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Sound Cards / Subsystems

A Turtle Beach Catalina sound card with 7.1 surround sound and optical S/PDIF connectors.

Most motherboards have built-in sound subsystems. However, many users may prefer an expansion card with better quality or more features, even if the motherboard has built-in sound.

A sound card has three functions. The most common is converting digitized sound into analog sound and sending that to speakers or headphones. The second is to convert analog sound into a digital format. The third is to create music using data in a Musical Instrument Digital Interface (MIDI) file.

Analog to digital conversion

Let's tackle digitizing sound first. Sound consists of waves of changing pressure in the air. A microphone converts these pressure changes into a voltage (or current) that varies proportionally to the original sound. So, if the sound pressure varies 100 times per second, the voltage will vary 100 times per second. If the sound becomes louder, the voltage variations will become greater. The analog-to-digital converter (ADC) in the sound card takes a sample of this voltage at regular intervals. It then produces a binary number proportional to the voltage. For example, if a particular sample is positive one volt, the ADC may produce the binary equivalent of 100 (01100100). If the voltage is minus one volt, the ADC may produce the binary equivalent of -100 (10011100). If the number has a maximum of eight bits, the ADC can handle voltages from 1.28 to +1.27 volts, which is more than enough for typical consumer line-level signals.[1]

The resulting sequence of numbers, representing the voltage as it changes over time, is stored in memory or long-term storage (hard disk, SSD, etc.). The actual numbers stored will usually be scaled to optimize storage rather than be as straightforward as the above example. Therefore, with typical consumer equipment, where the maximum line level is one volt peak-to-peak, +0.5 volts will likely be represented by a number near the binary equivalent of 127 (01111111), and -0.5 volts will be represented by something near the binary equivalent of -128 (10000000). Proportional numbers between these maximums will represent voltages between +0.5 and -0.5.

The above example, where the voltage is converted to an eight-bit number, is called eight-bit sound. This allows for only 256 voltage levels to be recorded, limiting the signal-to-noise ratio. CD-quality audio is sampled 44,100 times per second and is stored as 16-bit numbers. This takes slightly more than two samples per cycle at the highest audible frequency (20 kHz) and can divide the signal into 65,536 levels. Each channel results in 44,100 16-bit numbers stored every second. A stereo signal, therefore, requires 88,200 16-bit numbers for every second of audio.

Digital to analog conversion

The next thing to discuss is the reciprocal of converting analog audio into digital audio. That is converting digital audio into analog audio. This consists of stepping through the numbers stored in memory at the same rate at which they were originally sampled. The digital-to-analog converter (DAC) converts each number into the appropriate voltage, faithfully reproducing the original audio signal.[2]

Myths:

1. Digital audio is not a faithful representation of the original analog signal because the recovered signal has stairsteps from one voltage to another (called aliasing) where the original is a smooth wave.

Original analog waveform

Supposed recovered waveform after analog-to-digital conversion, then digital-to-analog conversion.

These supposed stairsteps do not exist in the recovered signal. For reasons too complicated to explain here (see AC Waveforms in AC Circuits in the Electronics Technology course at vocademy.net), such stairsteps contain frequencies (harmonics) far above the capability of the audio subsystem to convey. The very nature of the electronics eliminates them.[3]

Actual recovered waveform with proper sampling rate selection. For lower sampling rates, an antialiasing filter produces the same results.

2. At the highest audio frequencies, the recovered signal consists of triangle waves instead of sinewaves because the waves are only sampled twice per cycle.

For the same reasons mentioned above, the electronics cannot convey the high harmonic frequencies contained in such triangle waves. Therefore, the original waveshape is recovered.

3. Analog sound is "warmer" than digital sound and is superior.

Analog recordings are altered in frequency response and dynamic range to accommodate the available recording media. Many people prefer the altered sound, but it is not a faithful reproduction of the original. However, digital compression, such as MP3, may introduce undesirable artifacts (distorted sound or strange background sounds). Increasing the sample rate reduces such artifacts.

Music synthesis

Most sound cards and subsystems contain a music synthesizer that can create the sounds of various musical instruments with various levels of quality. Musical instrument Digital Interface (MIDI) files tell MIDI-capable keyboards and computer sound subsystems which instruments to select and how to recreate music with them. Using appropriate software and cables, computers can interact with MIDI-capable keyboards to play MIDI files. Computers can also play MIDI files directly with the sound subsystem via Windows Media Payer or similar programs. However, a stand-alone keyboard's synthesis quality is far superior to the music synthesis of most computer sound subsystems.[4] A MIDI-capable keyboard can also utilize computer software to create MIDI files. Such software often emulates a multi-channel recording studio.

Sound card connections

Sound cards typically have three miniature phone jacks for audio input and output. These are:

Microphone

The microphone input goes to a preamplifier to make up for the low signal level of a microphone.

Line in

The line in input bypasses the preamplifier and expects a signal typical of the line-out connections on consumer audio systems.

