Editing and Mixing Sound: Optimising your Recordings

Now I've covered most of the basics for getting a sound source recorded, we can begin to touch on the mountain that is post-production. There are many stages involved in getting your sound from a 'raw' form (that is, straight from the recorder) into something that can be used and integrated into a production; be that a film, game, user interface for a vehicle etc. As this series of posts is going to cover quite a lot in terms of what you can do with sounds in post, I'll start with the step that all productions should start with - optimising recordings.

For the following post, I'll be using Audacity, just because it's a free tool that anyone can use on both Mac and PC. However, the methods shown can apply to most audio editing tools as well as DAWs (Digital Audio Workstations).

Step 1: Cut and Chop, but be wary of a Pop!
Inevitably when recording sound, there will be points of (almost) silence and sections of noise which will be of no use to you. The best thing to do with this sound is to either silence it or delete it all together. To silence the sounds you don't need, Audacity has a feature which allows you to 'Generate' it. The below image shows a sound recording with 2 pieces of audio we want, and some noise that we don't.

This is the editing window for Audacity. Here, we have the recording and I've selected
the noise that we want to remove.

With the default selection tool, you can drag along the desired area of which you want to 'create' the silence, or remove the unwanted noise.

With this noise still selected, we go to Generate and Silence. A small prompt appears
outlining the length of silence to generate (which should be the selected area).

The final product is shown below, now effectively without the unwanted noise.

The selected noise has now been removed from the recording.

You can (and should) take this further though. With all editing, there is the risk of interrupting a zero-crossing point. What is a zero crossing you ask? Well, when a sound wave fluctuates, it does so above and below a point of no movement - known as the zero crossing point. This is illustrated below, where the vertical line lies:

If you choose to remove a portion of your sound, you have to make sure that the start and end of the cut is on a zero crossing. Otherwise, you get what is shown below: a sharp jump in the waves fluctuation. The result of these non-zero crossing edits is a popping sound, which can cause a lot of issues for loops especially.

So how do you prevent such horrible audio gremlins? There are two things you can do which will both help keep the sound from popping and increase your work flow.

1. Zero Crossing Snap: In a lot of audio editing software, there is an option to only snap to zero crossing points. This means that, no matter where you click on an audio region, the cursor will only  snap on a point of no movement. This saves time, as you're not forever having to zoom in and fine tune the selection at a waveform level. This is something I would recommend using a lot of the time, with only a few cases where you might need otherwise, like creating loops that start and end above or below the zero crossing point.

2. Fading: Particularly for editing noise and unwanted audio either side of that you wish to keep, an easy way to clean up zero crossings and unnatural sounding cuts is to apply short fades. Obviously, this means a fade in before the desired audio and a fade out at the end of it. For splicing audio together, you can use this same technique on both bits of audio, and overlay one on the other ever so slightly so an almost 'seamless' transition occurs. Wherever possible, I would always try to bake these fades in, to save strain on a DAW session file. Fades can take up vital CPU power, so having all these edits in the audio file itself really helps put the work elsewhere for your processor, such as the Automation of a track.

Separating Sounds
When you've made a clear separation for your sounds by deleting unwanted audio, it's now time to create individual files for these. As far as I'm aware, Audacity doesn't have the most intuitive process to save separate sounds, but Adobe Audition (which I use a lot of the time) allows you to separate sounds and save them accordingly with very little input. Regardless, we'll go over how to process these sounds in Audacity.

So at this point, we have 2 sounds that were recorded within the same take. The first thing we'll need to do is cut one sound out (or copy it if you're wary of data loss). The next thing to do is create a new session and add a mono or stereo track, depending on the type that you copied from; stereo in this case, as the recording was made in stereo.

When this has been created, we simply paste the audio into it and we have our sounds separated. Now, if you've recorded lots and lots of sounds in one session that you'd like to chop up, this is obviously going to take a bit of time. Unfortunately, Audacity doesn't have any tools to compensate for this, but another method is available where you can keep a single session and not have to chop up all your audio. This does require you bouncing the file directly out, but it means you can separate everything much quicker. All you have to do is select the audio you want separating and then Export Selection (make sure you have the format WAV or PCM selected, I'll explain why later).

Removing Unnecessary Audio
This can apply to every kind of production, but is especially important for Game Audio. Due to the memory limitations on consoles and mobile devices, every aspect of a game must be optimised to the nth degree, including audio. The first means of optimising a sound in this instance is removing any waste audio at the start and end of it. The most effective way of doing this is by selecting the silence up to the start of the waveforms movement and then zoom in to fine tune the selection.

