In signal processing, an audio converter or digital audio converter is a type of electronic hardware technology which converts an analog audio signal to a digital audio format, either on the input (Analog-to-digital converter or ADC), or the output (Digital-to-analog converter, or DAC). They are common in numerous technologies —notably in computer sound cards, digital cellular phones, portable recording devices, and digital audio workstations (DAW). Once converted to digital format, digital audio signals and file formats can be processed in any of a number of ways as allowed by software —including converting to audio CD or MP3 formats.
Different types of converter units can operate at different resolutions which largely determines the resulting sound "quality." Depending on their quality and cost, converters also differ in their handling of:
"Resolution" generally refers to both bit and sample rates (though it may occasionally refer specifically to the sampling rate; being the more variable of the two.) For example, an inexpensive brand consumer sound card for computer has a typical operating range of around CD quality — 44,100 samples per second, (44.1 kHz), with 65,536 allowable values for volume (16 bits, giving values).
A single ("mono") channel (or "track") of digital audio can be visualized as a waveform, and is composed of a series of sampled sound pressures. Each sample has been quantized to one out of a set of discrete values depending on the bit depth. The playback of the waveform reproduces the sound pressure changes in sequence, producing complex sounds.
"Bit depth" determines the number of alternative values for sampled sound pressure that can be represented. The "sampling rate" (or frequency) refers to the number of samples (or "snapshots") per second, measured in hertz (Hz).
A stereo "track" is simply two isolated mono tracks within the same file, which are played back simultaneously. The illusion of spatial sound can be created through differences between the left and right channels. High-quality motion picture sound formats, like Dolby Surround and DTS simply use more mono channels to create more complex illusions of spatial depth.
Increasing the I/O capacity for either or both bit resolution and sampling rate will increase the quality of the sound. Multiple tracks can enhance the experience of spatial sound and sound "quality."
"Professional" sound quality refers to highest current sound qualities for available hardware. In the past, very high resolution hardware was also highly specialized and expensive. Today, many commonly available technologies are sufficient for generating adequate or above average quality digital audio signals for use in various recording or transmission applications.
Currently it's possible to purchase hardware using reasonably high quality digital converters for well under 1000 USD. In professional uses, extremely high fidelity converters are typically sold as components within specialized rack mount multi-channel (multitrack) recording units. These typically connect to a computer via parallel, proprietary or specialized (i.e. PCI card), fiber optic, USB, or Firewire connections. USB 2 and Firewire are the most common, though Firewire has lower latency and thus is more suited for multitrack applications.
At the barebones level, audio card converters take a nominal line-level input signal and is designed to be a transparent link between the mixing board and the computer. Once converted to digital format, the sound files can be processed in any of a number of ways as allowed by software, including converting to audio CD or MP3 formats.
Both the recording and playback of sound must consider the nature of the digital medium to produce undesirable noise. The noise produced by analog tape, by comparison can be pleasing, as it is of sufficient resolution to produce "warm" natural sounding random noise. Digital recording has traditionally had the problem of sounding "cold," mainly due to the quantization effect of digitizing a linear wave into a segmented series of adjacent samples. In the case of long wave sounds, such as bass, the wave is smooth, and therefore its playback does not "jump" from a low volume bit to an adjacent high volume bit. However, high pitched sounds like cymbal crashes and similar "white" noise can only be represented by a virtually random array of bits. This random array exposes the weakness of digital playback to produce digital artifacts in the form of a high-pitched "harshness."
While better resolutions produce better sound and less digital harshness, an important function of digital converters is to minimize the known problems endemic to the digital format. A common solution was to use separate vacuum tube amplifiers to "warm" the sound at both input output. This is generally expensive, and normalizes the sound to the range given by the tube amp, which negates the benefits of digital's deep low and clear high frequencies. The main solution has generally been to use an oversampling process, which can differ substantially depending on its application for input or output. Oversampling is the processing of sound input at higher sampling rates than the hardware actually performs. On input, this produces a digital wave which is still based on the input, but altered to record substantially fewer jumps which would produce audible artifacts. On the output, oversampling serves the same role as using a tube amp, whereby the adjacent bits are re-quantized at higher resolution to produce a smoother wave.
Given the capabilities of the digital realm to transfer exact files without analog loss, it is rare that multitrack digital to analog (DAC) converters are needed, with the exception of basic playback monitoring of stereo or specialized surround sound formats.