Analog recording vs.

digital recording
Analog recording versus digital recording compares the two ways in which sound is recorded and stored.
Actual sound waves consist of continuous variations in air pressure. Representations of these signals can
be recorded using either digital or analog techniques.

An analog recording is one where a property or characteristic of a physical recording medium is made to
vary in a manner analogous to the variations in air pressure of the original sound. Generally, the air
pressure variations are first converted (by a transducer such as a microphone) into an electrical analog
signal in which either the instantaneous voltage or current is directly proportional to the instantaneous
air pressure (or is a function of the pressure). The variations of the electrical signal in turn are converted
to variations in the recording medium by a recording machine such as a tape recorder or record cutter—
the variable property of the medium is modulated by the signal. Examples of properties that are
modified are the magnetization of magnetic tape or the deviation (or displacement) of the groove of a
gramophone disc from a smooth, flat spiral track.

A digital recording is produced by converting the physical properties of the original sound into a
sequence of numbers, which can then be stored and read back for reproduction. Normally, the sound is
transduced (as by a microphone) to an analog signal in the same way as for analog recording, and then
the analog signal is digitized, or converted to a digital signal, through an analog-to-digital converter and
then recorded onto a digital storage medium such as a compact disc or hard disk.

Both analog and digital systems have limitations. The bandwidth of the digital system is limited,
according to the Nyquist frequency, by the sample rate used. The bandwidth of an analog system is
limited by the physical capabilities of the analog circuits and recording medium. The signal-to-noise ratio
(S/N) of a digital system is limited by the bit depth of the digitization process. In an analog system, other
natural analog noise sources such as flicker noise and imperfections in the recording medium.

Main differences
It is a subject of debate whether analog audio is superior to digital audio or vice versa. The question is
highly dependent on the quality of the systems (analog or digital) under review, and other factors which
are not necessarily related to sound quality. Arguments for analog systems include the absence of
fundamental error mechanisms which are present in digital audio systems, including aliasing,
quantization noise, and the absolute limitation of dynamic range. Advocates of digital point to the high
levels of performance possible with digital audio, including excellent linearity in the audible band and
low levels of noise and distortion (Sony Europe 2001).

Accurate, high quality sound reproduction is possible with both analog and digital systems. Excellent,
expensive analog systems may outperform digital systems, and vice versa; in theory any system of either
type may be surpassed by a better, more elaborate and costly system of the other type,[citation
needed] but in general it tends to be less expensive to achieve any given standard of technical signal
quality with a digital system, except when the standard is very low. One of the most limiting aspects of
analog technology is the sensitivity of analog media to minor physical degradation; however, when the
degradation is more pronounced, analog systems usually perform better, often still producing
recognizable sound, while digital systems will usually fail completely, unable to play back anything from
the medium. (See digital cliff.) The principal advantages that digital systems have are very uniform
source fidelity, inexpensive media duplication (and playback) costs, and direct use of the digital 'signal'
in today's popular portable storage and playback devices. Analog recordings by comparison require
comparatively bulky, high-quality playback equipment to capture the signal from the media as
accurately as digital.

Early in the development of the Compact Disc, engineers realized that the perfection of the spiral of bits
was critical to playback fidelity. A scratch the width of a human hair (100 micrometres) could corrupt
several dozen bits, resulting in at best a pop, and far worse, a loss of synchronization of the clock and
data, giving a long segment of noise until resynchronized. This was addressed by encoding the digital
stream with a multi-tiered error-correction coding scheme which reduces CD capacity by about 20%, but
makes it tolerant to hundreds of surface imperfections across the disk without loss of signal. In essence,
"error correction" can be thought of as "using the mathematically encoded backup copies of the data
that was corrupted." Not only does the CD use redundant data, but it also mixes up the bits in a
predetermined way (see CIRC) so that a small flaw on the disc will affect fewer consecutive bits of the
decoded signal and allow for more effective error correction using the available backup information.

Error correction allows digital formats to tolerate quite a bit more media deterioration than analog
formats. That is not to say poorly produced digital media are immune to data loss. Laser rot was most
troublesome to the Laserdisc format, but also occurs to some pressed commercial CDs, and was caused
in both cases by inadequate disc manufacture. (Note that Laserdisc, despite using a laser optical system
that has become commonly associated with digital disc formats, is an old analog format, except for its
optional digital audio tracks; the video image portion of the content is always analog.) There can
occasionally be difficulties related to the use of consumer recordable/rewritable compact discs. This
may be due to poor-quality CD recorder drives, low-quality discs, or incorrect storage, as the
information-bearing dye layer of most CD-recordable discs is at least slightly sensitive to UV light and
will be slowly bleached out if exposed to any amount of it. Most digital recordings rely at least to some
extent on computational encoding and decoding and so may become completely unplayable if not
enough consecutive good data is available for the decoder to synchronize to the digital data stream,
whereas any intact fragment of any size of an analog recording is usually playable.

