the same company have the same bit depth, but different quality.
Let us note such a remarkable fact that the parameters of 2- and 4-channel codecs from the same manufacturer and from the same line are the same.
Since DACs are used much more often than ADCs, we will limit ourselves only to the DAC parameters. This corresponds to the parameters of the D-A codec path (from digital to analog).
Let's look at the basic audio quality parameters (they apply to any audio path; for more details on defining paths and their parameters, see the corresponding appendix). These parameters are given by codec manufacturers in the specifications, and for the maximum sampling frequency - 48 kHz for AC"97 codecs (since in this case the indicators are the highest).
DR (Dynamic Range), dynamic range.
This is the ratio of the strongest signal to noise in the presence of the latter (and noise is inherently independent of the signal). Measured in decibels. The larger the range, the better. Good values are considered to be 85 dB and above.
Note that this number should be considered only as an upper limit, a guideline. The fact is that, firstly, the measurement is made for a pure sinusoid. For a real signal, the parameter can be significantly smaller. Secondly, the measurement is made for the maximum signal. For a normal signal of average volume the ratio will also be smaller.
Unfortunately, most codec manufacturers do not provide DR, but an even more inflated parameter S/N (aka SNR) - signal-to-noise ratio, while noise is measured in the absence of a signal, i.e. in even more artificial conditions.
FR (Frequency Range), frequency range uniformity of reproduction
with the indicated spreads. This is the frequency range where the frequency response curve does not leave the specified scatter limits. The larger this range is for given scatter boundaries, or the narrower these boundaries are for a given range, the less the signal frequencies are distorted after passing through the codec. For good codecs this is 20-20"000 Hz at ±0.5 dB.
Note that more interesting is the spread boundary for the entire audio range of 20-20,000 Hz (as is customary in test utilities).
Obviously, FR is a kind of “squeeze” from the frequency response itself and does not guarantee the reliability of reproduction, which is determined by the shape of the frequency response itself, and not just by the dispersion values.
THD+N (Total Harmonic Distortion plus Noise), total harmonic distortion plus noise.
THD expresses the level of distortion generated by the codec itself (which is proportional to the signal). In THD+N, noise (by definition, independent of the signal) is also included here. Both coefficients are expressed as percentages. The lower the coefficient, the better. A good THD+N value is 0.02.
Sometimes THD or THD+N are expressed in decibels. The relationship between percentages and decibels is given by the following table (which is easy to continue due to its “periodicity”).
| THD+N% | THD+N dB |
|---|---|
| 1 | -40 |
| 0.9 | -40.9151 |
| 0.8 | -41.9382 |
| 0.7 | -43.098 |
| 0.6 | -44.437 |
| 0.5 | -46.0206 |
| 0.4 | -47.9588 |
| 0.3 | -50.4576 |
| 0.2 | -53.9794 |
| 0.1 | -60 |
| 0.09 | -60.9151 |
| 0.08 | -61.9382 |
| 0.07 | -63.098 |
| 0.06 | -64.437 |
| 0.05 | -66.0206 |
| 0.04 | -67.9588 |
| 0.03 | -70.4576 |
| 0.02 | -73.9794 |
| 0.01 | -80 |
| 0.009 | -80.9151 |
| 0.008 | -81.9382 |
| 0.007 | -83.098 |
| 0.006 | -84.437 |
| 0.005 | -86.0206 |
| 0.004 | -87.9588 |
| 0.003 | -90.4576 |
| 0.002 | -93.9794 |
| 0.001 | -100 |
2- and 4-channel codecs
Although the specification states one codec, there is actually no 6-channel codec. When 6-channel adapter in it 2 codecs are used, one of which is 4-channel, and the second is 2- or 4-channel (both the AC"97 controller and the AC-link bus allow this).
Note that the use of 2 codecs is redundant, since both, for example, have a line input, while only one is used. Instead of a second codec, a decoder is sufficient. But its mixer is also redundant, since there is no need to mix anything, and the volume can be controlled on the digital section of the path. Therefore, some adapters use a DAC (usually from Philips) instead of a second codec, connecting it not via AC-link, but via a simpler I2S bus.
Current situation with codecs
Currently, all sound cards (of those reviewed), from the cheapest to the most expensive, use codecs of approximately the same quality. Hence the important
advice: if the card will be used to play music CDs and audio files (compressed files such as MP3, MIDI, WAV files), then the cheapest cards (with the required number of outputs) are sufficient.
In the case of expensive cards, you pay for an advanced one digital processing sound (3D sound, MIDI creation compositions, etc.), as well as for digital ports.
