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!

Signal Levels and voltages

a) Microphones and Mic Level

The voltage level generated by a microphone is typically about 2 millivolts. Levels can vary between 2mV (0.002 volt), to as much as 1V or more depending on how loud the source is, and how sensitive the mic is. A loud guitar amplifier or drum kit can generate a signal of several volts.

A millivolt is 1/1000 volt, and 1 volt = 1000 millivolts

A preamplifier is required to bring a mic level signal up to usable Line Level. It is important that a preamp has a large gain range to accommodate the widely variable signal levels that can be fed into it from a microphone. Many preamps have a pad switch or gain control pot to prevent overloading as can happen with the large voltages generated when recording amplifiers, drum kits etc.
Another way of stating the measurement of mic level signal sometimes found is the dBu. Mic levels can vary from about -52 dBu to +2.2 dBu

b) Line Level

Line level has a much higher voltage than mic level. This stronger signal enables it to be carried over longer distances than mic level. It is a medium level output ( less than speaker level), and is the common output level for compressors, Equalizers, reverbs, delays and other signal processors, as well as microphone preamplifiers.

The two normal line level standards used are:

Consumer or semi-pro (-10 dBV) .316 volts, this is the level commonly found in such equipment as CD players.

Professional (+4 dBu) 1.23 volts, found in mixers, signal processors etc

Two reference voltages are common. These being decibel volts (dBV) for consumer applications, and decibels unloaded (dBu) for professional applications.

c) Instrument Level

Instrument level is generally somewhere between mic level and line level, though it can vary greatly. Typical values are between 0.1V and 1Volt for passive guitar pickups, and up to 1.75V for active pickups. Expressed in dBu these are from -17.7 dBu and +7 dBu. Therefor there is no absolute standard for instrument level, as it can range from a few millivolts where passive or piezo pickups are found, to several volts on instruments with active pickups and on-board preamps.

d) Speaker Level

Speaker level is the strongest signal. It can vary between 3 volts to 90 volts or more, depending on the power output from an amplifier. To produce a listening level of 85 dB in a theatre would require a speaker output level of about 4 volts.

Impedance

An often overlooked factor that greatly affects the sound and performance of devices is impedance compatibility and impedance matching. Impedance, measured in ohms, is the resistance of an electrical current in a circuit such as an audio signal circuit.

The input impedance of the device you are plugging into should be at least 7 to 10 times higher than the output impedance of the source of the signal (electric guitar, keyboards, microphone, etc) in order to obtain the optimum sound quality and the highest voltage transfer.

To illustrate this, if a mic’s impedance is 200 ohms, the input impedance of the device that you plug the mic cable into should be 1400 to 2000 ohms. Typically the input impedance of a mic preamp as found in a mixer is around 1500 ohms to 2400 ohms. The impedance of an electric-guitar pickup is High-Z at between at 20K to 40K ohms (20,000 to 40,000) ohms, so to match input impedances, the guitar amp, DI box input, or instrument input that you plug into should be at least 280K to 400K ohms. Examples here, are a Peavey 5150 guitar amp Hi-Z input is 470k ohms, a Mackie Onyx instrument input or Countryman DI input is 1.0M ohm. ( 1 million ohms).

Microphones with a rating of 150 ohms to 300 ohms are classed as low impedance. Above 600 ohms and up to 2000 ohms is considered medium impedance, while above 10,000 ohms is high impedance. Most microphones are low impedance( or low Z), and all mics that have XLR (3-pin) connectors are low Z. Low impedance microphones can be used with long cable runs of up to hundreds of feet without picking up hum or losing any of the high frequency content of a signal. A high-Z mic will lose treble frequencies and pick up hum if the cable is longer than 3 metres or 10 feet or so .

Balanced or Unbalanced?

Unbalanced cables use a single centre conductor, and a shield. Audio signals travel along the centre wire and the shield of the cable. This means that any interference that the shield picks up, will find it’s way into the audio signal. This noise picked up on unbalanced cables can range from hums to radio interference.

