What is a tad on a sound card. Professional sound cards. PC-D-A Path Parameters
The line input is an analog input for 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, radio tape recorders, etc.
Appointment
Line-in (stereo) is a standard interface for receiving from various audio devices. This connector is used to connect devices that are provided That is, this is the input of a device whose input signal level is proportional to the output level of the device with which it is connected. Use these jacks to connect to a sound carriage a guitar, radio, CD player, etc., the output of which does not require additional processing.
Connector design
On personal computers, the line input is represented by a blue jack (female) connector. This slot is located on the panel at the back of the system unit. Most laptops do not have a line-in, but there are microphone and headphone jacks. They are usually located on the front or left of the laptop.
Computer line-in setting
If you connect an external device to your PC to overwrite sound, this requires some setup. This procedure can be done with standard utilitiesthat are in each operating system, or using professional software. First of all, check the drivers for your sound card. It often happens that they are not installed correctly. To do this, go to the "Device Manager" and find your sound card there. If it is functioning properly, then the drivers are installed correctly. If not, then you need to re-install. Insert the plug of the connected device into the line-in (blue). After that, through the "Toolbar" go to "Sounds and Audio Devices" and select the "Audio" tab. Next, in the recording section, find the connected device and open its volume menu. Then the mixers for setting will be highlighted. Set them according to your preference. That's it, you can start capturing sound.
What is a multichannel line-in for?
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 (5.1 or 7.1 standard), which will provide high-quality surround sound. The device to which the speakers are connected must have a built-in decoder for multichannel sound and, accordingly, the necessary connectors.
Conclusion
To summarize, we note that the system of connectors corresponding to the line in and out, allows you to create a whole network of different acoustic devices that will work together. They are able to complement each other and amplify acoustic signals.
This is a device for input / output of audio (analog signal) to / from a computer.
Audio card - inputs, outputs, connectors
Distinguish connectors and inputs.
Connector- this is a socket located on the sound card itself, or brought out on a small piece of cable from the card. Most often, one connector provides connection to one input or output sound card. But, number of connectors sound card does not always match with the number of inputs and / or outputsthat are visible in the operating system or in the settings of the music editor.
This is because the physical input on the sound card can be switched between the line-in and microphone-in jacks. You will not be able to write both the microphone and the line-in at the same time, but only one by one.
The figure shows one mono channel of an A / D converter sound card. After the ADC, the digitized signal goes to the computer, to the music editor.
Connector types:
RCA (tulip)
TRS 1/4 "(big jack) 
TRS 1/8 "(mini jack) 
XLR 
Sometimes one connector provides a connection to one of several inputs. For example, there is a combo XLR (professional mic) and TRS (1/4 "- 1/4" jack) jack.
Combo XRL-TRS jack
This jack is used to save space and allows you to plug in either a microphone, synthesizer, or guitar.
Audio card - analog inputs
Line-in - this is an input that allows you to enter signals into the sound card standard level (about 250 millivolts or 0.25 volts). These are the outputs of a mixer, synthesizer, tape recorder, CD-player, etc.
Microphone input is a sound card input equipped with a microphone preamplifier.
Microphones can be of two types: dynamicand condenser (electret).
At the output of the microphone, the signal is very weak, so we will not hear the microphone connected to the line-in. Some microphones sometimes have a built-in preamp.
No power is needed for dynamic microphones. Amateur electret microphones are often powered by a battery located in the microphone itself.
Professional condenser microphones require so-called 48 V phantom power supply.
Phantom Power- this is the supply of supply voltage through the same wires through which the sound goes.
Therefore, if you are going to use a condenser microphone, there are 2 options:
1. Purchase an audio card with a microphone input (with a built-in microphone preamplifier), equipped with phantom power.
2. If the card has only line outputs, you will have to additionally purchase a microphone preamplifier that provides phantom power.
Instrument input (high resistanceor Hi-Z) - This is a high impedance input for connecting a guitar or bass. The increased (compared to the line input) impedance allows it to be more fully matched with guitar pickups (uniformity of frequency response).
The number of inputs required is determined based on how many audio sources you want to record simultaneously. You can record one instrument at a time, or you can simultaneously record several parts: a vocal group or an acoustic drum kit, in which different drums and cymbals are recorded through several separate microphones.
Audio card - analog outputs
Line outis an output that allows you to output standard level signals (250 millivolts) from the sound card.
Usually a sound card has 2 mono line outputs or one stereo. But if you do 5.1 / 7.1 - phonogram, then the number of required line outputs can increase accordingly.
Monitor output - This is a line-out to the musician's headphones when recording, or to the amplifier of concert monitors on stage during a performance.
If you are going to work with a sound card at a concert, then the main line outputs are connected to the acoustics going into the hall. It is desirable to have additional monitor outputs for sound output on stage so that the musicians can hear what they are playing. If you are using an external mixing console, then these outputs can be in it. Then in the audio card - they are superfluous.
INSERT - these are the connection points for hardware external sound processing modules (reverb, vocal and guitar processors, etc.). is a combination of the sound card output to the external processing module and the input to the audio card from the processing module (already processed signal).
Analog inputs and outputs can be balanced (symmetrical) and not balanced (not symmetrical).Balanced inputs are less susceptible to noise than unbalanced ones, so there is less noise when using balanced inputs).
Word clock are inputs and outputs for synchronizing multiple sound cards. If you are planning to work with multichannel recording, then perhaps You you will have to use 2 or more at the same time audio cards (maybe the same, or maybe different). Synchronization possible with dedicated inputs and outputsWord clock (bayonet type) or by SPDIF.
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word Clock cable and connector (BNC, bayonet).
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Basic parameters of sound adapters
Functional diagram of the sound adapter
When considering architecture, you do not need to distinguish between software and hardware of the adapter and consider them as a whole.
Block diagram
Let's consider each block in detail.
Analog ports
Turtle Beach Santa Cruz map is shown as an example. Note that a port is a logical concept, and a connector is a physical one, since one connector can combine several ports (this is practiced due to the lack of space on the card's bracket; modes are switched through the management utility).
External connectors - cards brought out to the bracket (bracket) or the outer edge of the motherboard in the case of an integrated adapter. Made in the format stereo mini jack.
Internal connectors - located on the map itself. They are usually made in MPC format.
There is an AC "97 specification that indicates 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.
Mandatory. Designed to play or record an analog signal (stereo or mono) from the line-out of other, usually external analog devices, for example, an audio player, radio, VCR, etc.
Colored blue.