PC-D-A path parameters
We will limit ourselves to the more relevant case of playback (but everything said below can be repeated for recording). Codec parameters describe the D-A path of the codec, which is only part of the overall end-to-end PC-D-A (digital playback to line-out) path. The quality of the latter is more important and depends on the quality of the wiring printed circuit board, and from power filters, and from wiring elements such as capacitors and an operational amplifier.
For reference: the basic requirements for the PC-D-A path according to MS PC"99 are
Alexey Lukin.
AC-link bus
The AC-link bus carries out bidirectional data transfer between the AC"97 audio controller and the AC"97 audio codec. Allows you to work with 12 data streams (incoming and outgoing) with a bit depth of up to 20 bits and a sampling frequency of 48 kHz. In most implementations, the bus frequency is fixed at 48 kHz. This means that if the audio file also has a sampling rate of 48 kHz, then one sample is transmitted for each bus clock cycle. If the sampling frequency is different, for example, 44.1 kHz for CD audio tracks, then preliminary resampling is performed in the audio controller (by the SRC block, see below). Theoretically, this introduces additional errors. Version AC"97 2.2 provides an optional mode for transmitting streams with a frequency of 44.1 kHz and any other without conversion. That is, the bus frequency changes depending on the stream frequency parameter. This option should obviously be supported by both the audio controller and audio codec .
In AC"97 2.0, stream transmission with a sampling frequency of 96 kHz was optionally added.
AC"97 controller
Here SRC (Sampling Rate Converter) is a frequency converter for PCM streams.
It is used by the stream processor when mixing streams at different frequencies, and when preparing data for the AC-Link and SPDIF buses, which both typically operate at 48 kHz. Standard values for PCM files are 48, 44.1, 32.0, 22.05, 16.0, 11.025, 8.0 kHz.
The controller has the ability to return the digital stream to the main memory for subsequent redirection to an external digital bus ( USB type). This is called “Digital loopback”; it also operates at 48 kHz.
The diagram omits the already interesting DOS audio support block, as well as the MPU-401 and gaming ports. PC"99 Audio recommends using USB ports instead.
DirectMusic API and support DLS.
Hardware accelerated decoding for theatrical audio formats.
Upmixing.
Also optional I2S input digital port. Unlike SPDIF, this interface suffers virtually no jitter. Designed for connection with DVD drives.
The controller has inside itself digital amplifier. Like any transistor amplifier, it is sensitive to overloads and produces high distortion in this mode. For some controllers, this “red zone” begins with fairly low Wave fader settings on the Windows Universal Mixer. Therefore, it is recommended not to touch this slider after installing the adapter drivers (the gain is then equal to unity), but to use only the general volume control.
Ideally, within the operating gain range, the controller should not introduce distortion during recording or playback. However, in practice this is not the case. For example, M. Lyadov’s comparison of three cards with different controllers and the same codecs (Genius Sound Maker 5.1, Philips Acoustic Edge 5.1, Creative SB Live! 5.1 cards; SigmaTel STAC9708 codec) revealed that the controller in Creative SB Live! 5.1 creates large non-linear distortions at the front output.
DSP and HSP audio controllers
Hardware acceleration of effects, acceleration of 3D sound, implementation of a digital equalizer, etc. entirely depends on whether the audio DSP (audio digital signal processor) is built into the controller.
If there is no DSP, the audio controller is called HSP ( h ost-based s ignal p rocessing), i.e. used instead of DSP CPU. This, of course, makes the audio controller cheaper, but it presupposes a certain level of CPU performance. It is known that an adapter on an HSP controller can take up to 20% of processor resources. With DSP, delay-free audio processing is guaranteed, even with a Pentium-166MMX processor.
Some audio DSPs are even made reprogrammable to support new technologies and improvements (and bug fixes).
Advice: Consider the presence of DSP in the controller as a significant plus.
According to the AC"97 2.2 specification, the optional SPDIF output port must be placed in the codec (only SigmaTel announced such codecs). But in practice, the port is built into the audio controller (and often the SPDIF input port is also placed there).
Some controllers also have I2S ports, which makes it easier to connect additional DACs.
Ports and their combination
The practice of combining ports entails tangible inconveniences, forcing you to engage in “plugging”. This is due to the fact that an obsolete and bulky MIDI/game port is still placed on the sound card bracket. This port has nothing to do with sound, and takes up a lot of space (remember that PC"99 recommends using devices connected via USB).
