Analog sound versus digital sound compares the two ways in which sound is recorded and stored. The information contained in a sound wave is retained as a signal, which over time can vary continuously in amplitude. This signal can be recorded either digitally or on an analog format. Sound is a disturbance of mechanical energy that propagates through matter as a wave. ... A digital system is one that uses discrete values (often electrical voltages), especially those representable as binary numbers, or non-numeric symbols such as letters or icons, for input, processing, transmission, storage, or display, rather than a continuous spectrum of values (ie, as in an analog system). ... An analog or analogue signal is an allergy continuous in both time and amplitude. ...

Main differences

An analog recording is one where the original sound signal is modulated onto another physical signal carried on some media or substrate such as the groove of a gramophone disc or the iron oxide surface of a magnetic tape. A physical quantity in the medium (e.g., the intensity of the magnetic field) is directly related, or analogous, to the physical properties of the sound (e.g., the amplitude, phase, etc.). In telecommunications, modulation is the process of varying a periodic waveform, i. ... Manufacturers put records inside protective and decorative cardboard jackets and an inner paper sleeve to protect the grooves from dust and scratches. ... Compact audio cassette Magnetic tape is a non-volatile storage medium consisting of a magnetic coating on a thin plastic strip. ...

A digital recording is produced by first converting the physical properties of the original sound into digital information (stored as bits) which can then be decoded for reproduction. The conversion process can be susceptible to noise and imperfection. However, the nature of the physical medium is immaterial in recovery of the encoded information as long as the individual bits can be recovered. This article is about the unit of information. ...

Accurate, high quality sound is possible with both analog and digital systems. The principal advantage that digital systems have over analog ones is one of lower cost. High-quality open-reel analog tape and related hardware is expensive to buy and maintain. With digital systems, high quality sound output is achievable with very low manufacturing cost and in mass-produced devices (Rumsey & Watkinson 1995). This is because analog systems require high-quality mechanical and electronic performance all the way through the audio-chain - recording, production, and finally playback by the consumer. Digital systems are only dependent on the electronic performance of the equipment, and because the signal information is conveyed as a digital (binary) code, any noise or distortion generated by the equipment is easier to reject.

Imperfections in the mechanical performance of the analog equipment may cause distortions like wow and flutter. Some of these distortions can be prevented using timebase correction, as is done in VHS tapes. Time-instability in digital systems (jitter) can degrade system performance. After a signal has been converted into a digital format, application of error-correcting codes helps to prevent data loss and/or corruption. This allows digital formats to have a higher resistance to media deterioration than analog formats. It is possible for poorly produced digital media to result in data loss. Laser rot was most troublesome to the Laserdisc format, which used digital audio, and was caused by inadequate disc manufacture. There can occasionally be difficulties related to the use of consumer recordable/rewritable compact disc. This may be due to poor quality CD recorder drives or low quality discs. Wow is a relatively slow form of flutter (pitch variation) which can affect both gramophone records and audio cassettes. ... Flutter: In electronics, rapid variation of signal parameters, such as amplitude, phase, and frequency. ... Timebase correction is a technique to reduce or eliminate errors present in all analog recordings on mechanical media, including video tape recorders and videocassette recorders, caused by mechanical instability. ... Bottom view of VHS cassette with magnetic tape exposed Top view of VHS cassette with front casing removed The Video Home System, better known by its abbreviation VHS is a recording and playing standard for analog video cassette recorders (VCRs), developed by Victor Company of Japan, Limited (JVC) and launched... In telecommunication, jitter is an abrupt and unwanted variation of one or more signal characteristics, such as the interval between successive pulses, the amplitude of successive cycles, or the frequency or phase of successive cycles. ... This article does not cite any references or sources. ... Not to be confused with disk laser, a type of solid-state laser in a flat configuration. ...

Unlike analog dubs, digital copies and regenerations are exact clones. They can be made infinitely without degradation, unless DRM restrictions apply or mastering errors 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. Digital Rights Management or Digital Restrictions Management (DRM) is an umbrella term for any of several arrangements which allows a vendor of content in electronic form to control the material and restrict its usage in various ways that can be specified by the vendor. ... Audio mastering is the process of preparing and transfering recorded audio to a medium for future duplication. ...