Besides it, two additional internal line inputs are required on the sound card (and in the case of an integrated adapter, optional).
Stereo CD Audio input.
Mandatory. 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 modern CD / DVD drives with IDE interface under Windows OS (starting from version 98) can digitally read CD-DA audio tracks, 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 line input (AUX-In).
Mandatory. It transmits analog sound from an FM or TV tuner card, or other internal devicessuch as a second CD drive, DVD drive, or MPEG2 decoder card. Painted blue.
PC-Beep.
Optional monaural. Plugs into the system board and allows you to route signals for the system speaker to external speakers through the line-out. The connector consists of two needle contacts.
Note that the signals can be fed to the line inputs simultaneously - the analog mixer will combine them.
Do not think that recording is done through the input ports. For example, you can send a signal from a cassette player to the line-in and listen to it right there (without starting a soft 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 amplification mode (20 dB Boost).
Used, for example, to insert voice comments into a clip of a photo album or chat on the Internet. Painted red or pink.
The input is of poor quality and is only suitable for voice recording. The fact is that a normal mic preamp starts at $ 50, so it would dramatically raise the price of the adapter. The famous sound engineer E. Petrov even recommends removing the built-in amplifier (by working with a soldering iron), as a result of which there are only 2 modes: an additional line input and a 20 dB amplifier for an electrodynamic microphone.
Karaoke lovers or vocal recording enthusiasts 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;
as an internal board on a bracket at the rear of the case;
in the form of a port module that comes 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 a 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.
Front stereo line output.
Mandatory. Designed to connect front (i.e. front) acoustic speakers... Made in stereo mini-phone jack format. Painted green.
Rear stereo line output.
Present in 4- and 6-channel adapters. Used to connect rear speakers.
It is usually made in the format of a stereo mini-phone jack and is then painted black. To save space, can be routed to the compact G9 complex connector.
Center speaker and subwoofer output ("Theater port").
Only present in 6-channel adapters. Two-channel.
It can be made in the format of a stereo mini-phone jack, but to save space it can be connected to a compact complex G9 connector (or combined, see below).
A small number of channels can be compensated for by a digital output port (through which several channels are transmitted), but this makes the adapter itself, and especially acoustics, rather expensive, and, moreover, has an effect only on expensive acoustics.
Modern adapters allow you to do more sophisticated upmixing and downmixing, for example, lay out 6-channel sound to 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 decision is addressed to those who do not want or do not have the opportunity to clutter up their workplace many speakers and tangled with wires.
4-channel adapters designed for a 4-5 component speaker system (the fifth is a subwoofer) and already give a full sound in games.
6-channel adapters give full sound not only in games, but also in home theaters (movies on DVD). The acoustics are 6-component and differs from 4-channel by adding only one (center) speaker, and it provides much more possibilities. Therefore, preference should be given to 6-channel adapters over 4-channel, especially since the difference in price between adapters is also minimal.
Headphone out
Usually combined with line-out to front speakers. According to AC "97, the output has an impedance of 32 Ohms, so headphones must also be matched with the same impedance. A stereo mini-phone jack is used.
Telephone port
This is an optional bi-directional internal port (MPC3 format) called TAD (Telephone Answering Device). It is connected with 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 the sound from the modem. Painted red. Note that at the present time, unfortunately, there are no hardware PCI modems priced up to $ 100, adapted for the domestic telephone network.
Connector colors
According to the PC "99 specification, the audio connectors have the following colors.
Connector colors
| Connector | Colour |
|---|---|
| Outdoor on the map | |
| Line-in | Blue |
| Line-out front / headphones | Green |
| Rear line-out | The black |
| Microphone | Red |
| MIDI / Game | Golden |
| Internal on the map | |
| CD Audio-In | White |
| AUX-In | Blue |
| TAD | Red |
| Loudspeakers | |
| Loudspeakers | Brown |
| Subwoofer | Orange |
Analog mixer
The adapter contains analog input and output mixers, volume controls, complemented by a complete mute of each channel (Mute). Mixers with tone control are rare. The explanatory diagram is for the case of a 2-channel card.
The mixer capabilities are part of the standard Volume Control application - volume control. According to AC "97, it is possible to close any channel and adjust the volume of not only analog, but also digital sources.
Instead of having separate controls on the stereo channels, this application uses overall volume and channel balance. Here:
Volume control - line output. Located in the mixer. Another common name is Master Volume.
Wave - digital channel for Wave files.
ADC and DAC
ADC (ANDtax cifrovoy pconverter, Analog to Digital Convertor, ADC) is used to digitize analog audio (usually recording to a file).
DAC (Cifro- andtax pconverter, Digital to Analog Convertor, DAC) converts digital audio to analog (de-digitization).
The main ADC parameters are the sampling rate and 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 rate defines the "bumps" rate, and the bit depth defines the maximum number of levels in bars (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 should be no less than those of the played PCM file.
Note that “from the inside” modern mainstream ADCs and DACs are so-called single-bit ones, 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.
As a guide: CD Audio is a digital audio format on music CDs that is the recognized standard for Hi-Fi quality, using 44.1 kHz and 16 bits.
Mainstream adapters record and play no more than 16 bit PCM files. The frequency is usually 44.1, 48 kHz, but even lower multiples are used for sound of unassuming quality (22.5 kHz, etc.). 48 kHz is standard for computer audio, digital interfaces, DVD audio, and should be preferred (unless the file is a copy of the audio track).
When passing through the digital part of the adapter, the sound may undergo some processing, and the results of intermediate calculations go beyond the 16-bit grid. Therefore, it is highly desirable for a 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 overwhelming 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 that of 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 not high. Therefore, for recording, 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.
We also note that the same manufacturer has ADCs with the same bit depth, but different quality, so high bit depth is not a guarantee of quality, but only a necessary condition. There are more objective parameters like signal-to-noise ratio. These are discussed in the codec section below, as the DAC / ADC are now implemented integrated into the codec.
It is important to note that it is the DACs that mainly determine the sound purity of the adapter. At the same time, the relative cost of a DAC / ADC in an adapter is low.
Digital equalizer
Not present on all adapters. The main purpose is to compensate for the lack of acoustics. Affects all output streams the same.
In such advanced software players as WinAmp, WMP, Apollo digital equalizers are built in (you only need a sufficiently powerful processor). However, the hardware equalizer distorts the sound less. The equalizer is characterized by the number of bands (usually 10), the number of preset presets and the ability to memorize user settings.
