A sound card (also known as an audio card) is a computer expansion card that facilitates the input and output of audio signals to/from a computer under control of computer programs. Typical uses of sound cards include providing the audio component for multimedia applications such as music composition, editing video or audio, presentation/education, and entertainment (games). Many computers have sound capabilities built in, while others require additional expansion cards to provide for audio capability.

A typical sound card includes a sound chip, usually featuring a digital-to-analog converter, that converts recorded or generated digital data into an analog format. The output signal is connected to an amplifier, headphones, or external device using standard interconnects, such as a TRS connector or an RCA connector. If the number and size of connectors is too large for the space on the backplate the connectors will be off-board, typically using a breakout box, or an auxiliary backplate. More advanced cards usually include more than one sound chip to provide for higher data rates and multiple simultaneous functionality, eg between digital sound production and synthesized sounds (usually for real-time generation of music and sound effects using minimal data and CPU time).

Digital sound reproduction is usually done with multi-channel DACs, which are capable of multiple digital samples simultaneously at different pitches and volumes, or optionally applying real-time effects like filtering or distortion. Multi-channel digital sound playback can also be used for music synthesis when used with a digitized instrument bank, typically a small amount of ROM or Flash memory containing samples corresponding to MIDI instruments. A contrasting way to synthesize sound on a PC uses "audio codecs", which rely heavily on software for music synthesis, MIDI compliance, and even multiple-channel emulation. This approach has become common as manufacturers seek to simplify the design and the cost of sound cards.

Most sound cards have a line in connector for signal from a cassette tape recorder or similar sound source. The sound card digitizes this signal and stores it (under control of appropriate matching computer software) on the computer's hard disk for storage, editing, or further processing. Another common external connector is the microphone connector, for use by a microphone or other low level input device. Input through a microphone jack is often used by speech recognition software or for Voice over IP applications

CD-Rom drives are connected to the sound card using a four pin analog connector. If you use a computer's CD player (instead of Windows Media) you need to have this wire connected to the sound card. This four pin wire is one way only, meaning it can only pull audio from the CD drive to the sound card and not the other way around. This means that the headphone jack on the front of the CD-drive will not allow you to listen to any of the computer sounds, only the audio from the drive itself.

There are essentially two ways of making sound on a PC.

The first way is to recreate a wave form of sound using an analog device by digitizing it so that it can be reproduced later. The conversion of analog to digital is performed by the ADC (Analog/Digital Convertor). The wave form is digitally stored in "samples". The quality of the sounds are based on the bits per sample and the sampling rate. (See sample rates listed below). The sampling rate: the more samples the better the sound.

The second way is to program using MIDI.

MIDI (Musical Instrument Digital Interface; IPA: /ˈmɪdi/) is an industry-standard protocol that enables electronic musical instruments, computers and other equipment to communicate, control and synchronize with each other.

MIDI files contain only information on the notes to play the instrument making the sound and the timing of the notes. Each sound card has a variety of of prerecorded sounds the MIDI file can use. The more instruments (voices) a sound card has, the better it is thought to be. MIDI files are very small, but can't reproduce voices. Two kinds of sound creation in MIDI is FM synthesis and Wavetable Synthesis.

MIDI does not transmit an audio signal or media — it simply transmits digital data "event messages" such as the pitch and intensity of musical notes to play, control signals for parameters such as volume, vibrato and panning, cues and clock signals to set the tempo. As an electronic protocol, it is notable for its success, both in its widespread adoption throughout the industry, and in remaining essentially unchanged in the face of technological developments since its introduction in 1983.

At first, the beep acted primarily as a signal or a warning. Later, developers created music for the earliest PC games using beeps of different pitches and lengths. This music was not particularly realistic -- you can hear samples from some of these soundtracks at Crossfire Designs.

Fortunately, computers' sound capabilities increased greatly in the 1980s, when several manufacturers introduced add-on cards dedicated to controlling sound. Now, a computer with a sound card can do far more than just beep. It can produce 3-D audio for games or surround sound playback for DVDs. It can also capture and record sound from external sources.

In this article, you'll learn how a sound card allows a computer to create and record real, high-quality sound.

Analog vs. Digital

The most basic sound card is a printed circuit board that uses four components to translate analog and digital information:

• An analog-to-digital converter (ADC)

• A digital-to-analog converter (DAC)

• An ISA or PCI interface to connect the card to the motherboard

• Input and output connections for a microphone and speakers

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ADCs and DACs

An analog-to-digital converter measures sound waves at frequent intervals.

