- Analog sound vs. digital sound
Analog sound versus digital sound compares the two ways in which
soundis recorded and stored. Actual sound waves consist of continuous variations in air pressure. Representations of these signals can be recorded in either digitalor analog formats.
analog recordingis one where the original sound signal is modulated onto another physical medium or substrate such as the groove of a gramophone discor the iron oxide surface of a magnetic tape. A physical quality in the medium (e.g., the intensity of the magnetic field or the path of a record groove) is directly related, or analogous, to the physical properties of the original sound (e.g., the amplitude, phase, etc.)
digital recordingis produced by converting the physical properties of the original sound into a sequence of numbers, which can then be stored and played back for reproduction. The accuracy of the conversion process depends on the sampling rate (how often the sound is sampled and a related numerical value is created) and the sampling depth (how much information each sample contains, which can also be described as the maximum numerical size of each sampled value). However, unlike analog recording which depends critically on the long-term durability of the "fidelity of the waveforms" recorded on the medium, the physical medium storing digital samples is essentially immaterial in playback of the encoded information so long as the original sequence of numbers can be recovered.
It is a subject of debate whether or not 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, [http://www.findarticles.com/p/articles/mi_m1430/is_n5_v17/ai_16368605/pg_1] and supposed limitations in 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. One of the most limiting aspects of analog technology is the sensitivity of analog media to physical degradation. The principal advantages that digital systems have is very uniform source fidelity, inexpensive media duplication (and playback) costs, 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.
Analog equipment imperfections may cause distortions like wow, flutter,
tape hissor when the medium becomes worn (as in the case of a vinyl record), surface noise. Some of these distortions can be addressed using timebase correction, as is done in VHStapes, filtering or high-quality components. Time-instability in PCMdigital systems ( jitter) may be audible on some signals, particularly sinusoids (Rumsey & Watkinson 1995, Dunn 2003:34). As of 2008, all audiophile and consumer grade digital systems now encode the clock (which if independent from the bit stream is the source of jitter) into the coded data itself.
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 microns) 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."
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 rotwas most troublesome to the Laserdiscformat, and was caused by inadequate disc manufacture. 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 or low-quality discs.
Unlike analog duplication, digital copies are usually exact clones, which can be duplicated indefinitely without degradation, unless DRM restrictions apply or
masteringerrors occur. Digital systems 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 mechanically pre-fixed speed and channels of practically all analog systems.
There are also several non-sound related advantages of digital systems that are practical. Non-sequential (random) access, owing to their disk or memory-based nature, makes editing much easier and allowing the listener greater flexibility when choosing tracks. Most digital systems also allow non-audio data to be encoded into the digital stream, such as information about the artist, track titles, etc.
Noise and distortion
In the process of recording, storing and playing back the original sound wave analogy (in the form of an electronic signal), it is unavoidable that some signal degradation will occur. This degradation is in the form of linear (changes to the amplitude or phase response 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.
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). In doing this, the original sound wave can now be described using only numbers - as digital information. The number of the sample is an analog of time, and the magnitude of the sample is an analog of pressure at the microphone (Watkinson 1994). When the original signal is converted into binary numbers (1's and 0's, called 'bits') further additions of noise and distortion (in the form of digital errors) can be rejected at every stage of processing. Error correctioncoding, essential when transferring digital audio over noisy channels, helps to eliminate bit errors. 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.
For electronic audio signals, sources of noise include (unavoidable) mechanical, electrical and thermal noise level in the recording and playback cycle (mechanical transducers (microphones,
loudspeakers), amplifiers, recording equipment, mastering process, reproduction equipment, etc). Whether an audio signal is, at some stage, converted into a digital form will affect how much noise is added. The actual process of digital conversion will always add some noise, however small in intensity.
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 bits with which a signal is allowed to have on quantization 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.
* Note that a
decibel(dB) is one-tenth of a bel. It is a somewhat strange concept that characterizes the logarithmic nature of human senses. Now to make it more complex, the amplitudes discussed in this article are voltage levels. To convert a voltage level ratio to a bel, simply divide them and calculate the logarithmto base 10. Then multiply by 10 to get decibels. Unfortunately, Ohm's Lawcomes into play; the power of the sound is approximately the square of the voltage level. The human hearing range is around 120 dB.
In order to meet the theoretical performance of a 16 bit digital system, for a 0.5 V
peak to peakinput 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 by part of the whole recording system and medium. 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. Typically anything below 14 bits can lead to 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 Camcorderaudio (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 gave SNR performances closer to that of an ideal "6" or "4" bit PCM digital converter).