Stream processor
As you can see from the diagram, this is the central unit, the services of which are used by other units. Therefore, all modern adapters (except, maybe 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 bundle, this is equivalent to hardware acceleration for the MS DirectSound API.
The block controls the digital processing of audio streams. Among the processing functions: 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 done so quickly that the “gaps” between the blocks of the mixed streams are invisible to the ear (unlike software mixing).
An example of routing is directing the flow of violin sound from MIDI to a 3D block. The output stream describes a violin circling overhead.
The parameter of a thread processor is the number of simultaneously hardware accelerated threads, but the number itself is not so important for choosing, the main thing is the very fact of such acceleration. Note that if the application lacks hardware threads, then DirectSound provides for the addition of an unlimited number of software-processed threads (the power of the central processor would be enough). “Multithreading” is used, for example, in games, as well as when editing multiple recordings.
MIDI block
Only 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 played back are MIDI files in which the score (musical notation) of an orchestral work is recorded.
MIDI files are a thousand times smaller than regular PCM files (for good quality stereo it is 10 KB / min compared to 10 MB / min). This compactness is used in games, for network applications and karaoke (see, for example, the karaoke server http://www.fcenter.ru/www.karaoke.ru). In karaoke, the ability to play a MIDI file with a freely selectable tempo and key is valuable.
The parameters of the MIDI block are the number of hardware and software synthesized instrument voices (polyphony). A 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 a select few listeners can distinguish more voices.
The MIDI block uses an effect processor to apply effects (changing the sound of instruments to give more expression). Although effects can be applied to any stream, they first appeared in MIDI. Therefore, the possibility of overlaying effects is often quoted as MIDI parameters.
MIDI standards
The development process of MIDI 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. There is an international standard GM1 (General MIDI Level 1), its further development GM2 (General MIDI Level 2) and proprietary GM1 extensions: GS (General Synth), XG (Extended General). A MIDI block must necessarily support one of these standards.
Wavetable synthesis
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 chunks). Banks, of course, also conform to a specific MIDI standard.
One or more banks of different sizes can also be supplied with the adapter; smaller banks are used with a small amount of computer memory.
Banks are of very different quality and differ in the parameters of digitizing samples (for example, there are 14-, 16- and 18-bit samples), completeness of samples (whether a full note is included or only an attack and a piece of constant level, which is then cyclically played to obtain a large length) , the presence of sample variations (sharp and soft sound extraction), the fullness of the pitch of the voice (the number of base notes from which the rest are derived). An interesting parameter of both the bank and the synthesizer itself is the number of synthesizer envelope phases. 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 not available, but you can focus on the total size of the bank: the larger it is, the better. The bill goes to megabytes. For reference: a good quality bank from Yamaha of XG standard with 3 times compressed 18-bit samples, takes 4 MB.
Downloadable Sounds (DLS) technology
This state-of-the-art technology overcomes the shortcomings of MIDI standards such as fixed instrument kits and variability in sound across different adapters. Numbers of 128 instruments can be dynamically (loaded into memory, where the name comes from) to assign any instrument (called “loaded”). Analogous to 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 loudness, expression, envelope parameters, etc., which eliminates ambiguity.
The DLS Level 1 (DLS1) specifications from 1997 and DLS Level 2 (DLS2) from 1998 are known. MS DirectX 8.0 already supported the DLS2 synthesizer and contained its software implementation. The DLS2 bank was 3.3M in size with 16-bit samples, 226 melodic and 9 percussion instruments.
A modern adapter should support at least DLS1, and DLS support seems to be more important than support for MIDI standards. The PC Specification 99 Audio recommends hardware support for DLS.
Soft WT-synthesis
Pure software synthesis using the wave table does not require so many resources: the CPU power of 300 MHz with MMX support is quite enough for this. Therefore, such parameters as the number of hardware-accelerated voices have now become irrelevant (and even 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 is available that supports DLS2. By default, the Roland GM / GS Sound Set bank is 3.3 MB.
More advanced software XG synthesizers Yamaha S-YXGxxx are supplied with sound adapters based mainly on software processing (HSP adapters) (they can also be found even on the driver disc that comes with motherboard; I met 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 details in the corresponding appendix), but the card requires some support.
You can view the CPU load in Windows 2000 by turning on the player and looking at the CPU column in the 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.
Bank Format Compatible
The MIDI block is compatible with a specific bank format. So, cards on the Creative CT5880 audio controller use banks of a specific format from Ensoniq. Most attractive is DLS compatibility. In this case, you can usually convert banks of other formats to DLS.
To use Yamaha's S-YXGxxx soft synthesizers, the adapter only requires trivial support for 16-bit 44.1 kHz PCM playback.
However, the Nemesys GigaXXX soft synthesizers using hard disk banks require support from the GSIF quick interface card (see Glossary).
MS DirectMusic
If the sound adapter supports hardware acceleration DirectMusic, then this increases its appeal to 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 with Windows 98SE and later.
Effect processor
Sound effects originally appeared in MIDI and were about changing the sound of instruments to give more expression. The main ones are chorus and reverberation... The total number of effects is quite large, for example, 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 an effect specific to MIDI is breath for wind instruments). The effect processor does all this. Its hardware implementation is included only in adapters with onboard DSP (see below). In the absence of DSP, a few simple effects are applied in a software way.
The hardware capability of the adapter to apply effects is expressed in the following:
what are these effects;
how many of them can be applied to all streams (instruments) at the same time;
how many channels the effects can be applied to individually.
Modern adapters support a sufficient number of effects, so that, as a first approximation, you can ignore these parameters.
3D block
What a 3D block does
This block deals with the support of positioned 3D sound (for short, just 3D sound). Ideally, this means that you can hear a point source like a mosquito squeak. 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: the creation of a point source of sound in an infinite space and the imposition of environmental effects on the sound stream created by it ( room reverberation, occlusions and obstructions), since the games are mainly held not in open areas, but indoors. It can be a resounding echo of steps in the hall, an echo of rocks, etc.
Note that the overlay of environmental effects (such as a concert hall, a room with carpets) can bring even ordinary stereo audio files to life.
Currently, adapters use a small number of 3D technologies. The main part of the API of these technologies is standard (open). Some technologies have small API extensions to this standard part. The standard part of the API consists of the components DS3D, I3DL2, EAX2, which are supported (by the 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 functions of the DirectX libraries. The DirectX audio component (DirectX Audio) has minimal capabilities, operates with low quality sound (with 8-bit and 22 kHz parameters) and has a slow software mixer for 3D streams.
Despite the fact that most of the API is 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 various acoustic configurations for 3D sound are shown in the following table.