Here, I've zoomed in to fine tune the selection. It's far too easy to delete too much by
accident, so always listen to the sound as you're deleting.

There are some instances you must be wary of though. In the past, I've removed a little too much audio either side, which has created a distinctive 'jump' in volume. More often than not, this occurs when removing audio from the end of a sound, as natural reverberence can be mistaken for unwanted noise. To avoid this, always listen to the sound as you're deleting; if you delete too much, just go back on yourself (undo is your friend!).

This is a good example of too much audio selected for deleting.

If for whatever reason the sound you have has a 'jump' in volume that you can't go back on, there is another solution - a short fade in or out, as I discussed earlier. For example, you have a sound which has a very short build up before the body of the sound (a ray gun possibly?). Unfortunately, when the sound was separated from a session with a few takes, the build up was cut off slightly and now sounds unnatural  To help improve this, you can select a short portion of the start and apply a 'fade in' to it.

If the cut off is at the start, select a small portion and apply a simple Fade In...

...Similarly if the end is cut off, apply a short Fade Out.

With other productions, such as sound for film or music, it's not a huge issue if silence is left in the file, as automation or gating can remove this from the final mix down. However, new technologies in the near future (I'm looking at you, Pro Tools 11) will mean than removing silence WILL have a benefit on CPU usage, at least in the post-production stages. This is due to the way in which plugins currently work to process sound: when the session is playing, all plugins are in constant use, regardless of there being sound in the track to play, which is quite a large waste of processing power. In this new software, the DAW will now look forward in a track to see exactly where a plugin needs to turn on and off, vastly reducing usage.

Now that the sound has gone from being recorded to separated to optimised with the removal of unwanted sound, we can now move on to the volume side of the waveform.

Step 2: Gain and Normalisation causes much less Frustration
Thankfully, this next step doesn't take too long to accomplish, but has a lot of issues that need to be avoided. Basically, when you have your sound recorded, the signal strength generally doesn't take advantage of the headroom available. This is for good reason though, as it's used as a safety net in case the sound decides to jump in volume. However, with the sound recorded, we can now adjust this gain level to take advantage of that space left behind.

The easiest, quickest and safest way to achieve this is by using the normalisation function on your audio editor. This will take the entire waveform (or a selection) and bring the gain up until the highest peak hits the limit that you set. For example, the below image shows a waveform on the left which is unchanged. If we normalise to 0dB, it will increase the volume until the highest peak is exactly at 0dB, which gives us the waveform on the right.

This way, the waveform is increased in volume, but avoids distortion.

The second method is by increasing the gain manually. This does give you a little more freedom and is great for when your sound source has a little more noise floor than anticipated, but runs the risk of distorting the waveform... and once the sound is distorted in a destructive editing environment, you can kiss it good bye. This is why you should ALWAYS account for backups, save your sessions often and save them separately for progress!

Step 3: Simple EQ, which is easy to do
So just from listening to your sound source, you should have an initial impression of the frequency range it has and what you'll want to gain from it in production. This is where you can use that impression to help you get a feel for what frequencies will ultimately be removed and kept. At this stage however, we just want some very simple EQ that will remove the 'invisible' frequencies. By this, I mean the frequencies outside of what the source uses. Well how is this possible? Surely if you record a source with a range between 500Hz and 5kHz, you'll only pick up those frequencies? Unfortunately, the nature of a recording states otherwise - your microphone will literally pick up everything it is capable of. This is why we have shock mounts for condenser microphones, to help prevent deep rumbles and sharp movements coming out on the recording. In fact, no matter how well you set up a microphone, there will always be some form of unwanted frequency that needs removing in some capacity; this is why the HPF (high pass filter) is your best friend. This will bypass all that sub-frequency content that would otherwise come back to haunt you in post production.

Now I've covered that, let's look at a real world example. Here again, I have my recorded sound, now cleaned up with all unwanted noise removed and gained correctly.

By going up to Effect and selecting Equalization from the toolbar, you can add a very simply HPF that will remove any unwanted low frequency content. Below you'll see the interface which has been built for Audacity. A lot of modern DAWs have a simple button which adds a HPF; all you have to do is adjust at what frequency it starts to cut and how quick the volume slopes off. With this though, you get a simple line tool which you can adjust by adding points and moving these around. I've drawn 2 points on this and left 1 at 0dB around 100Hz and the other dragged all the way down to remove all those frequencies and below.