Greater than 16 bits
Each additional quantization bit theoretically adds a notable 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. One aspect that may prevent the performance of practical digital systems from meeting their theoretical performance is jitter. This is caused by deviations in the sampling of the waveform from ideal performance, and is usually expressed as a time value. Random jitter alters the noise floor of the digital system. It has been shown that a random jitter of 5 ns (nanoseconds) may be significant for 16 bit digital systems (Rumsey & Watkinson 1995). Systems of greater than 16 bits need performances higher than this (lower jitter meaning levels less than 5 ns) to meet their theoretical noise floors. Audibility tests have shown that the detection threshold for random jitter in musical signals is several hundred nanoseconds [http://www.jstage.jst.go.jp/article/ast/26/1/26_50/_article] .
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 recordmay surpass 70 dB. Analog studio master tapes using Dolby-A noise reduction can have a dynamic range of around 80 dB (Stark 1989).
"Rumble" is a form of noise peculiar to turntables. Because of imperfections in the bearings of turntables, the platter tends to have a slight amount of motion other than just the desired rotation. That is, besides its 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 massalso 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 ideal value (Driscoll 1980).
The digital equivalent of flutter is periodic jitter, which is caused by instablities in the sample clock of the converter (Rumsey & Watkinson 1995). The sensitivity of the converter to periodic jitter depends on the design of the converter. Periodic jitter produces modulation noise. Practical research by Benjamin and Gannon involving listening tests found that the lowest level of jitter to be audible on test signals was 10 ns (rms). With music, no listeners in the tests found jitter audible at levels lower than 20 ns (Dunn 2003:34).
frequency responseof audio CD is sufficiently wide to cover the entire audible range, which roughly extends from 20 Hz to 20 kHz. 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, aliasingof these higher frequencies may occur. This is where these higher, inaudible frequencies alias to frequencies which are in the audible range. To prevent aliasing, it is not necessary to design a brick-wall anti-aliasingfilter - that is a filter which perfectly removes all frequency content above (or below) a certain range. Instead, a sample rate is usually chosen which is above the theoretical requirement. This is called oversampling, and allows a less severe anti-aliasing filter to be used.
High quality open-reel tape frequency response can extend from 10 Hz to well above 200 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 cassettes may have a response extending up to 14 kHz at full (0 dB) recording level (Stark 1989).
The frequency response for a conventional LP player might be 30 Hz - 20 kHz +/- 3 dB. Unlike the audio CD, vinyl records 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). 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. 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 the same every time it is played.
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
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).
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 deficencies in some early digital recordings, where the digital release was inferior to the analog version.
Overload conditions and dynamic range
There are some differences in the behaviour of analog and digital systems when high level signals are present, where there is the possibility that such signals could push the system into overload. With high level signals, analog
magnetic tapeapproaches saturation, and high frequency response drops in proportion to low frequency response. The audible effect of this can be reasonably unobjectionable (Elsea 1996). In contrast, some digital PCM recorders can show non-benign behaviour in overload (Dunn 2003:65). The ‘softness’ of analog tape clipping allows a usable dynamic range that can exceed that of some PCM digital recorders (i.e. the coding scheme used in Compact Disc).
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 problems.
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 bitof the digital system (Stuart n.d.:3).
Overload conditions and dynamic range
In principle, PCM digital systems have the lowest level of nonlinear distortion at full signal amplitude. The opposite is usually true of analog systems, where distortion tends to increase at high signal levels. A study by Manson (1980) considered the requirements of a digital audio system for high quality broadcasting. It concluded that a 16 bit system would be sufficient, but noted the small reserve the system provided in ordinary operating conditions. For this reason, it was suggested that a fast-acting signal
limiteror ‘soft clipper’ be used to prevent the system from becoming overloaded (Manson 1980:8).
With many recordings, high level distortions at signal peaks may be audibly masked by the original signal, thus large amounts of distortion may be acceptable at peak signal levels. The difference between analog and digital systems is the form of high-level signal error. Some early analog-to-digital converters displayed non-benign behaviour when in overload, where the overloading signals were 'wrapped' from positive to negative full-scale. Modern converter designs based on sigma-delta modulation may become unstable in overload conditions. It is usually a design goal of digital systems to limit high-level signals to prevent overload (Dunn 2003:65). To prevent overload, a modern digital system may compress input signals so that digital full-scale cannot be reached (Jones et al. 2003:4).
