2011 was a really interesting year, i was in a dark place like so many others for a good amount of time. saesons changing, personal development, ordering the newfound information. All of this had been hard to deal with. I had a lot of good times too and was in some really good places, met some really good people and had some fun experiences.

I try to get on with my life, but again like so many others; cannot help but be reminded of the current state of Human Civilization. To me its seems so obvious the we could be living in a much better world in every possible way. We all know it will never be perfect, but i believe we also know that it could be so much better than this for everybody and together we could set a more promising path for the future. It doesn’t look like we are heading the right way at the moment. It looks like more and more control of the people of the world is gradually being implemented, though i actually see it as being pretty rediculous right now.

This year has been the realisation of getting older, the realisation of my own mortality. I care so much about the world, I have so much love to give to its people. I want to see wonderful things in my lifetime and i want to be able to experience it with everybody else. The current pace of progression in human development is being slowed by the economic system(s) that currently operate around the world. Its not a question of weather its better, last longer, is more effective etc… its seems to be about whats the best way of making the most money for as long as possible…So why release something that never wears out for example? because it would not be the best way to make money. and that’s pretty much the bottom line. If it wears out, you have to come back or pay to get it repaired. That principle can be applied to everything in society and as a result of not creating the best we can, we create so much waste.

Sometimes i look at it like this: It all FICTION… everything we create in society – unless it something the directly affects us all, its a created problem by an individual or a group of men/women – usually as a result of a lack of understanding, awareness, acceptance and/or tolerance on someones part..Whats real is being alive, having experiences, learning, growing, developing and prospering as people until out time is up. The real issues are the ones that affect us all!

If there is an infinate number of ways that it could have turned out here on earth, surely it doesn’t have to be this way and surely this can’t be the best possible world we could be living in. I cant help but empathise with those born in less fortunate circumstances than me own and imagine what it would be like to have an imagination and dreams but no way of realising them. I think a lot of people in society feel like that sometimes, but those less fortunate don’t even have the opportunity.

There is such a division between those who have and those who have not in society – This seriously affects the quality of life for every individual alive. Whether the situation for each individual promotes a positive or negative quality of life is depending on where they born, the immediate family they were born into and the limitations on them to have relevant opportunities to make money. Unfortunately the system we is clearly not fair, nor does it present equal opportunities. I know there are some really great peopl alive today that have amazing inventions set to potentially revolutionize thye world in so many ways, however i also know that a lot of these technologies being released, put into practice or used would greatly threaten so many businesses that currently operate. So it is in the current system that it is in the interset of the businesses that stand to be affected by these new technologies/ideas to stop,/prevent them from or discredit said technologies/ideas. Unless there is a way for those affected business to capitalise from the new technologies/ideas directly, it is not in the businesses interest for them to be implemented.

So that’s kind of the the general feeling. This year I’m going to do what i can to spread the word through my music in a way that is positive and non threatening to individuals own thoughts beliefs and/or perspectives. Take responsibility for myself as an individual that i make a difference every day just by being here and all the people i come into contact with throughout my life. In this way, i feel we truly are connected. Life wouldn’t be worth living if there wasn’t people to share your experiences with.

The greatest problems in the world arise from indifference in beliefs.

This is particularly demonstrated in the practice of Religion and the diversity of Religions throughout the world.

In my Opinion: Everyone has the right to believe and No one has the right to force their beliefs on to others.

A Belief is after all; simply a belief and therefore is pure speculation and/or hope of an idea in which a potential comfort is found.

To force a belief onto others is to try and convince them (as much as to further your own belief) that it is the truth.

In my opinion; we as a species have much to learn – We are so primitive – I Believe! Better time would be spent trying to find out the truth. So called belief can often provide us with the drive to find out.

SECRETS :

Secret Societies?

Why??

Can you not trust anyone else??

Or is it for an advantage over others created by ensuring they are kept in the dark to certain information???

In my opinion: I believe there should be no secrets; other than our own thoughts in our own heads, which we can choose share or to not at our own will. People should not feel like they should need to hide anything from anyone else. Nor is it courageous to seek power over others. It possibly suggests an underlying low sense of self-importance or a self belief of greater importance.

FAITH :

We should have more faith in other people, after all – We are all the same.

Yes we are all individuals in many respects. However these are minor differences when compared with the vast similarities.

So how can anyone be more important?

What is Beauty?? – Beauty is in the eye of the beholder; it doesn’t exist until something is appreciated in an emotional way.

How do you feel?

What is Love??

We all know Fear!!

GOVERNMENT/PEOPLE :

People (Humans) make mistakes. They are not perfect.

Each Imperfect person does a job.

We live in a system based on intended systems of control with no exception for human imperfections or mistakes.

People take bribes for personal gain. Everyone has a price. If one person can get away with this, whets to stop anyone else. – Anyone can be tempted.

How can we trust anyone to not try to gain based on the nature of the way the system works? The system promotes greed or at least – Personal and/or private gain.

We have to work to live, to earn the money, to provide.

Why??

Whets the point of devoting your life to something if you don’t necessarily enjoy it. Why do we not encourage those around to follow their passions fully? Unfortunately once again restricted but the necessity to keep the money flowing through the necessary channels.

In 50 years time they will look back at how primitive we are now.

GOOD/EVIL :

The system corrupts – When personal gain overrides morals.

Nobody really cares about anyone else’s problems unless they become affected by them.

We all want the same things in life fundamentally:

> Love

> Happiness

> Friends

> Family

> Fun

> Appreciation

> Fulfilment

And if two heads are better than one; think what we can achieve as a species if we really put our minds together.

There is nothing to fear but fear itself and without it, we could not know love.

The greatest payment of all is appreciation, respect and love.

Why do I pay for Education when Learning is the one thing that gives us each the opportunity to contribute to society and better the world?

WAR :

Why do we fight Wars when both sides are usually fighting for the same thing?

Their country!

But the people who started it – are not fighting?

They just funded it and allowed it to happen. Possibly playing war games in real life, with real people’s lives.

It’s just civil war on Earth between one species and one family, fighting amongst ourselves like children.

Still hoping for Maturity. Maybe this is adolescence

Ash

I think most people can agree weather results show global warming is occurng or not, burning crude oil and other non renewable fuels is not very logical.

Based on the level of technology in the modern day, we can easily come up with renewable green energies by working together to establish something practical.

Some of the main alternatives i am aware of include:

> Geothermal Energy
> Solar
> Wind
> Hemp Seed Oil
> Biofuels
> Hydro Power
> HHO (Gas)

By utalising theses kinds of energy we can help to preserve the earth for future generations and also start providing cheap energy for the population of earth and ultimately lead to free energy for all.