Balanced cables consist of two conductors that are twisted together and are covered by a shield wire that is usually braided. Each of the conductor wires are connected to an impedance at either end. The impedances of both conductors are the same. The audio passed on the two conductors is identical, except that on one of the conductors the phase of the signal is inverted at the source. The input of the equipment connected at the destination puts the inverted signal back into phase, and sums the two channels together. The shield of the cable does not carry the audio signal and should be grounded or earthed to the chassis of each audio device.

Unbalanced cable runs should be kept to 3 metres (10 feet) maximum length if possible, whereas balanced cables can be used for short runs or runs of several hundred feet. Guitar and instrument cables are usually unbalanced, as are the outputs of guitars etc. Professional Microphones usually have balanced Low impedance outputs, and professional signal processors commonly have balanced inputs and outputs. Where you have a choice between balanced and unbalanced generally choose the balanced option.

Getting Good Audio into Video Cameras

Many non-professional video cameras have a single 3.5mm mini-plug mic input. If you need to plug two microphones into a camera that only has a single mini-plug mic input, using a simple Y-cable will not work properly. The mic input on the camera is only designed to handle the impedance of one microphone, so attempting to patch two microphones into this input with a Y-cable will mis-match the impedance and result in really lousy audio quality. In this case the use of a camcorder XLR adapter which doubles as a mini-mixer and impedance matcher, or a field mixer is the answer. These use transformers to convert the audio from two or more mics into something usable by the camera’s mic input. The best camcorder adapters have the ability to supply phantom power to condenser mics and also have a mic/line switch that enables you to switch between mic and line level signal inputs.

Professional video cameras usually have 2 XLR mic inputs and deliver phantom power to condenser microphones. Many cameras still only record at 16 bit however, so use of a high quality audio recording device and good mic preamps recording at higher bit rates and separate from the camera is the most desirable option where possible.

Phantom Power: What is it and when do we need it?

Condenser microphones require the use of phantom power in order to operate. Ribbon microphones and dynamic microphones do not require phantom power. Tube condenser microphones normally have their own separate power supply and therefore also do not require phantom power to be supplied from a preamp, mixer or battery.

Phantom power is a DC voltage, usually +48V, but sometimes less, that is supplied down pin two and pin three of an XLR cable to power condenser microphones, active DI boxes and effects pedals.

Some Points to Note When Deciding Which Input to Use

The instrument inputs on audio devices are high impedance (Hi-Z), designed to match the impedance of a guitar/bass amplifier. An important point to note is that, signals fed to the instrument inputs go through the preamps where their low-level input signal is amplified to bring it up to line level. These inputs are usually unbalanced. Any coloration inherent in the preamp will therefore affect the sound of the signal.

The line inputs on audio devices are lower impedance than the instrument inputs, and are not usually amplified by the preamps, although in some circuits they do pass through the preamps without actually being amplified. Therefore, one would expect less coloration from the preamps than would occur when using instrument inputs. Line inputs are usually balanced inputs.

Use of DI boxes

Direct boxes are typically used where instruments or other devices that have only unbalanced 1/4″ outputs need to be connected to the XLR inputs of another device. The Direct Injection box or DI as they are commonly called, takes the high impedance and unbalanced signal from the instrument and converts it to a low impedance, balanced signal that can then be connected to the XLR input of the ensuing device.

The signal can be sent over long runs via microphone cable, with little or no signal loss due to it’s low impedance, and with greater rejection of interference than would be found using unbalanced cable. The low impedance of the signal sent from the DI (which is usually around 600 ohms), allows the mixing console or preamp to accept the signal at it’s XLR inputs which have been designed to cater for microphone and their low impedance outputs.

DI boxes are available in either passive or active models.

Passive models use a transformer to convert the high impedance signal from an instrument to the low impedance expected by a mic input at a mixer. They do not impart gain, and do not require powering. Because they are simple, the good ones are generally reliable and quiet.