Unlike analog duplication, digital copies are usually exact replicas, which can be duplicated indefinitely
without degradation, unless imposed DRM restrictions apply or mastering errors occur. Digital systems
often have the ability for the same medium to be used with arbitrarily high or low quality encoding
methods and number of channels or other content, unlike practically all analog systems which have
mechanically pre-fixed speeds and channels. Most higher-end analog recording systems offer a few
selectable recording speeds, but digital systems tend to offer much finer variation in the rate of media
usage.

There are also several non-sound related advantages of digital systems that are practical. Digital systems
that are computer-based make editing much easier through rapid random access, seeking, and scanning
for non-linear editing. Most digital systems also allow non-audio data to be encoded into the digital
stream, such as information about the artist, track titles, etc., which is often convenient. (However, it is
technically possible, and not difficult, to implement analog systems with integrated digital metadata
channels. In fact, it is possible to record digital metadata onto one track of an analog multitrack
recording using any home computer from the 1980s, such as a Commodore 64, that can record data on
cassettes.)

Noise and distortion

In the process of recording, storing and playing back the original analog sound wave (in the form of an
electronic signal), it is unavoidable that some signal degradation will occur. This degradation is in the
form of linear errors (consistent changes of the amplitude or phase within a specified passband) and
non-linear errors (noise and distortion). Noise is unrelated in time to the original signal content, while
distortion is in some way related in time to the original signal content.

Digital fundamentals
A digital recorder firstly requires the input of an analog signal; this signal may come directly from a
microphone pre-amp, but any analog audio signal can be converted. Measurements of the signal
intensity are then made at regular intervals (sampling) by the analog-to-digital converter. At each
sampling point, the signal must be assigned a specific intensity from a set range of values (quantization).
For doing this, the original sound wave can now be described using only numbers—as digital
information. Each sample can be given an ordinal number which signifies the point in relative time that
it represents, and the magnitude of the sample is an analog of pressure at the microphone (Watkinson
1994) (or, for an artificial sound signal, the pressure that would be at the microphone to correspond to
that sound.) When the original signal is converted into numbers (usually binary numbers, 1's and 0's,
called 'bits') further additions of noise and distortion, provided they are not great enough to cause
digital errors, can be rejected at every stage of processing; this is what is referred to as the regenerative
nature of digital signals. Digital errors, called bit errors in binary digital systems, are events of noise
and/or distortion which cause one number (or bit) to appear more like another number than like the
number it started out as. As long as a digital symbol appears closer to being what it began as than to
anything else, it can be regenerated. When raw digital errors cannot be avoided, error correction coding
can allow some of them to be detected and fixed. Error correction, essential when transferring digital
audio over noisy channels, helps to eliminate bit errors by comparing extra data against the main data to
detect limited numbers of digital errors, figure out which digital symbols (numbers) were changed, and
change them back. When playing back a digital recording, the digital information is converted back into
a continuous, analog signal by a digital-to-analog converter. This electronic signal is then amplified and
converted back into a sound wave by a loudspeaker (just as would be done with the analog signal
produced by an analog machine from an analog recording).

Noise performance
For electronic audio signals, sources of noise include (unavoidable) mechanical, electrical and thermal
noise in the recording and playback cycle (from mechanical transducers (microphones, loudspeakers),
amplifiers, recording equipment, the mastering process, reproduction equipment, etc.). Whether an
audio signal is, at some stage, converted into a digital form will affect how much effective noise is
added, due to the partial immunity to noise that the digital regenerative effect provides. The actual
process of digital conversion will always add some noise, however small in intensity; the bulk of this in a
high-quality system is quantization noise, which cannot be theoretically avoided, but some will also be
electrical, thermal, etc. noise from the analog-to-digital converted device.

The amount of noise that a piece of audio equipment adds to the original signal can be quantified.
Mathematically, this can be expressed by means of the signal to noise ratio (SNR). Sometimes the
maximum possible dynamic range of the system is quoted instead. In a digital system, the number of
quantization levels, in binary systems determined by and typically stated in terms of the number of bits,
will have a bearing on the level of noise and distortion added to that signal. The 16-bit digital system of
Red Book audio CD has 216= 65,536 possible signal amplitudes, theoretically allowing for an SNR of 98
dB (Sony Europe 2001) and dynamic range of 96 dB.