Please post your responses.
]]> http://heytukay.wordpress.com/2010/07/01/number-2/feed/ 0 heytukay http://heytukay.wordpress.com/2010/07/01/number-1/ http://heytukay.wordpress.com/2010/07/01/number-1/#comments Thu, 01 Jul 2010 03:04:33 +0000 Tu-kay Records http://heytukay.wordpress.com/?p=60 ]]> I think the first thing on the priorities list should be:

GOOD QUIALTY OF LIFE FOR ALL

As most people are aware people are dying in the world simply becase they cannot eat, whilst others can live a fairly wealthy lifestyle. We all fundamentally desire the same things in life; Happiness, Love, Friends, Family, Fun, Adventure and most importantly Freedom.

The main points to helping us work towards a good quality of life for all is ensuring that everyone can have the following:

> Food – High Quality Foods (untampered)
> Water – Clean and healthy
> Shelter - Decent Comfort (all seasons)

Other Key Factors Include:

> Healthcare – free healthcare o make sure we stay healthy for future generations

> Opportunities – opportunities for all to learn to experience and to have fun

> Employment (without being exploited) – to be a part of something and to benifit from your contribution

> Education – Everyone has the opportunity to learn to the highest standard by an approach suitable for the individual.

Any thoughts and feelings, please respond

Many Thanks

Ash

To export or bounce a track, this can usually be done from the file tab on the menu bar and the export followed by audio selection. When exporting audio information is important to appropriately set the right bit depth and sample rate at which to export.

CD Quality is: 16bits (bit depth) at 44,100Hz (sample rate). If exporting at this setting most if not all music playing devices will recognise the track. However if the track is exported at 24bit and a possibly higher sample rate. It would not be recognised by most CD players and audio devices. In this case you would probably just hear nothing for the duration of the track.

16 bits @ 44.1kHz would be the standard export factors and would ensure your music could be heard on most systems. The track would also need to be exported as a stereo interleaved format (as appose to left and right separately). This is to ensure that the left and right audio signals are exported as a connected (interleaved) pair.

When dealing with MP3s; tracks can be written to CD and converted to the right format or can be recognised by most hand held music devices such as iPods. Some CD players will not recognise MP3 CDs, therefore it is another important thing to be aware of when writing CDs and particularly MP3 CDs.

Tracks can usually be exported as various different file types including mp3. In most cases it is best to stick with the uncompressed WAV format as it can be recognised across platforms and devices worldwide and is an uncompressed audio format. WAVs can then be converted if necessary to a different format such as MP3, at the desired buffer size using an audio converter program. 320kbps is the highest quality MP3 and biggest MP3 file size. Any less the 120kbps and you can hear a distinct reduction in the audio quality.

Mastering

If mastering an audio session you might either chose to master the track within a mixing and producing session or alternatively; export the track at a higher bit rate such as 24bit at 44.1kHz and import the track into a new session to be mastered. Once then mastered the track could be exported in the appropriate format.

In a mastering session the main idea is to make only finalisation tweaks to help tweak the overall sound of the track and tidy up the overall sound to improve the overall quality as best possible and round off the mix in terms of frequency particularly.

When mastering you should pay particular attention to Overall levels, frequency proportions and the overall stereo image. When mastering any little change on an EQ or processing unit can change the sound exponentially, that is why it is important to try to get as close to the desired result as possible in the mix down stage. the mastering session can just be integrated to finalize the desired sound and implement the last methods of controlling the overall frequency content and dynamic.

Converting

Various formats including WAV and AIFF can be converted to MP3 at a buffer size of 120kbps-320kbps (kilobytes per second). The higher the amount the greater quality and file size of the MP3. By adjusting this accordingly you can find the right file size for the intended application For example you might export a lower quality mp3 for MySpace but a higher quality MP3 for you iPod when recording your own music. The demand for smaller file sizes is gradually decreasing overtime, therefore it is becoming progressively more common to deal with files and formats of bigger size included WAV and AIFF on occasion.

Writing to CD

Wavs can be written as can mp3s, will need to be converted to mp3 if intended for iPod or to post online and minimize file size. When writing WAV files to cd it will automatically convert it to the right format during the write process (likewise with mp3s).

Posting Online

Quite simply done by locating the file and uploading through the configured ways to sites such as MySpace or Reverbnation etc. Usually browse for the file, select it and click upload.

Which DAW software should you use?:

Though there is a range of various different Audio and music production software available; which one you chose to use is entirely subject to the user. Each of the programmes function in much the same way, often with only minor differences. Each of the DAWs listed below would be a good choice for recording. Pro Tools and Logic a the industries leading brands of software DAW. Logic is renowned for being more visually appealing and better for MIDI plug-ins and programming, whereas Pro Tools has a reputation for making greater use of audio-editing functionality. Listed Below are some demonstrations of the and tutorials of Cakewalk Sonar, Logic Pro and Pro Tools; each of which are well rounded (all-in-one) music production DAW software.

Cakewalk Sonar:

Logic Pro:

Pro Tools:

Saving Project:

Its is very important to make sure when you start a new project that you save in the right location (so you can find the file with ease) and appropriately titled; bearing in mind there may be later versions of the same track at a different stages (in different places). It is also important to back-up work frequently (save in multiple places) to avoid loosing any work or progress. It is common practice to allocate a specific drive or partition for all project files or data and to avoid saving projects on the desktop of any operating system. Countless people have lost work through one means or another; from saving and working from the desktop to not backing up work periodically.

Bit Rate/Bit Depth:

When initially starting a project, it is essential you set the project to record at the right bit-rate and sample depth. A good and common sample rate and bit depth would be 24 Bits (binary Digits) at 44,100 Hz (frequency – cycles per second). It is not uncommon for people to use higher bit rates for greater recording accuracy or for working in other areas such as music for film or television. a setting of 24bits and 41,000 Hz sampling depth would be a perfectly acceptable setting to deliver good quality signals with a good dynamic range. Increasing the Bit depth or sample rate increases the files size which when dealing with greater amounts of audio become progressively more CPU intensive.

Inserting Tracks:

This can usually be done in most (if not all) programs by accessing the toolbar (insert/audio track). The main two types of track dealt with when recording are Audio tracks and MIDI tracks. When inserting a new track; you must make sure the tracks input is configured to receive input from the relevant input channel from the soundcard (to which the instrument is essentially connected) and make sure the out put of the track is configure to the right output of the sound card (typically 1+2 – Left and Right) so you can hear the desired audio when recorded. Always ensure you have a good signal level coming into the computer (via soundcard) without it surpassing 0dB. This is to avoid any digital clipping or distortion.

Input Monitoring:

Input monitoring can be applied so you can hear what is being recorded as you play it or alternatively (if your using a mixing console) you can configure the setup to take a headphone feed from the desk rather than the soundcard and monitor the input signal directly. This defeats any latency issues; however you should always test to ensure a good recorded signal before committing as you are monitoring from the desk not what is being recorded. Provided everything is working correctly there should not be any issue.

Click/Tempo:

It is often common practice to set up a click track from the outset of a recording project. The speed would obviously depend on the project you were recording and whether there are any tempo/speed changes throughout the song; in which a tempo could be inserted at that point (if necessary) The easiest way to accurately set the tempo of a track is to get the performer to play it and tap the tempo (tempo tap) accordingly in the software DAW to match what they play. 1 beat per minute difference though hard to notice, makes all the difference to the overall speed and feel of the track. It is often important therefore to be specific about the tempo rather than just rounding (for example).