Line-in and line-out jacks on a c.1980 analog cassette deck. 

A cable with male RCA plugs on one end and a stereo miniature phone plug on the other is required to connect a computer to consumer audio systems.

Line out

The line-out output produces a signal suitable for connecting a line-in connection on consumer audio equipment or to typical computer speakers. This output may provide enough power to drive headphones without external amplification. However, this is not guaranteed if it is not labeled for headphones.

Notebook Computers

Notebook computers usually have a single miniature phone jack that operates as a headphone, line-out, and microphone-in connection. When something is plugged into this jack, the computer will usually pop up a window asking what was plugged in.

Sound cards may have other sound connections, which are:

Speaker out

This speaker-out jack, found on older sound cards, has a power amplifier that can drive older computer speakers that don't have built-in power amplifiers. This output can also drive headphones.

5.1 and 7.1 surround sound

These are for the extra rear, center, and subwoofer speakers of 5.1 surround sound and the side speakers of 7.1 surround sound. Depending on the card and speakers, the surround sound outputs may or may not need to be amplified.

Joystick/MIDI

This 15-pin D-sub connector is found on older sound cards and motherboards with built-in sound. It is not present on newer systems because USB has superseded it.

Digital out

This yellow miniature phone jack (not to be confused with an analog video connection) provides a digital audio signal for consumer audio systems with digital surround sound.

Early sound cards often had one or more proprietary CD-ROM interfaces. Subsequent cards had parallel ATA (IDE) interfaces. The proprietary interfaces were provided before parallel ATA became the standard for CD-ROM drives (before SATA). Once parallel ATA became the standard, sound cards often provided a parallel ATA interface to supplement the interface provided by multi-I/O cards, which were often populated with hard drives, leaving no room for CD-ROMs.

You may also find sound cards and motherboard subsystems with FireWire, S/PDIF, and other connections.

Resources

With today's plug-and-play configuration system, technicians don't need to manage I/O resources. If you encounter older systems (many are still in use in professional recording studios), here are the resources typically used by sound cards:

IRQ 5
I/O address 220
DMA channel 1

Some sound cards use 2 DMA channels. 16-bit SoundBlater cards typically used high DMA channels five or six.

Should you need to configure a sound card to work in real mode, you may need to put the following directive in the AUTOEXEC.BAT file:

Set BLASTER=A220 I5 D1 H6

This tells the driver what resources the Sound Blaster-compatible sound card uses.

The mixer

The mixer application provides separate controls for each device sound may come from. The number of controls depends on how many devices are available. The sound card or subsystem driver may alter the appearance of the mixer.

An example of the mixer application in Windows 10.

In Windows 10, the mixer application is started by right-clicking the speaker icon in the notification area and then clicking Open Volume Mixer. This has slider controls and mute buttons for each device sound may come from.

Before Windows 7, there were two modes: playback (default) and recording. In Windows XP, click options to change the mode. There is no recording mode window 7 and later, but some sound card drivers may add this function. To set the recording level, open the Control Panel and click Sound. Click the Recording tab to see the input devices.

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An example of the sound options with the Recording tab selected.

Double-click a device to see its properties, where you will find the levels tab.

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An example of an input device's properties with the Levels tab selected.

Troubleshooting

There are many points where the sound volume or mute state can be controlled. Any one of them can turn off sound. If you have no sound, check all of the following that are applicable before suspecting device or driver problems.

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1Line signals are used to convey analog audio between various pieces of audio equipment. For consumer audio equipment a common use is carrying analog audio from a DVD or Blu-Ray player to an audio amplifier. These are usually RCA connectors that are color coded red for the right channel and white or black for the left channel.
2Audio compression, such as MP3 and FLAC, significantly reduces the amound of memory required to store digital sound.
3If the sampling rate is equal to or greater than twice the highest frequency of the signal, the recovered signal is free of aliasing. The frequency equal to ½ of the sampling rate is called the Nyquist frequency. CD quality sound is sampled 44,100 times per second. Therefore, the Nyquist frequency is 22,500 Hz, which is beyond the range of human hearing and probably the range of the equipment reproducing the sound. For lower fidelity recording, where the Nyquist frequency is significantly below 20 kHz, an anti-alisaing filter is used to remove frequencies above the Nyquist freqnency and prevent aliasing. If the Nyquist frequency is above 20 kHz and the equipment can respond to frquencies above 20 kHz, aliasing may be present. Even though the human ear can not hear the harmonic frequencies contained in such aliasing, the harmonic frequencies may interfere with each other creating artifacts with frequencies below 20 kHz. Therefore, an antialias filter may be used even if the Nyquist frequency is above 20 kHz.
4Musical keyboards typically use recorded, digitized samples of actual instruments for music synthesis. Despite some problems concerning changes of timbre with pitch, such keyboards create fairly realistic sounds. High-end computer sound cards often use samples of real instruments, but computer sound subsystems typically use FM synthesis, which usually produces laughingly terrible facsimilies of musical instruments.
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