The downside of the headphones is the feeling that the sound source is closer than it really is.
Applied technologies
Here is a list of the main technologies:
technology by Creative (does not have a special name);
Sensaura 3D from Sensaura;
Q3D (QSound3D) from QSound Labs.
Creative technology is used only in adapters manufactured by the company itself. In contrast, Sensaura and QSound Labs do not manufacture adapters, but license their technologies to adapter manufacturers.
The technology from Creative is proprietary and little is known about it. However, its EAX API has become the de facto standard.
Q3D technology from Qsound is based on average listening results and does not require time-consuming 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 to the standard ones. These are the following Sensaura 3D components:
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 sweeping by. 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 is highly dependent on the shape and size of his auricles and head, as well as individual hearing (especially for the high frequencies). All technologies are designed for some average ear. By adjusting the parameters for yourself, 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 environment effects implementation component is called Sensaura EnvironmentFX. It is API compatible with EAX2 (the de facto standard), but sounds a little different (which does not mean worse).
Technologies also differ in different schemes for playing 3D sound on 4 speakers (we will call them 4-schemes).
| Technology | 4-diagram |
|---|---|
| Creative 5880 | panoramic |
| Creative EMU | accurate |
| Q3D | variable |
| Sensaura 3D | accurate |
panoramic (panning):
3D sound is calculated and sent to the front speakers, while the rear speakers simply duplicate the front ones. This is the most primitive scheme.
variable (transition):
3D sound is calculated and goes to those speakers that are closest to the sound source. The “remaining” columns simply duplicate the “main” ones. This is a slight improvement to the panoramic layout.
accurate:
3D sound is calculated and sent separately to the front and rear pairs of speakers.
All technologies also allow 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 maps, arranged the technologies in this descending order:
implementation on headphones
1.
Q3D
2.
Sensaura 3D
3.
Creative.
implementation for 2 columns
1.
Sensaura 3D
2.
Q3D
3.
Creative.
implementation for multichannel 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.
Adapter 3D 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 for more varied games. Additional streams will be created programmatically. PC Specification 99 Audio recommends hardware support for 8 sound sources. Hardware 3D acceleration is determined by the onboard DSP.
4-diagram
The technology determines, in principle, the scheme of using 4 columns, but for convenience this parameter is taken out separately.
Home theater
DVD movie soft players support 6-channel (or even 7-channel) audio playback on sound adapters. The image is output to a large TV, the sound is output to a multi-component speaker system. By placing the speakers not only in front of the listener, but also in the back, you can create a surround sound (Surround Sound), for example, the effect of a passing train.
Cinema surround sound can be said to be 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 the 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 is highly compressed. This is currently the most common format.
DTS
6-channel audio, less compressed than AC-3 and therefore better quality. Requires a lot of resources when decoding. Newer and less common.
DTS-ES
DTS extension to 7 channels. The first products appeared only in Q4 2001.
In 6-channel audio, the output channels are assigned as follows:
1,2 - to the front speakers
3.4 - to the rear speakers (everything is like in games)
5.6 - to the center speaker and subwoofer, respectively.
Dialogues (i.e. speech) are displayed on the center column, which increases their intelligibility. The placement of the subwoofer can be rather arbitrary due to the hearing immunity to the location of the bass source.
In 7-channel audio, the additional channel goes to the center rear speaker.
6ch sound is often recorded as 5.1ch, 7ch as 6.1ch, where “.1” stands for subwoofer.
Adapter modes
There are the following operating modes of the adapter, or rather its home theater unit:
through (pass-thru)
the track is transmitted unchanged through the digital output port of the adapter to the acoustics, where it is decoded.
The adapter must have a digital output port and the speaker must have a corresponding digital input port and a specific decoder format. The presence of a decoder dramatically (approximately twice) raises the speaker system. With the advent of new formats, the possibility of upgrading the speaker system remains in question.
However, the considered solution is justified when the corresponding speaker system already exists.
digital
the track is unpacked into channels, the channels are decoded into PCM streams and then packed into 3 streams. These streams are fed to the corresponding digital output ports of the adapter.
The adapter and the speaker must each have 3 digital ports. However, an expensive decoder is not required in acoustics.
This mode is available only in Creative SB cards starting from the Live! 5.1 and so far only for the AC-3 format.
Compared to analogue mode, better sound transmission is produced.
analog
the track is completely unpacked, decoded and converted into the appropriate number (6 or 7) of analog output channels, which are fed to the adapter ports.
The adapter and speaker must have an appropriate number of analog ports.
This is the cheapest option. All work can be done with a soft DVD player and therefore format support depends only on the player used. For example, DTS is supported by InterVideo's WinDVD player since v3.0.
In all cases downmixing possible for 2- and 4-channel acoustics and headphones. However, this leads to distortion, as conversions are performed with compressed audio that “does not like it”.
From the adapter side it is possible optional hardware acceleration for processing (unpacking, decoding) for specific formats.
The “theater” adapter is sometimes supplied with soft DVD player (it usually comes with a DVD drive) and / or remote control (remote control). The latter is more typical for execution with a port module.
A high-quality home theater implementation is not so much about acoustics (from $ 200), a large wide-screen TV (about $ 2,000), but about a separate room with an area of \u200b\u200b20 meters. The room should not be cluttered, and the walls should be covered with carpets for sound absorption.
Upmixing
This is a “multi-channel” audio layout into more channels 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 sound to 6 speakers.
Examples of upmixing technologies are:
CMSS (Creative Multi Speaker Surround)
Upmixing of stereo sound for 4 or 6 channels. It is 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 of stereo sound for 4 or 6 channels.
MP3 unit is optional. Performs hardware acceleration of compressed MP3 decoding when playing files. Along with MP3, less common formats such as WMA, OGG can also be supported. The block allows you to reduce the processor load by several percent, which is insignificant. Only some sound cards have such a block.
Digital ports
Transmission in digital is practically unaffected by pickups and interference. The cable in the case of a multi-channel speaker system becomes thinner and the connectors are 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 electrical SPDIF ports are used. Therefore, the same format is used in the vast majority of maps. However, there are also optical SPDIF ports in home music equipment. To connect to such equipment, you need to choose cards with an optical port. Sound cards with an additional port module have an especially large set of ports.
The 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 to complement the AC-Link bus (see below).
For more information on digital interfaces, see the corresponding appendix
IEEE 1394
Optional digital bi-directional outdoor port. Provides up to 400 Mbps throughput, PnP, hot-plug capability.
