From Wikipedia, the free encyclopedia

The tone or style of this article or section may not be appropriate for Wikipedia. Specific concerns may be found on the talk page. See Wikipedia's guide to writing better articles for suggestions.(December 2007)

This article needs additional citations for verification. Please help improve this article by adding reliable references. Unsourced material may be challenged and removed. (July 2007)

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

Contents 1 Main differences 2 Noise and distortion 2.1 Digital fundamentals 2.2 Noise performance 2.3 Digital dither 2.4 Greater than 16 bits 2.5 Analog systems 2.6 Rumble 2.7 Wow and flutter 3 Frequency response 4 Sound quality 4.1 Subjective evaluation 4.2 Early digital recordings 4.3 Higher sampling rates 4.4 Super Audio CD and DVD Audio 4.5 Analog warmth 5 Was it ever entirely analog or digital? 6 Hybrid systems 7 See also 8 References

An analog recording is one where the original sound signal is modulated onto another physical medium or substrate such as the groove of a gramophone disc or 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.)

A digital recording is produced by converting the physical properties of the original sound into digital information (called bits), 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 the sampling depth (how much information each sample contains). However, unlike analog recording, the physical medium for storing digital samples becomes immaterial in recovery of the encoded information as long as the individual bits can be recovered.

Accurate, high quality sound reproduction is possible with both analog and digital systems. The principal advantage that digital systems have over analog systems is lower costs for end users. With analog recordings, consumers must use high-quality playback equipment to accurately separate the signal from the background without picking up noise or distortion. With digital, only the signal is encoded, so playback equipment can be much less expensive for a given quality. (Incidentally this same principle applies to digital video and still photography.)

Imperfections in the mechanical performance of the analog equipment may cause distortions like wow, flutter, tape hiss or record surface noise. Some of these distortions can be prevented using timebase correction, as is done in VHS tapes, filtering or high-quality components. Time-instability in digital systems (jitter) also degrades system performance, which is why most use time-coding on the medium itself to prevent it. Digital information can also be somewhat self-correcting. After a signal has been converted into a digital format, error-correcting codes (looking for particular types or sizes of digital bits & bytes, and 'correcting' if not found) help to prevent data loss and/or corruption. This allows digital formats to have a higher resistance to 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, 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 discs. This may be due to poor-quality CD recorder drives or low-quality discs.

Unlike analog dubs, digital copies and regenerations are exact clones. They can be made indefinitely 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.

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, owing to their disk or memory-based nature. In production, this makes editing much easier. It also allows the listener greater flexibility when playing back recordings. Most digital systems also have the ability to encode non-audio information into the digital stream, such as information about the artist, track titles, etc.

Also, whereas digital formats retain a sample rate, analog does not.

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 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.

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 a SNR of 98 dB (Sony Europe 2001) and dynamic range of 96 dB.

• 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.

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).

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 suggested that undithered 14 bit recording or 13 bit dithered recordings were suitable for high-quality FM radio broadcasting (Croll 1970).

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. 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.

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].

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).

"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).

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).

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.

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. Large, sudden deviations in the amplitude of response at different frequencies will have phase shifts associated with them, which are very audible. Some analog tape manufacturers specify frequency responses up to 20 kHz, but these measurements may have been made at low signal levels (Stark 1989). 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.

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.

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).

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. Some recording engineers like Jack Renner of the Telarc record label produced digital recordings which were well-received by critics for their sound quality (Greenfield et al. 1986). Some analog recordings were remastered for CD, but problems were occasionally identified with these releases. For instance, violins that once sounded well-balanced on analog (vinyl) disc would sound too aggressive on CD. 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. Other faults in recordings were more noticeable, like background noise.

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].

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 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 an oversampling design and a process called sigma-delta modulation. The sample rate of the recorder is 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). 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 [4]. 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" [5]. 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], [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".

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.

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 [9]. Digital systems may use techniques which emulate the behaviour of analog (continuous) systems, e.g. dither.

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).

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.

• Audiophile
• Audio quality measurement
• Audio system measurements
• History of sound recording
• Psychoacoustics

• 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.
• Borwick, J. et al (1994). The Loudspeaker and Headphone Handbook, 2nd edition. Edited by John Borwick, with specialist contributors. Section 11.7 'Experimental Procedure', by Floyd Toole, pages 481-488. Focal Press. ISBN 0 240 51371 1.
• 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.
• Hawksford, M.O.J. (2001). "SDM versus LPCM: The Debate Continues", 110th AES Convention, paper 5397.
• 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.
• Lesurf, J. "Analog or Digital?", The Scots Guide to Electronics. Retrieved October 2007.
• 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.
• Watkinson J. (1994). An Introduction to Digital Audio. Section 1.2 'What is digital audio?', page 3; Section 2.1 'What can we hear?', page 26. Focal Press. ISBN 0 240 51378 9.