From home movies and stunt videos shot with just an iphone, to profesionally produced short films, the content on video sharing web sites keeps growing rapidly. A plethora of videos on just about any subject can be found quickly in searches. The choice of flicks beckoning you to watch them are often vast. Some people see putting their movies online as a chance at attaining a bit of fame (or infamy), many even hoping that maybe their video will go viral. Many of these videos are of pretty average quality, and many have really bad audio. On a bit of a higher level than some of these amateur videos is the semi-pro short film. Not quite in the same league as a major budget Hollywood production, but hopefully very watchable and of good overall quality. One thing that most videos benefit from is the use of appropriately chosen music. Using music in video and film can really set the tone and the mood, and add interest to the experience of watching.

If you want your video or movie to be successful, it needs to have something to set it apart from the others in competition to it, and therefore attract viewers. Using well chosen music can certainly be of assistance in attaining this goal. Ask yourself “what is it about your particular video that will attract people to start watching it, and then ultimately watch it through to it’s completion without getting bored?” You don’t want potential viewers flicking to the next video half way through watching yours! Having a good sound track can really help to capture peoples’ attention and their imagination. Most people love music, and often half the pleasurable experience of watching a video is in the hearing of the sound. This can include music, foley and sound effects. Videos and movies with good audio will attract more viewers. It can be a great idea to have some kind of musical soundtrack or bed music in your movie. However, just using a song by your favorite artist, may cause you more trouble than it’s worth, because in doing so you are likely to be breaching copyright, and may even run the risk of having the video removed from circulation. With the crack down on copyright breaches and more vigilant copyright policing, it is a wise move to be sure you have the legitimate right to use any music before you put it in your video.

If you do want to use a particular song by a famous artist, you often need to clear the use of this music with a number of sources. You need to find out who controls the rights to the music, and negotiate with them to use it. This can be both expensive and time consuming. An alternative is to find bands or artists online who are looking to promote their music. You may find music that you like by artists who are just happy to see their music in use somewhere for the exposure it may gain them. If you do find something you think is suitable, contact the artist or their publisher and tell them what you are trying to do, while asking them if you can use their music in your video or film. You will still need to be sure that you have some kind of contract agreement with them that spells out what you are going to use the music for, what you intend to do with it, and define any cost or license terms and agreements so that both parties are covered.

Another alternative is to get a composer to create something specifically for your film or video. Established composers are likely to charge you for this service, and their rates can vary considerably depending on their level of experience or expertise. The best and most experienced ones are likely to charge fairly heavily for this, so that may rule them out for you. However, you can also often find composers who are just starting out and are wanting to get their music heard. Many will do this either for free or cheaply. This may be viewed by them as a chance for exposure or to gain experience. Both parties need to carefully define what their expectations are, and a contract or agreement should be written out covering these expectations.

A third alternative to finding music for your video is to use royalty-free music. This music has been created, published and licensed for use in videos, movies, films, games, TV shows, gaming, and other mulimedia applications, and grants you license to use it. The use of royalty free music can be a practical solution for solving any difficulties surrounding music use and copyright issues. Finding royalty-free music and downloading it to use is far quicker, and often much cheaper than the previously mentioned choices. Whether the videos, movies or other productions are professional or amateur ones, you can legitimately obtain suitable music that is royalty-free and use it in your creations. The music does still have copyrights, but you are given license to use it when you download it. The best royalty-free music libraries have tracks available for download, that can be used anywhere, in any production, and as many times as you like. The good ones are also easy to search when looking for music, in the genre or style you want, and you can quickly and simply play previews of the music on offer. Once you have found a suitable song, you pay a small fee for the license to use that particular track. You download it, and it’s your’s to use in your video, film, game, or multi-media apllication. This leaves you with no worries around copyrights and infringements, and with no ongoing additional costs. Using royalty-free music can be an affordable solution for getting good quality music for video or film. Royalty-free music libraries such as the Rocksuresoundz.com site have separate licenses available for both commercial or non-commercial productions. Non-commercial licenses are cheaper than commercial ones, and therefore it doesn’t matter if your production is one with money-making potential, or is a private or non-commercial one, there are royalty-free license options to cover your particular needs.

The one disadvantage of using pre-created music is that if you are making a film where you want music changes or accents to co-incide with camera shots or cuts is the issue of timing. Because the music already has a set pattern to it, it may be difficult to co-ordinate the music with the changes in shots. An accent in the music may not fall precisely where a desired accent in the video falls. Therefore, a fourth good alternative, particularly for longer movie or short film productions, and one that may be particularly applicable where budget is an issue, is to use a mixture of royalty-free music in places where the timing issues are not so critical, and music specifically composed for your film in places where the timing of accents, changes in scene etc are more important. You can then create a soundtrack for your video that works throughout.

Summary

Wherever you decide to get your music from, it is important to have contract agreements or licenses that clearly define and cover the requirements of all parties involved, and leave no doubts as to what can and can’t be done with the music. This will hopefully leave you with peace of mind, and won’t lead to difficulties further down the line when your film or video is released. For most video production needs the use of royalty-free music is the simplest solution, but for film needs you may need to also use some specially composed pieces to suit the movie.

By Tony Koretz
© copyright April 2012

In this brief tutorial I would like to introduce you to the concept of recording nature and wildlife sounds. This includes capturing the sounds of individual birds and animals, as well as general stereo natural ambiences. I aim to give an introduction to the type of equipment you willl need in order to get started in this field. In no way do I pretend that this will be a comprehensive article covering all angles on the subject, but it should give you some basic guidelines to get you started.

Microphones

The most obvious thing that you will need is a microphone or microphones. No single microphone type is going to cover all your needs for recording nature, wildlife and ambiences.
The recording of individual birds or animals requires a different microphone and different technique from the recording of background and natural ambiences. Here are some guidelines that will hopefully help you determine your microphone needs.

Recording Individual Subjects

For the recording of individual bird or animal sounds, you want a microphone that is very directional in it’s pick up pattern. You want the microphone to pick up sounds from in front of it, while largely rejecting sounds to the side and behind it. There are two types of microphone that are designed to do this effectively. They are the shotgun microphone, and the parabolic dish microphone. These are most often held in a pistol type grip, or in the case of a shotgun microphone, may be held on a boom or fish pole. Most often individual subjects are recorded in mono.

