Digital to analog converter

Overview

The term "digital to analog converter" is used to describe a device that accepts a digital audio input and outputs an analog audio signal that is re-constructed from the digital code. This code is typically linear PCM format; but may also be other formats such as DSD or I2S (typically used internally in digital to analog converter units). A digital to analog converter must be used to listen to a digital audio signal. The term can be used to describe the actual digital to analog converter IC or circuit, or an entire unit that incorporates all of the necessary support circuitry to accept the encoded digital audio signal in one or more formats and output line level analog audio signals.

For brevity, the term "DA converter" or "DAC" will be used interchangeably with "digital to analog converter" in the following discussion.

History

Prior to the development of practical digital audio recording systems; DA converters were used primarily in industrial control applications. These early converters were limited either by the converter technology at the time or by the amount of data that associated system could handle to much lower resolution than typically used to encode audio. The resolution both in the amplitude domain (typically voltage of the input waveform) and time domain of these converters was often quite limited when compared to contemporary digital audio standards.

Before storage of the huge amount information generated by CD quality AD converters became practical, the earliest application of DA converters in music recording was in "outboard" equipment such as digital delay or effects processors. Largely because the output of these early units was mixed in with the original (unprocessed) source at a low level as an ambient effect; the less-than high fidelity quality of the converters was acceptable. Even with the noise and distortion present in analog recordings, the perceived quality of the analog tape recordings was far better than the signal processed through these early converters. One of the more popular early digital delay units employed a novel for of digital encoding "sigma-delta" where, in contrast to the "linear PCM" format where each "sample" of the analog input waveform is represented by a digital word made of a number of bits; sigma-delta encoded only one bit at a relatively high sample frequency. Compared to the relatively inaccurate PCM-based units, most recording engineers felt that the sigma-delta digital delay unit sounded closer to the source.

With the introduction of Compact Disc technology by Sony/Phillips in the early 1980's came the standard of recording audio in 16 bit linear PCM format. DA converter technology was still evolving at the time and even though many DA converters were nominally "16 bit" they were not truly accurate to 16 bit resolution. Contemporary DA converters are typically "24 bit." The sample frequency capability of DA converters has also increased since the original CD format of 44.1 kHz was introduced; with contemporary DA converters supporting output sample frequencies as high as 384 kHz. Although there are a number of advantages to DA conversion at sample frequencies higher than 44.1 kHz, these advantages are gained at sample frequencies of 88.2 or 96 kHz. Increasing the sample frequency beyond 96 kHz will degrade the conversion accuracy in the audio frequency range, while the only advantage is the ability to reproduce supersonic frequencies beyond the range even dogs can hear.

Basics

In order to make a useful digital audio system; the method used to encode and decode the analog audio signal must: 1.) Be reciprocal for encoding (recording) and decoding (playback). 2.) Must be able to "re-construct" the original analog information to a minimum level of accurately. 3.) Ideally incorporates a "standard" that facilitates interchange between systems made by different manufacturers.

To achieve (1), contemporary digital audio systems use a method referred to as "sampling" which, in a manner analogous to film or video cameras, takes a contiguous series of "snapshots" of the audio waveform at a specific frequency (the sample frequency). Analog audio derives its name from the manner in which the acoustic pressure variation of the original sound is represented by a voltage waveform with the same variations- the voltage variation is "analogous" to the pressure variation at every point in time. Although it is possible that at specific points in an audio system the signal is represented by current variations as versus voltage variations; the analog signal is typically a voltage waveform when it is transmitted from one piece of audio equipment to another.

The digital "words" are recorded in sequence as a file, and can be stored or transmitted without change to the information. In order for the playback DA to accurately reconstruct the voltage waveform; it must output the voltage of each sample at exactly the same voltage level and exactly the same relative time. This means the sample frequency must be very close to the same frequency and, more importantly, the sample clock must have extremely even time periods for each sample. This where the discussion of "jitter" comes in- jitter is the term that is used to describe short-term variations in the clock cycle period caused by real-world issues common to the transmission of very high frequency signals over signal conductors (cables or even signal "traces" on printed circuit boards). Although voltage (amplitude domain) accuracy has increased dramatically since the early days of digital audio; the performance of even extremely accurate converters can be compromised by inaccurate clocking of the conversion either during AD conversion, during DA conversion, or both.

A typical DA converter is actually a system made up of a number of stages: a.) The digital input circuitry b.) The digital signal processor c.) The clock circuitry d.) The DA converter e.) The level-shifting stage f.) The output filter g.) Line output stage