There are also several advantages of digital systems that are not related to sound quality but are of practical value. Most digital media have non-sequential (random) access, due to their disk or memory-based nature. In production, this makes editing it easier. It also allows the listener greater flexibility when playing back recordings. Digital systems have the ability to encode non-audio information into the audio stream such as information about the owner, track titles, etc. The terms storage (U.K.) or memory (U.S.) refer to the parts of a digital computer that retain physical state (data) for some interval of time, possibly even after electrical power to the computer is turned off. ...

Also, whereas digital formats retain a sample rate, analog does not. The sampling frequency or sampling rate defines the number of samples per second taken from a continuous signal to make a discrete signal. ...

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.

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). In doing this, the original sound wave can now be described using only numbers - as digital information. 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 correction coding, 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. In signal processing, sampling is the reduction of a continuous signal to a discrete signal. ... 4-channel stereo multiplexed analog-to-digital converter WM8775SEDS made by Wolfson Microelectronics placed on X-Fi Fatal1ty Pro sound card An analog-to-digital converter (abbreviated ADC, A/D or A to D) is an electronic integrated circuit (i/c) that converts continuous signals to discrete digital numbers. ... Quantized signal Digital signal In digital signal processing, quantization is the process of approximating a continuous range of values (or a very large set of possible discrete values) by a relatively-small set of discrete symbols or integer values. ... In computer science and information theory, error correction consists of using methods to detect and/or correct errors in the transmission or storage of data by the use of some amount of redundant data and (in the case of transmission) the selective retransmission of incorrect segments of the data. ... In information theory, the noisy-channel coding theorem establishes that however contaminated with noise interference a communication channel may be, it is possible to communicate digital data (information) error-free up to a given maximum rate through the channel. ... In electronics, a digital-to-analog converter (DAC or D-to-A) is a device for converting a digital (usually binary) code to an analog signal (current, voltage or electric charge). ...

Noise performance

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. A loudspeaker is a device which converts an electrical signal into sound. ...

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 audio CD has 2¹⁶= 65,536 possible signal amplitudes, theoretically allowing for a SNR of 98 dB (Sony Europe 2001) and dynamic range of 96 dB. The phrase signal-to-noise ratio, often abbreviated SNR or S/N, is an engineering term for the ratio between the magnitude of a signal (meaningful information) and the magnitude of background noise. ...

• Note that a decibel 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 logarithm to base 10. Then multiply by 10 to get decibels. Unfortunately, Ohm's Law comes 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 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 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. Practical digital converters may show considerable deviation from ideal performance. For other uses, see Decibel (disambiguation). ... Bel can mean: A unit of measurement for proportions and ratios; see Decibel and dB(A) The title of a Semitic god; see Bel (god) A Celtic deity; see Belenus Hindi name of the Bengal Quince tree or its fruit. ... In mathematics, if two variables of bn = x are known, the third can be found. ... Logarithms to various bases: is to base e, is to base 10, and is to base 1. ... A voltage source, V, drives an electric current, I , through resistor, R, the three quantities obeying Ohms law: V = IR Ohms law states that, in an electrical circuit, the current passing through a conductor between two points is proportional to the potential difference (i. ... An AC voltage (or other alternating signal) measured by looking at its maximum positive and maximum negative voltage. ... PCM is an initialism which can have different meanings: Phase Change Material Pulse-code modulation, a way to digitally encode signals representing sound and their video counterparts Potential Cancer Marker Communist Party of Mexico Plug Compatible Manufacturer Power-train control module, a computer in a car which controls the car... Parts per million (ppm) is a measure of concentration that is used where low levels of concentration are significant. ...

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 Camcorder audio (or DAT in long-play, 32khz 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). PCM is an initialism which can have different meanings: Phase Change Material Pulse-code modulation, a way to digitally encode signals representing sound and their video counterparts Potential Cancer Marker Communist Party of Mexico Plug Compatible Manufacturer Power-train control module, a computer in a car which controls the car... Sony DV Handycam A camcorder is a portable electronic device for recording video images and audio onto an internal storage device. ... DAT can mean: day after tomorrow, a J-Pop band. ...