In the case of a live recording a click track would not be necessary, though as a result the performance would need to be good, tight with as little performance mistakes as possible as editing would be harder due to the fact that nothing has been aligned or played to the grid from the outset. Editing capability is reduced and difficulty is increased.

Headphones:

When recording it is always necessary when using microphones to use headphones, particularly in the case of a home or bedroom studio due to the fact the recording often takes place in the same room as the mixing/control. When recording using microphones therefore it is always necessary to use headphones to a) hear the track and b) isolate the sound of the instrument within the room so its the only thing picked up by the microphones.

Environment’s:

Each environment has different properties to another. Some rooms are reflective and some rooms are dead (designed to diffuse sound quickly with high amounts of absorption in environment) in varying proportions.

Its is not uncommon for bands to record in strange environments to help define a particular sound. People have been known to record in churches, temples, tombs, cathedrals, concert halls and pretty much any environment should be considered if appropriate to the music and/or style you intend to record.

Microphone Choice:

There is a extremely wide choice of microphones available at varying price, each with a different characteristic and often made up of different quality components (depending on make, model and price). For the purpose of the accuracy, relevance and appropriation of this page; i will suggest more widely used, available and affordable microphone choices.

It is always important to familiarise yourself with the sound and characteristics of a good range of microphones; to make sure you have a good idea of what microphone would be appropriate for each recording session/task, based on the desired sound you wish to capture (particularly in terms of timbre or tone).

Small diaphragm microphones are often better for recording high end sound sources due to the small size of the diaphragm itself and the nature of the size or wavelength of the frequencies. A larger diaphragm condenser microphone would typically be better therefore at producing lower frequency, due to the larger size of the diaphragm.

When recording any instrument it is important to take into account not only what microphones will be used and how they will be positioned in accordance to the instrument, but also the situation of the instrument/performer in the room (the best sounding place in relation to the sound of the environment in which you are recording.

Microphone choices are entirely subject to the users preference, though the simplest thing to be aware of is that typically microphones are suited to a task, just biased on the style of the microphone (whether its large or small diaphragm) or how high sound pressure levels it can withstand. For example you wouldn’t use a ribbon microphone inside a kick drum, due to the high pressure sound levels damaging the ribbon by witch the microphone operates.

It is also essential to make sure all instrument are in tune before record them. If using a capo it is often a good idea to retune the instrument once the capo is positioned; to assures that all instruments will be in tune with it.

Drums:

Kick: AKG D112 & Shure SM57/Small diaphragm condenser

It is not uncommon to use two microphones for the kick drum. An AKG D112 has been rated for years as the best value for money kick drum/low frequency instrument microphone and is use heavily throughout the industry. The AKG D112 would be positioned inside the kick drum accordingly. The other microphone used would typically be to capture the clarity of the drum. By blending the two and adjusting the positioning accordingly to counter any phase issues a good kick drum sound can be achieved.

It is very important to ensure the kick is appropriately dampened before recording too as often the sound of the skin can heard flapping, by dampening the drum with a pillow or duvet you can improve the precision of the drum.

Snare: Shure SM57 x 2

I personally always like to use two snare microphones. One on the top (to capture the initial hit and overall attack of the drum) and one on the bottom to capture the harsh snare sound of the bottom. Two of the same microphone would be used (two SM57s ideally) for sound and recording consistency and the bottom microphone would need to have the phase reversed so the two microphones were working with each other in positive phase.

This can be understood by the direction of the snare drums skin when you hit the top with a stick. The top skin is hit away from the stick, which in tern causes the bottom skin to also move away in the same fashion as the top (based on the hit). This means that the top skin is moving away from the microphone on top and the bottom skin is moving toward the microphone at the bottom. By reversing the phase of the bottom you put the microphones into positive phase so both skins are moving away together reinforcing the snare sound resulting in a more punchy snare.

This is also know as the reverse phase snare drum technique.

Toms: Shure SM57, Sennheiser e604

Generally hoping to use as directional microphones as possible to avoid the spill from the cymbals etc. It is often common practice to ensure the drum kit is also set up with microphone placement consideration as well as the drummers playing considerations. The Sennheiser e604 is one among many different types of Tom specific microphones and it works really well. The Shure SM57 is also heavily used throughout the industry on tom toms and snare drums (particularly)

Overheads: Shure SM81 x 2, Shure KSM 137 x 2, Rode NT4 x 1, AKG 414B x2

Glyn Johns Overhead technique:

This technique has been used on much of the Led Zeppelin drum recordings from the 70s. The technique is particularly good for phase due to the equidistant positioning of the two microphones from the snare drum and kick drum.

The first microphone is be positioned directly above, pointed at the snare drum (around 4′ above) and the second microphone is position over the right shoulder of the drummer (assuming he is right handed) pointed towards the snare drum. The easiest way to make sure they are the same distance is by using a microphone lead. hold one end in the middle of the snare drum, then mark with your other hand the distance (on the lead) of the microphone, then simply move the marking hand towards the other microphone. The poisition can then be adjusted accurately.

Another good feature of this techniques is the fact that due to the equidistant positioning you can move the microphone (pivoting from the snare) anywhere around the drum kit to achieve the right sound. Bear in mind that when using multiple microphones as well as the two overheads the relative positioning of each of the microphones an in accordance to each other increases the potential phase. This is why it is all the more important when recording drums to ensure microphones are well positioned, balanced and properly auditioned seperately and together.

Trash: Shure SM57/SM58, Neumann TLM 103, or any crappy microphone for an already raw sound

The trash microphone is often a dirty sounding microphone position around the middle of the the kit next or near the drummer (facing in the same direction). This microphone is then typically used to add some raw dirtiness to the track or often delays are used on this microphone in particular to create movement in the drum sound and help to fill the space in a more rhythmic sense.

Ambient/Room: AKG 414B x2, Shure KSM 137, Shure SM81, Ribbon microphones (good for natural sound)

Bass: Direct Injection (D.I), AKG D112 (maybe SM57 to capture amp/room clarity of bass)

Bass is generally quite straightforward to record, it is not uncommon to take a clear line (D.I) signal for the natural sound of the bass guitar (if using a nice sounding good quality bass) or a woody tone as well as an amplified and/or potential room sound. This is so the signals can be blended to find the right overall bass sound and tone. It is not uncommon for bass to be recorded simply D.I due to its low frequency nature and its application within the song (much of the high-end frequencies are often rolled off, the rest of the sound is often just shaped to compliment the track further).

It is important as always to choose the right amp (if using one) for the sound you are intending on archiving or possibly even audition amps to find the best one (if you have time).