In order to meet the theoretical maximum performance of a 16 bit digital system, for a 0.5 V peak to
peak input line signal, a PCM (pulse code modulation) quantizer would require an equivalent minimum
input sensitivity of just 7.629 microvolts. For an analog recorder, this is equivalent to a 15.3 ppm
sensitivity for the whole recording system and medium.[citation needed] With digital systems, the
quality of reproduction depends on the analog-to-digital and digital-to-analog conversion steps, and
does not depend on the quality of the recording medium, provided it is adequate to retain the digital
values without an excessive error rate (exceeding the capacity of any error correction mechanisms used
in the system).

Typically anything below 14 bits can lead to perceptibly reduced sound quality, with 80 dB of SNR
considered as an informal "minimum" for Hi-Fi audio. However, it is uncommon to find digital media
specified for less than 14 bits, except for older 12-bit PCM Camcorder audio (or DAT in long-play, 32 kHz
mode) and the output from older or lower-cost computer software, sound cards/circuitry, consoles and
games (typically 8 bit as a minimum and standard, though trick sample output methods for generally
non-PCM hardware [e.g. FM synthesis cards including the Adlib card] gave SNR performances closer to
that of an ideal "6" or "4" bit PCM digital converter).

Each additional quantization bit theoretically adds 6 dB in possible dynamic range, e.g. 24 x 6 = 144 dB
for 24 bit quantization, 126 dB for 21-bit, and 120 dB for 20-bit.

Analog systems
Consumer analog cassette tapes may have a dynamic range of 60 to 70 dB. Analog FM broadcasts rarely
have a dynamic range exceeding 50 dB. The dynamic range of a direct-cut vinyl record may surpass 70
dB. Analog studio master tapes using Dolby-A noise reduction can have a dynamic range of around 80
dB.
Rumble
"Rumble" is a form of noise characteristic of poor or worn turntables. Because of imperfections in the
bearings of turntables, the platter tends to have a slight amount of motion besides the desired rotation
—the turntable surface also moves up-and-down and side-to-side slightly. This additional motion is
added to the desired signal as noise, usually of very low frequencies, creating a "rumbling" sound during
quiet passages. Very inexpensive turntables sometimes used ball bearings which are very likely to
generate audible amounts of rumble. More expensive turntables tend to use massive sleeve bearings
which are much less likely to generate offensive amounts of rumble. Increased turntable mass also tends
to lead to reduced rumble. A good turntable should have rumble at least 60 dB below the specified
output level from the pick-up (Driscoll 1980:79-82).

Wow and flutter

Wow and flutter are the result of imperfections in the mechanical performance of analog devices. Wow
and flutter are most noticeable on signals which contain pure tones. As an example, 0.22% (rms) wow
may be detectable by listeners with piano music, but this increases to 0.56% with jazz music. For LP
records, the quality of the turntable will have a large effect on the level of wow and flutter. A good
turntable will have wow and flutter values of less than 0.05%, which is the speed variation compared to
the mean value (Driscoll 1980). Wow and flutter can also be present in the recording, as a result of the
imperfect operation of the recorder, but in commercial published recordings the inbuilt wow and flutter
is usually very low, much lower than the wow and flutter generated by the playback equipment.

Frequency response
The frequency response of audio CD is sufficiently wide to cover the entire normal audible range, which
roughly extends from 20 Hz to 20 kHz. (Hearing varies among individuals, and some can hear
frequencies slightly beyond these limits.) Commercial and industrial digital recorders record higher
frequencies, while consumer systems inferior to the CD record a more restricted frequency range.
Analog audio is unrestricted in its possible frequency response, but the limitations of the particular
analog format will provide a cap.

For digital systems, the maximum audio frequency response is "hardcoded" by the sampling frequency.
The choice of sample rate used in a digital system is based on the Nyquist-Shannon sampling theorem.
This states that a sampled signal can be reproduced exactly as long as it is sampled at a frequency
greater than twice the bandwidth of the signal. Therefore a sampling rate of 40 kHz would be enough to
capture all the information contained in a signal having frequency bandwidth up to 20 kHz. The difficulty
arises in removing all the signal content above 20 kHz, and unless this is done, aliasing of these higher
frequencies may occur. The result then is that these higher, inaudible frequencies alias to frequencies
which are in the audible range, producing a kind of distortion. To prevent aliasing, it is not necessary to
design a brick-wall anti-aliasing filter - that is a filter which perfectly removes all frequency content
above (or below) a certain cutoff frequency. (It is impossible to build a filter with a perfectly square
cutoff characteristic, as the filter would have an impulse response which is a sinc function and so is not
causal.) Instead, a sample rate is usually chosen which is above the theoretical requirement. This is
called oversampling, and allows a less severe (and less expensive) anti-aliasing filter to be used.