It's worth mentioning here that you don't really want to cut too much. Later on in post-production or in another project, you may want to use a certain frequency range that was removed in this step. What you want to do is hear how the EQ affects the sound before applying it, as the whole point is NOT to have the HPF affect the sound obviously. It may seem very strange to say that after all I've explained, but it's more of a cleaning exercise than an attempt to make your source sounds better. There are two simple examples of where this is a clear benefit later on in production:

1. Layering sounds: When you get to a point in the production where multiple sounds start to pull together and play along side each other, these sub-frequencies start to build up if you haven't removed them. They can cause a lot of problems with the bottom end, so you can save some time by removing these now.
2. Pitch shifting: If you don't remove these frequencies and decide to pitch shift your sources up, you might start to hear what was once too low to hear naturally. E.g. if you have an 'invisible' frequency at 20Hz and pitch shifted your sound up 3x, this would turn into noise at 60Hz, which is well within hearing range.

You can also use a LPF (low pass filter) to cut out a lot of the higher frequency content. This is best for being used on any low frequency sounds that don't have any mid or high frequency content initially, as you don't want to remove anything you might need in the future. Again, it's a cleaning exercise to make your life a little easier later in the post-production stages. Removing unwanted low frequencies is more important at this stage though.

Now that we've got our basic source all cleaned and ready to go, we can bounce the sound ready for use in a production.

Step 4: Bouncing your Sound and the King is Crowned
This is very important to get right for any production. At this stage, you don't want to loose any quality from the recording session, which should have been done at as high a quality as possible. The biggest mistake made here is thinking that mixing down or bouncing a sound to mp3 or another lossy format is ok, as long as it has a high Kbps rate. This is not good! Only when your production has gone through a final mixing and mastering stage are you even allowed to think about mp3 or lossy, which I choose not to condone anyway. When bouncing your sound, use a lossless format like .aiff or .wav.

As long as you've taken the steps above, this should be very easy to do. In audacity, all you are required to do is Export the sound from the file menu, and make sure the format is WAV (or PCM [Pulse-Code Modulation] as some software shows it) as shown below.

I want to mention something that I feel is very important before I conclude this blog post. Consider how you name your sounds and where you bounce them, as you'll probably be saving a lot of them. It's far too easy to save sounds with odd names or ones that have no meaning. Before you know it, you're taking a significant amount of time to find what you need. It's therefore best to come up with a naming scheme that will allow for quick and easy searching of specific sounds. Let me explain what I mean.

First, you may want to start the name of the file with what kind of sound it is. If it's an ambience, you may put an 'A' at the start. Next, you may have many different types of ambiences, like mechanical or forestry. For this, you can add a '_Mec' for mechanical, or '_For' for forestry. Then finally if you have a few ambience tracks with the same feel, you can have a number for each one. The final name therefore would be along the lines of 'A_Mec1' or 'A_For3'. Another example for an SFX of a big gun with natural reverb could be 'SFX_Gun_Big_Verb1'. You get the idea.

Conclusion: I hope what I've covered has made sense for the most part. As said earlier: at this stage, you don't really want to be altering what your source sounds like; you just want to clean it up and make it easy to work with once you get to your production. When we come to the DAW stage, with adding sounds and creating track groups, we can really open the doors to EQ, compression, effects and all the lovely bits of Sound Design that really make it worthwhile. Also, many apologies for the section titles; on reflection, they're in line with what a teacher might put on some slides to make their subjects more interesting. Please feel free to slap me through the internet.

Next time, I'll attempt to cover what is the bigger picture of a production and how to frame your mind for mixing, levels, avoiding things like over-compression and the dreaded dynamic range which even I struggle with. It is all in the planning!

Alex.

Recording Sound - Experimentation!

The very best thing about recording sound is that you're not bound by what you can achieve. I've talked about quite a lot over the past few posts, regarding recording techniques and best practices. However, at no point was I trying so much at telling you what to do. This is more of a loose guide for recording techniques. I say loose, because you can pretty much do what you like, as long as it sounds good. The key is, it doesn't matter how you record it; if it sounds good, it IS good!