Acoustic Guitar:

A typical stereo microphone technique for recording an acoustic guitar is to position a small diaphragm condenser microphone pointed at around the 12th-14th fret of guitar (sometimes sounds good angling the microphone towards the sound hole/bottom of fret board) and a large diaphragm condenser microphone positioned near the bridge of the guitar, marking the bridge/sound hole. Microphones should be positioned 1-3 inches away though it is important to adjust and re-position accordingly to achieve best sound for style/make/model of guitar as well as the reflective properties of the room and respective microphone choice.

When positioning multiple microphones in accordance to one another; balancing them symmetrically to the instrument and the room, often solves any phase issues.

It is important to find the best position for clarity of notes, fret and pick noise, combined with low-frequency warmth and mid-range from the sound hole; to suit the nature of the style of music but more importantly; the sound you are hoping to achieve.

Another good technique is to record a D.I signal from the guitar and position a small diaphragm condenser microphone at the 12th fret angled between the 12th fret and the sound hole. The D.I captures the dry warmth and body of thee guitar, whereas the microphone captures the natural (room effected) sound and clarity of the notes.

Typical microphone choice – 12th fret: Shure SM81/Shure SM57/Shure KSM 137. Bridge/Hole: AKG 414B/Rode NT2A/Neumann TLM103.

Electric Guitar: Shure SM57 (close just off centre from cone), AKG 414B (distant at at ratio 1:3)

It is often good practise to select the right amp for the job depending on the genre (some amps are associated with certain sounds – Randall = metal), some amps have multiple speakers in the cabinet, it is often best just to concentrate on one, as it keeps the microphones in a good vicinity to each other particularly when recording using multiple microphones.

It is not uncommon to use a dynamic microphone like an sm57 on just off centre left or right of the cone between 3 and 6 inches away (clean and clear without being too harsh on the high frequencies and not too clinical) and a distant microphone, often a condenser microphone at a distance of about 1.5 to 2 feet with the intention of blending the different timbres of each microphone to achieve the desired sound. It is also important to be aware that if you are using valve amps and equipment; the valves need to warm up to give the better, warmer sound quality (greater depth/warmth particularly on low frequencies).

Usually if you position close and distant microphones at a ratio of around 1:3, the distances would be expected to be positioned well in terms of phase, however in a recording situation it would be expected of those involved to audition microphones and adjust microphone setups accordingly to tweak the sound and achieve a specific tone or timbres; based on the overall setup and recording environment.

In some situations multiple amplifies are used with microphones on each to find the right overall blend desired. The microphones could also be panned accordingly to create left and right blends. This could be configured at the mix down stage, or a blend could be achieved and recorded as a combined signal.

Vocals:

When recording vocals it is important to first choose the right microphone for the vocalists voice. This can be done by again auditioning microphones. Where the microphone is in the room can also effect the sound of that microphone; it is therefore often a good idea to find the best position for the microphones in the room and set up and sound treatment accordingly and audition the microphones from that position (to find the best one for the sound of the vocalists voice based on the relative position in the room and the sound treatment).

When recording vocals you must be aware of the proximity effect (increased bass response due to distance from the microphone being to little) and pops, sibilance or any pronunciation issues (this is why a pop shield should be used). Another method of countering sibilance issues, is to use an elastic band to fix a pencil central to  the diaphragm of the microphone so as the singer sings into the microphone the pencil splits the air,; reducing any sibilance. This can also be achieved by positioning the microphone off axis for a less direct sound again reducing any pronunciation issues.

It is common practise if you don’t have access to a vocal booth; to use foam tiles or something similar to position around the room to increase the diffusion of sound within the room. This will aid in more clear sounding vocal with a reduced room reflections being picked up by the microphone.

It is common practice in a recording situation to expect and encourage the artist’s to play as many time as you see fit (as the producer) in order to capture the desired quality performance. This is not only to assure you capture the best performance but also to allow the performer to warm up and to make sure you capture enough takes to compile a comp or compilation of different audio takes into one super well played/performed take for use in the final mix of the song.

It is by using this technique you can get the best overall sound of each part, though this would be done more so in the editing stage or between sessions to aid in the progression and again to avoid having to rerecord parts later because of a lack of editing and compiling early on..

When multi-tracking lots of different microphones and signals at the same time, you are limited by the amount of inputs to your soundcard and/or mixing desk. for example if you have an eight channel soundcard (eight inputs) you can only record a maximum of eight tracks at the same time (onto eight different tracks in the software DAW). To capture more than eight in this situation you would have to have another soundcard or input device allowing for greater inputs. When multi-tracking is limited by the amount of inputs, overdubs are the only way (recording each bit separately) is the only option, though if you have an eight input soundcard you could use eight inputs for one instrument at a time (if needed).

To multi-track each microphone or signal would simply be plugged into its own independent channel in the soundcard and then each track would need to be configured correctly to be received from the right input/output of the soundcard. When you hi222ecord; every track will record at the same time onto the relevant signal/microphone track in the software DAW. This is a good technique to use (providing you have the means) to record live bands (with the intention of capturing amore convincing live sound). In this situation there would be a greater need usually to control the reflections of the room in which you are recording and to set or position things in the recording room to actively manage the propagation of sound and help control the spill of each instrument into the other microphones (intended for other instruments etc).

Overdubbing is not only a good method of adding or recording the relevant parts of the song; but also a good means to create blends and thicken out the sounds of parts in the song. These parts can be later experimented with at the mix-down stage.

Various parts you might choose to overdub particularly include the guitars (for depth at different parts as the song progresses) and the vocals (backing vocals and ad libs again to thicken out the sound but also to aid in a greater harmonic and melodic content of the song).

It is good to experiment with overdubbing to find the right sound you are looking for. Different overdubs can be panned differently to enhance the stereo image also, particularly when dealing with mono signals.

Processing can be used such as EQ when recording or maybe a little reverb in the vocalists headphones for them to hear. However it is important to be aware that if you record an altered signal (EQ applied) it is destructive to the recorded sound source, meaning if it is set incorrectly, it will be recorded that way. Whereas if you send reverb to the vocalists headphones only but record a clean signal; as much or as little can be added afterwards at the processing/mix down stage.

It is important therefore to only EQ and/or process the recorded audio if you as the producer or engineer are sure about what you are doing and are willing to commit the settings to the recorded audio. A lot of producers chose not to do this and many in fact only apply effects/EQ/processing at the later editing and processing stage, due to greater flexibility in the desired sound and no potential mistakes slowing or lengthening a recording process (particularly if working to deadlines).

• Never rush a session

• Always be prepared to go back and record something again if its not right

• Expect to have to work to deadlines

• Effectively plan to make the best use of studio time.

• Try to each recording session positive

• Aim to not be too destructive to the recorded audio unless you are absolutely confident it is accurate in terms of the sound you intend on achieving.

• Don’t over work people (but expect to have to record multiple takes, maybe issue breaks when you feel necessary)

• Plan an appropriate path of process order to ensure a logical build up of the track – this helps for example when there is a need to edit during the recording process, so that things can be recorded and edited before moving on the the next part, furthermore ensuring an easier gradually progressing to the recording session itself (without having to record again due to bad planning).

