Comparison of analog and digital recording

This article compares the two ways in which sound isrecorded and stored. Actual sound waves consist of con-tinuous variations in air pressure. Representations ofthese signals can be recorded using either digital or analogtechniques.An analog recording is one where a property or charac-teristic of a physical recording medium is made to varyin a manner analogous to the variations in air pressureof the original sound. Generally, the air pressure varia-tions are first converted (by a transducer such as a mi-crophone) into an electrical analog signal in which ei-ther the instantaneous voltage or current is directly pro-portional to the instantaneous air pressure (or is a func-tion of the pressure). The variations of the electrical sig-nal in turn are converted to variations in the recordingmedium by a recording machine such as a tape recorderor record cutter—the variable property of the mediumis modulated by the signal. Examples of properties thatare modified are the magnetization of magnetic tapeor the deviation (or displacement) of the groove of agramophone disc from a smooth, flat spiral track.A digital recording is produced by converting the phys-ical properties of the original sound into a sequence ofnumbers, which can then be stored and read back for re-production. Normally, the sound is transduced (as by amicrophone) to an analog signal in the same way as foranalog recording, and then the analog signal is digitized,or converted to a digital signal, through an analog-to-digital converter and then recorded onto a digital storagemedium such as a compact disc or hard disk.Two prominent differences in functionality are thebandwidth and the signal-to-noise ratio (S/N); however,both digital and analog systems have inherent strengthsand weaknesses. The bandwidth of the digital system isdetermined, according to the Nyquist frequency, by thesample rate used. The bandwidth of an analog system isdependent on the physical capabilities of the analog cir-cuits. The S/N of a digital system is first limited by thebit depth of the digitization process, but the electronicimplementation of the digital audio circuit introduces ad-ditional noise. In an analog system, other natural analognoise sources exist, such as flicker noise and imperfec-tions in the recording medium. Some functions of thetwo systems are also naturally exclusive to either one orthe other, such as the ability for more transparent filteringalgorithms[1] in digital systems and the harmonic satura-tion of analog systems.

1 Overview of differences

It is a subject of debate whether analog audio is supe-rior to digital audio or vice versa. The question is highlydependent on the quality of the systems (analog or dig-ital) under review, and other factors which are not nec-essarily related to sound quality. Arguments for analogsystems include the absence of fundamental error mech-anisms which are present in digital audio systems, includ-ing aliasing, quantization noise,[2] and the absolute lim-itation of dynamic range. Advocates of digital point tothe high levels of performance possible with digital au-dio, including excellent linearity in the audible band andlow levels of noise and distortion.[3]

Accurate, high quality sound reproduction is possiblewith both analog and digital systems. Excellent, expen-sive analog systems may outperform digital systems, andvice versa; in theory any system of either type may besurpassed by a better, more elaborate and costly systemof the other type, but in general it tends to be less ex-pensive to achieve any given standard of technical signalquality with a digital system, except when the standard isvery low. One of themost limiting aspects of analog tech-nology is the sensitivity of analog media to minor physi-cal degradation; however, when the degradation is morepronounced, analog systems usually perform better, oftenstill producing recognizable sound, while digital systemswill usually fail completely, unable to play back anythingfrom the medium (see digital cliff). The principal advan-tages that digital systems have are a very uniform sourcefidelity, inexpensive media duplication, and direct use ofthe digital 'signal' in today’s popular portable storage andplayback devices. Analog recordings by comparison re-quire comparatively bulky, high-quality playback equip-ment to capture the signal from the media as accuratelyas digital.

1.1 Error correction

Early in the development of the Compact Disc, engineersrealized that the perfection of the spiral of bits was crit-ical to playback fidelity. A scratch the width of a hu-man hair (100 micrometres) could corrupt several dozenbits, resulting in at best a pop, and far worse, a loss ofsynchronization of the clock and data, giving a long seg-ment of noise until resynchronized. This was addressedby encoding the digital stream with a multi-tiered error-correction coding scheme which reduces CD capacity byabout 20%, but makes it tolerant to hundreds of surface

1

2 2 NOISE AND DISTORTION

imperfections across the disk without loss of signal. Inessence, "error correction" can be thought of as “usingthe mathematically encoded backup copies of the datathat was corrupted.” Not only does the CD use redun-dant data, but it also mixes up the bits in a predeterminedway (see CIRC) so that a small flaw on the disc will affectfewer consecutive bits of the decoded signal and allow formore effective error correction using the available backupinformation.Error correction allows digital formats to tolerate quite abit more media deterioration than analog formats. Thatis not to say poorly produced digital media are immuneto data loss. Laser rot was most troublesome to theLaserdisc format, but also occurs to some pressed com-mercial CDs, and was caused in both cases by inad-equate disc manufacture.[note 1] There can occasionallybe difficulties related to the use of consumer record-able/rewritable compact discs. This may be due to poor-quality CD recorder drives, low-quality discs, or incorrectstorage, as the information-bearing dye layer of most CD-recordable discs is at least slightly sensitive to UV lightand will be slowly bleached out if exposed to any amountof it. Most digital recordings rely at least to some ex-tent on computational encoding and decoding and so maybecome completely unplayable if not enough consecutivegood data is available for the decoder to synchronize tothe digital data stream, whereas any intact segment of anysize of an analog recording is playable.