So to that end, always use your ears as well as your equipment. A good example of this is when it comes to the mixing stage and you're removing problematic frequencies: you can find the frequency graphically on your screen, but it's always best to fine tune which frequency by looking away from the screen and listening. Let your ears guide you! Mixing will come at a later stage though, this post is very much about recording - the first step in your sound journey.

The main consideration for experimentation is having the time. You won't have much time to experiment if you're on the set of something being filmed, or with the band in the studio, so make sure you take plenty of time to do it. Thankfully (aside from equipment and software cost/rental), it's free to record sounds. You can do it over and over to your hearts content. And if you don't have posh equipment, that's fine too! You can use whatever microphones you have to experiment, such as phones,  mp3 players, or even poundland PC microphones; my number one go-to when I started experimenting 5 or 6 years ago. Why not take advantage of that time and be a bit whacky? After all, the most famous of Sound Designers only found some very classic sounds by experimenting (a lot found by accident!).

Hitting Things

When I say hitting anything with
anything, I mean it.

The easiest way to experiment with recording is to hit things... with other things. You'll be surprised what hidden gems of sounds will come out of some household items when you hit them in the right way with the right item! And you can literally do it with anything; just point a microphone towards it and record.

For one of my final year modules at university, I actually do just that for a composition. The basis was, create a percussive or musical piece out of one or several ordinary objects. It was only when I came home one weekend that I had the idea of recording my brothers VW golf mkII! I was literally smacking things, winding windows up and down, slamming doors and the glove box, revving the engine. But the part that really sticks out for me was a spare exhaust part he had in the garage. It had a very bell like quality which was a lot more concise and 'singular' than I thought it would be. That is, there weren't any clashing frequencies that made it dissonant. So I suspended it from a guitar stand and whacked it with some drum sticks. Below is the entire piece I created for the module, and you'll hear the exhaust near the end, which is used as a bell/glockenspiel type instrument:


For all the sounds, including this particular one, I imported them into Kontakt 4 and used the built effects and loops to design the sound for the piece. The exhaust was the easiest really, just because I only had to detune it slightly to match middle-C, and the rest did itself. Love it when that happens!

Microphone Manoeuvring

Sometimes you can get a completely
different sound depending on the angle

This is where your possibilities for record become endless. I think the mistake a lot of people make when approaching the recording of a source is trying to get it in one position which sounds 'the best', and keeping it there for the entirety of the recording. Especially for sound design, I would always recommend trying at least a few different angles, more so for a source that is being recorded as a 'sound'; i.e. not a source being recorded to represent itself in music or film.

The item in the picture to the left is an electric can opener. Items like this are great for all sorts of clunks, machinery and a wide range of engine-type sounds. One may choose to record from a distance, which will exclude a lot of the lower frequency content and get that clunky small-machine type sound. However, if you put the microphone very close, you get a much deeper intimate sound, good for slowing down and using as ambience for a factory setting.

The below track contains several different clips of the can opener, recorded from different angles. As you'll hear, the sound changes can be quite dramatic, and would compliment many different applications of the sound. Below is a list (and illustration) of how each short clip was recorded:

1: From above, front.

2. From above, back.

3. From above, side. (Number 4
Was the mic in the same position
with the can opener pushed against
the mic stands leg)

5. Mic pushing activator down.

6. Mic pushed against back opening.

7. Mic pushed against side.

What really gets the experimenting going is when you combine this item with others to enhance the sound.

For example, placing the can opener on a wooden bench and then using a contact microphone to record the bench as the can opener is working, you get an aggressive bassy sound, because of the direct transfer of energy. Similarly, you can place it on a hollow structure made of wood or plastic, and record with a normal condenser inside; this should give a resonance to the sound.

This tool of sound was from
the Eden Project. Big boomy
sound!

Tubes are also a widely used item for changing the properties of an otherwise normal sound. Depending on it's size and shape, the sound will bounce around inside, which will exaggerate a certain frequency range. The smaller the tube, the higher the frequency. This is usually recorded from an open end.

You can also put a membrane on the end of a tube and attach items to this, which will create a unique resonant sound. The image to the right shows one such item, which has a long spring attached to a membrane on this tube (made of wood). The resultant sound, when the spring moves, is much like a rumbling thunder.

Sound tools like these are very useful if you're stuck on resources or time; they allow you to create sounds that would otherwise need to be taken from an SFX library, or need a lot of planning to record properly.