Reference Material:

When mixing music it always important to have or define a good understanding of the sound you hope to define. It is always a good idea therefore to have a range of music that in someway demonstrates a sound close to that which you hope to achieve. This is known as reference material. By using reference material it can help to put a mix in perspective from the get go by comparing your mix to your reference material.

Balancing:

Stereo Image:

The stereo Image is by shaped by the effective use of balancing (in terms of panning and the effective use of EQ) to define a good  stereo Spread (Left + Right).  A good stereo image helps to accurately shape the perspective, the environments and/or relevant positioning of various parts in the mix. When working in MONO the stereo image does not apply as it is 100% equal on all speakers. This may bee seen for music intended to played in clubs and specific venues based on the sound systems and or listening environments. Some music is mixed to be appropriate for these venues and in some cases it is necessary for mixes to be in mono to give the best listening experience.

Frequency Spectrum:

The human hearing range starts at a low frequency of 20Hz and ends at the high frequency of 20,000Hz (20kHz) though we lose the high frequencies in modern times at an early age due to frequent noises in our various environments from the day we are born.

As a deterrent from loitering or from crime committed by young potential offenders in certain areas; speakers were put up that generated a frequency somewhere between 18kHz -20kHz at certain (problem) times.  Only younger people could hear the frequency and it is incredibly annoying.

There is an application on the iPhone that can functions in much the same way, intended to play practical jokes on certain people.

The frequency spectrum can be most easily understood and put into perspective through the use of EQ. Our ears are more susceptible to frequency boost, rather than cut. Therefore  if someone boosted a frequency of something, you would be more likely to be aware that something had been been boosted, whereas if it was cut by the same amount, you would not notice it so significantly. It is common practice therefore; to first exercise subtractive synthesis to subtract the main unwanted frequencies and help to first equalize the recorded sound. It is always good to be open to boosting to where necessary after all the rules are thereto be broken if it sounds good.

EQ can be used in various different ways to shape the sound of anything. You could use as many EQs as your computer can handle on one thing if you so desired. However i would recommend (and i believe this to be a sensible working principle) to keep your EQ use to as minimal as possible, for the sake of being very accurate with what you do on each. Any slight difference makes all the difference, this is why it is important to really listen and bypass frequently to hear whether the sound quality improves or declines as desired. Though the option is always there to use another if it feels necessary. When using digital software DAW-based plug-ins and particularly numerous EQs on one thing; it can make the sound of something become over-processed in a digital sense. As more and more are added the sound begins to sound less natural.

Another thing to be aware of when mixing anything using a software DAW is how you listen when you can see the audio data on the screen as well as the EQ or various effects plug-ins virtual control surfaces. It can be easy to edit things based on what it looks like your doing on the screen rather than listening to the actual sound coming out of the speakers. Over time by familiarising yourself with the frequency spectrum; you will be able to tune your ears in to certain frequencies in a mix to isolate what you like and what you don’t like about it (in terms of frequency) as well as how each part works well with other parts in the mix.

If you don’t know what to listen for; you won’t be able to hear it.

If its obvious; anyone can hear it.

Analyzer:

An analyzer is a type of plug in that can often be found within each of the available software DAWs and is a good means of seeing in the form of a real-time digital graph; what frequencies and amplitudes are contained within a signal. By effectively listening and/or using an analyzer, a good understanding of the frequency spectrum can be learned.

Parallel Compression:

parallel compression is a good technique that can often fatten up a signal and generally achieve a thicker blend of sound from one or more signals. This can be achieved by creating a new bus and applying compression. Then aux-send each of the desired signals you wish to parallel compress to the newly created and compressed bus. Then adjust the compression settings as well as the sends amounts and fader proportions until a satisfactory blend is achieved.

Side Chaining:

the act of side chaining is a technique that often features within gates and compressors amongst other plug-ins. In the case of a compressor on a bass, a kick drum could be side-chained to the compressor in order to create a ducking effect. When configured correctly (settings on compressor) each time the kick drum hits the compressor acts on the bass emphasizing the clarity of the kick drum and aiding in separation. This is also a good technique to create movement in a mix and often features in house music.

Automation:

Automation is the software DAW way of controlling and regulating the various settings such as volume, aux sends and panning (amongst others) of parts throughout the track. It can also be used to regulate the proportion of almost anything throughout the track in the mixing process. This include almost any feature within any plug-in such as Delays, Reverbs, EQs, Compressors, Gates and much, much more.

Busses:

Buses can be used to group channels as well as to place effects soles with the intention of sending tracks via auxiliary sends by desired amounts for example a reverb plug in. Effective use of plug ins can aid in a greater overall control of each aspect in the mix as you gradually group parts together. Once settings have been configured on drum channels for example you may wish to group snare top and bottom microphones to one bus, overheads to another, toms to another and the each of those busses to a master drum bus. This then gives you the control of each aspect in a more functional organised  method of practice.

Aux Sends:

Auxiliary sends on each channel give the option of sending a signal somewhere else by a desired amount independent of the original signal path. when using an effect on a bus, these can be used to adjust a good mix of each to the effect-applied bus.

Things can be sent via auxiliary Pre or Post Fader. If something is sent post fader then adjustments on the fader from which the auxilary is sent; affects the auxiliary sends amount. If something is sent pre fader then it remains unaffected by the fader on the channel from which the auxiliary is being sent.

Master:

The master fader can be used to apply compression as well as EQ and limiters if necessary or in fact any plug-in you desire, though it is recommended to only make minor alterations or tweaks to the overall sound of the mix.

It is common to apply compression and limitation to a mix and tiny tweaks on a EQ plug in to help control the overall sound and dynamic of a mix. Be aware that throughout a mixing process compression and processing gets applied to most aspects of a recording project. The when exported it is compressed again, then if you decide to convert it to mp3 it is again compressed. This is why it is important to always try to do as little as possible in processing the sound before achieving the desired sound. By aiming to achieve this you can assure avoiding over processing and potentially becoming destructive to overall sound quality of the mix.

Overall Levels:

It is good to aim for as high volume of something in the mix. This can be pushed to the liomit by controlling the overall dynamic with compression and limitation though over compression can turn a mix very linear. Therefore aim to get as higher level as possible with no peaks, but try to ensure a good dynamic is maintained throughout.

Plug-Ins

A plug-in is either a software instrument or a tool for processing, such as EQ, Compression, Delay, Reverb, Distortion and all other audio manipulation tool that can be used within a software DAW.

Plug-ins come in a variety of formats including: VST, VSTi (Instrument), RTAS, DX, (Direct X), DXI (Instrument) amongst others. some programmes recognise some or all of the different formats of plug-ins whereas others may only feature a few fundamental formats.

Within each of the software DAWs available there are a range of plug-ins included. Much of the plug-ins provided work well and provide good results. There are however many third party plug-ins available. Some good quality Plug-ins include: Waves bundles, Sonalksis, Sonnox, IK Multimedia and Sonitus amongst others. Each pack features a good range of different Processing plug-ins.

There are also a range of Plug-in Instruments available as mentioned in the editing section of the knowledge archive.