CD SPDIF (Digital Audio)
Optional digital internal input port, 2-pin, MPC format. Serves for digital connection of a CD drive and for playing music CDs CD Audio (these outputs are also optional for drives). The data format is the same as SPDIF, but 5V can be used instead of 1V.
SPDIF out
Optional digital output external port. Used for digital communication with a speaker system (including multicomponent). The difference in comparison with analog transmission is noticeable only in good speaker system... The digital port costs $ 10 or more for the adapter.
One port is enough for transmitting stereo or packed 6-channel streams.
Electrical connector format: stereo mini-phone jack for 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 audio output of MPEG2 decoder. Better quality than SPDIF.
Ports are anachronisms
MIDI port
is used to connect a musical MIDI instrument (usually a 4-6 octave keyboard) via an additional adapter costing about $ 20.
game port
optional, used to connect joystick-type game manipulators (both analog and digital due to the use of different contact groups).
Both ports are usually mated to a gold-colored DB-15 connector.
PCM file I / O unit
The PCM file input / output unit (Wave in / out) uses the PCI interface to exchange PCM audio files with the “outside world”, and it can do this simultaneously for several files.
PCI interface
The current version is 2.2. But 2.1 also applies, which is not much different.
It is more important that the adapter has a PCI Bus Master mode - a mode of data exchange via the PCI bus with a minimum participation of the processor. This makes it easier for software to implement many audio functions, such as sound synthesis. Relevant for adapters that do not have certain hardware blocks, for example, a MIDI synthesizer. PCI Bus Master is currently implemented in almost all adapters.
| Hardware composition AC "97 sound adapter |
Let's see how the architecture of the sound adapter is implemented in the Intel AC "97 specification, where AC is an abbreviation for Audio Codec, 97 - the year of adoption of the first version of “spec” - 1997. The current version is 2.2 from 2000.
Close-up
In the diagram, asterisks (*) indicate optional components. Internal ports of the codec 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. Hardware is implemented in the form of one or two chips.
AC "97 controller (Digital controller). Performs digital processing of audio streams, including mixing, MIDI synthesis, 3D, overdubbing. Performed as a single chip. Tests show that the controller can affect the color of the sound.
The AC-Link bus connects the codec to the controller. A bus drop to a CNR slot is used in integrated solutions.
The AC "97 specification specifically states that the codec and the 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 "harness" is placed between the ports of the codec and the connectors of the adapter. This is, firstly, an operational amplifier chip (Philips TDA1308 type), which protects the front line output from overloads. This allows you to connect to it and headphones that have a much lower impedance (according to the 32 Ohm standard) compared to the impedance of 1 kOhm for active speakers.
Note that other outputs (rear, theater) are usually not protected and therefore it is not recommended to connect headphones (only speakers) to them.
Input ports can also be protected (transistors, capacitors, chips, etc.). The importance of this strapping is not in doubt.
A harmful anachronism is the presence of a power amplifier in some cheap adapters (to use cheap passive loudspeakers). Not only is such an amplifier of low quality, it is also an additional source of noise and heat. Therefore, it is recommended to disable it (by means of a jumper).
Block diagram
In the diagram, the numbers show the connection of the ADC / DAC with the corresponding ports. If the names of the internal ports are enclosed in brackets (CD Audio, etc.), this means that they are optional. However, all the additional inputs and outputs shown in the upper part are present on the codec, and it is up to the manufacturer to install the appropriate 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 ( todiving / decoding; in English literature - codec, coder / decoder), derived from the name of its main components - ADC / DAC. The latter convert an audio signal from analog to digital (encoding) and vice versa (decoding).
Microphone input has a programmable gain, as well as an activated 20 dB boost mode. The latter is necessary for microphones dynamic type (as opposed to electret). An optional second microphone input allows simultaneous use of one headset microphone used for speech and a second quality desktop microphone.
Headphone out has an amplifier impedance of 32 Ohm, respectively, the same headphones must be chosen.
SPDIF output. The specification prescribes the transmission of audio streams with a frequency of 48 kHz, which guarantees compatibility with household appliances. If a PCM file with a sampling rate other than 48 is output via the interface, then it is transparently converted to 48 kHz before transmission via the AC-link bus. Support for other frequencies (ie no conversion, “bit exact”) is optional. Note that codecs with SPDIF port are still rare.
Mixer can have analog tone control. However, a software or hardware digital equalizer allows you to make more detailed adjustments.
ADC / DAC are 16, 18 or 20 bits (no higher), and the limitation is related to the AC-link bus, see below. However, writing (via the PCI interface) is done to no more than 16-bit PCM files.
The codec is full duplex, i.e. allows simultaneous recording and playback, and in different modes.
Codec quality parameters
The bit depth of the codec is understood as the bit depth of the ADC and DAC included in it. As already mentioned, the bit depths of codecs are far from complete parameters of their audio quality. Thus, the Cirrus Logic CS-4294-KQ and CS-4294-JQ codecs of the same company have the same bit depth, but different quality.
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 the DAC is used much more often than the ADC, we will restrict ourselves only to the DAC parameters. This corresponds to the parameters of the D-A codec path (from digital to analog).
Let's consider the main audio quality parameters (they apply to any audio path; for more details on the definition of paths and their parameters, see the corresponding appendix). These parameters are given by the manufacturers of codecs in the specification, and for the maximum sampling rate - 48 kHz for AC "97 codecs (since in this case the figures 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 essentially signal independent). Measured in decibels. The larger the range, the better. Good values \u200b\u200bare 85 dB or more.
Note that this number should be considered only as an upper bar, a reference point. The fact is that, firstly, the measurement is made for a pure sinusoid. For a real signal, the parameter can be much less. Second, the measurement is made for the maximum signal. For a normal medium loudness signal, the ratio will also be lower.
Unfortunately, most of the codec manufacturers cite not DR, but an even more overestimated S / N parameter (aka SNR) - the signal-to-noise ratio, while the noise is measured in the absence of a signal, i.e. in even more artificial conditions.
FR (Frequency Range), frequency range of uniformity of reproduction
with the specified spreads. This is the frequency range where the frequency response curve does not go beyond the specified boundaries of the spread. The larger this range at a given range of spread or the narrower these limits at a given range, the less distorted the signal frequencies after passing the codec. For good codecs this is 20-20 "000 Hz at ± 0.5 dB.
Note that the boundary of scatter for the entire audio range of 20-20 "000 Hz is more interesting (as is customary in test utilities).
Obviously, FR is some "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 only by the values \u200b\u200bof the spread.