Combining headphone output
In the case when a 6-channel sound card has output connectors in mini-jack format, a digital output and a game port, there is a catastrophic lack of space on it. This forces you to combine a digital port and (at least) one of the analog ones. One of the solutions found in the latest cards is the use of a compact G9 connector for the output ports (together with an adapter splitter cable).
Extra contacts do not add reliability, and the best solution would be the use of separate ports due to complete removal MIDI/game port.
When is an adapter not needed?
Listening to audio CD.
You can use the built-in headphone output on the front of the CD drive or the line-out at the rear. For IDE drives, it is possible to digitally read tracks and transfer them to USB speakers.In the last two cases, the connectors are easily accessible compared to the location on the rear wall of the case system unit. Such modules increase the cost of the card by about $50 (internal module) and $100-150 ( external module). Note that a separate microphone preamplifier costs about $60.
Other versions will be collectively referred to as integrated (into the motherboard). Their goal is to reduce the cost of the adapter.
The adapter is located on the system board and has a separate controller chip
Here the ports, codecs and controller are located on the motherboard, and the controller is made in the form of a separate chip.
Fundamentally, this design is no different from a sound card. The PCI slot is freed. Since the controller makes up the bulk of the cost of the adapter, low-cost controllers are used to achieve cost attractiveness.
Integrated adapter with controller in the system chipset
Audio controller is built into the South Bridge system chipset(in the case of a classic dual-bridge chipset architecture).
Almost all modern chipsets are like this. The controller, however, is an HSP and may not even have a hardware digital mixer. Usually the codec is 2-channel.
In the now common case, the codec is located on the system board. To upgrade to 6 channels, it is proposed to purchase a compact card for the AMR, CNR, ACR form factor slot, which houses additional codecs and ports. Such cards are just beginning to become widespread.
Thus, motherboard manufacturers offer the following upgrade steps.
- From the very beginning you get integrated 2-channel sound, of good quality and for very little money.
- If you need 6-channel sound, then you need to buy a CNR or ACR card.
- If you need effects, hardware support for 3D, etc., then instead of step 2, purchase a separate PCI card, and the built-in sound is disabled.
The sound card is an integral part modern computer, used to reproduce sound. In this chapter we will talk about the development and different types sound cards.
Sound card connectors
The sound card connectors are quite miniature and all look the same – small and round. To ensure you don't get them mixed up, the PC99 Design Guide has been adopted, which clearly specifies the color for each connector:
salad - linear output. You can connect a speaker system, headphones, or a stereo amplifier to this connector, which allows you to boost the signal if you are using a powerful speaker system. Some sound cards have two outputs - one for the left channel and one for the right;
blue – linear input. Used to record audio that comes from an external source (such as music center) to the hard drive ;
pink – input for a microphone or monophonic signal (which is a microphone signal). If the sound card does not have a microphone, you can connect it to the line input;
yellow – game port, used to connect a joystick or MIDI device (synthesizer). Sometimes a 15-pin D-shaped connector is used instead of this connector.
Built-in sound cards (most of them) do not have a game port. However, for it to appear, it is not at all necessary to change the sound card - just connect the game port via USB ( this device connected separately). Yes, modern joysticks and synthesizers can be connected via USB, so there is no need for a gaming connector as such.
When using a separate sound card (in the form of an expansion card), it is advisable to connect it to the CD/DVD drive using a special cable (Fig. 12.1) - otherwise you will not be able to play AudioCD. True, modern operating systems can do without this cable, but you should know about its existence. In Fig. Figure 12.2 shows a separate sound card. The port for connecting the CD is located on the inside of the board itself, that is, the cable will run inside the case and not outside.
//-- Rice. 12.1. Cable for connecting the sound card to CD-ROM --// 
//-- Rice. 12.2. Separate sound card --//
In Fig. 12.3 shows a typical CD-ROM drive with a description of the rear panel connectors. One of them is used to connect the sound card.
Rice. 12.3. CD-ROM with a description of the rear panel connectors
In Fig. Figure 12.4 shows a sound card connected to a computer via USB. To make this sound card work, you need to do the following:
remove the drivers for the existing sound card (this can be done in Device Manager) - as a rule, this will be an internal sound card;
Rice. 12.4. Sound card(USB)
Disable the internal sound card in SETUP (see below);
boot Windows;
connect the USB sound card and install drivers if necessary.