Digital dither

In digital recording, quantization of the original analog signal results in quantization noise. Unlike the noise floor in analog systems, quantization noise is non-random in nature, and is more audibly disturbing. Dithering can be used to hide quantization noise. Dither reduces the amount of low level distortion in digital recordings but increases the amount of background noise by a few dB. Early tests in the 1970s by the BBC using a 10 bit PCM system suggested that undithered 14 bit recording or 13 bit dithered recordings were suitable for high-quality FM radio broadcasting (Croll 1970). Later research highlighted difficulties experienced by digital recorders due to idle-channel noise, with such noise showing less variation in recorders using dither (Ely 1978). This article or section should be merged with Dither An illustration of dithering. ... The British Broadcasting Corporation, which is usually known as the BBC, is the largest broadcasting corporation in the world in terms of audience numbers, employing 26,000 staff in the United Kingdom alone and with a budget of more than GB£4 billion. ... The abbreviations FM, Fm, and fm may refer to: Electrical engineering Frequency modulation (FM) and its most common applications: FM broadcasting, used primarily to broadcast music and speech at VHF frequencies FM synthesis, a sound-generation technique popularized by early digital synthesizers Science Femtometre (fm), an SI measure of length...

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 of the advantages in using digital recorders with more bits is the ability to directly record uncontrolled microphone signals. 19 bits has been shown to be necessary to capture some high-quality signals for broadcast (Manson 1980). The benefits of using digital recorders with greater than 16 bit accuracy can be applied to the 16 bits of audio CD. This may be done using dither and noise shaping. More noise is present in recordings using noise shaping, but the noise is present in less audible frequency regions, thus improving the subjective dynamic range. Similar to dither, noise shaping is a bit reduction technique used to minimize quantization error. ...

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 [1].

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 (Stark 1989). Typical 60-minute Compact Cassette. ... A gramophone record, (also phonograph record - often simply record) is an analog sound recording medium: a flat disc rotating at a constant angular velocity, with inscribed spiral grooves in which a stylus or needle rides. ... Dolby NR is a noise reduction system developed by Dolby Laboratories for use in analogue magnetic tape recording. ...

Rumble

"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 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). A bearing is a device to permit constrained relative motion between two parts, typically rotation or linear movement. ... A 4 point angular contact ball bearing A ball bearing is a common type of rolling-element bearing, a kind of bearing. ... This article or section is in need of attention from an expert on the subject. ...

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).

Frequency response

The frequency response of 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 sampling 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. 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 filter - that is a filter which perfectly removes all frequency content above (or below) a certain range. Instead, a sampling rate is chosen above the theoretical requirement. This allows for a less severe filter to be used. In addition to this, other methods can be used to try and increase performance, for example, oversampling. The sampling frequency or sampling rate defines the number of samples per second taken from a continuous signal to make a discrete signal. ... The Nyquist-Shannon sampling theorem is the fundamental theorem in the field of information theory, in particular telecommunications. ... This article does not cite any references or sources. ... Properly sampled image of brick wall. ... In signal processing, oversampling is the process of sampling a signal with a sampling frequency significantly higher than twice the bandwidth or highest frequency of the signal being sampled. ...

High quality open-reel tape frequency response can extend from 10 Hz to well above 200 kHz. What must be thought of, however, when looking at specified frequency responses is the linearity of that response. Large, sudden deviations in the amplitude of response at different frequencies will have phase shifts associated with them, which are very audible. 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. What is also important is the recording level at which the frequency response measurements are made. Some analog tape manufacturers specify frequency responses up to 20 kHz, but these measurements may have been made at low signal levels (Stark 1989). What is important is the linearity of the response over the quoted range and what design compromises have been made in order to achieve this response (Driscoll 1980). This article is about a portion of a periodic process. ...

High-quality metal-particle cassettes may have a response extending up to 14 kHz at full (0 dB) recording level (Stark 1989). A vinyl record player can have a frequency response extending to 20 kHz, and unlike the audio CD, does not require a cut-off in response above this. 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. The low frequency response of vinyl records is restricted by rumble noise (described above). The frequency response for a conventional LP player might be 30 Hz - 20 kHz +/- 3 dB. This compares with the CD system which 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). 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.