THD + N (Total Harmonic Distortion plus Noise), total harmonic distortion plus noise.
THD expresses the level of distortion generated by the codecs themselves (which are proportional to the signal). THD + N also includes noise (by definition, independent of the signal). Both ratios are expressed as a percentage. The lower the ratio, the better. A good THD + N value is 0.02.
Sometimes THD or THD + N are expressed in decibels. The relationship between percent 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 claims one codec, there really is no 6-channel codec. When 6-channel adapter in it 2 codecs are used, one of which is 4-channel, and the second 2- or 4-channel (both AC "97 controller and AC-link bus allow it).
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, in some adapters, instead of the second codec, a DAC (usually from Philips) is used, with its connection not via AC-link, but via a simpler I2S bus.
Current situation with codecs
Currently, all sound cards (of the reviewed ones), 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 sound 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 advanced digital audio processing (3D sound, creation of MIDI compositions, etc.), as well as for digital ports.
PC-D-A Path Parameters
We will restrict ourselves to a more relevant case of playback (but everything said below can be repeated for recording). Codec parameters describe path D-A codec, which is only part of the total pass-through path PC-D-A (digital reproduction to line out). The quality of the latter is more important and depends on the quality of the PCB layout, and on the power filters, and on the strapping 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
Alexei Lukin.
AC-link bus
The AC-link bus provides bi-directional 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 capacity of up to 20 bits and a sampling rate of 48 kHz. In most implementations, the bus frequency is fixed at 48 kHz. This means that if the audio file has a sampling rate of 48 kHz as well, then one sample is transferred for each bus cycle. If the sampling rate is different, for example 44.1 kHz for CD audio tracks, then pre-sampling is performed in the audio controller (SRC block, see below). In theory, this introduces additional errors. The AC "97 2.2 version provides an optional mode for transmitting streams with a frequency of 44.1 kHz and any other mode 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 the audio codec. ...
In AC "97 2.0, an option was added to stream transmission with a sampling rate of 96 kHz.
AC "97 controller
Here SRC (Sampling Rate Converter) is a PCM stream frequency converter.
It is used by the stream processor when it mixes streams with different frequencies, as well as when preparing data for the AC-Link and SPDIF buses, which both usually operate at 48 kHz. The standard values \u200b\u200bfor PCM files are 48, 44.1, 32.0, 22.05, 16.0, 11.025, 8.0 kHz.
The controller has the ability to return a digital stream to the main memory for subsequent redirection to an external digital bus (USB type). This is called a “digital loopback” and it also operates at 48 kHz.
The scheme omits very few people who are already interested in the DOS audio support unit, as well as the MPU-401 and game ports. PC "99 Audio recommends using the USB ports instead.
DirectMusic API and support DLS.
Hardware accelerated decoding for cinema sound formats.
Upmixing.
Optional also I2S input digital port. Unlike SPDIF, this interface practically does not suffer from jitter. Designed to communicate with DVD drives.
The controller has a digital amplifier inside. Like any transistor amplifier, it is sensitive to overloads and produces high distortions in this mode. For some controllers, this “red zone” starts with fairly low values \u200b\u200bfor the Wave fader 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 one), but to use only the general volume control.
Ideally, in the operating range of amplification, 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; SigmaTel STAC9708 codec) revealed that the controller in Creative SB Live! 5.1 produces large harmonic distortion at the front output.
DSP and HSP audio controllers
Hardware acceleration of applying effects, acceleration of 3D sound, implementation of a digital equalizer, etc. entirely depends on whether the audio DSP (audio digital signalnew processor).
In the absence of DSP, the audio controller is called HSP ( host-based signal processing), i.e. instead of DSP, a central processor is used. This, of course, makes the audio controller cheaper, but it assumes a certain level of CPU performance. It is known that the adapter on the HSP controller can take up to 20% of the processor's resources. With DSP, lag-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 a significant plus.
According to the AC "97 2.2 specification, an optional SPDIF output port must be placed in the codec (such codecs were announced only by SigmaTel). In practice, the port is built into an audio controller (and the SPDIF input port is often placed there as well).
Some controllers also have I2S ports, which makes it easier to connect additional DACs.
Ports and their combination
Perceptible inconveniences are caused by the practice of port matching, forcing you to engage in "poking". This is due to the fact that an obsolete and bulky MIDI / game port is still placed on the sound card bar. This port has nothing to do with sound, and it takes up a lot of space (recall that PC "99 recommends using USB devices).
Headphone output combination
In the case when a 6-channel sound card has output connectors in the mini-jack format, a digital output and a game port, there is not enough space on it. This forces the digital port to be combined with (at least) one of the analog ports. One of the solutions found in the latest cards is to use a compact G9 connector for the output ports (together with an adapter splitter cable).
Extra pins do not add reliability and the best solution would be to use separate ports by completely removing the MIDI / game port.
When an adapter is not needed
Listening to audio CD.
You can use the built-in headphone out on the front of the CD drive or the line out on the back. 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 system unit. Such modules cost about $ 50 (internal module) and $ 100-150 (external module) for the card. Note that a standalone mic preamp costs around $ 60.
Other versions will be referred to collectively as integrated (on 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, with the controller being made as a separate chip.
Basically, this performance is no different from a sound card. PCI slot is freed. Since the controller accounts for the bulk of the cost of the adapter, inexpensive controllers are used to achieve price attractiveness.
Integrated adapter with controller in the system chipset
The audio controller is built into South Bridge system chipset (in the case of the classic two-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 on the motherboard. To upgrade up to 6 channels, it is proposed to purchase a compact card for a slot in the form factor AMR, CNR, ACR, on which additional codecs and ports are located. Such cards are just beginning to spread.
Thus, manufacturers motherboards offer the following upgrade steps.
- Right from the start, you get integrated 2-channel audio, good quality and very little money.
- If 6-channel audio is needed, then a CNR or ACR card is purchased.
- If you need effects, hardware support for 3D, etc., then instead of item 2, a separate PCI card is purchased, and the built-in sound is disabled.
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Experienced users will probably remember the days when computers could only "beep". Times are changing, and now it is difficult to imagine a computer without the ability to play and record sound. The quality of playback / recording can be different, although lately it has varied from normal to gorgeous (we agree that these terms are somewhat unscientific, but understandable).
The quality of sound emitted by a computer depends on two devices: a sound card and acoustic (computer, multimedia) speakers.
Sound card (sound card, sound accelerator, « sound ear» ) - a device that allows a computer to play and record sound information.