1.2 Duplication

Unlike analog duplication, digital copies are exact repli-cas, which can be duplicated indefinitely[note 2] withoutdegradation. This made Digital rights management moreof an issue in digital media than analog media. Digi-tal systems often have the ability for the same mediumto be used with arbitrarily high or low quality encodingmethods and number of channels or other content, un-like practically all analog systems which have mechan-ically pre-fixed speeds and channels. Most higher-endanalog recording systems offer a few selectable record-ing speeds, but digital systems tend to offer much finervariation in the rate of media usage.There are also several non-sound related advantages ofdigital systems that are practical. Digital systems thatare computer-based make editing much easier throughrapid random access, seeking, and scanning for non-linear editing. Most digital systems also allow non-audiodata to be encoded into the digital stream, such as infor-mation about the artist, track titles, etc., which is oftenconvenient.[note 3]

2 Noise and distortion

In the process of recording, storing and playing back theoriginal analog sound wave (in the form of an electronicsignal), it is unavoidable that some signal degradation willoccur. This degradation is in the form of distortion andnoise. Noise is unrelated in time to the original signalcontent, while distortion is in some way related in time tothe original signal content.

2.1 Noise performance

For electronic audio signals, sources of noise include me-chanical, electrical and thermal noise in the recordingand playback cycle. The actual process of digital con-version will always add some noise, however small in in-tensity; the bulk of this in a high-quality system is quan-tization noise, which cannot be theoretically avoided, butsome will also be electrical, thermal, etc. noise from theanalog-to-digital converted device.The amount of noise that a piece of audio equipment addsto the original signal can be quantified. Mathematically,this can be expressed by means of the signal to noise ra-tio (SNR or S/N). Sometimes the maximum possible dy-namic range of the system is quoted instead. In a digitalsystem, the number of quantization levels, in binary sys-tems determined by and typically stated in terms of thenumber of bits, will have a bearing on the level of noiseand distortion added to that signal. The 16-bit digital sys-tem of Red Book audio CD has 216= 65,536 possible sig-nal amplitudes, theoretically allowing for an SNR of 98dB.[3] Each additional quantization bit adds 6 dB in pos-sible SNR, e.g. 24 x 6 = 144 dB for 24 bit quantization,126 dB for 21-bit, and 120 dB for 20-bit.With digital systems, the quality of reproduction dependson the analog-to-digital and digital-to-analog conversionsteps, and does not depend on the quality of the record-ing medium, provided it is adequate to retain the digitalvalues without error.

2.2 Analog systems

Consumer analog cassette tapes may have a dynamicrange of 60 to 70 dB. Analog FM broadcasts rarely havea dynamic range exceeding 50 dB, though under excel-lent reception conditions the basic FM transmission sys-tem can achieve just over 80 dB. The dynamic range of adirect-cut vinyl record may surpass 70 dB. Analog studiomaster tapes using Dolby-A noise reduction can have adynamic range of around 80 dB.

2.3 Rumble

“Rumble” is a form of noise characteristic caused by im-perfections in the bearings of turntables, the platter tends

3.3 Aliasing 3

to have a slight amount of motion besides the desiredrotation—the turntable surface also moves up-and-downand side-to-side slightly. This additional motion is addedto the desired signal as noise, usually of very low frequen-cies, creating a “rumbling” sound during quiet passages.Very inexpensive turntables sometimes used ball bearingswhich are very likely to generate audible amounts of rum-ble. More expensive turntables tend to use massive sleevebearings which are much less likely to generate offensiveamounts of rumble. Increased turntable mass also tendsto lead to reduced rumble. A good turntable should haverumble at least 60 dB below the specified output levelfrom the pick-up.[4]:79–82

2.4 Wow and flutter

Wow and flutter are a change in frequency of an analogdevice and are the result of mechanical imperfections,with wow being a slower rate form of flutter. Wow andflutter are most noticeable on signals which contain puretones. For LP records, the quality of the turntable willhave a large effect on the level of wow and flutter. Agood turntable will have wow and flutter values of lessthan 0.05%, which is the speed variation from the meanvalue.[4]Wowand flutter can also be present in the record-ing, as a result of the imperfect operation of the recorder.

3 Frequency response

3.1 Digital mechanisms

The frequency response of the standard for audio CDs issufficiently wide to cover the entire normal audible range,which roughly extends from 20 Hz to 20 kHz. Commer-cial and industrial digital recorders record higher frequen-cies, while consumer systems inferior to the CD recorda more restricted frequency range. Analog audio’s fre-quency response is less flat than digital, but it can vary inthe electronics.For digital systems, the upper limit of the frequency re-sponse is determined by the sampling frequency. Thechoice of sample rate used in a digital system is basedon the Nyquist-Shannon sampling theorem. This statesthat a sampled signal can be reproduced exactly as longas it is sampled at a frequency greater than twice thebandwidth of the signal. Therefore, a sampling rate of40 kHz would be theoretically enough to capture all theinformation contained in a signal having frequency band-width up to 20 kHz.