Active models contain a preamp and impart gain to the signal. They require powering from a battery, phantom power, or sometimes even an AC outlet. Instead of a transformer, an active DI box usually uses a FET (Field Effect Transistor) which has a very high input impedance. Active DI boxes usually incorporate switches to pad or control gain, ground lift, external power or battery source selection, and sometimes a mono or stereo mode selection. They are therefore more versatile than passive units.

In Summary

Hopefully this tutorial article will be of some benefit to those of you baffled by audio signal types and their associated jargon. There is much more than could be written about this topic, but I hope this gives you a useful overview on the subject.

Written by Tony Koretz
© copyright January 2012

How many times have you sat down to watch a video or budget movie, only to find that the camera work and picture editing is great, but that the sound is so bad it takes your attention away from the picture? You might even stop watching it altogether, or at least turn the sound way down. Watching a TV show, video or film should be a “complete experience,” where the picture and audio combine to produce a meaningful and well balanced whole.

However, it has often been the case that in television, cinema, and especially in amateur or semi-professional productions, that the sound has been seen as secondary to the picture. I don’t believe it should be that way. People will often put up with mediocre picture if the sound is good, but good picture with poor sound may see the ‘off switch” flicked by many viewers. So let’s set about giving  a bit of an outline as to how you can go about acheiving good results with your audio.

Use A Separate Recording Device

‘What is wrong with recording sound using your cameras built in mics’ you ask? Well generally the quality of the built in mics and preamps in a camera is pretty average. Cameras also can have tapes or hard disks running, and other electronics that make noise which can often be picked up by the microphones. Another drawback is that to get good sound recordings, you may need to be closer to the source than the camera will allow while maintaining the right picture perspective. The best solution is to use a separate audio recording device, that is away from the camera.  Many portable recording devices have built in microphones that are far superior to camera or camcorder mics. You could use the built in mics of the recording device and get better results than your camera  mics, but a still better way  is to use a separate microphone (or multiple mics) plugged into a portable recorder to capture sound. At a pinch you could mount a video mic to the camera, and record the audio that way. It should give you better results than the built-in camera mics, but still not really up to the standard of dedicated audio recorders and microphones. Cameras often use mini-jacks for plugging in microphones. Some smaller portable sound recorders also use this type of plug. The connections of these can be a bit unreliable, sometimes working lose, pulling out or crackling with movements. The best devices generally have XLR inputs, which lock the cable in place, and provide a much more reliable connection. Larger professional cameras normally use XLR inputs. Microphone signals are of a very low voltage, and require preamping to bring them up to a useable level. Sometimes as much as 60 or 70dB of gain is required to do this. It is important to have quality preamps in order to do this well. Poor preamps can add noise, distort signals, and generally do nasty things to the sound! Many cameras have very poor onboard preamps. A good portable recorder will have far better preamps than those found in a camera.

Another thing to consider is that audio metering in cameras is often poor, and also it’s hard to monitor these at the same time as concentrating on filming. Automatic Gain Control is also something to be avoided if at all possible, as it can introduce compression that may not be of the best quality.

A really high-end recorder like those made by Nagra or Sound Devices will have preamps that rival the best studio preamps in quality, but even quality lesser devices than these will  generally better preamps than a camera. Try to use a separate recording device rather than the camera if at all possible.

If you have a couple of mics you can record them onto separate tracks in the recording device and mix these to suit later. Sometimes a field mixer (a separate device from the portable recorder) can be handy in the field where more than one source needs to be blended or mixed on the spot. However, recording everything to separate tracks if possible is going to give you maximum flexibility at mixdown later. Traditionally a field recorder and and field mixer were distinct devices,and the sound recordist carried both, but these days it is common to see devices that include both the recording and mixing functions in a single unit.