High quality open-reel tape frequency response can extend from 10 Hz to well above 20 kHz. The
linearity of the response may be indicated by providing information on the level of the response relative
to a reference frequency. For example, a system component may have a response given as 20 Hz to 20
kHz +/- 3 dB relative to 1 kHz. Some analog tape manufacturers specify frequency responses up to 20
kHz, but these measurements may have been made at low signal levels (Driscoll 1980). High-quality
metal-particle compact cassettes may have a response extending up to 14 kHz at full (0 dB) recording
level (Stark 1989). At lower levels, cassettes typically are limited at the upper end to around 17 kHz for
the best machines, due to the nature of the tape media and the tape speed chosen by Philips for the
format (which was originally designed for dictation.)

The frequency response for a conventional LP player might be 20 Hz - 20 kHz +/- 3 dB. Unlike the audio
CD, vinyl records (and cassettes) do not require a cut-off in response above 20 kHz. The low frequency
response of vinyl records is restricted by rumble noise (described above). The high frequency response
of vinyl depends on the record itself and on the cartridge. CD4 records contained frequencies up to 50
kHz, while some high-end turntable cartridges have frequency responses of 120 kHz while having flat
frequency response over the audible band (e.g. 20 Hz to 15 kHz +/-0.3 dB).[1] In addition, frequencies of
up to 122 kHz have been experimentally cut on LP records.[2]

In comparison, the CD system offers a frequency response of 20 Hz – 20 kHz ±0.5 dB, with a superior
dynamic range over the entire audible frequency spectrum (Sony Europe 2001).

With vinyl records, there will be some loss in fidelity on each playing of the disc. This is due to the wear
of the stylus in contact with the record surface. A good quality stylus, matched with a correctly set up
pick-up arm, should cause minimal surface wear. Magnetic tapes, both analog and digital, wear from
friction between the tape and the heads, guides, and other parts of the tape transport as the tape slides
over them. The brown residue deposited on swabs during cleaning of a tape machine's tape path is
actually particles of magnetic coating shed from tapes. Tapes can also suffer creasing, stretching, and
frilling of the edges of the plastic tape base, particularly from low-quality or out-of-alignment tape
decks. When a CD is played, there is no physical contact involved, and the data is read optically using a
laser beam. Therefore no such media deterioration takes place, and the CD will, with proper care, sound
exactly the same every time it is played (discounting aging of the player and CD itself); however, this is a
benefit of the optical system, not of digital recording, and the Laserdisc format enjoys the same non-
contact benefit with analog optical signals. Recordable CDs slowly degrade with time, called disc rot,
even if they are not played, and are stored properly.

Analog advantages
It can be argued that analog formats retain some inherent advantages over digital formats. The relevance
of these advantages depends on the quality of specific digital or analog equipment. The advantages of
analog systems are summarised below:
  Absence of aliasing distortion
 Absence of quantization noise
 Behaviour in overload conditions

Aliasing
Unlike digital audio systems, analog systems do not require filters for bandlimiting. These filters act to
prevent aliasing distortions in digital equipment. Early digital systems may have suffered from a number of
signal degradations related to the use of analog anti-aliasing filters, e.g., time dispersion, nonlinear
distortion, temperature dependence of filters etc. (Hawksford 1991:8).

Jitter
One aspect that may prevent the performance of practical digital systems from meeting their theoretical
performance is jitter. This is the name given to the phenomenon of the variations in spacing of the
discrete samples in time within the stream of samples that make up a (decoded) digital signal. This can
be due to timing inaccuracies of the digital clock. Ideally a digital clock should produce a timing pulse at
exactly regular intervals. Other sources of jitter within digital electronic circuits are data-induced jitter,
where one part of the digital stream affects a subsequent part as it flows through the system, and power
supply induced jitter, where DC ripple on the power supply output rails causes irregularities in the timing
of signals in circuits powered from those rails.

The accuracy of a digital system is dependent on the sampled values, known as quantised values, which
exist in the amplitude realm, but it is also dependent on the timing regularity of the discrete values which
exist in the temporal realm. This dependency on accuracy of discrete values in the temporal realm is
inherent to digital recording and playback and has no analog equivalent, though analog systems have
their own temporal distortion effects (pitch error and wow-and-flutter).

