Early digital recordings

Early digital audio machines had disappointing results, with digital converters introducing errors that the ear could detect (Watkinson 1994). Record companies released their first LPs based on digital audio masters in the late 1970s. CDs became available in the early 1980s. At this time analog sound reproduction was a mature technology.

There was a mixed critical response to early digital recordings released on CD. Compared to vinyl record, it was noticed that CD was far more revealing of the acoustics and ambient background noise of the recording environment (Greenfield et al. 1986). For this reason, recording techniques developed for analog disc, e.g., microphone placement, needed to be adapted to suit the new digital format (Greenfield et al. 1986).

Some analog recordings were remastered for digital formats. Analog recordings made in natural concert hall acoustics tended to benefit from remastering (Greenfield et al. 1990). The remastering process was occasionally criticised for being poorly handled. When the original analog recording was fairly bright, remastering sometimes resulted in an unnatural treble emphasis (Greenfield et al. 1990).

Higher sampling rates

CD quality audio is sampled at 44.1 kHz (Nyquist frequency = 22.05 kHz) and at 16 bits. Sampling the waveform at higher frequencies and allowing for a greater number of bits per sample allows noise and distortion to be reduced further. DAT can store audio at up to 48 kHz, while DVD-Audio can be 96 or 192 kHz and up to 24 bits resolution. With any of these sampling rates, signal information is captured above what is generally considered to be the human hearing range.

Work done in 1980 by Muraoka et al. (J.Audio Eng. Soc., Vol 29, pp2–9) showed that music signals with frequency components above 20 kHz were only distinguished from those without by a few of the 176 test subjects (Kaoru & Shogo 2001). Later papers, however, by a number of different authors, have led to a greater discussion of the value of recording frequencies above 20 kHz. Such research led some to the belief that capturing these ultrasonic sounds could have some audible benefit. Audible differences were reported between recordings with and without ultrasonic responses. Dunn (1998) examined the performance of digital converters to see if these differences in performance could be explained [4]. He did this by examining the band-limiting filters used in converters and looking for the artifacts they introduce.

A perceptual study by Nishiguchi et al. (2004) concluded that no perceivable difference could be found between music signals with and without frequency components above 21 kHz. They were, however, unable to say whether or not some subjects could perceive a difference, and felt that further evaluation tests were necessary [5].

Super Audio cd and dvd Audio

The Super Audio CD (SACD) format was created by Sony and Philips, who were also the developers of the earlier standard audio CD format. SACD uses Direct Stream Digital, which works quite differently from the PCM format discussed in this article. Instead of using a greater number of bits and attempting to record a signal's precise amplitude for every sample cycle, a Direct Stream Digital recorder works by encoding a signal in a series of PWM pulses of fixed amplitude but variable duration and timing. The competing DVD-Audio format uses standard, linear PCM at variable sampling rates and bit depths, which at the very least match and usually greatly surpass those of a standard CD Audio (16 bits, 44.1 kHz).

A Direct Stream Digital (DSD) recorder uses sigma-delta modulation. Originally DSD recorders operated at 64 times the Nyquist rate (44.1 kHz), at around 3 MHz. The output from a DSD recorder alternates between levels representing 'on' and 'off' states, and is a binary signal (called a bitstream). The long-term average of this signal is proportional to the original signal. In principle, the retention of the bitstream in DSD allows the SACD player to use a basic (one bit) DAC design which incorporates a low-order analog filter.

There are fundamental distortion mechanisms present in the conventional implementation of DSD (Hawksford 2001). These distortion mechanisms can be alleviated to some degree by using digital converters with a multibit design. Historically, state-of-the-art ADCs were based around sigma-delta modulation designs. Oversampling converters are frequently used in linear PCM formats, where the ADC output is subject to bandlimiting and dithering (Hawksford 1995). Many modern converters use oversampling and a multibit design.

In the popular Hi-Fi press, it has been suggested that linear PCM "creates [a] stress reaction in people", and that DSD "is the only digital recording system that does not [...] have these effects" (Hawksford 2001). A double-blind subjective test between high resolution linear PCM (DVD-Audio) and DSD did not reveal a statistically significant difference [6]. Listeners involved in this test noted their great difficulty in hearing any difference between the two formats.