A Two Man Job

The answer to acheiving the best levels and quality of audio lies in having a second person, who is away from the camera, to be responsible for the audio recording. Their job is to monitor the sound, and not worry about having to concentrate on the filming as well. If you are really lucky you might even have a third person to man a boom pole, leaving the second person to just monitor the recording levels. The boom operator concentrates on microphone postioning. Unless it is an interview, news report, or a documentary style production, you don’t really want to see someone holding a mic or wearing a lavalier mic. To get clear speech you will want the  boom operator to be able to get the microphone just out of shot above the voice. Operating a boom pole is an art in itself. The difference in level between whispering and shouting is huge, so someone needs to have eagle eyes watching the metering at all times for best results. Especially since we are not using AGC (automatic gain control). That’s the job of the person reponsible for the audio recording. Some recording devices have good limiters to catch rogue peaks before they clip, but some other devices have lousy limiters where you can hear the sound truncating abruptly. Manual gain control and/or light compression usually gives more natural results. However, compression is best left to someone who is experienced and knows what they are doing. Novices should probably not try to use compressors while recording, as the results of poor compression can’t be undone. Set levels manually, and if you have a limiter of a known high quality, you could use that to catch any extreme transients. If a limiter is being triggered constantly the results will not be pleasant. Backing off the gain will produce more natural and pleasing audio than sound that has been heavily limited or compressed.

What Type of Microphone should I Use?

In most cases for use in location recording a directional shotgun mic mounted to a pole is the best option. There are a number of models of shotgun microphones available, ranging from relatively cheap, to very expensive at the high end. The Sennheiser MKH416 is the traditional standard mic here. Rode Australia makes some good model shotgun mics for those with a lower budget. The mic can be plugged directly into your portable sound recorder if it has in-built mic preamps. Shotgun mics are very directional and have good rejection of sound to the sides. You will need some kind of shockmounting to prevent handling noise, and if outdoors some sort of wind protection as well. Low rumbles and wind noise can really marr a recording. A blimp or zeppelin style shockmounted microphone enclosure, covered with a fluffy fur type windjammer is ideal for outdoor shoots. A foam windsock alone over a microphone capsule is not sufficient to block out wind noise and rumble. For recording general of atmos and ambience, as well as shotgun mics, fixed polar pattern (such as cardiod) condensor mics can  be useful. If you want  a stereo atmos sound there are various ways to position mics for this, but I won’t go into those here. An example would be an X-Y pattern. I find small diaphram condensor mics handy for this. Special stereo mics are also an option. If your recorder has onboard stereo mics of sufficient quality you may even want to use those.

Sync’ing Video and Audio

Recording of picture and video separately requires some way of sync’ing otherwise it will result in a shambles. If you have multiple cameras and an audio recorder, you will need to be able to keep these perfectly sync’ed so that you can align them during editing. The simplest way to do this is to use a clapper board, or someone using a single handclap in view of all the cameras, and with the audio rolling in record. It is important that everyone can see the clapperboard or hand clap, and that there is only a single clap at the start of each recorded clip, to minimize the risk of confusion as to the start of the cut. It will then be easy to align all the audio tracks and film in the edit suite. A clear procedure of ‘calls’ that everyone on the shoot is familiar with is essential to maintining an orderly shoot. Traditionally in film an asistant would call “turnover,” at which point the sound recordist would call back “speed” once he had hit the record button. The camera operator starts his camera rolling and calls “turnover,” and the sound recordist replies back “speed,” before the cameraman calls “mark” for the clapperboard to be clapped. Then the dirctor calls “action”. In our case, so that everyone knows all devices are in record, we should follow a similar procedure. I prefer to use terms words like “roll sound” or “roll film”, and having worked in television, I am much more used to the term “cue” rather than “action”. Whatever words you use, make sure that everyone involved is clear as to their meaning. It’s also a good idea to take notes on each scene that could make things easier later when editing. This is Known as “slating.”

Timecode

The recording of picture and sound and their syncing on a location shoot using type above method should work well in most situations, though it is possible for small drifts in timing that may require correction due to the different internal time clocks of all the devices. If you are using analog tape or film it is most likely to be noticeable. Our clapperboard only gives us the beginning point of our film, and so that is our main reference, but nothing solid as timing reference from there on in. In television or larger film productions, a method of time aligning using time code is the norm, where everything is sync’ed to one clock, keeping things tightly aligned frame to frame, throughout the record process. SMPTE/EBU timecode uses a string of eight numbers that make everything easily identifiable on a timeline. The numbers are hours, minutes, seconds and frames. For example 03:24:37:16 read on a timecode would mean three hours, twenty-four minutes, thirty-seven seconds and sixteen frames. The hours are in 24 hour format, the minutes and seconds as normal from 00 to 59, and the frames from 00 to 29. It is not within the scope of this article to go into depth on understanding timecode, as it can be quite complex to understand and deserves it’s own separate explanation.

