Point of View

We asked Australian musician and sound engineer, Steve Newton to answer the following question...

July 2010   What is the difference between digital and analogue recording?

by Steve Newton with help from Robin Gist

I'd like a dollar for every time I've been asked this question and a dollar for every different point of view on the subject. It seems like a simple question and if there was a simple answer it might be that analogue IS the waveform and that digital is a SAMPLE (or a numerical representation) of the waveform. This, however, is not telling us much about the audible differences and only fuels the hot debate over what's the best format to record to. One thing is very clear and that is that people are pretty much either in one camp or the other. I'm not going to try to say one platform is better than the other here, just that they have known, measurable differences. So let's explore what the differences are in a bit more detail.

Analogue recording setups generally include microphones that "hear" or detect the air pressure changes from a sound source and then generate a small, changing voltage over time at the mic's output socket (a waveform). This waveform is fed into a mic pre-amplifier and that same waveform is then amplified up to "line level" (the sort of level that most audio equipment works with) and this is what is fed into an analogue tape recorder, sometimes via a compressor and/or equaliser (EQ) that modifies the waveform somewhat. The resulting waveform that is recorded is still the original air pressure changes represented as a changing voltage over time from the mic with some slight changes due to the compressor or EQ.

To record this waveform, the recording device ultimately routes the electrically changing signal to a small coil of wire built into a housing, which is otherwise known as a record head. Thanks to the almost magical phenomenon of electromagnetism, the changing electrical signal in the coil produces a varying magnetic field or flux around the head. A very sensitive magnetic material - usually ferric oxide (rust) bonded to a plastic backing medium (recording tape), is passed over the head at a constant speed. As the tape passes over the head and through the varying magnetic field (a direct analogue of the original waveform) a representation of the field is stored in the tapes magnetised particles.

Analogue tape recorders have separate heads that are used in the playback function. When the machine is made to play back the recording, the magnetic tape is moved over the replay head. As the magnetic pattern in the tapes particles moves past the head, a small voltage is created in the coil in the playback head that is then amplified back up to line level (electromagnetism works both ways!). This line level waveform/signal is then sent to a more powerful amplifier that drives your room speakers connected to it and the result is a waveform moving the speaker coil in a way that resembles the original waveform that the mic detected in the first place. It might be said that the original pressure wave was sent through various electrical circuits, converted to a varying magnetic field and stored magnetically, then replayed and sent to a device that creates air pressure changes (a speaker) and is the same waveform all the way.

With a digital recording setup some things are still analogue but not all. You still have the same analogue microphone to "hear" the air pressure changes from the sound source that generates a voltage at the mic's output socket. This signal still needs to be amplified up to line level with an analogue mic pre-amp and then this signal is then sent to the all-important analogue to digital converter (A to D).

An A to D converter can either be in a computer, inside a stand-alone digital audio recorder or can be just a converter on its own. Think of an A to D converter as like a cold meat slicer that chops the variable analogue signal into slices or "snapshots" of the analogue signal's voltage level at that instant in time. The number of slices you take per second is called the samplerate, and the higher the samplerate, the better the data resolution. A digital clock drives the A to D chip to sample at typically 44,100, 48,000, 96,000 or 192,000 times a second that results in a continuous data stream of 16, 24 or 32 bit binary "words" that is then stored on a hard drive (again using electromagnetism) or alternative digital storage media (memory sticks etc).

To convert our data stream back to an analogue signal, we need a digital to analogue converter (D to A). The D to A chip takes the data stream and generates a voltage on it's output. As the binary numbers pour in, the voltage on the output keeps changing and presto an analogue waveform! It is, in theory, just like the original analogue waveform that was sampled in the first place. This waveform/voltage/signal is sent to an amplifier usually in the converter first to get it up to line level. Then this goes to the main power amplifier that drives the room speakers. The speakers modulate the air pressure in the room and there is the sound again: produced by sampling a waveform, representing is as a binary number and storing the resulting data stream, NOT the actual wave. This is for me, the real difference between analogue and digital!

