Basic Principles of Audio Recording

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This is the first part of a series of short articles written by a sound engineer with many years' experience. The idea is to explain in plain language how to make a quality sound file.

Contents

Digital recording, a brief overview

Recording to a computer these days is cheap and relatively easy. In 1997 recording software cost cost me 150 Pounds. These days there are much better free and open source applications to cover all that one can do in a studio. Audacity is one of the best. I'll try to refer to Audacity as much as I can so that what I write can be tried by all those who have the inclination.

Digital and tape, what's the difference?

Tape

  • Magnetic tape records sound as a continuously variable magnetic field along the length of the tape; all of us over teenage are familiar with cassettes and possibly eight track cartridges. Magnetic tape has advantages and disadvantages against digital.
  • Tape recordings degrade over time.
  • Even the best tape has inbuilt hiss.
  • Tape has to be moved extremely accurately both in position and speed, requiring very high quality hardware. Just one part out of adjustment can ruin a recording.
  • Tape distorts the sound; admittedly it distorts the sound in a musically pleasing way such that Pink Floyd, Elton John and Kate Bush have never recorded in digital studios.
  • Tape forgives high recording levels; the nature of magnetism is that when tape is over-saturated it makes music sound nicer, so much so that in studios it is deliberately over-driven to achieve this richly pleasing effect.

Digital

Where the waves on tape are stored as field strength, digital recordings are stored as a long series of numbers, which is what computers excel at. In fact storing and shifting numbers is the only thing a computer can do, but they can do it very fast, so much so that the numbers can be coloured spots on a screen or voltages on a speaker. Once you start shifting the numbers quick enough, the pictures move and the speaker sings: video and audio.

How it works

The ever changing voltage caused by a sound vibration from microphone or mixer is presented to a tiny measuring circuit called an Analogue to Digital converter. At predetermined intervals the circuit measures the microphone voltage and assigns a number as its value. At CD quality this voltage is measured 44,000 times per second. You may have heard mention of 16 bit and 24 bit and 32 bit sound; this refers to the accuracy to which the measurements are taken. A 16 bit binary number equates to about 65,000 in old money. CD is 16 bit 44k, therefore every 44th of a millisecond a measurement is taken and stored as a number between 0 and 65,000.

To play back that recording you do quite the opposite, every 44th of a millisecond a number is taken from memory and presented to another circuit, the Digital to Analogue converter, which then produces a voltage on its output in proportion to the number. The ever changing numbers produce an ever changing voltage which drives the speaker and you've got your sound vibrations back.

Digital recordings when looked at really closely do not look like a smooth curvy wave, they look like a series of steps, but the steps are so small and the duration of them so short that you don't hear the steps, it smooths out to an uncannily accurate reproduction, which is why even the smallest cheapest MP3 player sounds much better than the best cassette player.

Simple really, when someone sensible takes all the waffle away, isn't it?

Benefits and risks of digital recording

Digital recordings are very accurate, the accuracy determined only by the quality of the DA and AD converters.

But there is a risk: if the signal going in exceeds the measuring capacity of the converter it can't possibly get a higher number than 65k or a lower one than zero. Digital does not forgive overdrive. Digital distortion will make you throw off your headphones; it is about as pleasant as the sound made by a scallywag dragging a sharp key along the side of your brand new car.

Consequently when making a recording it is imperative to see to it that the signal never reaches and crosses 0dB DFS (Digital Full Scale).

On digital equipment zero decibels is the measure of the highest level, all other values are expressed as minus decibel numbers, all the way down to minus 96dB in the case of 16 bit or CD quality.

When I record to digital tape, I record with a maximum peak value of -12 dB; in film soundtracks the value is more like -20dB. This leaves room for unexpected peaks to remain undistorted they can be compressed later, but clipping (going over the maximum level) can't be fixed easily.