Below are 3 clips I recorded with the tube sound tool. The first clip uses the same microphone above: at the start, the mic is half way into the tube, and for the end of the clip, moved out but still facing the opening. The second clip is me pulling the spring tight against the edge of the tube, which is recorded from outside the opening and then inside. For the final clip, I found that shouting into the tube caused a spring reverb effect. However, due to the tubes resonance, you get some very interesting sounds at different pitches.

However, I digressed slightly from the original topic, but I thought it was worth noting.

Be Random!
Well, what else can I say? Just use your imagination! Seriously, the amount of times I came across a sound I needed by changing the angle an object was at, or rolling it on the ground, putting it inside something, submerging it in water, putting water inside it, putting my phone in developer mode to stick the vibrate on constantly and holding it against said item... You can literally do anything to get the sound you want!

Be Forth-Thinking
If you plan on using any of the sound you're experimenting with in a production, always always always think ahead about what the sound will be used for. If you're doing a film project that involves a lot of out-of-the-ordinary sounds, you'll more the likely have to slow down or speed up a lot of them, so record these sounds at a high sample rate (96kHz). This means the detail is retained when slowed down.

Also think about composite sounds. When creating a machine say, you won't usually have a single recording for it. You'll have layers, each creating different elements of the frequency content or components of the machine. For example, a bassy engine sound might come from the can opener. Then for movement, you may have a car starter motor ticking over and pitch shifted. Then for a robot-like voice, you can record your voice and play it back through a speaker, with the cone attached to a membrane on a metal tube, which will give it a metallic sound. See what I did there?

Conclusion: If you have a recording device, or microphone and interface, you have the tools to create endless sounds. Even with free software such as Audacity, you can pitch shift, speed up, slow down, apply EQ and multitrack to create and shape any sound you can possibly imagine. As long you let your ears guide you, the possibilities are endless. Go out there and capture some noise!

Next time, we'll get stuck into some simple post production bits - audio editing and optimising, simple EQ and making sure you get the most out of your raw sounds.

Alex.

Recording Sound - The Room, Mic Placement & Gain

So there's your sound source. What's the most effective way of going about recording it? Todays post will go into the details of how important it is to consider the room you're recording in, where to best place your microphone and what levels are appropriate for what you need to get out of the sound. As we've covered the different types of microphones (Mic Choice Part 1) as well as their respective polar patterns (Mic Choice Part 2), we can incorporate how these will affect our choices of distance, angle and gain levels.

So why can't I just record everything in my bedroom? Well... you can. There's nothing stopping you getting the most out of the sounds you need with a reasonable representation, which can be enhanced in post production. However, taking some simple steps to prepare your room, or choosing a room dependant on the source can improve your recording immensely. Below, I'll go over what you might use for the appropriate desired effect on your recordings.

Dampened and Clean - If you plan to go the route of a reflection/reverberant-less recording that will allow you to add any and all effects afterwards, you'll want to use a room that is smaller and with as little hard surface as possible. The most typical setup for a studio recording is to have a reasonably sized room (4l x 6b x 3.5h metres) that incorporates lots of foam tiling for sound absorption and texturing that would reflect sound in all directions (to reduce reflections and possible phasing on microphones). There's no reason you can't get this effect at home though. A bed duvet pinned against the length of a wall will provide a sufficient amount of dampening. Some even choose to keep egg cartons and line their walls with them, which also helps. The thing to remember, which even I've forgot to consider in the past, is your floor and ceiling. These reflect sound as much as walls, especially when you have a laminate flooring. The easiest way to resolve this is by having carpeted flooring and the means to pin a duvet or carpet-like material to the ceiling, as well as your walls. And if you're budget can take it, definitely invest in some high-density acoustic foam tiles - you can easily buy a pack of these from ebay for £35/$55 which would be sufficient for even a professional environment.

Reverberant - It may be the case that you want to record your source with some natural reverb, which is perfectly fine if you know that this is the desired effect for your production. As reverb has unlimited potential in sound and effect, it's good to play around with this if you haven't had experience before. I'll go over a few examples and what they might add to your recording.

Music: If you're a guitarist who wants to have a huge reverberant sound for a lead solo part, you might want to have your amp and mics in a large church-like hall. The time at which the sound reverberates is usually between a second or two, creating a large sustained sound. This can also be true if you want a large drum sound, but i'd advise to have at least a little dampening to control this; otherwise, the sound isn't as precise, which won't help in the mixing stage.