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 in which it was designed for. 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. Audio compression is a form of data compression designed to reduce the size of audio files. ... A codec is a device or program capable of performing encoding and decoding on a digital data stream or signal. ...

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 (Toole 1994).

Early digital recordings

Analog sound reproduction was already a mature technology when digital recording and compact discs first appeared. Early digital recorders were designed at a time when the need for applying dither was not widely appreciated. Recordings made without appropriate dither suffer from signal distortion at low signal levels (Hicks 1995).

Some early digital recordings were criticised for their sound quality. One explanation for this was that engineers had learned to place microphones in such a way as to improve fidelity when producing analog recordings. Due to the extra resolution of the audio CD, such techniques were no longer appropriate. For instance, violins that once sounded well-balanced on analog (vinyl) disc would sound too aggressive on CD. Other faults in recordings were more noticeable, like background noise. Even so, some recording engineers like Jack Renner of the Telarc record label were more aware of these problems than others, and early on were able to produce recordings of excellent sound quality (Greenfield et al. 1986). Jack L. Renner (born April 13, 1935) is an American classically-trained musician and recording engineer, best known as chairman, CEO and chief recording engineer of the Telarc International Corporation. ... Telarc International Corporation is a Cleveland, Ohio based independent record label, founded in 1977 by two classically trained musicians and former teachers, Jack Renner and Robert Woods. ...

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 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 [2]. 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 [3]. CD may stand for: Compact Disc Canadian Forces Decoration Cash Dispenser (at least used in Japan) CD LPMud Driver Centrum-Demokraterne (Centre Democrats of Denmark) Certificate of Deposit České Dráhy (Czech Railways) Chad (NATO country code) Chalmers Datorförening (computer club of the Chalmers University of Technology) a 1960s... MHZ redirects here. ... The Nyquist frequency, named after Harry Nyquist or the Nyquist–Shannon sampling theorem, is half the sampling frequency of a discrete signal processing system. ... Digital audio tape can also refer to a compact cassette with digital storage. ... The DVD-Audio logo. ... Psychoacoustics is the study of subjective human perception of sounds. ...

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 to the (PCM) format discussed in this article. Instead of using a greater number of bits depth and attempting to record a signal's precise amplitude for every sample cycle, a DSD 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). Super Audio CD (SACD) is a read-only optical audio disc format aimed at providing much higher fidelity digital audio reproduction than the compact disc. ... Sony Corporation ) is a Japanese multinational corporation and one of the worlds largest media conglomerates with revenue of $66. ... Philips HQ in Amsterdam Koninklijke Philips Electronics N.V. (Royal Philips Electronics N.V.), usually known as Philips, (Euronext: PHIA, NYSE: PHG) is one of the largest electronics companies in the world, founded and headquartered in the Netherlands. ... Direct-Stream Digital (DSD) is an encoding technology to store audio signals on digital storage media and is used for the super audio compact disc (SACD). ... PCM is an initialism which can have different meanings: Phase Change Material Pulse-code modulation, a way to digitally encode signals representing sound and their video counterparts Potential Cancer Marker Communist Party of Mexico Plug Compatible Manufacturer Power-train control module, a computer in a car which controls the car... Pulse-width modulation of a signal or power source involves the modulation of its duty cycle to either convey information over a communications channel or control the amount of power sent to a load. ... Linear Pulse Code Modulation used in communications (or LPCM) is a format that is a popular choice in music production. ...

A Direct Stream Digital recorder uses an oversampling design and a process called delta-sigma modulation. The sample rate of the recorder is 64 times the Nyquist rate, at around 3 Mhz. Delta-sigma modulation is a derivative of delta modulation. The delta modulation process has been described as being digital at a microscopic level, but as having analog attributes at the macroscopic level (Hawksford 1991). The output from a Direct Stream Digital 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. The Delta-Sigma (ΔΣ) modulation is a kind of analog-to-digital or digital-to-analog conversion. ... The Delta modulation (DM or Δ-modulation) is an analog-to-digital or digital-to-analog signal conversion. ... A bitstream or bit stream is a time series of bits. ...