In the Editing Suite

Once we have our video and audio recorded and we get them back to our editing software,  we need to align the “clap” from each camera and audio track to give the start point of the clip. Cut all tracks at the point of the initial transient of the handclap and work from there. Go to the end of the take and check to make sure everything is still aligned there too. Hopefully it will be. If a single timecode clock has been used it almost certainly will be.  However, if we have not used timecode, and just the “clap” method, if it has been a long scene, it is possible that timing errors may have crept in. If you find further along that sync has slipped, try to find a clearly identified event further into the scene, and cut and re-align here. Plosives sounds such as “B’s” and “P’s” can be good because they are clearly visible in the audio, so you could use them as your reference, so long as the person speaking is also clearly visible in your picture. Sounds such as “p” and “b” should occupy only one frame, which should make re-alignment relatively easy. Otherwise you will need to find another clearly evident transient in the sound that is also clearly visible in the picture, and use that for aligning.

Summing Up

As you can see here I am passionate about sound and I get irritated by poor sound in video, film or television. Sound quality in youtube is often even worse. Let’s all do our part to improve things. I hope in this article that I have given you a few basic ideas to get you thinking about improving the sound of your next production. Your aim should be to give the viewer a good listening and viewing experience, or one that doesn’t distract them from the topic if it’s a documentary or similar. So don’t cut corners and take short cuts with your audio if at all possible.

By Tony Koretz

© copyright 2011

People often see the words “royalty free” bandied round, and they notice the number sites offering royalty-free music, but  are confused by what this actually means. My aim here is to give a brief , but hopefully easy to understand explanation as to what the term royalty-free acutually means, and how it’s use can be applied.

“Royalty free” as it applies to music, is the right to use music without having to pay a royalty fee. The use of certain intellectual properties require users to pay a royalty fee to the owner, creator or copyright holder of that property.

Royalty-free does not mean the product is free or public domain, it just means you don’t have to pay royalties.

Royalty free or buyout music, and it’s use is governed by the licensing agreement that you should receive and agree to when you purchase the right to use some specific music. The type of license will determine how  the music may be used, whether you can use it in a commercial production or only in a personal or non-commercial production, and what the limitations of use are. License agreements can vary considerably. It always pays to read the fine print.When purchasing a piece of music, you buy the rights to synchronize the song with your productions. The music and it’s copyright will however, always belong to the copyright holder of the product. Generally you pay a one-off fee to purchase the use of the music, and  no more additional fees are required. As long as you abide by the conditions of the license agreement you should have no trouble.

So what exactly are royalties?

Royalties are fees that are paid to the rights owners (such as record labels, publishers, composers and artists) for the use of their work. The money paid to composers by publishers for the right to use their songs, is usually determined by an agreed percentage of the price per copy sold. Often only a small percentage of the continuing sales of the music goes to the composer or artist. Payments received for use of or the right to use intellectual rights are called royalties.

OK, what about cue sheets and broadcast royalties?

One thing that can be a confusing issue is royalties for the broadcast of shows on television that use music ( that may or may not be licensed as “royalty-free” music). Television stations pay royalties to the Performing Rights Societies  for the right to broadcast music in their shows. Music usage in Television and radio show broadcasts are tracked by documents known as  cue sheets. It is important for cue sheets to be submitted so that the composer of the songs receives the royalties from the broadcasting of their music. The broadcasters still pay their royalties to the performance rights societies regardless of whether or not a Cue Sheet is submitted. If no cue sheet has been submitted, the royalty money paid does not go to the person who wrote the music. Instead, it goes into a general fund, which is then distributed to whomever is topping the charts at the time. The composer of the music is entitled to the broadcast royalties accumulated by the use of their music in the broadcast, but this cannot be tracked unless cue sheets have been submitted. This is why most sellers of royalty-free music still require cue sheets be submitted when the music is to be used in a broadcast show.