Shotgun microphones employ a hypercardioid or super cardioid capsule set back in an interference tube that cancels frequencies to the sides, while focusing on those to the front of the microphone. The longer the barrell, the more directional the mic is. The shorter shotgun mics are reasonably compact and easy to handle, and the better models produce very good sound recordings. Generally it is best to have these set in shock mounts to prevent handling noise, thumps and bumps. Some examples of quality shotgun mirophones are: Sennheiser MKH416, Rode NTG-3, AKG C568EB, Sanken CS-2, Audio Technica BP 4071.

Parabolic dish microphones gather sounds from a larger area and focus them on the microphone, through the use of a reflector, making the mic very directional. Because the area of sound collected by the dish is larger than that of a shot gun microphone, it in effect delivers more sound to the microphone capsule. Because of this they have great sensitivity, and increased signal output, and enable low noise operation. This allows the possibility of recording quieter and more distant sounds than those of a shotgun microphone. Cardioid or omni microphone capsules are aimed at the reflecting dish, and collect the sounds gathered by the dish. Disadvantages of parabolic dish microphones are that the sound may be sharper and less natural sounding than those achieved by a conventional microphone, and unless the dish is very large they are not suitable for recording low frequency information. They are also somewhat large and unwieldy. The best known models are made by Telinga.

Recording Ambiences And Natural Background Sounds

For the recording of ambiences, we require different microphones and a different technique than for the recording individual subjects. Often times we wish to record these in stereo, and so stereo micing techniques come into play.

Stereo recording with spaced omni microphones requires the use of two identical microphones with omni capsules used as a spaced pair. Positioning is critical when using a spaced pair, and it is a case of experimenting as to the placement and distance apart of the microphones.
Spaced omni recordings tend not to have as strong a center image as XY recordings do,and can seem to suffer from a “hole in the middle.” So hence this technique is not particularly suitable for gathering individual sounds. However, on the other hand, if you are gathering background ambiences that are likely to be used mixed in with sharper, more defined sounds recorded with a shotgun or parabolic microphone, then spaced omnis may be perfect. Having a bit of a “hole-in-the-middle” of your natural ambience track can make it easier to position a separate sound in the forground and center, without it competing with the background ambient sounds.

X-Y stereo micing is a common way of recording stereo sounds, and in field recordings it is often a good choice, particularly if you don’t want to have microphone stands placed a distance apart. The idea is to set up a pair of cardioid microphones (usually on a specially made stereo bar and on a single mic stand) crossed at approximately at right angles (or slightly wider angles)to each other.
This achieves a nice wide stereo image from in front of the mics, while rejecting sounds that emanate from the rear of the microphones. To keep good mono capability it is advisable to have the microphone capsules crossing next to each other. This means either having one microphone placed above the other, or on the same plane and head to head. A disadvantage of this technique over using a pair of spaced omnis is that cardioid capsules tend to color their off axis response more than omnis do, and when recording wide stereo ambience backgrounds this may not be particularly desirable.

Notes On Polar Patterns And Practicality

Omni microphones will only give you good stereo if you separate them by some distance. Having two omni microphones placed close together will really only yield a recording of two near identical mono sounds unless there is some type of barrier placed between the mics. While having omni mics set as a spaced pair may give you good results, this setup will also be less mobile, so having X-Y mics either on a bar or held in two pistol grips may be more practical in many situations. An affordable small diaphram condensor microphone that is available in matched pairs, with screw-on capsules of different polar patterns, and can be used in either the spaced pair of omni or cardioid X-Y configuration is the Octava MK-012.

Stereo microphones on the other hand can be handheld and still have a wide pickup pattern. They can be handy for recording ambiences such as forests or the surf at a beach. They can also be used to record streams and rivers. The additional of mono microphones placed strategically can also be useful, and there exists the possibility of blending these with the stereo microphone recording later.
Models of stereo mics that have shotgun amd M/S capsules in one microphone include Audio-Technica BP4029, Sanken CSS-5, and Sennheiser MKH418S. They utilise a hyper-cardiod shotgun capsule to capture the central “M” (Mid) detail, and a separate figure of eight capsule within the barrell of the microphone to capture the “S” (Side) signals.The Rode NT-4 is a stereo microphone that has two capsules placed in an X-Y pattern. Stereo microphones usually have a special 5-pin XLR connector to send the signal to two separate mic preamp inputs.

Wind Protection

Whatever microphone configuration you decide to go for, one thing that is common to all when you are doing field recordings is the need for wind protection for the microphones. Nothing can spoil recordings more than microphone distortion caused by wind in the capsules. For most situations the use of a blimp, such as those made by Rycote for example, and a fluffy windjammer (also often known as a “Deadcat”) is the best solution. However, sometimes you can get away with just have fluffy windjammers made of fake fur placed tightly over the microphones. Foam wind shields on their own do not offer enough protection outdoors in the wind. Parabolic dish microphones usually have their own wind protectors.

One thing that is common to most nature and wildlife recording is that you can roll off the low frequencies ( unless recording crocodiles or something of that nature) to minimize unwanted noise from handling and wind etc.

Field Recorders And Portable Recording Devices

There are many portable digital recorders available these days that capture very good quality recordings. It is even possible to use the internal microphones in some of these and still get pretty good sonic results. If you do use the internal microphones found in portable recorders you will need a fluffy windjammer placed tightly over the microphones to prevent wind noise. However, better results are normally still gotten by plugging condensor microphones into these recorders, either utilising their internal mic preamps, or plugging microphones into separate preamps, and then feeding the signal to the recorder via analog, s/pdif or AES/BU digital, firewire or USB connections depending on what the recorder includes.
The smaller pocket sized recorders usually only have 3.5 m mini jack connections, while the larger semi pro and professional models include XLR microphone connections. The latter type of connector is preferable because they lock in place, are much more secure and less prone to pull out accidentally. The larger models usually have better built-in microphones, better preamps and better metering as well. Separate level control for each mic input is a desirable attribute.
Good mid-priced recorder models include the Marantz PMD661, Sony PCMD-50, Tascam DR-100, and Fostex FR-2 LE. The 7 series Sound Devices (702, 702T and 722) are rather pricey, but are some of the best professional recorders on the market.