3.2 Analog mechanisms

High quality open-reel machines can extend from 10 Hzto above 20 kHz. The linearity of the response may be

indicated by providing information on the level of the re-sponse relative to a reference frequency. For example, asystem component may have a response given as 20 Hzto 20 kHz +/- 3 dB relative to 1 kHz. Some analog tapemanufacturers specify frequency responses up to 20 kHz,but thesemeasurements may have beenmade at lower sig-nal levels.[4] Compact cassettes may have a response ex-tending up to 15 kHz at full (0 dB) recording level (Stark1989). At lower levels usually−10 dB, cassettes typicallyrolls-off at around 20 kHz for most machines, due to thenature of the tape media caused by self-erasure (whichworsens the linearity of the response).The frequency response for a conventional LP playermight be 20 Hz - 20 kHz +/- 3 dB. Unlike the audioCD, vinyl records and cassettes do not require a cut-offin response above 20 kHz. The low frequency responseof vinyl records is restricted by rumble noise (describedabove). The high frequency response of vinyl depends onthe cartridge. CD4 records contained frequencies up to50 kHz, while some high-end turntable cartridges havefrequency responses of 120 kHz while having flat fre-quency response over the audible band (e.g. 20 Hz to15 kHz +/−0.3 dB).[5] In addition, frequencies of up to122 kHz have been experimentally cut on LP records.[6]

In comparison, the CD system offers a frequency responseof 20 Hz–20 kHz ±0.5 dB, with a superior dynamic rangeover the entire audible frequency spectrum.[3]

With vinyl records, there will be some loss in fidelity oneach playing of the disc. This is due to the wear of the sty-lus in contact with the record surface. A good quality sty-lus, matched with a correctly set up pick-up arm, shouldcause minimal surface wear. Magnetic tapes, both ana-log and digital, wear from friction between the tape andthe heads, guides, and other parts of the tape transport asthe tape slides over them. The brown residue depositedon swabs during cleaning of a tape machine’s tape path isactually particles of magnetic coating shed from tapes.Tapes can also suffer creasing, stretching, and frillingof the edges of the plastic tape base, particularly fromlow-quality or out-of-alignment tape decks. When a CDis played, there is no physical contact involved, and thedata is read optically using a laser beam. Therefore nosuch media deterioration takes place, and the CD will,with proper care, sound exactly the same every time itis played (discounting aging of the player and CD itself);however, this is a benefit of the optical system, not of dig-ital recording, and the Laserdisc format enjoys the samenon-contact benefit with analog optical signals. Record-able CDs slowly degrade with time, called disc rot, evenif they are not played, and are stored properly.[7]

3.3 Aliasing

Technical difficulty arises with digital sampling in thatall high frequency signal content above the Nyquist fre-quency must be removed prior to sampling, which, if not

4 4 DIGITAL ERRORS

done, will result in these ultrasonic frequencies “foldingover” into frequencies which are in the audible range,producing a kind of distortion called aliasing. The dif-ficulty is that designing a brick-wall anti-aliasing filter,a filter which would precisely remove all frequency con-tent exactly above or below a certain cutoff frequency,is impractical.[8] Instead, a sample rate is usually chosenwhich is above the theoretical requirement. This solutionis called oversampling, and allows a less aggressive andlower-cost anti-aliasing filter to be used.Unlike digital audio systems, analog systems do not re-quire filters for bandlimiting. These filters act to preventaliasing distortions in digital equipment. Early digital sys-tems may have suffered from a number of signal degrada-tions related to the use of analog anti-aliasing filters, e.g.,time dispersion, nonlinear distortion, temperature depen-dence of filters etc. (Hawksford 1991:8). Even with so-phisticated anti-aliasing filters used in the recorder, it isstill demanding for the player not to introduce more dis-tortion.Hawksford (1991:18) highlighted the advantages of dig-ital converters that oversample. Using an oversamplingdesign and a modulation scheme called sigma-delta mod-ulation (SDM), analog anti-aliasing filters can effectivelybe replaced by a digital filter.[8] This approach has sev-eral advantages. The digital filter can be made to have anear-ideal transfer function, with low in-band ripple, andno aging or thermal drift.

3.4 Higher sampling rates

CD quality audio is sampled at 44.1 kHz (Nyquist fre-quency = 22.05 kHz) and at 16 bits. Sampling the wave-form at higher frequencies and allowing for a greaternumber of bits per sample allows noise and distortion tobe reduced further. DAT can sample audio at up to 48kHz, while DVD-Audio can be 96 or 192 kHz and up to24 bits resolution. With any of these sampling rates, sig-nal information is captured above what is generally con-sidered to be the human hearing range.Work done in 1981 by Muraoka et al.[9] showed that mu-sic signals with frequency components above 20 kHzwereonly distinguished from those without by a few of the176 test subjects (Kaoru & Shogo 2001). Later papers,however, by a number of different authors, have led toa greater discussion of the value of recording frequen-cies above 20 kHz. Such research led some to the beliefthat capturing these ultrasonic sounds could have someaudible benefit. Audible differences were reported be-tween recordings with and without ultrasonic responses.Dunn (1998) examined the performance of digital con-verters to see if these differences in performance could beexplained.[10] He did this by examining the band-limitingfilters used in converters and looking for the artifacts theyintroduce.A perceptual study by Nishiguchi et al. (2004) concluded

that “no significant difference was found between soundswith and without very high frequency components amongthe sound stimuli and the subjects... however, [Nishiguchiet al] can still neither confirm nor deny the possibility thatsome subjects could discriminate betweenmusical soundswith and without very high frequency components.”[11]

Additionally, in blind tests conducted by Bob Katz, re-counted in his book Mastering Audio: The Art and theScience, he found that listening subjects could not discernany audible difference between sample rates with opti-mum A/D conversion and filter performance. He positsthat the primary reason for any aural variation betweensample rates is due largely to poor performance of low-pass filtering prior to conversion, and not variance in ul-trasonic bandwidth. These results suggest that the mainbenefit to using higher sample rates is that it pushes con-sequential phase distortion out of the audible range andthat, under ideal conditions, higher sample rates may notbe necessary.[12]

4 Digital errors

4.1 Quantization

0123456789

101112131415

An illustration of quantization of a sampled audio waveform us-ing 4 bits.