The demand for sound among computer users is so great that almost all motherboards released today contain an integrated sound card (similar to an integrated video card).
As a rule, the capabilities of the built-in sound card are sufficient to meet the needs of most users. If you are not an ardent music lover or a fan of complete immersion in a computer game (this is when the walls are shaking from your shots, and the neighbors are calling the police), then you will have enough integrated sound and you will not have to buy a more powerful sound card, which is a separate expansion card ( fig.5.2).
Figure: 5.2.
If you do decide to purchase a sound card, then study the following information.
Sound card connectors and slots
Look at the back of the system unit. Regardless of the type of sound card, you will likely see multi-colored connectors (Figure 5.3).
Figure: 5.3.Built-in sound card audio connectors
Line-out
(line-out) - a stereo output to which powered speakers or headphones are connected. On cheap video cards there is only one such output, but there are sound cards with two or more line outputs (if it is designed to connect more than two speakers). You may also encounter the line outputs as two mono jacks (labeled right and left).
By the current standard, the line-out connector is usually lemon-colored (don't ask us why, we don't know). However, some manufacturers (apparently also not understanding why lemon) do not adhere to this rule. For example, one of the authors has this connector green, while the other has all the connectors black. And the connectors of professional and even semi-professional sound cards can be gold-plated. Better focus on the line-out icon or read the instructions for your sound card if the icon is missing.
In addition, using this connector, you can connect various musical equipment (for example, a tape recorder or a stereo system) to your computer to play music through the speakers of the tape recorder or center, as well as to record on them.
Line-in
(line-in) - stereo input for connecting other playback devices. It is necessary if you are going to record sound from other devices to your computer. The Line-in connector is usually blue.
Mic-in
(microphone) - a monaural connector that is used to connect simple microphones and further record voice (or other sounds) on a computer. This connector is usually red or pink.
Most conventional sound cards have only these three connectors, but the more advanced and expensive ones boast several additional ones.
MIDI / gameport(joystick port) - A rectangular connector that allows you to connect a game device (joystick) or, for example, a synthesizer keyboard. This connector is usually yellow.
Speaker-out / Subwoofer(output to speakers / subwoofer) - stereo output, unlike Line-out, it has an amplifier. This connector can be used to connect passive speakers (without an amplifier) \u200b\u200bor a subwoofer. Some users think that connecting powered speakers to this output will improve the signal. However, it is not. As a result, the sound quality will surprise you unpleasantly. This connector is orange.
S / PDIF(Sony / Philips Digital Interface Format) - a connector that can be located both on the outside of the sound card and on the board itself (that is, inside the system unit). It allows you to connect external audio devices to your computer, such as a DVD player or home theater. The signal through such a connector is output in digital form, which eliminates the appearance of noise inherent in analog devices.
Just in case
Some sound cards, to save space on the external panel, have one special socket, to which an external device with several connectors is connected at once: S / PDIF, Line-in / out and MIDI. Don't be alarmed if you don't find familiar connectors, just look in the box from the sound card, there must be an additional device somewhere.
CD-in(MPC-3 CD-input) - a special connector that allows the audio card to transmit information from a CD-ROM drive in analog format. If the sound card manufacturers followed all the rules, its color is black or white.
MPC-3 Aux-in(external device input) - connector for connecting other devices (for example, a second CD-ROM drive). Outwardly very similar to CD-in.
MPC-3 Modem-in / out(input-output for connecting a modem) - this connector is used to connect a modem. This connector is green. This is usually unnecessary unless you intend to listen to the crackle of the modem through your speakers or plan to host teleconferences over the Internet.
Connector for connecting various daughter cards- has the most big size... Very similar to an IDE (remember how a hard drive?). Daughter cards connected to the sound card to expand its capabilities. This is used by specialists working with sound.
Before we look at the "internals" of a sound card, let's examine the basic concepts of digital audio recording.
Converting the original analog signal to discrete
Analog signals are signals whose parameters change continuously (and not abruptly) over time, that is, the number of values \u200b\u200bof some parameter of an analog signal (for example, amplitude) is, in general, infinite. Sound waves (sound) are typical analog signals.
A discrete signal, in turn, is described by a finite number of parameters separated in time. The process of signal digitization is reduced to replacing the continuous function of parameters with a finite set of so-called samples- measurements of the signal parameter value, made through certain intervals time (fig. 5.4).
Figure: 5.4.Digitizing an analog signal
As you can see in fig. 5.4, \u200b\u200ball the points of the curve located between samples (for example, between samples 5 and 6) will not fall into the digitized version of the signal. Thus, we can conclude that the more samples there are, the more accurately (better) the signal is digitized.
Definition
The number of samples per unit of time is called the sampling rate. The sampling rate is measured in kilohertz.
Conversion of an analog signal to digital is carried out by a special device called analog-to-digital converter(ADC). To convert a digital signal back to analog, use digital-to-analog converter(DAC).
What parameters determine the quality of the digitized sound? We have already found out that the quality depends on the sampling rate. There is even a special Kotelnikov theorem, according to which the sampling frequency should be twice the oscillation frequency of the highest component of the audio signal. Considering that the highest-pitched sounds that the human ear can recognize have a frequency of about 20 kHz, then good quality digitized sound is achieved using a sampling rate of 44 kHz.
You probably know that sound waves are characterized by the following parameters: pitch (that is, the frequency of vibrations), loudness (that is, the amplitude of vibrations) and timbre (sound quality that gives the sound an individual color). The timbre of the sound depends on the composition of the sound signal. The fact is that the real sounds around us consist of a huge number of superimposed sound waves. The sound wave having the lowest frequency in such a "sound packet" is called basic tone... The sound waves of the "packet" propagating at higher frequencies are called overtones.
It is interesting
Are there sound sources that are capable of producing sound waves of a strictly defined frequency? Yes, there are. These are tuning forks used to tune musical instruments.
Why are we telling all this? And to the fact that when sampling sound, not one parameter of an audio signal is measured at all (as shown in Fig. 5.4), but many. The number of sound characteristics measured during sampling is influenced by a digitizing parameter called bit depth digitized sound. The depth of 16 bits is considered qualitative. With such a bit depth, digitizing the sound allows you to fix 2 16 \u003d 65 536 sound characteristics (frequency and amplitude of the fundamental tone and overtones).
Another parameter that determines the quality of the digitized sound is number of audio channels... The most common two-channel stereo recording nowadays creates the illusion of a sound panorama. Sound sources seem to be separated in space. More sophisticated systems (5 or more channels) create a surround sound effect.
