Digital recording techniques compared to Analogue recording techniques
Created | Updated Jan 8, 2003
First things first, all statements in this entry are entirely subjective and are true.
Most (55.5674321%) Sound Engineers agree that Digital audio does not sound as good as Analogue audio. This is due to the constraints of technology and physics.
How the B****Y hell does it work? (By request)
Bits And Pieces
A Binary number is either a 0 or a 1. The lowest useful level of encoding possible is 2BIT (Binary digITs).
There are four levels of loudness in 2BIT recording:
01
10
11 (Loudest or most detailed)
4BIT encryption has 16 levels of loudness (the range of loudness from 0 to the loudest possible signal), 16BIT has 65,536 levels of loudness. The more BITs a recording signal has the more levels of loudness. The sound quality of a digital recording is dependant of the level of encoding used.
Sampling Rates
The sampling rate also affects the quality of the recording, the better the dynamic range of the recording. If a sampling rate is too low it will intercept a waveform too few times to make a proper analysis of the sound. The sample will not intercept the signal enough times to build an acurate representation of the sound recorded.
The normal dynamic range of the human ear is 100-20000Hz, so ideally the sampling rate will be at least double the upper value, allowing it to capture and analyse the full (human) audible range.
This means that the sampling frequency will able sample enough of the waveform of the highest possible pitch and 'guess' the correct frequency and timbre of the sound recorded.
8BIT sound sampled at 22000Hz, usually used in computer games and on the web (because af space constrictions), sounds naff. A sampling rate of 200,000Hz and an encryption of 96BIT would be good, but there is no computer powerfull enough, nor a medium efficient and big enough, to record a sound at that level.
How Digital Audio Is Recorded
This is going to be quite hard to explain, but here goes!Digital audio works by sampling sound into binary code. To save space and simplify the process the sound is usually encoded into 16BIT PCM (Pulse Code Modulation) and sampled at 44100Hz(Hertz). This is the configuration most commonly used for Compact Discs.
First the sound is sampled by using PAM (Pulse Amplitude Modulation). The amplitude of the waveform sampled is measured by each sample taken.After the sound is sampled it goes through a process called quantising. Basicly each set of indefinite samples collected is given a definite value, dependant on the BIT rate of the sample (remember that a 16BIT sample will have 65,356 levels of clarity, or definition). The amplitude of each sample that is taken is either shortened or increased to fit in to the nearest level of clarity. There is an error ratio, usually in the range of half a level (this means a sample could either jump or slip a level). If any errors occur at BIT rates of 8+ and sample rates equal to or greater than 220500Hz, they frequently do at low sample rates and bit levels, you would not necessarily notice (as there are so many levels oF clarity). When you have less levels of clarity to play around with then half a level error is quite a lot, take the 2BIT for example. At anything less than 8BIT where there is supposed to be silence in a track there is noise. This is because the quantising process has infact superimposed itself over the characteristics of the wave!
To stop the sound in lower BIT rated tracks from appearing quite so prominantly we use dither, a process which adds a small amount of noise to make the quantising process unpredictable. This means that instead of a fixed set of variables that the quantiser usually utillises to sort amplitudes, they change for each sample. A triangular sampler usually produces the best results.
The sample is now refered to as PCM.
Aliasing
This is a concept which requires a neatly annotated diagram, but I'll try!Imagine a low frequency sine wave, also imagine eight samples taking place along the wave form. The samples measure the amplitudes of the waveform and, when processed, the digital waveform matches the original sound.
Same conditions, but a frequency roughly eight times higher than the original wave. The digital recording still sounds like the original, low frequency sound wave!
This is because although the sampling rate was the same the sample rate was too low to accurately predict the second, higher frequency correctly, aliasing has taken place!
To beat this, a given fact is the Nyquist Theory. This holds that to reduce the likelyhood of aliasing a sample rate should be at least two and a half times the frequency as that of the highest sound you wish to record. This means that the digital process will sample a sound at least twice along it's wavelength, giving an accurate sample.
Ok how do I now get sound out, smarta**e?
Ahh, re-sampling and re-construction. This employs the use of a low-pass filter. Again using the Nyquist theory we assume that the filter must have a frequency half that of the sampling rate. When a binary impulse is passed through the filter it produces a 'ripple'. Each time an impulse passes through another ripple is created. Each impulse has a different amplitude, so each ripple is of a different size. When the output of the filter creates a waveform it simply 'joins up' the 'ripples', like a piano note is made of the fundamental and its harmonics. The pitch is not perfect however, so the higher (sometimes barely audible) frequencies are reduced in amplitude. The recorded sound now is almost exactly the same as the original. Again low levels of noise, or dithering, is used to negate any aliasing that may occur.Analogue Audio Is Easy
Analogue Audio, when recording on magnetic tape, transfers sound into a voltage and polarises the magnetic paricles in a way which is (more or less) analogous to the sound produced, the sound is 'sampled' instantaneously.