A signal is recorded digitally by an analog-to-digital con-verter, which measures the amplitude of an analog sig-nal at regular intervals, which are specified by the samplerate, and then stores these sampled numbers in computerhardware. The fundamental problem with numbers oncomputers is that the range of values that can be repre-sented is finite, which means that during sampling, theamplitude of the audio signal must be rounded. This pro-cess is called quantization, and these small errors in themeasurements are manifested aurally as a form of lowlevel distortion.Analog systems do not have discrete digital levels inwhich the signal is encoded. Consequently, the origi-nal signal can be preserved to an accuracy limited only

4.3 Jitter 5

by the intrinsic noise-floor and maximum signal level ofthe media and the playback equipment, i.e., the dynamicrange of the system. With digital systems, noise addeddue to quantization into discrete levels is more audiblydisturbing than the noise-floor in analog systems. Thisform of distortion, sometimes called granular or quanti-zation distortion, has been pointed to as a fault of somedigital systems and recordings (Knee & Hawksford 1995,Stuart n.d.:6). Knee & Hawksford (1995:3) drew atten-tion to the deficiencies in some early digital recordings,where the digital release was said to be inferior to theanalog version.The range of possible values that can be represented nu-merically by a sample is defined by the number of binarydigits used. This is called the resolution, and is usuallyreferred to as the bit depth in the context of PCM au-dio. The quantization noise level is directly determinedby this number, decreasing exponentially as the resolu-tion increases (or linearly in dB units), and with an ade-quate number of true bits of quantization, random noisefrom other sources will dominate and completely maskthe quantization noise.

4.2 Dither as a solution

An illustration of dither used in image processing. A randomdeviation has been inserted before reducing the palette to only16 colors, which is analogous to the effect of dither on an audiosignal.

It is possible to make quantization noise more audibly be-nign by applying dither. To do this, a noise-like signal isadded to the original signal before quantization. Dithermakes the digital system behave as if it has an analognoise-floor. Optimal use of dither (triangular probabilitydensity function dither in PCM systems) has the effect ofmaking the rms quantization error independent of signallevel (Dunn 2003:143), and allows signal information tobe retained below the least significant bit of the digitalsystem (Stuart n.d.:3).Dither algorithms also commonly have an option to em-ploy some kind of noise shaping, which pushes the fre-

quency response of the dither noise to areas that are lessaudible to human ears. This has no statistical benefit, butrather it raises the S/N of the audio that is apparent to thelistener.Proper application of dither combats quantization noiseeffectively, and is commonly applied during masteringbefore final bit depth reduction,[12] and also at variousstages of DSP.

4.3 Jitter

One aspect that may degrade the performance of a digitalsystem is jitter. This is the phenomenon of variations intime from what should be the correct spacing of discretesamples according to the sample rate. This can be dueto timing inaccuracies of the digital clock. Ideally a digi-tal clock should produce a timing pulse at exactly regularintervals. Other sources of jitter within digital electroniccircuits are data-induced jitter, where one part of the dig-ital stream affects a subsequent part as it flows through thesystem, and power supply induced jitter, where DC rip-ple on the power supply output rails causes irregularitiesin the timing of signals in circuits powered from thoserails.The accuracy of a digital system is dependent on the sam-pled amplitude values, but it is also dependent on thetemporal regularity of these values. This temporal de-pendency is inherent to digital recording and playbackand has no analog equivalent, though analog systems havetheir own temporal distortion effects (pitch error andwow-and-flutter).Periodic jitter produces modulation noise and can bethought of as being the equivalent of analog flutter (Rum-sey & Watkinson 1995). Random jitter alters the noisefloor of the digital system. The sensitivity of the con-verter to jitter depends on the design of the converter. Ithas been shown that a random jitter of 5 ns (nanoseconds)may be significant for 16 bit digital systems (Rumsey &Watkinson 1995). For a more detailed description of jit-ter theory, refer to Dunn (2003).Jitter can degrade sound quality in digital audio systems.In 1998, Benjamin and Gannon researched the audibilityof jitter using listening tests (Dunn 2003:34). They foundthat the lowest level of jitter to be audible was around10 ns (rms). This was on a 17 kHz sine wave test sig-nal. With music, no listeners found jitter audible at levelslower than 20 ns. A paper by Ashihara et al. (2005) at-tempted to determine the detection thresholds for randomjitter in music signals. Their method involved ABX lis-tening tests. When discussing their results, the authors ofthe paper commented that:

'So far, actual jitter in consumer productsseems to be too small to be detected at least forreproduction of music signals. It is not clear,however, if detection thresholds obtained in the