There are a number of reasons why delta sigma modulation is often used in digital audio. For inputs where the signal level is low, a multi-bit quantizer will be exercising only a small part of its operating range. With resolutions of 16 bits and above, differences in the electronic levels between quantization levels are minute, making multi-bit quantizers sensitive to nonidealities in the electronics of the system. Sigma-delta modulation allows a single-bit quantizer to be used instead of a multi-bit design. With converter designs based on SDM, the one-bit quantizer constantly exercises its full operating range with all signal inputs, thus making the quantizer less sensitive to non-idealities in the electronics of the system.

The oversampling in SDM allows digital filters to be used in the ADC stage. Early on, this was of significant importance to the audio CD format, because digital recorders that operated at 44.1 kHz required very steep analog cut-off filters which were difficult to implement. The digital filters used in SDM-based recorders offer much better attentuation of spurious aliases in the audible range. Therefore SDM is frequently used in linear LCM formats as well as DSD. The difference is that the original bitstream is retained with DSD, but with linear PCM, the bitstream is subject to bandlimiting and dithering (Hawksford 1995). There is a larger amount of audio production equipment designed to support linear PCM rather than DSD, but production tools have been developed which directly support DSD. DSD does have greater storage requirements than linear PCM, but has the advantage that the bandlimiting and dithering needed with linear PCM is not required. 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. A double-blind subjective test between high resolution linear PCM (DVD-Audio) and DSD did not reveal a statistically significant difference [4].

Analog warmth

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" [5]. Dub producer Adrian Sherwood has similar feelings about the analog cassette tape, which he prefers because of its warm sound [6]. Adrian Sherwood Adrian Sherwood (born 1958) is a British record producer best known for his work with dub music as well as for remixing a number of popular acts such as Coldcut, Depeche Mode, Primal Scream, Pop Will Eat Itself, and Sinéad OConnor. ...

Those who favour the digital format point to the results of blind tests, which demonstrate the high performance possible with digital recorders [7], [8]. 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". Herbert von Karajan (April 5, 1908 – July 16, 1989) was an Austrian conductor. ...

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 correction on playback. Video is the technology of capturing, recording, processing, transmitting, and reconstructing moving pictures, typically using celluloid film, electronic signals, or digital media, primarily for viewing on television or as video clips on computer monitors. ... Timebase correction is a technique to reduce or eliminate errors present in all analog recordings on mechanical media, including video tape recorders and videocassette recorders, caused by mechanical instability. ...

Despite this some musical artists, such as Mission of Burma, still make it a point to record fully analog albums. Mission of Burma is a post-punk band from Boston, Massachusetts, USA comprising guitarist Roger Miller, bassist Clint Conley and drummer Peter Prescott, with Bob Weston (originally Martin Swope) as tape manipulator and sound engineer. ...

Hybrid systems

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. E.g. while vinyl records and common compact cassettes are analog media and use quasi-linear mechanical encoding methods (e.g. spiral groove depth, tape magnetic field strength) without noticeable quantization or aliasing, there are analog non-linear systems that exhibit effects similar to those encountered on digital ones, such as aliasing and "hard" dynamic floors (e.g. frequency modulated audio on VHS tapes, PWM encoded signals). A gramophone record, (also phonograph record - often simply record) is an analog sound recording medium: a flat disc rotating at a constant angular velocity, with inscribed spiral grooves in which a stylus or needle rides. ... The Compact Cassette, often referred to as audio cassette, cassette tape, cassette, or simply tape, is a magnetic tape sound recording format. ... Look up Tape in Wiktionary, the free dictionary. ... Magnetic field lines shown by iron filings In physics, a magnetic field is a solenoidal vector field in the space surrounding moving electric charges and magnetic dipoles, such as those in electric currents and magnets. ... Properly sampled image of brick wall. ... Bottom view of VHS cassette with magnetic tape exposed Top view of VHS cassette with front casing removed The Video Home System, better known by its abbreviation VHS is a recording and playing standard for analog video cassette recorders (VCRs), developed by Victor Company of Japan, Limited (JVC) and launched... Pulse-width modulation of a signal or power source involves the modulation of its duty cycle to either convey information over a communications channel or control the amount of power sent to a load. ...