The reason why digital audio is thought to be worse than analogue is that digital signals 'lose' most of the signal that is recorded, it simply records changes in voltage in a given period of time (1/41000th of a second) with a certain amount of clarity(16BIT or with 65,536 levels of perception).
The only problem with analogue recording is that you will always incur noise from the equipment you are recording. Noise comes from mechanical peices moving, hum from cables etc. The information is carried as a change in voltage down a cable and is analougous to the original waveform. The distance the cable travels degrades the signal, and as the information is a part of the signal, this is also means the sound is degraded.
Digital signals do not rely on the quality of signal between devices as the information is sent as a definite off or on (0 or 1) and so is a seperate entity to the signal.
Compare this with analogue and you see the pros and cons of both systems. Analogue technology records all the signal all of the time -this means that all the information included in the sound is recorded (subject to degredation)! Digital recording captures the sound without (much) noise, but the depth of sound is found wanting.
Perhaps it would be useful to think of analogue as taking an infinite (an amount pointless to measure) number of samples a second at an infite level of clarity, but the quality of the signal is affected by the way the information is transferred and how that information is read/written/interpreted. A digital signal, although it has a better delivery system the actual system only records the 'bullet points' of the signal rather than the 'full conversation'.
Compare analogies?
Perhaps it is even easier to think of the two systems of water delivery. A well could represent digital audio, the bucket size the BIT depth, the length of string the amount of times you are able to bring the bucket to the top of the well. A stream could represent analogue audio. The stream constantly flows but there is never an adequate way to retain the water, as collecting the water usually stirs up silt - muddying the water/signal.
Copies, Copies, Copies
Oh yes this part. Imagine copying a video (home recording of course!), the copy looks quite good, in fact almost indistinguishable from the original. Say you lose the original and you need another copy, you copy the copy. The second copy is not as good as the first, and nowhere as clear as the master tape. Imagine doing this 500 to 1000 times. Even using top of the line equipment degredation occurs. Now imagine copying a high quality MPEG-1 layer file in exactly the same way. No appreciable loss in quality. Why?(okay, yes I know I'm supposed to be dealing with audio but this is my entry, so there).
Perhaps the most obvious benefit of digital audio is the fact that the information recorded is generally very hard to corrupt when compared to analogue, and a good deal quicker. Why is this?
Remember that digital information is measured as pulses of voltage, either off or on. There is very little chance of a misunderstanding between one peice of digital kit and another as these deal with definite values.
This 'understanding' between digital machines means that a copy of a master recording will be a clone of the original recording (well not exactly, but enough to fool you, more later).
As the information is binary it very simple to simply copy the information in bulk - an extremely fast process not needing to be done in real time.
Analogue is another matter. The information contained within an analogue system is bulky and unwieldly. Basicly whatever machine is recording it has to wait for a signal from the output device before it can record the signal. This can only happen in real time.
Even if connections between analogue machines are the same as between digital machines, the sound is still only part of the signal (not, as in digital, a separate entity) and still suffers from interference. This interference is caused by proximity to magnetic fields, 'dirty' power even loose connections. Digital signals are still subject to these fluctuations, but as digital relies on pulses of information, rather than a steady stream of information, digital signals rarely incur any penalties (as long as the pulses are powerfull enough to bigger than the background noise).
Clones?
A copy of a digital signal is extremely accurate, as we've discussed. However, it will include some incredibly minute mistakes. These mistakes could be due to any circumstance, buffer underrun, energy spikes, dust on laser etc. Whatever can go wrong will. Unlike analogue copying, a digital copy will be specially coded in a technique first formulated and performed by Phillips (after all they did invent the CD). Basicly samples are not sent in sequential order, instead they are sent and recieved in an apparently random order in a process known as interleaving. This means that if the sample is damaged in some way then the next sequential sample is in another physical space to the first and so on. The CD player is then able to 'fill in gaps' a kind of 'persistance in hearing'.Why The Digital Revolution
The reasons why there has been a digital revolution are:- Digital information is easy to store, and make changes to.
- Digital technology is easy to store, cheap to buy and easy to use.
- The techniques used to encode digital signals mean that the information carried by the signal (0's and 1's) is forever separated from the signal itself. This means no whirr from recording/playback equipment, no noise from the disc etc.
-Record companies are able to charge large amounts for a 'New' technology - even though they now cost less to produce and distrobute than vinyls/cassettes.
-New advances are now able to be made using digital technology therefore increasing sales of new playback devices and utillities increase as the defunct media are put out to pasture. The companies make more money more quickly (then blame everyone else when the market collapses under it's own weight).
-CDs look nice, can be used as coasters/frisbees when they get scratched and confuse guinea pigs.
-It is relativly easy to copy, edit and pirate digital information.
Digital equipment rarely wears out, compared to analogue.
For more info please see A533170 and A543782.