Periodic jitter produces modulation noise and can be thought of as being the equivalent of analog flutter
(Rumsey & Watkinson 1995). Random jitter alters the noise floor of the digital system. The sensitivity of
the converter to jitter depends on the design of the converter. It has been shown that a random jitter of 5
ns (nanoseconds) may be significant for 16 bit digital systems (Rumsey & Watkinson 1995). For a more
detailed description of jitter theory, refer to Dunn (2003).

Quantization noise
Analog systems do not have discrete digital levels in which the signal is encoded. Consequently, the
original signal can be preserved to an accuracy limited only by the intrinsic noise-floor and maximum
signal level of the media and the playback equipment, i.e., the dynamic range of the system. With digital
systems, noise added due to quantization into discrete levels is more audibly disturbing than the noise-
floor in analog systems. This form of distortion, sometimes called granular or quantization distortion, has
been pointed to as a fault of some digital systems and recordings (Knee & Hawksford 1995, Stuart n.d.:6).
Knee & Hawksford (1995:3) drew attention to the deficiencies in some early digital recordings, where the
digital release was said to be inferior to the analog version. The quantization noise level is directly
determined by the number of bits of quantization resolution, decreasing exponentially with it (or linearly in
dB units), and with an adequate number of true bits of quantization, random noise from other sources will
dominate and completely mask the quantization noise.

Counter-arguments
Aliasing distortion
The mentioned disadvantages of digital audio systems have been the subject of discussion. With regard
to aliasing distortion, Hawksford (1991:18) highlighted the advantages of digital converters which operate
at higher than the Nyquist rate (i.e., oversampling converters). Using an oversampling design and a
modulation scheme called sigma-delta modulation (SDM), analog anti-aliasing filters can effectively be
replaced by a digital filter. This approach has several advantages. The digital filter can be made to have a
near-ideal transfer function, with low in-band ripple, and no aging or thermal drift.

Quantization
It is possible to make quantization noise more audibly benign by applying dither. To do this, a noise-like
signal is added to the original signal before quantization. Dither makes the digital system behave as if it
has an analog noise-floor. Optimal use of dither (triangular probability density function dither in PCM
systems) has the effect of making the rms quantization error independent of signal level (Dunn 2003:143),
and allows signal information to be retained below the least significant bit of the digital system (Stuart
n.d.:3).

Sound quality
Subjective evaluation
Subjective evaluation attempts to measure how well an audio component performs according to the
human ear. The most common form of subjective test is a listening test, where the audio component is
simply used in the context for which it was designed. This test is popular with hi-fi reviewers, where the
component is used for a length of time by the reviewer who then will describe the performance in
subjective terms. Common descriptions include whether the component has a 'bright' or 'dull' sound, or
how well the component manages to present a 'spatial image'.

Another type of subjective test is done under more controlled conditions and attempts to remove possible
bias from listening tests. These sorts of tests are done with the component hidden from the listener, and
are called blind tests. To prevent possible bias from the person running the test, the blind test may be
done so that this person is also unaware of the component under test. This type of test is called a double-
blind test. This sort of test is often used to evaluate the performance of digital audio codecs.

There are critics of double-blind tests who see them as not allowing the listener to feel fully relaxed when
evaluating the system component, and can therefore not judge differences between different components
as well as in sighted (non-blind) tests. Those who employ the double-blind testing method may try to
reduce listener stress by allowing a certain amount of time for listener training (Borwick et al. 1994:481-
488).

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 significant difference was found
between sounds with and without very high frequency components among the sound stimuli and the
subjects... however, [Nishiguchi et al] can still neither confirm nor deny the possibility that some subjects
could discriminate between musical sounds with and without very high frequency components."

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 [5]. Listeners involved in this test noted their great difficulty in
hearing any difference between the two formats.
Analog warmth
Some audio enthusiasts prefer the sound of vinyl records over that of a CD. Founder and editor Harry
Pearson of The Absolute Sound journal says that "LPs are decisively more musical. CDs drain the soul
from music. The emotional involvement disappears". Dub producer Adrian Sherwood has similar feelings
about the analog cassette tape, which he prefers because of its warm sound [6].

Those who favour the digital format point to the results of blind tests, which demonstrate the high
performance possible with digital recorders[7]. The assertion is that the 'analog sound' is more a product
of analog format inaccuracies than anything else. One of the first and largest supporters of digital audio
was the classical conductor Herbert von Karajan, who said that digital recording was "definitely superior
to any other form of recording we know". He also pioneered the unsuccessful Digital Compact Cassette
and conducted the first recording ever to be commercially released on CD: Richard Strauss's Eine
Alpensinfonie.