Headphones

It is a good idea to include a good pair of headphones in your field recording kit. Try to use a pair of studio grade headphones that exhibit good sound reproduction and seal well over your ears if possible. It helps to be able to hear what you are recording while you are out and about. Some studio headphones that are designed for acoustic music recording have quite a low output volume which may mean quiet sounds are not so easy to hear. Obtaining some phones with a higher output may be better suited to outdoor recording.

Summary

Regardless of what type of microphone and setup you use, the closer you can get to the subject the better the signal to noise ratio will be. At 100 meters distance, even with a parabolic microphone, you will still have more self noise, background noise and preamp noise to deal with than you would if the same subject was 10 meters away. Get as close as you can without disturbing the subjects you are trying to record. Also consider having some weather protection available for yourself and your equipment, because you don’t want to be caught out having a bunch of expensive equipment unprotected in a surprise downpour of rain.

By Tony Koretz
© copyright April 2012

Brief Introduction to How Microphones Work
Microphones convert acoustical energy or sound waves into electrical energy, thus reproducing the audio signal that we hear. They are therefore transducers, in that they convert energy from one form to another. Microphones have different ways of converting acoustical energy, but one thing that is common to all types, is that they have a diaphragm. This diaphram is a thin membrane that mimics the human ear, and is a piece of material that vibrates when struck by sound waves. The transducer elements of the microphone are housed in the mic capsule. In a typical handheld micophone the capsule is found in the microphone’s head.

Microphone Types

Microphones are usually primarily categorised according to their transducer principle. For example, they may be classed as condenser, dynamic, ribbon, etc.
They can also be labelled according to their directional characteristics, diaphragm size, how they are intended to be orientated (end- or side-address), or used ( boundary, boom, lapel etc).

Dynamic Microphones

Dynamic, or moving coil microphones work like a speaker in reverse. In these microphones the diaphragm is attached to an induction coil, and as the diaphragm vibrates responding to incoming sound waves, that coil moves backwards and forwards past the magnet. The variable current that is created in the coil is then channeled along the wires in the microphone.
Dynamic microphones are robust and can take a bit more of a beating than other types. They do not require phantom power. Most hand-held microphones are of the dynamic type. However, they do have a more restricted frequency response than other types of microphones, and can’t reproduce very high frequencies with accuracy. Often this doesn’t matter too much, but if you want to capture an instrument or voice that has lots of tonal detail found in it’s upper harmonics, using dynamic microphones may not be the best choice. They also require more gain to obtain a useful level of signal than condensor mics. The downside of this is, that the more gain you need to use to get a good signal level, the more the noise potentially also add to the signal rises. For close micing this is not usually an issue, but for distant recording, or the recording of quiet sources it may become a problem.

Condensor Microphones

Condensor microphones are also sometimes known as capacitor microphones.
A condenser or capacitor is an electronic component that stores energy in an electrostatic field. Condensor microphones use capacitors to convert acoustical energy into electrical energy. One of the two plates that is found in the capacitor is made of a very light material and acts as a diaphram.
The frequency response and transient response of these mics is considerably better than most dynamic mics. They are therefore able to reproduce the character and subtleties of an instrument or voice with more accuracy and detail. The output of these microphones is also usually higher than dynamic mics, but they are correspondingly more sensitive to loud sounds, where they may distort the signal, and are also more easily damaged. Condensor mics require the input of phantom power in order to operate. This power is supplied by batteries or by an external source of phantom power.

Large Diaphragm Condensor Microphones
Large diaphragm condensor microphones (LDCs) generally tend to “flatter” the sound source to some extent, and are often chosen for studio vocals, or instrument recording where a warm sound is desired. They are perhaps not as clinically accurate, or have as flat a response as small diaphram condensor mics, but in many cases the flattery may be more appealing and musical than simply flat. They are very sensitive to plosives such as “P’s” and “Bs,” and when used for speech or vocals it is advisable to have a “Pop” filter infront of the microphone. Many LDCs have switches to change the polar patterns of the mic’s pickup, included on them.

Small Diaphram Condensor Microphones
Small diaphragm condensor microphones (SDCs) are often the best choice when you want a natural sound to be captured on sources that have a wide frequency range, or that require a rapid response to transients. They also tend to have the flattest response to all frequencies, and that includes bass, as well as high ones. Microphones of this type are often chosen for recording acoustic stringed instruments, choirs, cymbals, and the recording of concerts. Good SDCs can reproduce a wide range of sounds with great accuracy and detail. Some SDCs are available with screw-on capsules of different polar patterns.

Tube or Valve Microphones
Tube microphones are usually a type of condensor mic that combines capacitor capsules with valve preamplifiers. Valves (or tubes as they are known in some places) are often used to provide a subjective warmth to the tonality of sound. They often sound very smoothe and warm, but generally cost more than most other condensor microphones. They also tend to be more susceptible to damage They require their own external power supply, with a special cable that runs from the power supply to the microphone, and cannot be powered by a conventional phantom power source. They also tend to be noisier than solid state microphones.

Electret Microphones
The electret microphone is another type of capacitor microphone. Most lavalier microphones, consumer cam-corder microphones and many other low cost microphones are electret microphones. The diaphragm is usually quite thick in these mics and has a permanent electrical charge attached to it.
Another type is known as the back-electret microphone, and these have the permanent electrical charge attached to a back plate rather than to the diaphram This means that a thinner and more sensitive diaphram can be used. These are often quite good sounding mics that are relatively cheap to buy .Electret and back electret microphones require a very low voltage input from a preamplifier, and often can be run on batteries. One downside is that the electrical charge does weaken over time and eventually the capsule needs to be replaced.

Ribbon Microphones

Ribbon microphones employ a thin strip of metallic foil that is suspended in front of a magnetic plate. Upon entering the capsule, the sound waves cause vibrations in the foil or “Ribbon”, and they produce fluctuations in the electrical current, which creates the audio signal. The ribbon is a delicate mechanism, with great sensitivity. These microphones generally reproduce a wonderful rich sound over a wide range of frequencies. Ribbon mics tend to capture high frequencies in great detail, but without the brittleness of some condensor microphones. They are also generally the most sensitive to damage of all the microphone types. They do not require phantom power, and some may even be damaged if it is introduced to them inadvertently. Older ribbon microphones particularly, were very delicate, and also costly to buy. However, nowadays less expensive and hardier ribbon mics have found their way into production. Most ribbon microphones are made with a bi-directional polar pattern.