In the days of vinyl records, the instruments were recorded to 24 track tape resulting in tape hiss. When it came time to mixdown, the tape signals were passed through processors and effects units and the mixing desk, picking up electronic noise along the way. The mix then went down to a stereo master tape picking up more tape hiss. The master was then taken to the pressing plant where it was passed through yet more processors in the mastering process picking up yet more electronic noise until it was cut to a master pressing disc, which was then used to press records, which ended up on your turntable. The fact that the resultant record sounded extremely clean and nice explains why a decent analogue studio costs a million pounds, but an equivalent Pro Tools set up costs fourteen thousand.

Decibels (dB) an explanation

Audio, whether floating through the air or as an electrical voltage, is measured in decibels, which is one tenth of a bel. When processing sound it is useful to understand decibels, which are not like other measurements. For a start the decibel is logarithmic, because human hearing is logarithmic.

If I give you a ten pound load to carry, you would rate it as some value of heaviness. Then if I gave you another ten pound load to carry, it would feel twice as heavy. Hearing is not like this. Ten watts of audio power playing in a room could be measured at some position as 90dB, twenty watts would register a level of 93dB at the same spot, not 180dB.

That is the rule of thumb. For every 3dB up, double the actual power; for every 3dB down divide by two. This measuring in air is called SPL (sound pressure level) and is a consistent measurement: 90dB SPL is the same volume wherever it occurs.

  • 85dB SPL is the recommended limit for long term industrial exposure without protection
  • a jackhammer is 110dB SPL
  • a jumbo jet on take off is 120dB SPL
  • Motorhead once played a gig above 120dB SPL. (Long live King Lemmy!)
  • 140 dB SPL causes instant and permanent deafness.

When talking about signal levels, decibels are used in different contexts. In an analogue channel of a mixer, dB is a relative measurement where 0dB represents the upper limit of signal strength for that channel but with some leeway above it.

In digital recordings the decibel is referred to as dB DFS (Digital Full Scale) where 0dB DFS is the absolute upper limit and all values are measured in minus values, there is no such thing as a positive dB in Digital Full Scale.

When talking about “line level” electrical signals that pass audio between equipment, 0db V (voltage) is defined as 775 millivolts. Perversely, pro equipment has its 0dB point at a line level of +4dB V and consumer equipment has a line level of -10dB V.

Confused? I am, but the important bit is the part about logarithmic hearing, just remember that part and you'll have a feel for what decibels measure.

Noise Prevention

There are many effective ways of removing background noise once you have got it, but of course, it is better if you can avoid the noise in the first place.


Before reading this, it is useful to read Post Production 1 EQ and hum. That article tells how to identify problem noise. If you know the nature of the noise, you can go back to the recording setup and perhaps prevent it being recorded in the first place.

In The Karate Kid Mr. Miyagi once said, “Best way to avoid punch, no be there”.

So how do we avoid noise getting in?

In a typical studio, obscene quantities of money have been spent on ensuring that noise-producing things cannot get on to the recording. Any machine that produces noise lives in the machine room, and that includes computers with their fans. Studios have very, very long mouse and keyboard cables, and that is not a humorous quip, it is hard fact. In my college days, when it came time to burn a mix to a CD I had to go through two isolation doors into another room to open the CD drawer.

At home we can't go that far, but if we do have an identifiable intrusion on a test recording we can go some way to tracking it down and curing the problem before making a whole book's worth of recordings.

Electrical mains hum

There is often only one kind of hum, that produced by the mains power supply. In America it is 60Hz, in Europe 50Hz. So how does the mains get into our recordings? In a very similar way to the way that generators make electricity and motors consume it.

When a current flows through a wire, it makes a magnetic field around it which pulses in time with the current. The magnetic field can travel across empty space. When a pulsing magnetic field encounters a wire, it makes a current in the wire which pulses in time with the field.

So there in a nutshell you can see how the mains can get into mic cables and the like.

One thing that can prevent this is a metal case. If an electrical device has a metal case, the case has to be earthed for safety's sake, and there is a side benefit. A magnetic field can't get into or out of an earthed metal case. An earthed metal case is a Faraday cage, after the physicist and electrical engineer Michael Faraday.

So how does this apply to us ordinary folk?