So what is a Cue Sheet then?

A cue sheet is a document that lists all the music used in a project, allowing Performance Rights organizations like APRA, BMI, ASCAP, and SESAC (and others worldwide) to properly compensate composers for performances of their work. Cue sheets detail the length of the music used, the track’s title, the composer and publisher of the music, and where it was used in the film, documentary or television show. It also has some general information about the project. Broadcasters such as TV stations, Radio stations, etc are responsible for paying performance fees to the performing rights societies (PRS). Then the PRS then pay the composers. Broadcasters are required to pay a regular fee to the PRS. As a purchaser of royalty-free music you only need to report what music was used, via the cue sheet system. The composer is then able get their rightful portion of that fee. As the music license purchaser (you) are not required to make any payments to PRS. Webcasts and podcasts do not require the submission of cue sheets.

By Tony Koretz

© copyright 2011

In order to create post production mixes that translate well to consumer systems, calibration of  your system’s  playback  level is important. It is also important to do this if you are creating music that will be passed on to other editors for use in film, TV or video. I intend to give a brief guideline here for calibrating your listening environment in a small to medium sized room, such as an edit suite. The tuning of a room is beyond the scope of this article.

For film work (usually done in larger rooms), a pink noise at reference level should be used to produce a sound pressure level of 85dBC for each of the front channels. Taking readings  for left, right and center speakers. For surround speakers, a lower reading of 82dBC is used. Generally -20dBFS is used as the “0″ VU reference level point.

For television and video work, which is often done in smaller rooms, a lower reference level is used. This is most often done at 79dBC for each of the speakers in a smaller edit suite room. In mid sized rooms such as pre-mix rooms this level is often set a bit higher, for example 82dBC.

The procedure for calibrating the playback level is as follows:

You will need a sound pressure level (SPL) meter to take readings from the mix position in the room. Set the meter to a  “C” weighting curve, and a slow response time. You should hold the meter in the same position at the place where you normally sit when mixing, when you are taking the reading from each speaker. It’s a good idea to hold the meter at arms length and at chest level to prevent measuring sound that may be reflected from your body.

In your DAW, insert a pink noise source on a mono track, with the level set at -20bDFS. This will allow you 20 dB of headroom in your mix before clipping occurs. In effect -20 dB is where the average level of your mix will sit. Set your noise generator to RMS rather than peak, so that you obtain an average rather than a peak reading. Set your channel fader at unity gain. First pan the signal to the left speaker. Set the pink noise running, and turn your speaker up or down until you get a meter reading of 79dB at the mix position. Do this for the right speaker, and each of the other speakers (if more than two are used). This should then be your normal “working” level.

It is then a good idea to mark the pot or fader with a marker pen or tape, so that you know exactly where they should be set to always obtain a 79dB reading.

Glossary of Terms used

What does dBFS mean?

The term dBFS relates to audio levels. Decibels relative to full scale is often abbreviated as dBFS. This is used to measure amplitude in digital systems, which have a defined maximum level before peaking occurs.

The maximum possible digital level is 0 dBFS. Clipping will occur at this point.

What does dBC mean?

The term dBC relates to decibels relative to the carrier.

What is Pink Noise?

A pink noise signal contains all the audio frequencies (20 Hz to 20 kHz) distributed equally by the octave within the hearing range of humans. These range from the lowest tones to the highest tones we can hear, and  are combined at the same time, and with the same density. Each octave has the same amount of energy as the ones adjacent , and decreases at the rate of 3dB per octave.

What is Unity Gain?

Where both the input and output voltage and impedance of a device are at  the same level. This is called  unity gain, and is most commonly marked as “0″ on your console or DAW fader.

What is SPL?

The abbreviation SPL stands for sound pressure level. SPL is usually measured at 1 watt and 1 meter in front of a speaker.

By Tony Koretz

© copyright 2011