Microphone polar patterns

A Microphone’s polar pattern or directionality, relative to it’s axis, refers to it’s sensitivity to sound as it relates to the direction or angle from which that the sound arrives. Put simply, it is the way that the micophone hears the sound that hits it.

Omni directional microphones generally produce the most natural sound, but since they pick up sound from all around them, and this is not always desirable, they should be used in rooms that are well treated or contolled acoustically. They can also be used outdoors for gathering ambiences.

Unidirectional microphones come in several polar patterns, the most common of which is the cardioid microphone. The sensitivity pattern of a cardiod mic is heart-shaped. Hyper-cardioid microphones are similar but have a tighter area of sensitivity towards the front, and the lobe towards the rear has less sensitivity. A super-cardioid microphone is similar to a hyper-cardioid one, but has more frontal pickup and less rear pickup.
Because unidirectional microphones are relatively efficient at rejecting sounds form directions other than straight in front of them, they are often used for recording vocals and speech.

Bi-directional microphones which are commonly known as figure-of-eight microphones pick up sound with a high degree of sensitivity, from the front and back, but with low sensitivity, from the sides. They are therefore used in applications where sound you want sound to be recorded from the front and back of the micrphone, but not from the sides. Sound from right angles to the mic “nulls” well.

Special Purpose Microphones

Shotgun or Boom Microphones
The Shotgun microphone is a highly directional one. Often also known as boom mics on account of the fact that they are generally suspended by a boom, which is a special stand that enables the microphone to be hung above the subject, or fitted to a boom pole held by a boom operator. The object of these stands is to keep the microphone out of the camera shot on a film or TV set. The stands have adjustments for height and angle, and can also be swivelled. Portable boom poles are also often known as fish poles. Because they are the most highly directional microphones, shotguns pick up very little background noise. However, conversely because they are so highly directional, they must be positioned very carefully to keep sounds on axis.
The capsule element is placed at the back end of a tube that has slots cut along it’s sides; Much of the off axis sound is nullified by wave cancellation. Because they have such a narrow area of sensitivity and great off axis rejection, shotgun microphones are ideal for use on television and film sets.They are also very handy for the field recording of wildlife, or at sports events. The longer the shotgun tube the more directional the microphone is. Contrary to some misguided thinking, they are not like a telephoto lense. In other words, distant sounds will still sound distant, and not magnified as in the case of a telephoto camera lense. Ideally when recording a spoken voice they should be positioned just out of camera shot, and preferably no more than 18 inches (45 cm) from the subject’s mouth.

Lavalier Microphones
A lavalier microphone, lav or lapel mic is a small electret condensor mic or dynamic microphone. The are used in television or theatre, and for public speaking applications. Usually they employ small clips for attaching them to collars and other clothing. The mic cable is thin, and usually gets hidden by the subject’s clothes. Lav mics may be wireless microphones with a transmitter attached to a person’s belt, or cabled and running straight to a stage box or a mixer’s inputs. When lavalier microphones are concealed under clothing in order to be inconspicuous, scratching of clothing against the microphone is a potential problem. Taping the cable is to the person is sometimes done, and often times it is advisable to wrap the head of the microphone in moleskin, or find some other way of preventing clothes or hair from touching the capsule and therefore causing annoying scratching sounds every time the subject moves.

Microphone preamps that have a phantom power supplying +48V on the balanced signal pins are designed for use with condensor microphones. However, if you apply this voltage directly to an electret lavalier microphone capsule, it can be damaged to the point of being ruined in a nano-second. Electret Lavalier microphones usually have their own power supply that provides them with a low DC voltage (usually between 1.5V and 5V). A lavalier power supply usually looks like a barrel or an enlongated XLR connector, and usually contains a battery. If the mic is hard wired to the power supply, then on one end of this power supply is a wire that runs directly to the microphone. If it is not hard wired, but has a detachable power supply, then it will have a connector that mates to the microphone’s cable. With wireless lavalier microphones that are connected to a transmitter, it is usually the transmitter that supplies the low vottage current that the microphone requires to operate.

Boundary Microphones
Boundary microphones are somewhat unconventional, because they are designed to lay on a table, be placed on the floor or attached to a wall. They use the entire surface to facilitate the pickup of sound. The larger the surface on which they are placed, the better the sound is likely to be. The microphone capsule is incorporated into the body, so that it sits just a fraction of an inch above the surface. This eliminates phase cancellations that can happen to other types of microphones. Boundary mics are designed to avoid the coloration of the peaks and troughs and comb filtering that other microphones types placed close to hard surfaces introduce. Where they can really shine is in the picking up of ambiences, crowds of people, audience applause, stage sounds, and room tones. They are also often attached to the underside of the lids of pianos. But because they are so good at picking up ambiences, they are also prone to pickup air conditioning sounds, and other unwanted noise too. Boundary microphones are often referred to as PZM microphones. This is an abreviation for the term “Pressure Zone Microphone.” The PZM trademark belongs to Crown International, but the term has become widely used to describe boundary mics made by other manufacturers as well.
They are miniature condensor microphones, and are available in omni-directional or bi-directional configurations.

Power Supplies:Phantom Power and T-Power

Phantom Power
Condensor microphones require an external power source in order to operate. Phantom power is a DC voltage, that is supplied down pins two and three of an XLR cable to power condenser microphones, active DI boxes and effects pedals. The standard phantom power voltage supplied by
most mixers, preamps and consoles is +48V, however some manufacturers power supplies for their microphones may be less than that. Mixers and preamps usually have a switch for turning this power supply on or off. The switches may be on individual channels, or on groups of channels.

T-Power
T-Power ( also sometimes known as A-B power) was developed for use in portable applications, mainly for film and television sound equipment. It is sometimes still used in field equipment.
T-power is usually 12 volts, and only T-power mics should be connected to T-power inputs. Dynamic microphones and ribbon microphones may be damaged if T-Power is introduced to them, and normal condensor mics will not operate properly on it either.