Well, plastics technology is so good these days that there aren't many metal-cased electrical appliances in our sitting rooms any more, and although they are usually shielded inside, there might be one in your recording spot that is not so well-shielded.

So if you've found a mains hum, you could start by killing the power to unneeded devices. If this does the trick, you're laughing. But if not, it might be in the stuff that has to stay on, such as the computer itself. Now don't even bother. It's probably cheaper and less stressful to buy a noise-free computer than to modify one that isn't.

But there are two things that can be done at this point.

Microphone to sound card noise

If, and only if, you're using a mic that goes straight into the mic socket, you can gain a 200-fold improvement by reading Digital Recording, How to do it and getting a mini mixer.

If you're already working that way, there is a more recent innovation, the USB microphone. A USB microphone does not put out an audio signal, it puts out a digital bit stream, and digital bit streams don't care much about dirty magnetic fields and interference.

Let me draw a parallel. Take a Haynes manual for a car and use it until the pages have lots of oily finger marks. Now when you consult the book, you can no longer identify the engine parts in the photographs but the instructions can still be read although the page is dirty. The pictures are like an analogue signal picking up interference. The words are like a USB signal picking up the same interference. You reject the random dirt and only take in the information.

A lot of Librivox readers use USB mics for this reason and for reasons of simplicity: it's a lot less fuss than messing about with mixers.

Computer fan noise

If you find an unacceptable level of random white noise on your recordings, amplify a bit of it and have a good listen. If it's got a good random spread of frequency that sounds like hiss, it is likely to be electronic noise that is introduced in the sound card. But, if it's of a soft warm character, it is highly likely to be the windy sound of the computer fan.

While recording for Radio Mod I noticed this, when I got the room nice and quiet and recorded a solo female performance I could hear the soft noise of the fan through the sensitive mics, so rather than change the mics for less sensitive ones and lose the nuances of the pretty voice I turned off the PC and recorded to tape, a totally silent digital studio tape recorder which didn't have a cooling fan. I then bounced it to computer with out any mics active, problem solved.

If your PC tower is on the desk with a side-mounted fan facing into the desk, you can go a long way to reducing this. My situation: the tower case on the desk with left-facing fan sitting on the right. Cure: put my machine under the desk on the left and the noise would be down to acceptable levels. Feel free to experiment.

This short article does not in any way have all the answers, but it may help you on your way. Good luck and good hunting!

Digital recording: How to do it from scratch

To make decent recordings for LibriVox is easy. If you know that LibriVox exists you already have the most expensive bit of kit you'll need.

All you need to add to that is:

  • Recording software (You can't go wrong with Audacity.)
  • A microphone (not expensive, if it sounds OK then it is)
  • For some mics, a small mixer to act as a pre amp and gain control for the microphone. (Look up Behringer mixers, best value in the business)
  • A pop shield

Microphones

There are condenser mics and dynamic mics.

If you have a condenser microphone it needs power: either it has a battery compartment or it doesn't. If it doesn't, you need a mixer with a 48 volt phantom power button. This sends the mic power up the signal cable.

If you have a dynamic mic you don't need phantom power. A word of warning here, with most dynamic mics, if you switch on phantom power you will instantly need a new mic.

There are three types of connection for mics:

  • USB
  • 6mm or 3.5mm Jack (unbalanced)
  • XLR (balanced)

The USB type has a flat connector that plugs into the rectangular USB port on your computer. XLR is a substantial plug with three pins. 6mm Jack is the round silver plug that you would plug into an electric guitar; 3.5 mm Jack is the small round plug that fits into the sound card on your computer.

The reason for the different types is simply that 3 core XLR is a method of getting interference in the mic cable to cancel itself out. 2 core Jack is less expensive. The USB mic is particularly useful for LibriVox purposes, as it is Plug and Play and suffers less from background noise.

Mixer

Note: it is entirely possible that you won't need a mixer. I know of at least one laptop which could record a nice clean signal through the mic socket but in my experience the mic socket in my desktop machine was filthy with noise. USB mics also don't require a mixer.