Use of Effect

When using effects to manipulate the recorded sound it is good practice to first have a good idea of the sound you are trying to achieve.

By experimenting accordingly with the settings on the various plug-ins, the sound can be shaped in an unlimited number of ways. It is always a must when processing different parts within a mix to frequently listen to how the processed part sounds in context to the entire mix. When one aspect of a mix is changed, it can affect the rest of the mix exponentially. This applies particularly when using EQ to shape the sound of something in the mix.

Any change in one aspect of the mix can affect the overall sound and feel of a track.

When using any plug-in it is often common practice to first boost the effect in order to hear and accordingly shape it however intended; it can then be reduced to the desired amount.

It is often better to be subtle (particularly when using effects like delay and reverb), rather than swamping something in it. A little touch goes a long way though whatever rules stand in a production process; they are only intended as a guideline and are there to be broken, if it sounds good.

Plug-ins can be used as an insert on any channel of the mixer or one or more channels can be sent to an bus channel via auxiliaries on which an effect can be applied

Attack, Decay, Sustain & Release:

Attack , Decay, Sustain & Release are a common feature to many plug-ins and plug-in instruments. Attack is the rate at which the plug-in acts after the sound source surpasses the threshold or in the case of a plug-in instrument, how quickly the plug-in acts when the key is pressed/triggered.

Decay is the rate the plug-in reduces after the initial attack to the sustain level, Sustain is the amplitude to which the signal maintains before the key is released (so long as the key is held) and the release defines how long the plug-in continues to work after the sound signal level falls below the set threshold or in the case of a software instrument, when the key is released or the triggering triggering note ends.

Reverb

There are two different types of reverb; convolution (Real) and non-convolution (Fake).

Convolution reverb is created by capturing an impulse response of an environment, such as a balloon popping by using a stereo microphone technique (for a stereo reverb). Once recorded the impulse response can be used as a reverb setting within a convolution reverb plug-in such as Logic Space Designer or Audioease Altiverb.

Non-convolution is a purely digital based plug-in. All of the reverb settings within are generated by a series of algorithms.

When using reverb it is good to use a combination of convolution and non-convolution reverbs. When defining the overall environment to send each instrument by a desired amount it might be a good idea to use a organic natural sounding convolution reverb. Non-convolution reverb could then be used to tweak the overall amounts on specific channels, where necessary.

Above: Sony Oxford (Sonnox – non-convolution) Reverb Plug-in

The main settings on a reverb plug in include: reverb mix, early reflections, reverb time/RT60, size of environment, diffusion, high and low pass filters and sometimes EQ.

Equalization (EQ)

Equalization is one of the main plug-ins used within a recording/editing/mixing process. It can be used to manipulate the sound of anything throughout a project.

Our ears are more susceptible to boost rather than cut, meaning that it will be more obvious to you if a frequency has been boosted, than if it is cut by the same amount. Therefore it always better to cut rather than boost. Therefore when initially using EQ on something it is common practice to first take away what is immediately bad about the sound.

The best way to accurately find the offensive frequencies is to first boost a band and then cycle through the spectrum until you hear the frequency you are looking for. The frequency can then be reduced by the desired amount. As with any plug-in and particularly when using EQ it is important to bypass what you have done and listen to it before and after as well as in context with the overall mix. By following these steps you can accurately define each part within an overall mix to achieve the overall sound you are looking for. If something is not right you can always go back and edit it, or sometimes its best to delete what you have done and try again.

EQs can also be use as high and low pass filters; to either get rid of the lower unwanted frequencies – high PASS or to get rid of the higher frequencies low PASS.

Through practicing the use of EQ you can gain a good knowledge and understanding of the frequency spectrum.

Above is a Sonitus 4 Band (Paragraphic) EQ

The main settings within a EQ plug in include the Gain (Boost/Cut amount), Frequency, Q (Band Width), Filter Type (High-Pass, Low-Pass, Peak/Dip and Shelf). Different EQ plug-ins often have varying amounts of bands usually around 6 for which you can adjust all the covered settings.

EQ can be used on absolutely anything throughout a production process.

Compression

A compressor is used mainly to control the overall dynamic of the audio on which it is used. It can also be used to accent the attack or initial impact of a sound source (such as a snare or kick or to accent the quieter aspects of the sound source/recorded audio.

The compressor is based around the threshold. Once the threshold is set accordingly; any audio that passes above it will be compressed by the set ratio. If the ratio is 2:1 this means for every 2 dB the audio passes above the threshold it will be compressed to 1dB above. therefore the ratio indicates the input and ouput signals of the compressor.

By adjusting the attack, Knee, release and various other settings on the compressor you can adjust the overall severity of the compression.

When applying compression it is important to avoid over compression. when something has been over compressed it can lose dynamic and sound squashed and over processed. It is a good idea to always try to use just enough compression rather than too much and again bypassing frequently to hear the changes in context with the overall mix.

Above: Logic Pro 9 Compressor Plug-in

Other settings: Threshold, Attack, Release, Knee, Circuit Type, Limiter, Gain/Output

Gating

A gate can be used on an audio signal to reduce or cut out unwanted sound when the sound source is not in action. For example by using a gate on a vocal, you can cut out any unwanted sound picked up when they are not singing. By adjusting the gate settings accordingly you can make it sound as natural as possible. This can help to clean up a signal. The function of the gate is mainly based around the threshold as with the compressor. When the input signal passes over the threshold, the gate opens up allowing the sound through. When the sound input signal drops below the threshold the sound signal is cut off or reduced by the set amount.

Above: Sonalksis SV-719 Gate Plug-in

Other features of a gate include: dB Range (how much it reduces the sound source by), Attack, Hold, Release and filters (to further define the frequencies it doesn’t allow through)

Delay

Delays can be used in various different ways to help shape and define the sound of parts within a track as well as to help create movement in the overall mix. A delay can be set to delay by a note value or by a manually set time. Tight delays can help to define more width in the applied sound source and using a stereo delay such as the one below give the user the option of setting different delays left and right in different proportions and settings; means that they can be an effective way of creating movement around the sound source in a potentially rhythmical way.

Another common feature of a delay plug-in is a filter. This can be used to adjust the frequency range of the delayed signals to find the right blend with the original signal.

Above: Sonitus (Stereo) Delay Plug-in

Other features of a delay plug-in include: Feedback, Crossfeed and diffusion.

Arrange Window:

The arrange window is the main area in which you work when using a software DAW. The screenshot above is the arrange window from Cakewalk Sonar 8 Producer Edition. All of the digital audio and MIDI information is displayed here and can bee arranged (hence the name) and edited.

As the event meter (which can be seen as the vertical line on the left side of the pictured arrange window) passes over the audio and MIDI information; that information is translated into sound through the sound output via the speakers/monitors.