Summary
All the different types of microphones, and their varying polar charcteristics have their place and uses. Learning which mics best suit which purpose takes a bit of experience. I hope this tutorial has covered many of the basics to help with understanding the concept of microphone choice, and the different applications for which they are suitable.

Written by Tony Koretz © copyright March 2012

A BRIEF INTRODUCTION

The compressor is often one of the least understood pieces of equipment, or software in a sound engineer or amateur sound recordist’s arsenal.
In this tutorial I intend to present a brief but broad overview on the use of compressors in the recording and mixing of audio. The principles discussed here will applicable for any sound application, whether it be in music, the spoken voice, or general sound effects and foley. For an audio engineer, the compressor is a great tool to have in your bag of tricks, but it is one of those things that can either make or break your recordings and mixes. Used right, compressors can greatly improve audio tracks, but used wrongly they can ruin them, leaving them sounding squashed, lifeless and muffled. The human ear is able to detect sounds in a wide dynamic range – from quiet whispers to a huge explosions. However, our recording and playback equipment has physical limitations which mean we have to squash or compress these sounds into a smaller dynamic range in order to reproduce them. The purpose of this article is to give you some tips to help you get started in the use of compressors and limiters for controlling the dynamics of your sound.

DESCRIPTION

Here I aim to give you a quick outline on what a compressor or limiter does. Firstly, a Limiter cuts off the dynamics of the signal above a predefined threshold abruptly, while in contrast to this, a compressor gradually attenuates the signal above the predefined threshold.
Often times the same piece of gear can be used as a compressor or a limiter, or it may have two sections, a compressor section and a limiter section and both can be used together. A compressor is generally used to control the dynamics within the normal program level of the material, while a limiter normally has it’s threshold set above the average signal level, in order to catches peaks that rise above those levels. With digital equipment this is often necessary to prevent “overs” or digital clipping if the level exceeds 0 db. There are a number of different types of compressors and limiters available, either as hardware or software, but most have some, or all of the following adjustable features available for user control, though different manufacturers do use different names to label these functions at times.

THRESHOLD

The threshold control enables you to set a point above which the program material begins to have it’s level reduced in amplitude. Any signal which exceeds the threshold point in it’s volume will begin to be attenuated by an amount that is determined by the setting of the ratio control.

RATIO

When a signal exceeds the threshold, the setting of the ratio control will determine how “Hard” the signal above the threshold is attenuated. For example, if a ratio of 1:1 is used, it would mean that no attenuation is occurring at all. However, at a ratio of 2:1 the signal above the threshold will have it’s gain reduced.
At a 2:1 ratio, for every decibel of signal gain that happens, only half a decibel of increased output will actually occur. Two decibels of signal gain would result in one decibel of increased output, while ten dB of gain would result in a 5 dB increase etc.
The higher the ratio, the harder the gain reduction will be. In practice, any ratio above 8:1 is in effect pretty close to being labelled as a limiter, and not much extra signal gain will be output beyond the threshold at settings above 8dB.

ATTACK

The attack control determines how quickly the compressor will react to signals that exceed the threshold. You may think that instant response would be ideal, but in practice a slight delay in the time between the signal’s increase and the onset of compression will often result in a more musical and less muffled sound. For example with a kick drum, where you want to be able to hear the initial attack transient of the beater hitting the drum to give it some punch before the resonance of the drum occurs, and the signal is squashed you don’t want instantaneous compression to stifle this. Generally attack times are set somewhere between about 0.1 and 100 milliseconds. With a limiter however, you will generally want a very quick attack to catch the peaks before they slam into the 0 dBFS of digital recorders causing distortion, and maybe equipment damage.

RELEASE

The release control determines the amount of time it takes for the signal to return to it’s normal gain once the material has dropped below the threshold level again. Release can usually be set from times that are near instantaneous, to times of 5 seconds or more.

OUTPUT OR GAIN MAKEUP CONTROL

When we compress or limit a signal, the result is an overall level output that is lower than the original program material, and we normally use this control to bring the level back up to, or beyond the original level, as long as we don’t take it above the level where clipping occurs. Some compressors automatically apply makeup gain to compressed signals.

SETTING THE CONTROLS

So, having given a brief outline of what the controls do, I will now attempt to give you a few guidelines on how to use them. There are many ways that compressors can be used for interesting and creative effects, but that is beyond the scope of this tutorial.

Compression
Begin with a ratio of between 3:1 and 6:1 for the most natural sounding results, and then gradually lower the threshold until you acheive the desired amount of gain reduction. Remember, that the lower the threshold is set, the more the signal will be compressed. For voices and non percussive instruments try setting the attack time moderately fast, perhaps at around 10 milliseconds, and the release time set between 0.5-1 sec. If the attack time is too fast your, S’s and T’s will begin to disappear, and dynamic distortion will also be more likely to occur at fast attack times, particularly if coupled with a low threshold. Conversely, if the attack is too slow, the S’s and T’s may actually stand out too much. If the release time is too short, the level will appear to fluctuate and pump, as the compression goes in and out, but if set too long, then pumping side affects will become noticeable when loud passages are followed by quieter ones.
Percussive instruments can be treated either by using a faster attack time to catch the initial transients, or alternatively using a slower attack time will let the transients through while catching the main body of the sound. How you use these settings can be quite a big part of the whole creative process, and used for shaping the sound of an instrument to suit the rest of the material. The settings used may depend on the desired effect and the end result you are hoping to attain.

Levelling
When set up for levelling, a compressor is used to keep the overall signal at a more uniform level, while not affecting the short term peaks in the material. In order to acheive this, you should set the threshold relatively low (affecting more of the sound), the attack time relatively slow, and the release time slow as well.

Peak Limiting
Peak limiting can be used for tracking instruments during recording, or when mixing, and is often used in broadcast program feeds. To set up as a peak limiter, use a high ratio (8:1 or more), a high threshold, and a fast attack and fast release time.
Programme Limiting or Average Limiting
Applications for average limiting include the tracking of instruments and vocals during recording. This method is also often used to increase the apparent loudness of broadcast feeds and programs. Such things as “punching up” an on-air presenter’s voice are done this way. Often average limiting is done in conjunction with a peak stop limiter.
If you desire only to limit the average signal level of the audio and not to catch the peaks then set the attack time at 20 ms or slower.