6 5 DYNAMIC RANGE

present study would really represent the limitof auditory resolution or it would be limitedby resolution of equipment. Distortions due tovery small jitter may be smaller than distortionsdue to non-linear characteristics of loudspeak-ers. Ashihara and Kiryu [8] evaluated linearityof loudspeaker and headphones. According totheir observation, headphones seem to be morepreferable to produce sufficient sound pressureat the ear drums with smaller distortions thanloudspeakers.' [13]

On the Internet-based hi-fi website, TNT Audio, Pozzoli(2005) describes some audible effects of jitter. His as-sessment appears to run contrary to the earlier papersmentioned:

'In my personal experience, and I woulddare say in common understanding, there is ahuge difference between the sound of low andhigh jitter systems. When the jitter amount isvery high, as in very low cost CD players (2ns),the result is somewhat similar to wow and flut-ter, the well known problem that affected typ-ically compact cassettes (and in a far less evi-dent way turntables) and was caused by the nonperfectly constant speed of the tape: the effectis similar, but here the variations have a farhigher frequency and for this reasons are lesseasy to perceive but equally annoying. Veryoften in these cases the rhythmic message, thepace of the most complicated musical plotsis partially or completely lost, music is dull,scarcely involving and apparently meaningless,it does not make any sense. Apart for harsh-ness, the typical “digital” sound, in a word...In lower amounts, the effect above is difficultto perceive, but jitter is still able to cause prob-lems: reduction of the soundstage width and/ordepth, lack of focus, sometimes a veil on themusic. These effects are however far more dif-ficult to trace back to jitter, as can be caused bymany other factors.' [14]

5 Dynamic range

The dynamic range of an audio system is a measure ofthe difference between the smallest and largest amplitudevalues that can be represented in a medium. Digital andanalog differ in both the methods of transfer and storage,as well as the behavior exhibited by the systems due tothese methods.

5.1 Overload conditions

There are some differences in the behaviour of analogand digital systems when high level signals are present,where there is the possibility that such signals could pushthe system into overload. With high level signals, analogmagnetic tape approaches saturation, and high frequencyresponse drops in proportion to low frequency response.While undesirable, the audible effect of this can be rea-sonably unobjectionable (Elsea 1996). In contrast, digitalPCM recorders show non-benign behaviour in overload(Dunn 2003:65); samples that exceed the peak quanti-zation level are simply truncated, clipping the waveformsquarely, which introduces distortion in the form of largequantities of higher-frequency harmonics. The 'softness’of analog tape clipping allows a usable dynamic range thatcan exceed that of some PCM digital recorders. (PCM,or pulse code modulation, is the coding scheme used inCompact Disc, DAT, PC sound cards, and many studiorecording systems.)In principle, PCM digital systems have the lowest levelof nonlinear distortion at full signal amplitude. The op-posite is usually true of analog systems, where distortiontends to increase at high signal levels. A study by Man-son (1980) considered the requirements of a digital audiosystem for high quality broadcasting. It concluded thata 16 bit system would be sufficient, but noted the smallreserve the system provided in ordinary operating condi-tions. For this reason, it was suggested that a fast-actingsignal limiter or 'soft clipper' be used to prevent the sys-tem from becoming overloaded (Manson 1980:8).With many recordings, high level distortions at signalpeaks may be audibly masked by the original signal, thuslarge amounts of distortion may be acceptable at peaksignal levels. The difference between analog and digi-tal systems is the form of high-level signal error. Someearly analog-to-digital converters displayed non-benignbehaviour when in overload, where the overloading sig-nals were 'wrapped' from positive to negative full-scale.Modern converter designs based on sigma-delta modula-tion may become unstable in overload conditions. It isusually a design goal of digital systems to limit high-levelsignals to prevent overload (Dunn 2003:65). To preventoverload, a modern digital system may compress inputsignals so that digital full-scale cannot be reached (Joneset al. 2003:4).

5.2 Resolution

The dynamic range of digital audio systems can exceedthat of analog audio systems. Typically, a 16 bit analog-to-digital converter may have a dynamic range of between90 to 95 dB (Metzler 2005:132), whereas the signal-to-noise ratio (roughly the equivalent of dynamic range, not-ing the absence of quantization noise but presence of tapehiss) of a professional reel-to-reel 1/4 inch tape recorderwould be between 60 and 70 dB at the recorder’s rated

6.2 Digital filters 7

output (Metzler 2005:111).The benefits of using digital recorders with greater than16 bit accuracy can be applied to the 16 bits of audioCD. Stuart (n.d.:3) stresses that with the correct dither,the resolution of a digital system is theoretically infinite,and that it is possible, for example, to resolve sounds at−110 dB (below digital full-scale) in a well-designed 16bit channel.

6 Signal processing

After initial recording, it is common for the audio sig-nal to be altered in some way, such as with the use ofcompression, equalization, delays and reverb. With ana-log, this comes in the form of outboard hardware com-ponents, and with digital, the same is accomplished withplug-ins that are utilized in the user’s DAW.A comparison of analog and digital filtering shows tech-nical advantages to both methods, and there are severalpoints that are relevant to the recording process.

6.1 Analog hardware

θ

Phase shift: the sinusoidal wave in red has been delayed in timeequal to the angle θ , shown as the sinusoidal wave in blue.