Sound for TV/Film: A lot of sound effects and dialogue in this instance are recorded in a dampened/clean studio. However, you may not be able to bring a church bell or a huge crowd into your tiny recording studio. Instead, you can record them in their natural environment, or one which will reflect (pun not intended) the scenario on the screen. As long you perform some test recordings to ensure it sounds fine, you can save some time in post-production (as well as some CPU power from those plugins!).

Let's now consider the source sound. Much like with microphone choice, you have to understand how your source is creating sound physically, in order to find the best placement and gain level. For example, if you're recording an acoustic guitar, there isn't a lot of physical movement in the air apart from the musicians hand moving up and down to strum the strings. On the other hand, when you're trying to record a vocal, there's a considerable amount of air moving around with the singer breathing, shouting, as well as the troublesome popping caused by 'b's and 'p's. If you don't know what I mean, just place your hand about 2" in front of your face and say "Barry and Peter": you should feel the air shoot out more on the 'b' and 'p'.

Another consideration, especially for music, is how it may hinder the musicians ability to play their instrument. The worst thing you could do for any musician is to place a microphone or stand in a position that would mean they aren't comfortable and therefore can't perform in what should be their top form. From personal experience, this means microphones on drums that I would catch with my sticks when moving around the kit, or stands behind me that I would catch with my elbow. In short, always consider the musicians needs before placing microphones to get what may technically be the best sound; if they aren't playing their best, you wouldn't get the best sound from them anyway.

Linking in with musicians, but not exclusively relating to them, the proximity effect is also something you would need to think about. I say linking in with musicians, because the proximity effect can add a sense of intimacy with a vocal or guitar. I won't go into too much detail (as it's quite complicated), but due to the physics involved with the diaphragm and the short distance between the source and the microphone, a bass boost occurs naturally. You can test this yourself: just get someone to say a few words into one of your ears a few feet away, and then have them say it up close to your ear - you should hear it. This is used a lot in music and movies to emphasise a voice, but it's really up to you what you want to use it for: after all, if you use it and don't need it, you can (for the most part) EQ out the effect in post production.

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General Microphone Placement: I think the best (and only) way to describe microphone placement is through several different examples. These examples will work around a single instrument or source, and then detail different scenarios in recording them:

Vocals / Voice Over
As mentioned briefly, the human voice is quite varied in terms of both dynamic range and frequency. The way to work out the best microphone placement and gain is what you'll be using it for.
Music: I would always try and go for a studio-type acoustically treated room and have the microphone set up at mouth level with your singer. Generally you want the mic to be around 6" away. However, you won't have to worry about distance too much as an engineer, because the singer will (or should) move back and forth depending on the volume and effect. As with the proximity effect discussed earlier, they may move in to about 2", or for very loud singing, they may choose to back up to 8-10". Gain wise, you'll want to get them to sing a few lines repeated at as loud a level as necessary for the song or section, making sure the pre-fade level is touching about -3dB. This gives plenty of head room for short loud bursts. You can also consider the polar pattern, which would probably be cardioid to avoid capturing the room tone, but if you do want this, use a figure-of-8 pattern to control it.
ADR (Automatic Dialogue Replacement): This one has always fascinated me. I must clarify what this is first: all ADR is recorded in the studio after filming (hence dialogue replacement); this is common practice in many films. Basically, a film set is usually very noisy and, even though the sound is usually recorded regardless, you wouldn't get a seamless transition between shots. So instead, they record dialogue afterwards in a studio which gives them more control. So in terms of mic positioning, you want to try and record it as if you we're listening to them in the scene. For example, if they happen to be very close to the screen, you want quite an intimate recording similar to vocals as described above. If they're stood in the middle of a room a few metres away from the camera, you want to record them from a few feet away, usually above them. To get a better idea of this, just check out some DVD extras of films: Lord of The Rings has a dedicated feature on sound, and animated films usually have a section devoted to ADR (although not as direct as recording technique).