Makeup Gain
Once you have done your compressing, use the output control to make the gain up so that you raise the signal level back up to, or beyond the unprocessed sound. However, you must be careful not to push the signal back up beyond the clip level.

USING A COMPRESSOR AND LIMITER IN TANDEM

A very common thing to do is to place a compressor first in a signal chain to raise the overall level of the programme material, and follow it with a limiter set only to catch the peaks that the compressor misses, and thereby keep the level close to the zero decibel mark. However, another way is to reverse the order, and place the limiter first in the signal chain, followed by the compressor to massage the remaining material. The second method can often avoid holes being punched in the sound by the interaction of these two processors.

AVOIDING SIDE EFFECTS

Just like in the taking of medicine, there can be side affects to compression. In order to avoid unwanted side effects, don’t overdo the compression and limiting. Too much of a good thing can ruin your health ! Sometimes less is more, so be judicious and don’t slam everything to pieces by “over compressing”. Music and other audio can sound really lifeless if you squash them too hard, and can get fatiguing to listen to if there are no dynamics present. If you find you are getting distortion in the audio, then back off on the attack time a little, or set the threshold higher, and the ratio lower. If your audio starts sounding muffled, then slow the attack time down, and lower the ratio to let some of the transients pass through before the onset of “squashing” occurs.

CONCLUSION
So there you have it. Aside from being a little confusing to use if you don’t know what the controls do, once you understand them and have a play round with compressors, they can be fun tools and can really make or break your final mixes. Start conservatively and don’t overdo it. Repairing damage from poorly compressed audio is very difficult. Better to be safe than sorry until you have gained experience with different compressors, in different scenarios. But don’t be put off using them. When used well they can really make a great difference in the world of audio. So go to it and have fun doing some squashing!

By Tony Koretz
© copyright March 2012

Recording And Gain Staging

The old adage of recording signals as hot as possible really only applies to the analog domain, not the digital one. With analog reording this was because of the noise floor inherent in analog circuitry and tape, where you were fighting low level noise and hiss. This was also how we were taught to record when digital recording was relatively new, when the best we had was 16-bit audio. Back them everyone was trying to record as close as possible to 0 dB without clipping. However, now with 24-bit converters there is no need to record signals nearly so hot. You can get great recordings with the levels set much lower, and then you will also find it MUCH EASIER to mix later, because you are not trying to force so many loud signals into the headroom limitations of a DAW. If you have a number of tracks all recorded very loud, and approaching 0 dB, at mixdown you will find yourself running out of headroom, and will have to bring all your track levels down in order to compensate.

The professional +4 standard reference point of 0 dB in the analog domain ( 1.23 volts) is actually at -20 from 0 dBFS (top of the Digital scale) in your DAW meters. So with the reference level of an analog console being +4 (unity Gain or “O”) and equalling -20 in the digital world of your DAW, if you slam everything up too hot and close to zero in your software mixer, it will be like trying to jam all your signals on a console at maximum level into the master section. Not such a good idea. If we run a signal into the line input of a high-end console at a level where the channel meter reads 0 vu,( +4 or 1.23 volts) it will be at unity gain. A great console might be able to handle up to about +20 or even +24dB, which means we have about 16-20 dB of headroom above the 0 VU point before the signal turns to custard. Keeping our recording levels lower makes it much easier to mix things later. By the same token then, if we have -20 dBFS as our “0″ reference in our DAW, we then have 20 dB of headroom before we clip the signal. Clippng a digital signal sounds far worse than clipping an analog one, so steering well clear of that should be our aim. Digital clipping does not sound good. If you look at a clipped waveform up close, you will see it is square at the point of clipping. There isn’t any logical reason to record with levels even close to approaching the level where they clip. It certainly won’t improve the sound of a recording, and you run the risk of having little headroom left for mixing.

The dynamic range of 24-bit recording is theoretically about 144dB. With 16-bit it is only 96dB. With the dynamic range being 144 dB, and if the noise floor is about 120 dB, then the effective dynamic range or signal to noise ratio is 120 dB. That is plenty of room for recording with levels well below 0 dB before you introduce unwanted noise into the signal.
So, during the recording process trying to aim for an average of between -15 to -10 dB on the meters of the DAW tracks, with peaks not going above -6 dB is a good level. However, it is perfectly fine to record at -20 or even lower.

Mixing And Gain Staging

Effective use of gain staging when recording sets you up for a good mix. Keeping your gain staging at sensible levels during mixing is the secret to obtaining clean, clear, transparent mixes with plenty of headroom. Proper gain staging will not reduce punch in a mix. If each track is recorded at a level in your DAW which is not pushing them too hard, and you send various instruments to Busses or sub groups, you will find that you don’t clip the inputs of any plugin compressors or EQ’s that you place on the busses, and you have every opportunity to boost the outputs of those pugins or busses to make up for any perceived lack of punch you may have because your levels are lower. You can then have some cool punchy buss sub mixes, that can then be summed to your main stereo buss, without digitally distorting anything anywhere in the chain.
If you find that signals in your DAW have been recorded too hot for clean mixing with plenty of headroom, the use of trim plugins is the first place to go. Using one of these as your first plugin on each channel, you can adjust the level going into EQ’s, compressors and other processors. This is not the same as just pulling down the fader, because that only affects the signal level going to the master buss or summing busses. The trim plugin acts like the gain control on your preamp, whereas the fader is like the output volume control. Another advantage of this is that if you insert a hardware compressor or EQ into a channel it will be receiving a signal around the 1.23V (+4) that it was designed to operate at.

Definition: Headroom

Headroom is the term used to describe the difference between the maximum level that can be achieved without clipping a signal, and the average level of the signal.

Definition: Gain Staging

The term “Gain Staging” refers to the level of a signal beginning at its source, and ending at its final destination. Along the signal path, you may have several points where changes to the signal level can be made. Proper gain staging refers to setting sensible gain levels at these various possible adjustment points. This gives us our gain structure. An example of gain structure would be a vocalist who can sing louder or quieter, going into a microphone, then into a preamp, followed by a compressor, and then into the A/D converter to record. All points in that signal chain are places where the gain could be changed.