Many analog units possess unique characteristics that aredesirable. Common elements are band shapes and phaseresponse of equalizers and response times of compres-sors. These traits can be difficult to reproduce digi-tally because they are due to electrical components whichfunction differently from the algorithmic calculationsused on a computer.When altering a signal with a filter, the outputted signalmay differ in time from the signal at the input, which iscalled a change in phase. Many equalizers exhibit this be-havior, with the amount of phase shift differing in somepattern, and centered around the band that is being ad-justed. This phase distortion can create the perceptionof a “ringing” sound around the filter band, or other col-oration. Although this effect alters the signal in a way

other than a strict change in frequency response, this col-oration can sometimes have a positive effect on the per-ception of the sound of the audio signal.

6.2 Digital filters

Digital filters can be made to objectively perform betterthan analog components,[1][15] because the variables in-volved can be precisely specified in the calculations.One prime example is the invention of the linear phaseequalizer, which has inherent phase shift that is homo-geneous across the frequency spectrum. Digital delayscan also be perfectly exact, provided the delay time issome multiple of the time between samples, and so canthe summing of a multitrack recording, as the sample val-ues are merely added together.A practical advantage of digital processing is the moreconvenient recall of settings. Plug-in parameters can bestored on the computer hard disk, whereas parameter de-tails on an analog unit must be written down or otherwiserecorded if the unit needs to be reused. This can be cum-bersome when entire mixes must be recalled manually us-ing an analog console and outboard gear. When workingdigitally, all parameters can simply be stored in a DAWproject file and recalled instantly. Most modern profes-sional DAWs also process plug-ins in real time, whichmeans that processing can be largely non-destructive untilfinal mix-down.

6.3 Analog modeling

Many plug-ins exist now that incorporate some kind ofanalog modeling. There are some engineers that endorsethem and feel that they compare equally in sound to theanalog processes that they imitate. Digital models alsocarry some benefits over their analog counterparts, suchas the ability to remove noise from the algorithms and addmodifications to make the parameters more flexible. Onthe other hand, other engineers also feel that themodelingis still inferior to the genuine outboard components andstill prefer to mix “outside the box”.[16]

7 Sound quality

7.1 Subjective evaluation

Subjective evaluation attempts to measure how well anaudio component performs according to the human ear.The most common form of subjective test is a listeningtest, where the audio component is simply used in the con-text for which it was designed. This test is popular withhi-fi reviewers, where the component is used for a lengthof time by the reviewer who then will describe the per-formance in subjective terms. Common descriptions in-

8 7 SOUND QUALITY

clude whether the component has a 'bright' or 'dull' sound,or how well the component manages to present a 'spatialimage'.Another type of subjective test is done under more con-trolled conditions and attempts to remove possible biasfrom listening tests. These sorts of tests are done with thecomponent hidden from the listener, and are called blindtests. To prevent possible bias from the person runningthe test, the blind test may be done so that this person isalso unaware of the component under test. This type oftest is called a double-blind test. This sort of test is oftenused to evaluate the performance of digital audio codecs.There are critics of double-blind tests who see them asnot allowing the listener to feel fully relaxed when evalu-ating the system component, and can therefore not judgedifferences between different components as well as insighted (non-blind) tests. Those who employ the double-blind testing method may try to reduce listener stress byallowing a certain amount of time for listener training(Borwick et al. 1994:481-488).

7.2 Early digital recordings

Early digital audio machines had disappointing results,with digital converters introducing errors that the earcould detect (Watkinson 1994). Record companies re-leased their first LPs based on digital audio masters in thelate 1970s. CDs became available in the early 1980s. Atthis time analog sound reproduction was a mature tech-nology.There was a mixed critical response to early digitalrecordings released on CD. Compared to vinyl record,it was noticed that CD was far more revealing of theacoustics and ambient background noise of the recordingenvironment (Greenfield et al. 1986). For this reason,recording techniques developed for analog disc, e.g., mi-crophone placement, needed to be adapted to suit the newdigital format (Greenfield et al. 1986).Some analog recordings were remastered for digital for-mats. Analog recordings made in natural concert hallacoustics tended to benefit from remastering (Greenfieldet al. 1990). The remastering process was occasion-ally criticised for being poorly handled. When the origi-nal analog recording was fairly bright, remastering some-times resulted in an unnatural treble emphasis (Greenfieldet al. 1990).

7.3 Super Audio CD and DVD-Audio

The Super Audio CD (SACD) format was created bySony and Philips, whowere also the developers of the ear-lier standard audio CD format. SACD uses Direct StreamDigital (DSD), which works quite differently from thePCM format discussed in this article. Instead of using agreater number of bits and attempting to record a signal’s