Thus, the so-called "CD quality" are two-channel stereo recordings created with a bit depth of 16 bits and a sampling rate of 44 kHz.
Digitized audio is recorded in files with extension WAV... It is clear that the size of such a file depends, firstly, on the length of the audio track, and secondly, on the quality of digitization. These sizes are sometimes very large - one minute of high-quality digitized stereo recording takes about 10 MB. To solve this problem, many algorithms for compressing WAV files were invented, as a result of which files of other formats were obtained.
The most common of these is the MP3 format. It allows you to compress a WAV file more than 10 times almost losslessly. It is worth noting that "practically no loss" is a very conditional concept. Ardent music lovers will dispel this statement to smithereens and they will be right. To work with files of each format, a special compression-decompression program is used, called codec.
This is useful to know
If your computer does not want to play any sound file, most likely you do not have the necessary this format codec. You can find the codec you want on the Internet. Modern operating systems of the Windows family already contain built-in codecs for playing popular audio formats such as MP3 and WMA.
Codecs are characterized by bit rate - the number of compressed bits of an audio file that the codec decompresses in 1 second. The notorious quality of a CD is matched by a transfer rate of 128 Kbps.
Describing sound using parameters
You've probably heard of such a musical instrument as a synthesizer. It allows you to simulate the sound of various musical instruments and other sounds. This device is equipped with a special processor that processes MIDI sound files.
let's get acquainted
MIDI (Musical Instrument Digital Interface) is a digital interface for musical instruments. MIDI files have the MID extension.
Unlike WAV files, MIDI files contain information about the notes of the melody (pitch, duration, and intensity) that the synthesizer should play, as well as instructions on which instruments these notes should be played. It turns out a kind of music book for an electronic conductor "hidden" in the synthesizer processor.
MIDI files are significantly smaller than WAV files. Absolutely all sound cards support the MIDI format. The quality of its execution depends on the characteristics of the card processor. In obsolete sound cards, imitation of various musical instruments was carried out using the technology frequency modulation synthesis(or FM synthesis). The quality of the "similarity" of the reproduced sounds to the sound of real musical instruments was average.
Modern sound cards use the so-called technology wavetable synthesis(Wave Table), or technology wave synthesis(WT synthesis). It boils down to the fact that sound samples of real musical instruments are recorded in advance in the sound card's memory. These sounds are then used when playing MIDI tunes. The number of such recorded instruments is determined by the parameter polyphony... Good sound cards are equipped with 64-voice polyphony.
Midi today
MIDI technology is of course still very heavily used by musicians. With the help of the MIDI interface, you can not only transmit commands to play certain notes, but also synchronize work, turn on certain modes of digital musical instruments and devices. But for the average computer user, MIDI technology seems to have ceased to be of interest. This is understandable, no synthesis can replace the sound of a real instrument.
Now we can finally move on to examining the "inner world" of sound cards.
Internal components of the sound card
Sound cards contain the following elements.
Converters- they are on each stereo channel: analog-to-digital (ADC) and digital-to-analog (DAC) (there are more converters on expensive cards). The ADC processes the analog signal from the line-in or microphone and converts it to digital. A DAC, in contrast, converts a digital signal to analog and sends it to the line-out. The sound quality depends on bit depthsupported by the converter.
Generator clock frequency - issues synchronizing signals to the converters, thereby setting the speed of information processing (remember the concept of sampling frequency). The most popular sound cards today have a frequency of 96 kHz.
CPU- generates analog sound that we hear from the speakers from incoming MIDI commands. It is the processor that determines the capabilities of the sound card. He is "connected" between central processing unit computer, operating system and music playback software. The processor of the sound card takes on a lot of work related to sound processing (partially offloading the central processor).
Important characteristics of sound cards
Now let's look at the main characteristics of sound cards that are useful to pay attention to when buying.
Sound cards, like most internal devices, usually plug into a PCI slot on the motherboard.
The indicator that we have already drawn your attention to is sampling frequency(it is set by the clock generator). The higher this frequency, the more accurately the sound is digitized, which has a positive effect on the sound quality.
The next parameter is the number of audio channels. If you intend to reproduce sound through two speakers, then any sound card will work (provided that you are satisfied with the other characteristics). If you want to surround yourself with sound, you need a multi-channel sound card (5.1 or 7.1). Of course, you will have to purchase a corresponding set of speakers. By the way, most sound cards built into modern motherboards contain six audio channels (5.1).
The quantity signal to noise ratio(S / N) is measured in decibels. The higher the value, the better. We advise you not to be interested in cards with S / N below 90 dB.
Another characteristic - supported sample size."Sample size" shows how much information describes each sound, and therefore sets the maximum number of possible sound options. This applies to those interested in MIDI capabilities.
If you are interested in high quality sound, we also recommend choosing a sound card with good hardware acceleration... Most high-quality (and, mind you, inexpensive) sound cards now have 3D support.
Before we finish talking about sound cards, let us give a few tips on choosing them in case you are not satisfied with the built-in one.
Advice 1.Don't buy cheap sound cards. The difference in quality between them and the integrated one will be small (if at all), so there is no point in overpaying.
Advice 2.Do not buy sound cards from unknown manufacturers. As a rule, this does not bring joy to anyone except the manufacturers and sellers themselves. We advise you to opt for Creative products. This company produces sound cards of sufficient quality and affordable, which even musicians do not disdain.
A sound card (or sound card) is a computer component that processes and outputs sound. With it, you can listen to music, voice acting in movies and games, or even process the sound yourself using special programs. This can be digitizing recordings, eliminating noise, mixing, recording, adjusting the frequency range, etc. The audio card has connectors for speakers or headphones, a microphone, and a line-in for audio input from another device. All connectors are colored in different colors, under which there is an icon of the device to be connected. For example, the speaker output is colored green. The speaker plug is also colored green.
The sound card can be built into (integrated sound card) or so-called external, which is made in the form of an electronic expansion board inserted into a special slot on the motherboard.
Built-in sound card 
External sound card
In modern computers, the sound card is almost always integrated into the motherboard. For an ordinary user, the capabilities of an integrated audio card are quite enough, and external ones, due to their high characteristics, are more focused on music lovers, musicians, sound engineers. In addition, external sound cards are also produced as a separate device in a separate housing. They will connect to USB port ... Such devices can already be called professional.
Studio external sound card It should be noted that if you nevertheless decide to purchase a high-quality external sound card for installation in a computer, then the speakers (or headphones) must be of high quality, otherwise it will be a waste of money.