Reference Line Level
-20 dBFS = 0 dBVU = 1.23V

By Tony Koretz
© copyright March 2012

In this tutorial I am going to attempt to give a brief outline of some of the problems we can strike when doing audio post production or sound mixing work. If you are the one responsible for the project from the outset, including both the recording aspect, and the mixing and post production work, it makes sense to try to address the issues of unwanted noises before committing to tape (ok well more likely hard drive, but I’m sure you know what I mean). If you have received someone else’s recordings, and are responsible for the final mix/master, then it is already likely too late for that, but here are a few pointers that will hopefully help you with fixing the issues already in existence, or helping to avoid future ones that you have control over.

DC Offset Problems
One problem with using cheap soundcards or recorder inputs is they often add a DC voltage offset to the audio when recording. This sound sits at 0 Hz, which we can’t actually hear, but it offsets the wave so that instead of it being centered, it sits above the zero line. This means you that you will have less audio headroom before clipping occurs. Have a look at your tracks, and if tthe wave is not centered around the zero, then you need to apply a DC offset filter to it. If at the time you were recording, the meters in your DAW were reading even when you were not sending them any signal, that could likely be a sign of a DC offset problem. In most recording or editing software you will find that they have a functionlabelled something like “remove DC offset”. Use it if the problem is evident, but preferably do it prior to any compression or EQing. It should get rid of any unwanted offset.

Unwanted Hums and Buzzes
Hums and buzzes in your audio, especially from things like guitar amps can be a problem when recording. Try to elimante the sources of such unwanted sounds before undertaking the recording process. Try to avoid earth loops. Keep all your audio sources on one power circuit (or phase), and any lighting or other appliances on a different circuit. Make sure all your equipment is properly grounded. Check your cables, making sure all wires and shielding are firmly attached at the terminals. It’s amazing how often a loose wire can cause intermittent noises. If the problem is a guitar and amp comination, experiment with different proximities and angles of the guitar in relation to the amp. Facing in one direction you may experience hum, but if you turn your body and the guitar in another direction the hum may disappear. Single coil pickups are more prone to this problem than hum-bucking pickups ( hence the name). Use balanced cables where possible, and avoid long runs of unbalanced cables. Read the tutorial Audio Signals: Basic Guide to levels, Signal Types and Usesand the paragraph titled “Balanced or Unbalanced” for more information on this topic. Also try to avoid having power cables and transformers running near audio cables. Keep them at least a foot (30 cm) apart, if possible, and if they must cross, try to have them do it at right anges, or better still sit the power cable on something above the audio cables so they don’t touch. Hums can be dealt with by some programs and plugins such as The Waves X-Hum, but it is more difficult to remove buzzes from a recording without altering the tone of the recording. The center frequency of the hum is likely to be either around 50 Hz, or 60 Hz, depending on the voltage of the power supplied, and this varies from country to country. Sometimes you can use the “Learn” function of a noise reduction platform to select and remove buzzes and such like from good audio. If you have an area of audio that is essentially silent apart from the offending noise, you can use this section of the audio for the software to “learn the noise,” and then remove it from the rest of the track..

Hiss and white Noise
Hiss is often a result of incorrect gain settings when recording. Perhaps you had a quiet source, or a small a signal, and the gain on the preamp was set high to compensate for this. You may then pick up noise from the preamp, or amplify the self noise of a microphone. Perhaps you just have a noisy piece of equipment. Whatever the cause, use your denoiser, dehisser or noise removal plugin, software or hardware function before you compress, EQ or otherwise tamper with the audio. If you compress before you use noise removal, you will likely be boosting the noise along with the signal, but if you remove the noise first, the compressor then won’t find any white noise to boost! Higher and more aggressive settings on these programs often seriously mess with the signal, adding unwanted artifacts, and change the tone considerably. Try to avoid using overly aggressive settings, if at all possible, try mild settings first. Sometimes making a couple of passes with mild settings does a better job than one pass with a more extreme setting.

Removal of Clicks and crackles
Unwanted clicks and crackles can be a real bane of digital recordings at times. Some suggestions I can offer for minimizing them are as follows:
Defragment your hard drives regularly to ensure the computer runs fast enough to keep up with the high demands of recording, mixing and running video and audio together.
Adjust your DAW program buffers to avoid clicks
Use the latest drivers for your interfaces and other attached hardware
Use crossfades at edit points
Avoid clipping any equipment’s inputs while recording or any outputs while mixing
If you are unfortunate enough to discover clicks in your audio, use your DAW’s or plugin declicker, and try to get as close to the area of the waveform as possible by cutting the region either side of the problem area, and selecting just that small area of the region for processing. Sometimes you can get better results by manually removing clicks using methods such as re-drawing a waveform, with the pencil tool available in the software.

Summary
Hopefully you have found some useful answers to unwanted noise problems in this article. I would suggest if you are intending to do your own audio editing, post production, mixing or mastering, that you try to obtain a good software platform and suitable plugins for this task. Learning to use them well is something that does take a bit of experience, and a bit of ear training, but having good tools to work with means half the battle is already won.

Written by Tony Koretz
© copyright February 2012

There are good royalty-free music deals and bad royalty-free music deals.
I have decided it is about time to write an article addressing a worrying trend I have been observing: namely music libraries that are not allowing composers they list to be part of Performing Rights Organisations, or the music they list to be registered with any royalty collection agencies.

“Bad Deal” Royalty-Free Music Websites

There are some websites out there in the cyber world that are saying they are the only ones that are truly “royalty-free”..ie:no royalties, no cue sheets, no broadcast royalties for composers, and then selling their songs sometimes for as little as 99c. As if that is supposed to make it cheaper for people buying the license to use a song.
The truth is, that the buyer of a track license NEVER has to pay royalties for broadcasts anyway, as any broadcast royalties collected come from the broadcast royalty pool. All it actually saves them is the 5 minute job filing in of a cue sheet. It doesn’t save them any money. I am all for the promotion and use of Royalty-free music, but at Rocksuresoundz.com we would never even consider taking away the right of a composer to get their legitimate broadcast royalty compensations. My advice is to stay away from any arrangments that require a composer to forego those rights. Artists are often the first to be screwed by corporate greed and/ or cost-cutting. The website music libraries in question here will not accept works that are registed with any royalty collection agencies, PRO, PRS (Performing Roghts Organisations and Societies).What is happening here is they are selling a composers music for a small sum, and refusing to allow the composer to collect any broadcast royalties, even if their music gets thousands of plays in a big budget production. I am first and foremost a composer and musician myself., and believe as such, that we should be fairly and rightly compensated for our work, and this type of deal is not good for us at all.