Although those "hybrid" techniques are usually more common in telecommunications systems 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. Telecommunication involves the transmission of signals over a distance for the purpose of communication. ...

See also

• Audiophile
• Psychoacoustics

An audiophile, from Latin audire[1] to hear and Greek philos[2] loving, can be generally defined as a person dedicated to achieving high fidelity in the recording and playback of music . ... Psychoacoustics is the study of subjective human perception of sounds. ...

References

• Ashihara, K. et al (2005). "Detection threshold for distortions due to jitter on digital audio", Acoustical Science and Technology, Vol. 26 (2005) , No. 1 pp.50-54.
• Blech, D. & Yang, M. (2004). "Perceptual Discrimination of Digital Coding Formats", Audio Engineering Society Convention Paper 6086, May 2004.
• Croll, M. (1970). "Pulse Code Modulation for High Quality Sound Distribution: Quantizing Distortion at Very Low Signal Levels", Research Department Report No. 1970/18, BBC.
• Driscoll, R. (1980). Practical Hi-Fi Sound, 'Analogue and digital', pages 61-64; 'The pick-up, arm and turntable', pages 79-82. Hamlyn. ISBN 0 600 34627 7.
• Dunn, J. (1998). "The benefits of 96 kHz sampling rate formats for those who cannot hear above 20 kHz", Preprint 4734, presented at the 104th AES Convention, May 1998.
• Dunn, J. (2003). "Measurement Techniques for Digital Audio", Audio Precision Application Note #5, Audio Precision.
• Greenfield, E. et al. (1986). The Penguin Guide to Compact Discs, Cassettes and LPs, Penguin.
• Ely, S. (1978). "Idle-channel noise in p.c.m. sound-signal systems". BBC Research Department, Engineering Division.
• Hawksford, M.O.J. (1991). "Introduction to Digital Audio", Images of Audio, Proceedings of the 10th International AES Conference, London, September 1991.
• Hawksford, M.O.J. (1995). "Bitstream versus PCM debate for high-density compact disc", ARA/Meridian web page, November 1995.
• Hicks, C. (1995). "The Application of Dither and Noise-Shaping to Nyquist-Rate Digital Audio: an Introduction", Communications and Signal Processing Group, Cambridge University Engineering Department, United Kingdom.
• Kaoru, A. & Shogo, K. (2001). "Detection threshold for tones above 22 kHz", Audio Engineering Society Convention Paper 5401. Presented at the 110th Convention, 2001.
• Libbey, T. "Digital versus analog: digital music on CD reigns as the industry standard", Omni, February 1995.
• Lipshitz, S. "The Digital Challenge: A Report", The BAS Speaker, Aug-Sept 1984.
• Liversidge, A. "Analog versus digital: has vinyl been wrongly dethroned by the music industry?", Omni, February 1995.
• Sony Europe (2001). Digital Audio Technology 4th edn, edited by J. Maes & M. Vercammen. Focal Press.
• Manson, W. (1980). "Digital Sound: studio signal coding resolution for broadcasting". BBC Research Department, Engineering Division.
• Nishiguchi, T. et al. (2004). "Perceptual Discrimination between Musical Sounds with and without Very High Frequency Components", NHK Laboratories Note No. 486, NHK (Japan Broadcasting Corporation).
• Paul, J. "Last night a mix tape saved my life", The Guardian, September 26, 2003.
• Pohlmann, K. (2005). Principles of Digital Audio 5th edn, McGraw-Hill Comp.
• Rathmell, J. et al. (1997) "TDFD-based Measurement of Analog-to-Digital Converter Nonlinearity", Journal of the Audio Engineering Society, Volume 45, Number 10, pp. 832-840; October 1997.
• Rumsey, F. & Watkinson, J. (1995). The Digital Interface Handbook, 2nd edition. Sections 2.5 and 6. Pages 37 and 154-160. Focal Press.
• Stark, C. (1989). Encyclopædia Britannica, 15th edition, Volume 27, Macropaedia article 'Sound', section: 'High-fidelity concepts and systems', page 625.
• Toole, F. (1994). The Loudspeaker and Headphone Handbook, 2nd edition, Chapter 11: 'Subjective Evaluation'. Edited by John Borwick. Focal Press.