Your mixer doesn't have to mix anything, the smallest Behringer has one mic channel and costs peanuts. What it does though is give you a beautiful clean pre-amplifier, gain controls and a set of EQ controls to set the right tonal balance. No microphone has a flat frequency response and if yours does not suit your tastes you can adjust this with the EQ.

To record to the built-in sound interface on a PC, you'll need an output cable which will have two 6mm mono jacks at one end and one 3.5mm stereo jack at the other. The small jack will go into the line socket on your sound card, the 6mm monos will go into the unbalanced outputs from your mixer. You mixer may only have RCA connectors for unbalanced output, in which case get a cable to suit those. (RCA connectors are also known as phonos and are to be found on the back of CD players and on Playstations, one red, one white, the yellow on a Playstation is video.)

Pop shield

This is not essential but if you're close to the mic it will prevent popping on Bs and Ps. Say “Buh” and “Puh” to the palm of your hand and feel the blast of air. That blast is not sound it is wind and wind plays havoc with microphones, that's why TV and film recordists have a big hairy dog on a stick. The hairy dog is hollow with the mic floating in the middle. Sound can get through the hairy dogs furry coat but wind can't.

A pop shield is a disc of tenuous fabric suspended four inches in front of the mic, you can make one with a pop sock and a wire coat-hanger. Hey, maybe that's why they're called pop socks, they stop your mic popping.

Setting up the signal chain, when using a mixer

Put your mic cable in your mic channel. Connect your mixer output to the sound card and you're ready to set the signal.

There are at least three level controls on your mixer and there's a reason for them all to be there, it's called gain staging. Simply said, the level controls ensure that the signal level is optimum at each stage.

Bigger mixers have linear faders, smaller mixers have rotary faders: they do the same job. Faders are always labelled such that when set at 0dB there is room to move up as well as down which is where you get your freedom to adjust the mix if you're mixing. The 0dB setting is where the signal passes through unchanged in value.

Set your mic channel fader and master output fader to 0dB. Just where the mic connects there will be a small rotary control labelled trim, turn it right down. In this position, on the smallest inexpensive mixer you will be able to use the output meter to set the mic trim level. The output meter on my small Behringer is three LEDs, like a traffic light. Bigger mixers may have a bar graph or a needle meter.

At this point you get your sound source going into the microphone. e.g. Read a book.

While slowly turning up the trim control, watch the meter: if it's labelled in decibels you want to be bouncing around somewhere near 0dB. But don't peak too far over 0dB.

In the case of my traffic light meter you want some green light on most of the time with the occasional flash of yellow light; if the red light comes up the signal is too high.

Next look on the mic channel, there should be a red LED usually labelled 'Clip'. If this lights up, the channel is overloading, so turn down the trim a bit. You're nearly there.

Now turn down the master output fader and start looking at the recording meter on Audacity which should be on Record and Pause.

With your sound source still going, start turning up the output fader on the mixer. The reason for this is that 'line level' on a PC is lower than 'line out' on a mixer, so 0dB would be too loud. We have to come up at it from below. This is the gain staging process: setting the gain at each stage so it's the correct level for that stage.

If you find that the computer seems too sensitive, double click the speaker icon in the system tray. When the Windows mixer opens, click Options | Properties. Change the check box from playback to recording. The 'line in' fader may be all the way to the top, drag it down a bit to make the computer less sensitive to the mixer.

Back to the mixer: with Audacity running and in Record and Pause you'll have a meter to watch. Bring up the mixer output until the highest peak is reaching -12 dB on the recorder meter.

At this point, decide if you'll be monitoring the recording on headphones--probably not because all headphones leak sound and it will loop back through the mic. But you do need to do a bit whilst monitoring just to check that it actually sounds OK. While monitoring you can apply a bit of EQ on the mixer; for instance you can take a bit of low EQ off if you are coming out too boomy.

Monitoring must always be done from the end of the line so don't use the headphone socket on the mixer, use the 'line out' socket on the sound card.

If all is OK then you are ready to make your recording. You have just set up a signal chain like a professional would. See? I told you it was easy.

The rest of it is a breeze, just click off the pause and yap.

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