Connecting a separate sound card
To connect a separate sound card, you need to do the following:
press Win + Break (or execute the menu command: Start, Settings, Control Panel, System) to open the System Properties window;
in the window that appears, go to the Hardware tab and click the Device Manager button (Fig. 12.5);
//-- Rice. 12.5. Properties of the system --//
in the Device Manager window (Fig. 12.6), expand the Sound, video and gaming devices. Remove the sound card (in the picture – VIA AC’97 Enhanced Audio Controller);
restart your computer;
When booting, enter SETUP (usually the DEL key). Go to the Advanced section, then to Integrated Peripherals (sometimes there is an Integrated Peripherals section right in the root menu - it all depends on the BIOS version);
Disable the built-in sound card. Typically you'll want to set the Onboard AC97 Audio Controller option to Disabled;
save changes (F10) and turn off the computer;
unplug the power cable from the computer, remove the system unit cover and install a separate sound card in PCI slot or PCI-E (depending on the sound card);
turn on the computer's power, boot Windows and install sound card drivers, if necessary (what if Windows knows your sound card?);
restart your computer. That's it, the sound card is ready for use.
//-- Rice. 12.6. Device Manager --//
Line input is an analog input of an acoustic signal that does not require additional processing. This connector on a personal computer is intended for connecting acoustic equipment, such as CD and DVD players, radios, etc.
Purpose
Line input (stereo) is standard interface, designed to receive from various audio devices. This connector is used to connect devices that have a device. That is, this is the input of a device whose input signal level is proportional to the output level of the device with which the connection is made. These connectors are used to connect a guitar, radio, CD player, etc. to the sound carriage, the output signal of which does not require additional processing.
Connector design
On personal computers, the linear input is represented by a blue jack (female) connector. This socket is located on the panel at the back of the system unit. Most laptops do not have a line-in input, but do have jacks for connecting a microphone and headphones. They are usually located on the front or left of the laptop.
Setting up your computer's line input
If you connect an external device to your PC to record audio, you will need to specific setting. This procedure can be performed using standard utilities found in every operating system, or with the help of a professional software. First of all, check the drivers for your sound card. It often happens that they are installed incorrectly. To do this, go to Device Manager and find your sound card there. If it functions normally, then the drivers are installed correctly. If not, you will need to perform the installation again. Insert the plug of the connected device into the line input (blue). After that, through the “Toolbar”, go to “Sounds and Audio Devices” and select the “Audio” tab. Next, in the recording section, you should find the connected device and open its volume menu. Then the mixers for configuration will be highlighted. Set them according to your preference. That's it, you can start capturing sound.
Why do you need a multi-channel linear input?
This question can often be heard from people who have little interest in music. However, any music lover knows: thanks to this input, you can connect speakers to a DVD player, home theater or personal computer in multi-channel mode(standard 5.1 or 7.1), which will provide high-quality surround sound. The device to which the speakers are connected must have a built-in decoder for multi-channel audio and, accordingly, the necessary connectors.
Conclusion
To summarize, we note that the system of connectors corresponding to the linear input and output allows you to create a whole network of different acoustic devices that will work together. They are able to complement each other and enhance acoustic signals.
Dear newcomer to the information business, your first acquaintance with the device personal computer took place. Now you know what components the personal computer (PC) system unit includes and what external devices are connected to it. How are PC components and external devices physically connected? For this purpose they are used
Implemented in PC architecture backbone-modular principle of computer construction. The modular principle allows you to change the computer configuration and upgrade it. Installing additional expansion cards provides this opportunity. In addition to installing the sound cards, video cards, internal modems, etc. required by the user, it is possible to connect additional non-standard external devices (Web cameras, digital cameras, etc.).
The modular organization is based on the backbone (bus) principle of information exchange between devices. The backbone design principle is that all devices are controlled and exchange information through one common backbone (computer system bus), which includes three buses. One bus is for data exchange, another is for transmitting addresses, and the third is for control.
Schematically, a PC can be represented as follows:
Computer system bus (highway) can be simplified as a set of cables and electrical conductors on the PC motherboard.
The motherboard with the slots and buses used can be imagined:
North Bridge is a system controller. It is responsible for exchanging information with the processor, RAM and a video adapter (graphics controller).
South Bridge– This is a functional controller (input/output controller). Hard drives are connected to it through the appropriate connectors, optical drives, audio system, network card, keyboard, mouse, etc.
In reality, inside the PC system unit, the components are connected using slots (special connectors), cables, loops (flat cables), bundles of wires that end in connectors:
The motherboard itself looks like this:
External devices are connected to connectors and sockets located on the outside of the PC system unit (back and front sides) or laptop (sides or back):
The response connectors look like this:
Power cables(220 V)
power unit ASUS laptop
PS/2 plugs for connecting a keyboard (purple) and mouse (green).