Tools:

There are various different tools available to you when working within a software DAW. The main tool used within any program is the simple mouse pointer. This functions as always allowing you to select and move the audio and MIDI information as well as work the various controls in the program. When working specifically in the arrange window, there are various different tools available to you other than the mouse pointer. Some tools featured often include: Cut, Glue, Free Edit, Envelope, Draw and Fade/Cross-fade amongst others. Pro Tools has a multi-tool function built into it and most things can be achieved in Sonar simply using the mouse pointer. Logic has a good selection of tools with the option of setting a secondary tool and toggling it by holding down alt on the keyboard, however this often applies Sonar and Pro Tools in a slightly different way.

Shortcuts:

Shortcuts are often assigned to certain functions within the program(s) to allow for quick working, editing and navigation throughout the recording process. This can be as simple as the Spacebar operating Play/Stop to cutting, copying, pasting, selecting and the heavy majority of functions throughout the program(s).

Some of the DAWs if not all have the option of assigning preferred shortcuts to specific functions. The shortcuts can often be found from the menu bar or by searching the help section in the relevant program.

Working to a Grid:

When working to a click (ensuring the project has been set to the right tempo & time signature) it is important to capture as rhythmically tight and accurate as possible. This is to capture the best sounding performance overall and to make the editing process easier and more accurate. The tighter the audio recorded to the click, the easier the editing. It is therefore harder to edit live recordings in terms of copying and pasting parts (for example), as the recorded audio will vary slightly throughout due to human feel and inconsistency. When working with competent musicians and performers this becomes less of an issue.

Nudge:

Particularly when quantizing audio or just trying to adjust the overall groove of parts/cuts; nudge is a good method.

By using the relevant shortcuts, audio/MIDI parts can be nudged left or right by a small (set) amount such as 1 millisecond to adjust where they sit in conjunction to the grid and or click. So when quantizing audio certain parts can be nudged into the right place if there are inconsistencies in the resulting quantize.

Transport Bar:

Featured above is the transport bar from Sonar 8 Producer Edition. Most of (if not all) the software DAWs available have a transport bar.

The function of the transport bar is to allow the user to quickly and easily navigate the project throughout the recording editing and mixing process .It also often features some key project defining settings such as tempo, time signature, jump to or set markers and set and toggle loop features. On the right of the Sonar 8 transport bar above you can also see the CPU and Hard disk usage to monitor how demanding the project is on the computer you are working.

Copy & Paste:

Some songs can and have been constructed and arranged entirely through the process of copying and pasting repeated sections. It is a good way therefore to use a good first chorus acoustic guitar track  for every chorus throughout the song. It is important to be aware when copying and pasting heavily of the repetitive nature the song will have; this may be a technique you would use carefully, ensuring all the cuts are appropriately faded in and out where appropriate (as to not make it to obvious to the listener).

In a recording situation, given enough time you should always aim to get the best takes of each part of the song rather than copying and pasting, however it is a good technique to save time and can deliver good results.

Drum Replacement:

There are various features within each of the software DAWs available integrated for the purpose of analyzing audio, mapping the transients of files and either cutting them up, quantizing them or adjusting each of the transients by stretching them all accordingly. This includes the Audio-Snap Palette in Sonar, Beat Detective in Pro Tools and Flex Time in Logic.

When using sonar’s audio snap palette to replace drums you would do the following: Select the audio file, enable the audio snap feature (this will map the transients at the default setting), adjust threshold to reduce unwanted transients as best possible (usually about 70% is good), then select all enabled transients, Add them to the pool, copy as MIDI notes, Insert a MIDI channel, and paste the notes onto the new channel. The MIDI notes can then be adjusted accordingly or used to trigger a drum plug in or a sample.

The pasted MIDI notes should always be checked for inconsistencies and compared to the audio track the information was copied from and to make sure they are rhythmically correct.

Quantizing:

When using sonar’s audio snap palette to Quantize you would do the following: Select the audio file, enable the audio snap feature (this will map the transients at the default setting), adjust threshold to reduce unwanted transients as best possible (usually about 70% is good), then select all enabled transients, cut audio at transient markers, quantize to the relevant note value and cross fade between clips at about 2 millisecond intervals.

When quantizing multiple files such as multiple drum microphones it is common practice to first select the Bass drum and Snare drum audio (as they are the most predominant hits and often this technique works well when adjusting inconsistencies) and enable the audio snap feature. Then adjust the threshold, select the enabled transients and add them to the pool. Once this has been done, the you can simply highlight all of the drum microphones and cut them based on the transients you have just added to the pool. The cuts can then be quantized to the desired note value and cross faded at 2 milliseconds.

Fades & Cross Fades:

It is always important when cutting and editing audio to ensure all clips are faded in or out and cross faded accordingly. This is to prevent any clips, peaks and to assure the audio progresses smoothly and consistently.

When fading clips it is always best to fade to the nearest 0° crossing point or the beginning of a sound cycle; to furthermore assure a consistently smooth and flowing progression from one audio file/take to the next.

Above: The various curves that can be used to fade audio clips in and out at the desired speed and duration.

Above: The various cross fades that can be used to achieve the desired transition.

Bouncing:

Particularly when Comping audio it is often good practice to commit to editing work and bounce separate cuts into on new edited audio file. This is often to keep the project as simple as possible in terms of separate takes of audio and also to ensure a confident progression of work. It is important therefore when using this technique to make sure all cuts and transitions are good before bouncing files (so you don’t have to do it all over again if something is not right).

Normalizing:

Normalizing something increases the overall gain of it to the 0dB (based on the loudest bit). It is a good method of boosting the initial signal if it is weak. Some producers tend to stay away from normalizing as they feel it can be destructive to the audio. It is therefore important to ensure good signal levels when initially recording to avoid the need to use this feature.

Additional Plug-Ins:

There are a wide range of third party plug ins available; for example: East West Symphonic Orchestra is a good multi-channel plug in that features high quality orchestral samples and can be trigger via the MIDI information configured on a MIDI track accordingly. It is often a good tool among the wide range of varying third party plug ins; to help define a greater depth in overall tonality and harmony.

There are a range of different microphone techniques that could be used within a recording session as well as different microphone types and polar patterns. In a recording situation, the microphone choice, setting and placement would be based on the environment you were recording in, the nature of the sound or timbre you were hoping to achieve and the overall depth or clarity of the signal.

Further away from the sound source in a reflective environment (for example) would allow more reflections of the room to be picked up by the microphone and would give a more natural sound, where as close to the sound source will usually deliver a sound with reduced room reflections and a stronger, clearer, more up-front signal. By changing the polar pattern you can control from which direction the microphone will pick up the signal source and/or reflections/room sound.

Microphone Types:

Listed below are the most commonly used types of microphones (Particularly the dynamic and condenser (possibly due to affordability), as well as a featured make and model.

Dynamic (Shure SM58)

Condenser (AKG 414B)

Ribbon (Nady RSM2)

The Difference:

When choosing a microphone for a recording task; it is important to chose the right microphone diaphragm size. A large diaphragm microphone will react to the lower frequencies of the hearing spectrum more precisely, whereas a small diaphragm microphone (due to its size) and nature will often deliver a slightly narrower frequency spectrum accenting the clarity of the recording source.