Electric Guitar/Bass
If you're either a guitarist of bassist, these tricks are so easy for you to have a mess around with. Both are pretty much the same, but guitars amps tend to have a slightly more direct sound from the speaker.
Live: When you don't have much control over where the amp goes, you tend to have to put up with what you have (especially in smaller venues). If you can, try and have the speaker unit in a reasonably open space with the back exposed; otherwise, you'll get exaggerated peaks and troughs in the bass frequencies. However, the key to remember for live sound is that you will never have as much control as the studio; always concentrate on suiting the sound for the venue. So moving back to the guitar amp; if you have a single cone, place your dynamic mic facing towards the centre of the cone, either directly or at a slight angle. If you have a 4/6 speaker cab, pick a cone and mic it up the same way. After all, you'd expect the amp/cab to be doing most of the work.
Studio: Because you will have more control, you can be much more specific about your mic placement to change the sound to your needs. The best way to find the guitar sound you want is to work out where it will lie in the mix. Are you recording a rhythm part? Is it a scorching solo? Fat power chords? All of these encompass different microphone positioning, because moving the mic around the speaker cone can dramatically alter how the recording turns out. The centre of the cone is where you get a very precise sound because it's where the most movement occurs. On the other hand, the edge of the cone is where the least movement occurs, so you get a bassier rounder sound with less high frequency content. Therefore, if you want to record a rhythm guitar part for example, you'll want to keep the microphone away from the centre of the cone so it doesn't take all the attention in the mix. Similarly with a lead guitar part, you want to place it in the centre to capture as much of an intimate in-your-face sound as possible.

Loud One-Shots
This one isn't very specific, but describes basic technique for anything in the nature of loud sounds that could potentially cause distortion. One particular example is fireworks. They have a whizz and then a very loud short bang. I'd say record the firework from a normal distance (8-10m) and follow the firework with your microphone; swinging it upwards as it lifts off. Gain wise, I would recommend keeping it low for a test recording, check what level came out and then alter accordingly. If it was recorded right, you'll have a very quiet whizz and then the bang heading towards 0dB (ideally -0.5dB). If your recorder has a very low noise profile and there isn't too much back ground noise, you can use a transient designer to bring out the whizz and tail of the bang which will make it sound a lot sweeter. More on that in another post though!

Traffic/Public/Machinery
This all lies on the application you'd use it for. For sound and film (or music if you chose to include something like that), recording in stereo in the best way to go about it. If it's not the focus of the mix (which is usually the case), you can always get away with recording 2 mono tracks and summing them to stereo panned hard left and right. For games which would use point sources, you can either record in stereo and sum to mono/split the tracks, or record in mono. This is especially important for ambience which I'll detail in a later post when I get to game audio implementation in UDK.
Regardless of this, if it's background noise you want, try and record the sound from a distance. With the public, this is vital, as the listener is drawn to the human voice and a particularly apparent one will stick out like a soar thumb. Mic placement is pretty simple: just point it at the crowd; for gain, keep it at about -3dB so you get a healthy signal if possible. If you're pushing your recorder though, back it off a little to avoid a high noise floor, as you'll more than likely pull this sound down in the mix anyway. I'd also recommend a cardioid polar pattern, as hyper-cardioid may be too precise for a 'crowd' sound, where omni-directional wouldn't give you much control over what you were recording.

Conclusion: Microphone placement and gain is a bit of art in itself. You really have to be forward-thinking and reflective, based upon the image you're portraying or the overall mix. As long as you have a good plan, you should be able to make the right decisions. And before I forget, I would go over Drum mic placement and gain, but I think that's big enough for it's own post! So I'll leave it here for now.

Thank you very much if you've read this far, it was a big one today. Next time, we'll go over some experimental things and the endless possibilities of recording your sources! I'm rather excited about this one I must say...

Alex.

Recording Sound - Microphone Choice [Part 2]

Click here for Part 1

Coming straight from part 1, there's a little more to learn about microphones. Obviously, we now know that dynamic microphones are great for high volume sounds, where as condensers are great for their dynamic range and sensitivity. What we're going to touch on next is the variation in sizes and shapes, as well as the 3D pattern at which sound is picked up relative to the diaphragm.

Rode NTG-1 with a smaller
1cm Diaphragm

Microphones are designed to receive sound. However, The way in which this is achieved varies widely depending on the size and polar pattern of a microphone. So what does this mean? Well imagine a set of speakers. The speaker cones come in several different sizes to help represent the broad range of frequencies. Tweeters are usually about 1" in diameter and deal with anything from 7-22kHz, with 4-5" cones handle the mids at 400-6Khz, which leaves large 10" cones that will delivery the very deep 20-300Hz range. This is no different for microphones.