precise amplitude for every sample cycle, a DSD recorderuses a technique called sigma-delta modulation. Usingthis technique, the audio data is stored as a sequence offixed amplitude (i.e. 1- bit) values at a sample rate of2.884 MHz, which is 64 times the 44.1 kHz sample rateused by CD. At any point in time, the amplitude of theoriginal analog signal is represented by the relative pre-ponderance of 1’s over 0’s in the data stream. This digitaldata stream can therefore be converted to analog by thesimple expedient of passing it through a relatively benignanalog low-pass filter. The competing DVD-Audio for-mat uses standard, linear PCM at variable sampling ratesand bit depths, which at the very least match and usuallygreatly surpass those of a standard CD Audio (16 bits,44.1 kHz).In the popular Hi-Fi press, it had been suggested thatlinear PCM “creates [a] stress reaction in people”, andthat DSD “is the only digital recording system that doesnot [...] have these effects” (Hawksford 2001). Thisclaim appears to originate from a 1980 article by Dr JohnDiamond entitled Human Stress Provoked by DigitalizedRecordings.[17] The core of the claim that PCM (the onlydigital recording technique available at the time) record-ings created a stress reaction rested on “tests” carried outusing the pseudoscientific technique of applied kinesiol-ogy, for example by Dr Diamond at an AES 66th Con-vention (1980) presentation with the same title.[18] Dia-mond had previously used a similar technique to demon-strate that rock music (as opposed to classical) was badfor your health due to the presence of the “stoppedanapestic beat”.[19] Dr Diamond’s claims regarding dig-ital audio were taken up by Mark Levinson, who assertedthat while PCM recordings resulted in a stress reaction,DSD recordings did not.[20][21][22] A double-blind sub-jective test between high resolution linear PCM (DVD-Audio) and DSD did not reveal a statistically significantdifference.[23] Listeners involved in this test noted theirgreat difficulty in hearing any difference between the twoformats.

7.4 Analog warmth

Some audio enthusiasts prefer the sound of vinyl recordsover that of a CD. Founder and editor Harry Pearson ofThe Absolute Sound journal says that “LPs are decisivelymore musical. CDs drain the soul from music. The emo-tional involvement disappears”. Dub producer AdrianSherwood has similar feelings about the analog cassettetape, which he prefers because of its warm sound.[24]

Those who favour the digital format point to the resultsof blind tests, which demonstrate the high performancepossible with digital recorders.[25] The assertion is thatthe 'analog sound' is more a product of analog format in-accuracies than anything else. One of the first and largestsupporters of digital audio was the classical conductorHerbert von Karajan, who said that digital recording was“definitely superior to any other form of recording we

9

know”. He also pioneered the unsuccessful Digital Com-pact Cassette and conducted the first recording ever tobe commercially released on CD: Richard Strauss’s EineAlpensinfonie.

8 Was it ever entirely analog or dig-ital?

Complicating the discussion is that recording profession-als often mix and match analog and digital techniques inthe process of producing a recording. Analog signals canbe subjected to digital signal processing or effects, andinversely digital signals are converted back to analog inequipment that can include analog steps such as vacuumtube amplification.For modern recordings, the controversy between analogrecording and digital recording is becoming moot. Nomatter what format the user uses, the recording prob-ably was digital at several stages in its life. In case ofvideo recordings it is moot for one other reason; whetherthe format is analog or digital, digital signal processing islikely to have been used in some stages of its life, such asdigital timebase correction on playback.An additional complication arises when discussing hu-man perception when comparing analog and digital audioin that the human ear itself, is an analog-digital hybrid.The human hearing mechanism begins with the tympanicmembrane transferring vibrational motion through themiddle-ear’s mechanical system—three bones (malleus,incus and stapes)—into the cochlea where hair-like nervecells convert the vibrational motion stimulus into nerveimpulses. Auditory nerve impulses are discrete signallingevents which cause synapses to release neurotransmittersto communicate to other neurons (see here.) The all-or-none quality of the impulse can lead to a misconceptionthat neural signalling is somehow 'digital' in nature, butin fact the timing and rate of these signalling events is notclocked or quantised in any way. Thus the transformationof the acoustic wave is not a process of sampling, in thesense of the word as it applies to digital audio. Instead itis a transformation from one analog domain to another,and this transformation is further processed by the neu-rons to which the signalling is connected. The brain thenprocesses the incoming information and perceptually re-constructs the original analog input to the ear canal.It is also worth noting two issues that affect perceptionof sound playback. The first is human ear dynamic rangewhich for practical and hearing safety reasons might beregarded as 120 decibels, from barely audible sound re-ceived by the ear situated within an otherwise silent en-vironment, to the threshold of pain or onset of damageto the ear’s delicate mechanism. The other critical issueis manifestly more complex; the presence and nature ofbackground noise in any listening environment. Back-ground noise subtracts useful hearing dynamic range, in

any number of ways that depend on the nature of the noisefrom the listening environment: noise spectral content,noise coherence or periodicity, angular aspects such aslocalization of noise sources with respect to localizationof playback system sources and so on.

9 Hybrid systems

While the words analog audio usually imply that thesound is described using a continuous time/continuousamplitudes approach in both the media and the reproduc-tion/recording systems, and the words digital audio implya discrete time/discrete amplitudes approach, there aremethods of encoding audio that fall somewhere betweenthe two, e.g. continuous time/discrete levels and discretetime/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) behaviourat the microscopic scale,[26] and asynchronously operatedclass-D amplifiers even consciously incorporate continu-ous time, discrete amplitude designs. Continuous ampli-tude, discrete time systems have also been used in manyearly analog-to-digital converters, in the form of sample-and-hold circuits. The boundary is further blurred by dig-ital systems which statistically aim at analog-like behav-ior, most often by utilizing stochastic dithering and noiseshaping techniques. While vinyl records and commoncompact cassettes are analog media and use quasi-linearphysical encoding methods (e.g. spiral groove depth, tapemagnetic field strength) without noticeable quantizationor aliasing, there are analog non-linear systems that ex-hibit effects similar to those encountered on digital ones,such as aliasing and “hard” dynamic floors (e.g. frequencymodulated hi-fi audio on videotapes, PWM encoded sig-nals).Although those “hybrid” techniques are usually morecommon in telecommunications systems than in con-sumer audio, their existence alone blurs the distinctiveline between certain digital and analog systems, at leastfor what regards some of their alleged advantages or dis-advantages.There are many benefits to using digital recording overanalog recording because “numbers are more easily ma-nipulated than are grooves on a record or magnetized par-ticles on a tape” (Rudolph & Leonard, 2001, p. 3). Be-cause numerical coding represents the sound waves per-fectly, the sound can be played back without backgroundnoise.