The dynamic range of digital audio systems can exceed that of analog audio systems. Typically, a 16 bit analog-to-digital converter may have a dynamic range of between 90 to 95 dB (Metzler 2005:132), whereas the signal-to-noise ratio (roughly the equivalent of dynamic range, noting the absence of quantization noise but presence of tape hiss) of a professional
reel-to-reel1/4 inch tape recorder would be between 60 and 70 dB at the recorder's rated output (Metzler 2005:111).
The benefits of using digital recorders with greater than 16 bit accuracy can be applied to the 16 bits of audio CD. Stuart (n.d.:3) stresses that with the correct dither, the resolution of a digital system is infinite, and that it is possible, for example, to resolve sounds at -110 dB (below digital full-scale) in a well-designed 16 bit channel.
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
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 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-Audiocan be 96 or 192 kHz and up to 24 bits resolution. With these higher 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 in order to see if these differences in performance could be explained [http://www.nanophon.com/audio/antialia.pdf] . He did this by examining the band-limiting filters used in converters and looking 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 [http://www.nhk.or.jp/strl/publica/labnote/lab486.html] .
Super Audio CD and DVD Audio
Super Audio CD(SACD) format was created by Sonyand Philips, who were also the developers of the earlier standard audio CD format. SACD uses Direct Stream Digital, which works quite differently from the PCMformat 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 - and therefore strictly speaking an analogue signal - (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 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 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 [http://www.hfm-detmold.de/eti/projekte/diplomarbeiten/dsdvspcm/aes_paper_6086.pdf] . Listeners involved in this test noted their great difficulty in hearing any difference between the two formats.
Some audio enthusiasts prefer the sound of vinyl records over that of CD, this despite the apparent technical advantages of the digital format. 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" [http://www.findarticles.com/p/articles/mi_m1430/is_n5_v17/ai_16368605/pg_1] . Dub producer Adrian Sherwoodhas similar feelings about the analog cassette tape, which he prefers because of its warm sound [http://arts.guardian.co.uk/fridayreview/story/0,12102,1049363,00.html] .
Those who favour the digital format point to the results of blind tests, which demonstrate the high performance possible with digital recorders [http://www.findarticles.com/p/articles/mi_m1430/is_n5_v17/ai_16368607] , [http://www.bostonaudiosociety.org/bas_speaker/abx_testing2.htm] . The assertion is that the 'analog sound' is more a product of analog format inaccuracies than anything else. One early supporter 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".
Was it ever entirely analog or digital?
Complicating the discussion is that recording professionals often mix and match analog and digital techniques in the process of producing a recording. Analog signals can be subjected to digital signal processing or effects, and inversely digital signals are converted back to analog in equipment that can include analog steps such as vacuum tube amplification.
For modern recordings, the controversy between analog recording and digital recording is becoming moot. No matter what format the user uses, the recording probably was digital at several stages in its life. In case of
video recordings it is moot for one other reason; whether the format is analog or digital, digital signal processing is likely to have been used in some stages of its life, such as digital timebase correctionon playback.
While the words analog audio usually imply that the sound is described using a continuous time, continuous amplitudes approach in both the media and the reproduction/recording systems, and the words digital audio imply a discrete time, discrete amplitudes approach, there are methods of encoding audio that fall somewhere between the two, e.g. continuous time, discrete levels and discrete time,continuous levels.
While not as common as "pure analog" or "pure digital" methods, these situations do occur in practice. Indeed, all analog systems show discrete (quantized) behaviour at the microscopic scale [http://www.st-andrews.ac.uk/~www_pa/Scots_Guide/iandm/part12/page1.html] , and e.g. asynchronously operated class-D amplifiers even consciously incorporate continuous time, discrete amplitude designs. Continuous amplitude, discrete time systems have also been used in many early analog-to-digital converters, in the form of sample-and-hold circuits. The boundary is further blurred by digital systems which statistically aim at analog-like behavior, most often by utilizing stochastic dithering and noise shaping techniques.While vinyl records and common compact cassettes are analog media and use quasi-linear mechanical encoding methods (e.g. spiral groove depth,
tape magnetic fieldstrength) without noticeable quantization or aliasing, there are "analog" non-linear systems that exhibit effects similar to those encountered on digital ones, such as aliasingand "hard" dynamic floors (e.g. frequency modulated audio on VHStapes, PWMencoded signals).
Although those "hybrid" techniques are usually more common in
telecommunicationssystems than in consumer audio, their existence alone blurs the distinctive line between certain digital and analog systems, at least for what regards some of their alleged advantages or disadvantages.
Audio quality measurement
Audio system measurements
History of sound recording
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