AKG C414 with a larger
1" Diaphragm

The best way to think about this is setting up and recording a drum kit. The first part you may want to mic up is the snare drum. As the frequency range tends to be low to high mids (500Hz-5kHz), you won't need a large diaphragm microphone in order to fully replicate the sound realistically. Similarly, you won't want to use a pencil condenser microphone with a 1cm diaphragm; this would be more useful on the cymbals which create a lot of those higher frequencies. For the kick drum, which has a fundamental frequency ranging from 50-200Hz, the diaphragm on the microphone would need to be much larger: about 1" for most proprietary kick or condenser microphones. Some may even want to convert a medium sized speaker cone into a microphone, which would be able to pick up the sub frequencies of the kick (20-60Hz). However, I digress: can you tell which instrument I play yet?

The point is, before you pick a microphone, consider the frequency range that the source will be creating, or at least the frequency range that you'd like to use.

Now, polar patterns. Have you ever wondered why microphones are the shape and size they are? Why some are long and thin, where others can be cuboid-like or have the diaphragm exposed on 2 sides? These different designs have all been implemented for the purpose of creating a pick up pattern; better known as the Polar Pattern.

The Polar Pattern is a 3D shape (usually depicted in 2D) that highlights the area around a diaphragm that can pick up sound. There are 4 main patterns available for microphones, which are described below:

Omni-Directional Polar Pattern
credit:wikimedia.org

Omni-Directional - This is the simplest pattern available for a microphone and is the most basic to understand: it picks up sound from all directions equally. You can imagine the pick up pattern as being a large sphere around the diaphragm. This pattern can be found on a lot of vocal microphones and as a choice on many higher end condenser microphones.

Uses: As it works to pick up everything, it tends to give the most realistic representation of the sound, with both the source and reflections on the recording.

Cardioid Polar Pattern
credit:wikimedia.org

Cardioid - This pattern, much as the name may depict it to be, is shaped like a heart. Rather than picking up sound from all directions, it's only able to reach sounds on one side, with minimal 'leakage' of sound from the back. As you can see, there is a small amount from the rear, which is only due to the nature of a diaphragm being effected from either side regardless; for the most part though, this is a single direction pattern.

Uses: This pattern is particularly useful in both live and studio settings, where sound leakage may want to be avoided. For live purposes, you'll want to prevent as much leakage from other instruments as possible, so recording in this single direction very much helps. Similarly, if you want to record in a studio with as little reverberation as possible, the cardioid pattern should help with that.

Hyper-Cardioid Polar Pattern
credit:wikimedia.org

Hyper-Cardioid - Coming directly from the previous pattern, Hyper Cardioid is almost an exaggerated version of Cardioid, which is far more directional, leaving even less leaked sound or room tone to affect the source sound. However, what it gains in it's precision it looses in it's mono-directionality. I.e the pattern turns more into a figure-of-8 shape, which means sound is picked up from the rear. This shouldn't be significant enough to burden a recording though, especially with higher end microphones.

Uses: As this provides a more precise directionality, it is usually used for broadcast recording. For example, a lot of shotgun condenser microphones (such as the NTG-1 shown above) use this pattern so that the newscaster, who may be stood outside of a building or built up area, is picked up clearly at a distance without too much of the external noise being picked up.

Figure-of-8 Polar Pattern
credit:wikimedia.org

Figure-of-8 - Most common for condenser microphones which are exposed either side, figure of 8 is a fairly self explanatory recording pattern: sound is picked up equally from either opposite side of the diaphragm. By having this, two sources can be picked up either side of the microphone, without a large amount of spill that would otherwise occur on an omni-directional pattern. Similarly, a single source can be recorded, with a controlled amount of room tone to the desired effect.

Uses: One good example I read about recently was to use this pattern with a guitar amplifier either side of the microphone, with one guitar split and wired into both amps. As the distortion would be slightly different on either amp, this would create an almost doubled-up effect, which is naturally summed from either side of the mic to create a huge guitar sound. Also, for a lot of vocal work, you may want to record 2 singers either side of the mic for again the doubled up effect which is good for a pre-chorus emphasis.

Conclusion: These ideas are relatively simple to understand and are mostly common sense. However, knowing these simple steps and querying what your own microphones have can really change how you think about recording. Making sure the frequency response and directionality is accounted for can improve recordings greatly, especially with a combination of microphones and their respective polar patterns.

Next time, we'll go into the placement of microphones in relation to a source to make the most out of your recordings, as well as how multiple microphones can have adverse effects due to phase, with simple solutions to resolve this.

Thanks again for reading!

Alex.