10 See also• Audiophile• Audio quality measurement

10 13 BIBLIOGRAPHY

• Audio system measurements

• History of sound recording

11 Notes[1] Note that Laserdisc, despite using a laser optical system

that has become commonly associated with digital discformats, is an old analog format, except for its optionaldigital audio tracks; the video image portion of the con-tent is always analog.

[2] Unless imposed DRM restrictions apply.

[3] It is technically possible, to implement analog systemswith integrated digital metadata channels.

12 References[1] “Chapter 21: Filter Comparison”. dspguide.com. Re-

trieved 2012-09-13.

[2]

[3] Sony Europe (2001). Digital Audio Technology 4th edn,edited by J. Maes & M. Vercammen. Focal Press.

[4] Driscoll, R. (1980). Practical Hi-Fi Sound, 'Analogue anddigital', pages 61–64; 'The pick-up, arm and turntable',pages 79–82. Hamlyn. ISBN 0-600-34627-7.

[5] Technics EPC-100CMK4

[6] “mastering”. Positive-feedback.com. Retrieved 2012-08-15.

[7] Byers, Fred R (October 2003). “Care and Handling ofCDs and DVDs” (PDF). Council on Library and Infor-mation Resources. Retrieved 27 July 2014.

[8] Thompson, Dan. Understanding Audio. Berklee Press,2005, ch. 14.

[9] Muraoka, Teruo; Iwahara, Makoto; Yamada, Yasuhiro(1981). “Examination of Audio-Bandwidth Require-ments for Optimum Sound Signal Transmission”. Journalof the Audio Engineering Society 29 (1/2): 2–9.

[10] Dunn, Julian (1998). “Anti-alias and anti-image filtering:The benefits of 96kHz sampling rate formats for thosewho cannot hear above 20kHz” (PDF). Nanophon Lim-ited. Retrieved 27 July 2014.

[11] Nishiguchi, Toshiyuki; Iwaki, Masakazu; Ando, Akio(2004). Perceptual Discrimination between MusicalSounds with and without Very High Frequency Compo-nents. NHK Laboratories Note No. 486 (Report) (NHK).Retrieved August 15, 2012.

[12] Katz, Bob (October 3, 2007). Mastering Audio: TheArt and the Science (2nd ed.). Focal Press. ISBN 978-0240808376.

[13]

[14] “Jitter explained - Part 1.4 [English]". Tnt-audio.com.Retrieved 2012-08-15.

[15] John Eargle, Chris Foreman. Audio Engineering forSound Reinforcement, The Advantages of Digital Trans-mission and Signal Processing. Retrieved 2012-09-14.

[16] “Secrets Of The Mix Engineers: Chris Lord-Alge”. Re-trieved 2012-09-13.

[17] “Digital stress”. The Diamond Center. 2003 [1980].Archived from the original on 2004-08-12. Retrieved 17July 2013.

[18] Home. “AES E-Library » More on -Human Stress Pro-voked by Digitalized Recordings- and Reply”. Aes.org.Retrieved 2013-08-16.

[19] Are the Kids All Right?: The Rock Generation and Its Hid-den Death Wish, John Grant Fuller, ISBN 0812909704,pp130-135

[20] http://www.acoust.rise.waseda.ac.jp/1bitcons/1bitforum2002/Mark.pdf

[21] “Red Rose Music SACDs”. Redrosemusic.com. Re-trieved 2013-08-16.

[22] “Stereophile eNewsletter”. Stereophile.com. 2005-07-05. Retrieved 2013-08-16.

[23] Blech, Dominik; Yang, Min-Chi (8–11 May 2004).“DVD-Audio versus SACD Perceptual Discrimination ofDigital Audio Coding Formats” (PDF). Audio Engineer-ing Society. Retrieved 27 July 2014.

[24] James Paul (2003-09-26). “Last night a mix tapesaved my life | Music | The Guardian”. London:Arts.guardian.co.uk. Retrieved 2012-08-15.

[25] “ABX Testing article”. Boston Audio Society. 1984-02-23. Retrieved 2012-08-15.

[26] “Analog or Digital?". St-andrews.ac.uk. Retrieved 2012-08-15.

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• Nishiguchi, T. et al. (2004). “Perceptual Discrim-ination between Musical Sounds with and withoutVery High Frequency Components”, NHK Labora-tories Note No. 486, NHK (Japan Broadcasting Cor-poration).

• Pozzoli, G. “DIGITabilis: crash course on digital au-dio interfaces. Part 1.4 - Enemy Interception. Ef-fects of Jitter in Audio”, “TNT-Audio - online HiFireview”, 2005.

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14 External links• Skeptoid #303: Are Vinyl Recordings Better thanDigital? at Skeptoid

12 15 TEXT AND IMAGE SOURCES, CONTRIBUTORS, AND LICENSES

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