The big question is: "Which is better - analogue or digital?" Well, this is where it gets tricky. "Beauty is in the ear of the beholder". There is much to be said about recording actual wave energy and then using this stored energy to re-modulate the air to make sound again. Obviously though, the energy from the sound of a real drum for example live in a room, is more than gets stored when you try to record it, digital or analogue. To reproduce a sound with a speaker and amplifier, with that much energy, requires a lot of power. This is why there are all these amplifiers along the way. Only a small amount of the energy from an analogue sound in the air is stored to the recorder.

In the analogue domain there are lots of variables. The type of oxide on the recording tape for example affects the tone of the recording slightly due to the chemical coating being slightly more or less sensitive to the magnetic field and therefore slightly changing the waveform. Any variation in the speed of the tape player, slight slowing down or speeding up for example, also changes the waveform by effectively stretching or shrinking it over time (wow and flutter). The overall volume range that you generally have to work in is much less than in the real world. Analogue tape also comes with a relatively high noise floor due to friction noise from recording tape passing over the heads. This has been very cleverly hidden with modern noise reduction systems but they require very fancy encoding modifications to the waveform during the recording stage and are then changed back in the playback stage (decoding). This returns the waveform to how it was but can contain some slight artifacts. Often, these are thought of as tonal changes.

Digital recording is advancing all the time. Some of the first products available used low sample rates and low-grade electronic components and didn't do a very good job of trying to describe or capture complex waveforms. To record music, sample rates need to be high. This is very important. Some instruments in the orchestra have incredibly high frequency content. Cymbals have frequencies out to 35 or 40kHz. This is way above what humans can hear so what's the problem you ask?  The problem is that some frequencies modulate against other frequencies. For example, if two violin notes play together but are ever so slightly different in pitch, they will chorus or phase against each other. This inter-modulation happens right across the audio spectrum and is one of the things that give our brains acoustic detail. Even if we can't hear some sounds because they are out of our range, it doesn't mean they aren't having an effect on the frequencies we can hear. Many very expensive and high-grade pieces of analogue audio equipment have frequency responses far beyond human hearing and are easily able to record signals at extended high frequencies.

The standard "red book" CD digital audio format of 16bit/44.1kHz only lets you record or reproduce to 22kHz (Google Nyquist theory if you want to know more about this). As processing speeds and the capacity of large storage devices increase, we have seen and will continue to see better digital sound quality and cheaper storage. I like to record audio frequencies out to 40kHz. This means I set the sample rate of my A to D to 88.2kHz and 24-bit depth. The bit depth is what determines the available dynamic range of an A to D and 24 bits gives me about 118dB of dynamic range (far beyond any analogue recorder that I know of). Recording at 88.2kHz sample rate also allows for an easy "down conversion" to 44.1kHz for red book audio CDs.

Another difference between analogue and digital recording is that the production methods can be a bit different particularly with the ability to "cut and paste" in the digital domain. Often, a digital studio is smaller and many are home or project studios. This usually means that the instruments are recorded separately. If you have an analogue studio, you generally have physically bigger equipment and therefore a larger studio to work in. Recording drums, bass, guitars and keyboards at the same time you're getting vocals, gives the music a more integrated sound. The musicians work better with each other if they are in the same room together. Often this more cohesive recording sounds more like an old style analogue recording.

I think that these days both analogue and digital recording systems are of a very high standard and quality. Yes, some systems sound different to others but are still generally very good. In my opinion, it is down to the sounds that you record. A good microphone in the right spot on a quality instrument with a good player can sound very good, digital or analogue. The reproduction quality is only as good as the playback equipment to some degree anyway (think about MP3's!) and if that equipment is not set up in the right spot in the room or with wrong settings on the EQ, this would have a far greater effect on the quality of the sound anyway. 

To summarise, there are many differences between digital and analogue and the argument on what's better will go on for a while yet. Digital will get better with time but analogue will also, to some extent. My experience is that, seemingly redundant technologies still hang around long after their supposed "use by" date and at least for the moment, you – the engineer/producer/musician have a choice between the two formats.

Steve Newton
Manager,
ENREC STUDIOS

© 2010 Steve Newton