Here is an example to ponder:

Bad Royalty-free website:
A composer lists a song on one of the Suspect, bad deal websites. The site sells the song for 99c or $9-95 or even maybe even $19-95. A client comes along and uses the song in a documentary. The documentary gets played on television channels world wide. The maker of the documentary gets thousands of dollars off the production in sales and royalties. The composer gets his/her share of the original sale price, ( a few cents or dollars) minus the website fee, and cannot get any royalties from the broadcast plays at all. Is that a good deal? I think not! The only ones really benefitting from such a deal are the greedy management at the website that sold the music for a song ( pardon the pun) and are laughing all their way to the bank!

Fair Compensation Royalty-Free website
The composer lists the song on rocksuresoundz.com or another site that compensates composers fairly. We sell the song for $39-95. A client comes along and uses this song in their documentary. The documentary gets played on television channels world-wide. The maker of the documentary makes thousands of dollars off the production in sales and royalties. The composer gets his/her share of the original sale price, and also has the opportunity to collect broadcast royalties from the documentary every time it is aired in public broadcast. Now that’s a right and fair compensation for the composer, and it doesn’t cost the production company any extra money!

Broadcast Royalties
Broadcast royalties are a standard part of every professional composers income. Broadcast License fees are charged to broadcasters because they generate money by having a composer’s music used in broadcast, and so it’s only fair that that a composer receives something in return for that: Namely a broadcast royalty, as has traditionally been the case.
Suggesting that composers just be happy, and get used to not being compensated for broadcast usage of their music is unfair, and should not be a part of the industry in any way in my opinion.

“Bad Broadcaster Deals”: No Royalties For music Usage!

I have recently seen some advertising on a behalf of a broadcaster who wants to cut costs by using music by composers that are not members of any Performing Rights Sociey. They want to use only music that is not subject to any form of broadcast royalty collection. Their intention is to create a library of music that is devoid of any possible broadcast royalties, and are refusing to include music in the library by any composer who is a member of a PRO ( Performing Rights Organisation). This is a new and alarming development in the broadcast industry, and if it were to become standard practice it could have huge negative repercussions for composers’ and their income streams.
The broadcaster in question here appears to be trying to cut down on the fee they pay for the right to broadcast music. I can only speculate, but assume that this would be by declaring less usage to the PRO’s that they have traditionally paid their license fees to, and thereby lessen the fee they have to pay to the PRO’s.
If they don’t pay license fees to any royalty management/collection agencies, then they can’t broadcast anything that includes music created by members of royalty collection agencies at all. They either pay the fees to the Royalty collection societies or they don’t. All major films and TV shows will have been made using works by society members. If they play just “One” of those shows/films if they are not registered as a Broadcaster with the PRS companies then they are breaching copyright laws. So what is the deal that they are managing to cut that still allows them to play those shows but not pay royalties for music usage from their libraries?
This trend worries me both as a composer and as someone who supplies royalty-free music. . If the number of websites etc selling dirt cheap music while not allowing contributors to belong to any Royalty collection societies, and the number of broadcasters using music by non society members grows bigger, what will become of the industry?. If this model trend was to continue to grow, and corner the market, composers would slowly lose the rights that were fought for in the formation of those agencies in the first place, and unscrupulous broadcasters could basically pay them a pittance, declare only what they decide to declare in the way of music usage, payout no royalties, and no-one is there to police that. I myself see this as another screwing down of artists who generally struggle anyway.

Another possible scenario that could occur is this:
The broadcaster who is using non PRS music decides to make a particular TV show for broadcast within their country and on their network. The show goes on to become a massive success spawning the desire of Other broadcasters in other countries decide they want to broadcast the show in their territory also. So they contact the orignal broadcaster/show creator, and purchase the rights to broadcast the show on their own network in a specified territory. Again it succeeds in that territory also, so then other broadcasters around the world also buy ther rights to use the show. Then the show goes to air worldwide, and it becomes a global hit, broadcast on a number of different networks.
Question: Who then will be responsible to pay the per-play money owing to the composer of the music used in the show? The new broadcasters won’t have made the deal with the composers of the non PRS music, so won’t be liable to pay them for it, and the orignal broadcaster won’t want to pay money for music that is not being played on their own network.

Oh But it’s Only Composers That Suffer. Who Cares? It Helps Us Make Things Cheaper!
So, video makers and film makers are you going to also go for the cheapest option in music so you can make your productions on the cheap, and show no regard for the consequences to the industry long-term, or will you think beyond your own needs and look to the future of all artists? Because afterall, who and what would be next in line for culling then? What about broadcasters telling studios, producers and film makers their works won’t be used if they belong to any film and television royalty collection agencies, societies and collectives like the Motion Picture Licensing Corporation (MPLC), Association of International Collective Management of Audiovisual Works (AGICOA) or the SMPTE. What about refusing to use works or performances by actors who are members of SAG (the Screen Actors Guild), or the AAAA ( Associated Actors and Artistes of America)? If YOU USE MUSIC from companies that don’t allow the filling of cue sheets and the reporting of broadcast royalties, you are helping to kill the income of composers, and encouraging the proliferation of sub-standard non-professional music. Remember, your affiliations could be next.
Shall we shoot your foot first or mine?
To re-iterate: My advice is to stay away from any arrangments that require a composer to forego the rights to collect broadcast royalties. Don’t be a part of de-regulating and killing our industry.

In order to better understand broadcast royalties, read the following tutorial as it explains the concept further:
http://rocksuresoundz.com/2011/10/11/royalty-free-music-what-is-it/
Tony Koretz © copyright 2012

Thanks to all our new clients who took to using our products during the year… our first year of operating the rocksuresoundz.com website. To kick off the new year we have added a number of new products to the Production Music, Free loops and Sound Effects categories for January. Also a new tutorial titled “Audio Signals: Basic Guide to levels, Signal Types and Uses” is up in the tutorials section. More new stuff to come in February too, so keep watching this space!