The number of measurements per second, called the sampling rate, is measured in kHz. The faster a card's sampling rate, the more accurate its reconstructed wave is.

If you were to play your recording back through the speakers, the DAC would perform the same basic steps in reverse. With accurate measurements and a fast sampling rate, the restored analog signal can be nearly identical to the original sound wave.

Even high sampling rates, however, cause some reduction in sound quality. The physical process of moving sound through wires can also cause distortion. Manufacturers use two measurements to describe this reduction in sound quality:

• Total Harmonic Distortion (THD), expressed as a percentage
• Signal to Noise Ratio (SNR), measured in decibels

Digital sample rate conversion

There are at least two ways to perform digital sample rate conversion:

• (a) If the two frequencies are in a fixed ratio, the conversion can be done as follows: Let F = least common multiple of the two frequencies. Generate a signal sampled at F by interpolating 0s in the original sample. This will also introduce aliases at multiples of the baseband frequency. Remove these with a digital low pass filter, until only the signals with less than half of the output sample frequency remain. Then reduce the sample rate by discarding the appropriate samples.

• (b) Another approach is to treat the samples as a time series, and create any needed new points by interpolation. In theory any interpolation method can be used, though linear (for simplicity) and a truncated sinc function (from theory) are most common.

Although the two approaches seem very different, they are mathematically identical. Picking an interpolation function in the second scheme is equivalent to picking the impulse response of the digital filter in the first scheme. Linear interpolation is equivalent to a triangular impulse response; sinc() will be an approximation to a brick wall filter (it approaches the desirable "brick wall" filter as the number of points increase).

If the sample rate ratios are known, fixed, and rational, method (a) is better, in theory. The length of the impulse response of the filter in (a) is the same as choosing the number of points used in interpolation in (b). In approach (a), a slow precomputation such as the Remez algorithm can be used to compute the "best" response possible given the number of points (best in terms of peak error in various frequency bands, and so on). Note that a truncated sinc() function, though correct in the limit of an infinite number of points, is not the most accurate filter for a finite number of points.

However, method (b) will work in more general cases, where the sample rate ratios are not rational, or two real time streams must be accommodated, or the sample rates are time varying.

Normally, due to the mathematical operations employed, the output samples of sample rate conversion are almost always computed to more precision than the output format can hold. Conversion to the output bit size can be done by simple rounding, or more sophisticated methods such as dither or noise shaping can be employed.

 Example

CDs are sampled at 44.1 kHz, but a Digital Audio Tape, or DAT is usually sampled at 48 kHz. How can material be converted from one sample rate to the other? First, note that 44.1 and 48 are in the ratio 147/160. Therefore to convert from 44.1 to 48, for example, the process is (conceptually):

The bitrates in this section are approximately the minimum that the average listener in a typical listening or viewing environment, when using the best available compression, would perceive as not significantly worse than the reference standard:

 Audio (MP3)

• 32 kbit/s — MW (AM) quality
• 96 kbit/s — FM quality
• 128–160 kbit/s — Standard Bitrate quality; difference can sometimes be obvious (e.g. bass quality)
• 192 kbit/s — DAB (Digital Audio Broadcasting) quality. Quickly becoming the new 'standard' bitrate for MP3 music; difference can be heard by few people.
• 224–320 kbit/s — Near CD Quality. Sound is indistinguishable from most CDs.

 Other audio

• 800 bit/s — minimum necessary for recognizable speech (using special-purpose FS-1015 speech codecs)
• 8 kbit/s — telephone quality (using speech codecs)
• 500 kbit/s–1 Mbit/s — lossless audio as used in formats such as FLAC, WavPack or Monkey's Audio
• 1411 kbit/s — PCM sound format of Compact Disc Digital Audio

 Video (MPEG2)

• 16 kbit/s — videophone quality (minimum necessary for a consumer-acceptable "talking head" picture)
• 128 – 384 kbit/s — business-oriented videoconferencing system quality
• 1 Mbit/s — VHS quality
• 5 Mbit/s — DVD quality
• 15 Mbit/s — HDTV quality
• 36 Mbit/s — HD DVD quality
• 54 Mbit/s — Blu-ray Disc quality