LPT cable.The LPT (parallel port) port was mainly used to connect printers. Modern printer models provide connection to a USB port.
COM port (serial port) is mainly used to connect modems.
USB cable. The USB port was developed later than the above ports. Most peripheral devices are connected via the USB port: modems, printers, scanners, flash drives, portable hard drives, digital cameras, etc.
VGA cable. Used to connect a monitor.
Cable for connecting to the Internet (Intranet) ( RJ-45 connector)
Slot connector types, used on motherboard(ISA or EISA, PCI, AGP):
Slots with PCI connector (female):
and sound card withPCI connector (male):
PCI connectors used to connect an internal modem, sound card, network card, SCSI disk controller.
ISA slots (Mother). The ISA interface is deprecated. In modern PCs, it is usually absent.
PCISA FlipPOST diagnostic board with connectors PCI and ISA (male) PCZWiz company
Slot with AGP connector(dad is at the top, mom is at the bottom).
The AGP interface is designed to connect a video adapter to a separate bus, with output directly to system memory.
UDMA connector slot(father is on the right, mother is on the left).
Hard drives and more are connected to it.
It should be noted that each slot type has its own color. By opening access to the motherboard, you can easily find your way around. But it’s better that you don’t need it. But the cables that connect external devices to the PC “you need to know by sight.” Remember that the mother and father of the connector must be the same color. Always remember to match the colors of the male and female connectors or know what the colors of the connectors on the PC (laptop) case indicate.
Take, for example, a standard sound card:
Linear audio output to the speaker is always green.
The line input for audio amplification is always blue.
The microphone connector is always pink.
Match them with plugs:
The color design of the connectors will help you. True, colors among PC manufacturers are not unified. For example, some may have a purple keyboard connector, while others may have a red or gray one. Therefore, pay attention to the special symbols that mark the connectors. In this case, it will not be difficult for you to find out :
The interface cables for external devices are unique. You cannot insert it into another connector on your PC (the design and number of sockets are different). All this will help you move your PC (laptop) from place to place without prompting from anyone. You will be able to connect devices and cables to your PC correctly. I hope that the material presented will help you with this.
Now you know what it is PC ports, PC slots, PC connectors, PC cables. More detailed information about connectors and their use with excellent color illustrations can be obtained
If you are a beginner, regardless of age, please leave your comment. And if you are a pensioner, then mark this. After all, we are colleagues! We must help each other!
Connectors and their names.
Firstly. The connector consists of two parts. The nest is Where stick. The plug is what What stick.
monophonic jack, or TS. The sleeve to the black ring is ground, the tip is a signal
stereo jack, or TRS. The sleeve to the first ring is ground, the contact between the black rings (Ring) is the right channel or negative phase or power, the tip is the left channel or a signal with a positive phase
Mini-jack- like headphones for a player. Looks like a large jack, only smaller - 3.5 millimeters (1/8). Recently, a tiny 2.5 millimeter jack is sometimes used in mobile phones and players, but it is not called a mini-jack.
minijack, familiar to you from players
Tulip, or RCA jack- found on professional sound cards (for linear inputs and outputs), as well as on household VCRs and old VHS camcorders. Since there are usually two such connectors (left and right channels), the right channel connector is black if there are black and red connectors; and the right channel is red if there are white and red connectors. Initially, the “tulip” was developed back in the forties of the last century to connect radios to gramophones. The tulip is often used as a connector for the S/PDIF digital interface.
common tulips
also a tulip, but pretends to be a knight
XLR (less commonly XLR-3, “canon” or “canon”, more correctly Cannon - not to be confused with Canon)- usually found on microphones. Such a massive connector with three pins and a latch.
Every normal microphone has an XLR connector, and a cable is connected to it. The cable can end with either another XLR or an ordinary jack. If you don't have a mixer where you can plug in an XLR, you'll need a jack cable. You can connect the jack to a household sound card using an adapter from a regular jack to a mini-jack. Since the connectors for the line output, as well as the line input and microphone input are located on the audio speaker in close proximity to each other, often this adapter simply does not plug in next to the mini-jack going to the speakers. Therefore, you need to buy this adapter not in an all-metal shell, but in a plastic one. It can be cut with a knife on one side and wrapped with electrical tape. Then it will stick in without difficulty.
Colors of "household" connectors
You may have noticed that often the plugs and sockets have certain colors - for example, the microphone plug and the socket for it are pink, and the headphones are light green. This is not a whim of the manufacturers, but their adherence to the PC 99 standard. Below I present these colors and their descriptions.
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