Dynamic:

The dynamic microphone is the most robust and durable design of microphone. Its operating principle is based around a light metal coil that vibrates between a magnet when sound is introduced; translating the vibrations of the coil into fluctuations of voltage.

It is not as quick to respond as the condenser of ribbon microphone due to the nature of the mechanism, therefore the transients of a waveform cannot be as accurately represented by the vibrations of the coil and the voltage fluctuations of a dynamic microphone. This means that you will achieve a less-harsh sound in using a dynamic microphone often warmer in overall tone, where as a condenser would deliver a more natural sound with increased fidelity. Background noise or breathing sound for example would be much more easily picked up by the microphone. This also implies that dynamic microphones are better at handling high pressure sound waves such as the inside of a kick drum when beaten.

Condenser:

The Condenser Microphone operating principle is based around a thin polymer membrane mounted on a back-plate; with the freedom to resonate as sound is introduced thus creating fluctuations in voltage (changes in resistance) between the membrane and back-plate resulting in an electrical transduction of the physical vibrations.

They tend to be  fragile and simply dropping the microphone can be enough to break it. Over time dust can reduce the sound quality of condenser microphones, therefore it is often good practice to keep them in closed padded flight cases to preserve them.

A Condenser microphone also requires +48V (phantom) power to operate, some have the option of using a battery though it is preferred not to; due to potential hum when in operation.

Ribbon:

The ribbon microphone operates when sound is introduced to a ribbon element. A current is induced at right angles to both the ribbon velocity and magnetic field direction. As the sound wave causes the ribbon to move, the induced current in the ribbon is proportional to the particle velocity in the sound wave.

The ribbon microphone is the most fragile of them all. It can be easily damaged (the ribbon can get stretched) by high pressure sound waves. It is compared to the condenser in terms of the way it can pick up sound and the speed it can react to the transients of waveforms; though it is often praised for its predominance particularly in the high end of the frequency spectrum and its often gritty, rough sound.

Microphones can be placed close, distant, high up, low down, off-axis, inside outside and obviously anywhere/anyway depending on the way you intend to capture the sound source, based on the aspects of it. For example when someone stands in a room whilst somebody beats a kick drum; they would hear all aspects of the drums sound based on the reflective properties of that environment (in various proportions). A microphone on its own does not pick up sound in the same way; due to the fact it’s (usually) a mono signal, it has a different frequency response to our ears and different microphones react at different speeds to the transients of the waveforms (as well as the speed of our ears).

Recording Vocals:

When recording vocals; a certain distance from the microphone (depending on the microphone) should be maintained; to avoid any “proximity effect” issues. This applies particularly when using condenser microphones. It is common place to use a pop filter to avoid any “pop” (pronunciation) issues. This can be avoided further as well as sibilance issues by placing the microphone off-axis to the singer and achieve a less direct signal.

Pictured below are the main polar patterns to which most microphones operate (From Left: Cardioid, Hyper-Cardioid, Bi-Directional/Figure of Eight and Omni-Directional. The majority of microphones are fixed with a specific polar pattern (the most common being Cardioid) though some microphones (such as the AKG 414B) have switchable polar patterns to suit the desired application.

The central point of the horizontal line in the middle of the circle represents where the microphone is situated (facing upwards -plan view).

Therefore if a microphone has a Cardioid polar pattern it will pick up sound mainly form the front (the point at which the signal is strongest; assuming the sound source is central to the microphone), gradually reducing the sound picked up from the sides and only a small amount of sound would be picked up from behind the microphone. If a microphone has an Omni-Directional polar pattern; it will pick up sound equally from all directions.

Although there are a wide range of stereo microphone techniques you can implement in recording sessions; those listed below are the tried and tested techniques, that have been used throughout all kinds of recordings for the last half century.

The various microphone techniques can be used from anything from drum overheads or ambient microphones, to single instruments or even musical ensembles. This is obviously in respect to the sound you hope to achieve based on the characteristics of the microphones as well as the characteristics of the stereo microphone technique itself.

From Left: Coincident Pair, Near Coincident Pair, Spaced Pair

Coincident Pair:

The most clinical microphone that is typically good in MONO and defines an accurate stereo field (in terms of relative position of the sound source in relation to the microphones). The are position at a 90 degree angle to each other, aligned at the tips (diaphragm of the microphones)

This is a level (difference) based technique in terms of the amount of reflections of the room picked up and particularly the position of the sound source in relation to the microphones

Near Coincident Pair

Again the microphones are positioned at a 90 degree angle to each other, this time aligned at the bottom ends of the microphones rather than the tips. It has a less clinical characteristic than the coincident pair but still sounds good in MONO and again gives an accurate stereo field.

As with the coincident pair it is a level based technique, however due to the slight spacing of the microphones there is an increased time difference between the sound reaching each of the microphones.

Spaced Pair:

The spaced pair is a time based technique due to the spacing of the microphones, this is therefore why phasing becomes a greater issue when using this technique. It is common practice to ensure the microphones are no greater spacing than about 3 – 4′ (at equal distance from the sound source) to ensure an accurate stereo field with minimal phase issues.

A more recent stereo recording technique designed to capture sound as accurately as possible to the way which we hear the sound or our ears pick up sound. The polar patterns would be set to omni for this technique and the baffle or divider between the microphones acts as the human head between the ears (microphones).

Some microphones have been integrated into a moulded human heads with accurately shaped ears. This is in attempts to accurately capture sound as we hear it. This technique has been used to create music referred to as Binaural or binaural beats and is also used heavily for relaxation music such as wind and waves. This technique is good at giving the listener a sound very close to what they would actually hear being in the room at the position of the microphones hearing the sound source as it was played or hearing waves in relaxation music as if your actually sitting by the sea.

It is often a good idea to use microphones with a very good low end frequency response and to ensure the baffle is padded to absorb the high frequencies between the microphones and to create a shadow area such as our head between our ears. It is therefore a level based technique on the low frequencies (high frequency absorption of baffle) and a time based technique on the high frequencies (spacing). The result is a very accurate stereo field; captured in a similar way to how our ears pick up sound.

Typically a small diaphragm condenser microphone is pointed at the sounds source at the desired distance and another microphone typically a large diaphragm condenser microphone position at a 90 degree angle. The polar pattern of the large Diaphragm microphone would need to be a bi-directional or figure of eight whereas the small diaphragm microphone would typically be set to Cardioid (though can also be set to Omni or Bi-Directional). The signal from the large diaphragm (Bi-Directional Polar Pattern) microphone would then be split or duplicated in a software DAW and each signal (the original and the duplicate) would be panned left and light. The duplicated signal would then need to have the phase reversed to complete the technique; thus reinforcing the stereo image as seen in the picture below.

This gives an ultra-wide sound to the signal and good control over the amounts of the Mid Microphone to the side microphone signals. This also sounds good in Mono due to the nature of the balanced signals as well as the positioning of microphones in terms of angle, distance from sound source and polar patterns.