Phatchance's Comprehensive Studio Guide

[Phatchance]

Hey guys, I've written this guide for a couple of websites, I figured it would be useful to put up here. Hope someone gets something out of it, feel free to hit me up on myspace/my email/my MSN or whatever.

Phatchance's comprehensive studio guide, V2.0

Figure I might as well impart some of the information I wish I had known when I started recording, do what you will with this information in its entirety; I can make no claims to the absolute accuracy of anything stated, especially when my own opinion and understanding of things changes constantly. The guide has been revised a number of times since its inception, but it's still not flawless, and a lot of the writing is based on personal opinion and therefore is open to debate. All prices refer to rough market value of equipment in Australian Dollars as of 2006.

If you want to record with me, or have me mix a project, I charge very reasonable and flexible rates at my home studio in Sydney, if you want help with anything or just to discuss anything in the guide or let me know something I got wrong hit me up at chance.waters@hotmail.com

Since this guide has gotten quite large I'm adding a table of contents, simply hit ctrl + f and type in the section code to skip straight to the section you might be interested in, type in [home] to return to the table of contents.

[home] TABLE OF CONTENTS

[Part 1] - The Studio

[a] Studio Equipment

[a1] Microphone stands

[a2] Pop-filter

[a3] Headphones

[a3i] Monitor Headphones

[a3ii] HI-FI Headphones

[a4] Microphones

[a4i] Dynamic Microphones

[a4ii] Condensor Microphones

[a5] Preamp

[a6] Soundcards

[a7] Monitors

[a8] Software

[a8i] Sequencer

[a8ii] Plugins

[a8iii] Production programs

[b1] Tips for around the studio [just writing b would make it bold, yo]

[b2] Be aware of your environment

[b3] Record lower

[b4] Low-cut and high pass?

[b5] Microphone placement

[b6] Noise filters and noise gates

[b7] Use what you have properly

[b7i] It's not broken, idiot!

[b8] Microphone technique

[Part 2] – Mixing

[c] An introduction to mixing techniques

[c1] Equalisation

[c1i] Hardware equalisation

[c1ii] Software equalisation

[c1iii] Frequency descriptions

[c2] Compression

[c2i] Threshold

[c2ii] Attack

[c2iii] Release

[c2iv] Ratio

[c2v] Knee

[c2vi] Character and misc. Settings

[d] Panning

[e] Reverb

[e1] Natural Reverb

[e1i] Dealing with Natural Reverb

[e2] Artificial Reverb

[e2i] Pre-delay

[e2ii] Reverb

[e2iii] Equalisation and volume envelopes

[e2iv] Character and misc. settings

[f] Shameless self promotion

PART 1 – THE STUDIO

[a] STUDIO EQUIPMENT

[a1] Microphone stand: (cheapest you will get these is the 30 dollar mark); they are extremely important to your sound, as they assist in creating a relaxed and level delivery. A sturdy Metal Boom Mic. stand is a good investment, especially one with a lot of length for adjustment/placement; these can set you back up to $100.

[a2] Pop-filter: (you can get bad ones, or good ones, go a good one as it's about 10 dollars more, any popfilter with a gooseneck is much easier to use). The Basic function of the pop filter is to stop the percussive bursts of air formed when mouthing a 'p' sound from ruining a recording. A popfilter will also help to reduce 'sibilance' or the harsh hiss associated with vocal s’ and t’ sounds, remember that microphone placement and technique can be just as effective as a popfilter in warding off these two nasty elements. I personally would not suggest any recording without a properly positioned pop filter; remember to use distance and direction, as well as the popfilter and microphone grill to deal with these issues.

[a3] Headphones: There are two central varieties of headphones, Monitor headphones ('Monitor' in music when applied to signal chain elements such as Headphones or Speakers, means accurate - the purpose of 'Monitors' is to create an exceptionally accurate and linear depiction of the sound source. This is necessary in order to create an accurate mix that will translate well to a lot of different systems) and HI-FI headphones, which are designed to sound pleasant to the human ear.

No recording outside of an isolated booth to control room system should be attempted without all musicians and engineers wearing headphones. Using speakers to play your beat or instruments will create spillage and render the final recording hollow and muddy, it can also cause sibilant feedback loops which may totally destroy equipment and damage the human ear.

[a3i] Monitor headphones are an important step in the mixing process, whilst no mix should be laid out completely on headphones; headphones allow an analysis of the sound that goes beyond what can be reproduced on speakers. The mix is only partial though, and it exaggerates the stereo-image of the sound source, so it's better as a late-stage mixing tool. For mixing headphones I would suggest Roland RH-200's, ATH-M40's or any of the high-range Sennheiser Monitors, 555's etc. These can set you back from $130-$2000.

[a3ii] HI-FI headphones are also extremely important, they allow the performer to hear the beat and their vocal mix clearly, without this sound spilling into the microphone, thus the signal is much cleaner, and the HI-FI boosts create more energy, allowing the performer to get more into their delivery. For monitoring (listening) I would suggest HI-FI headphones such as the Roland RH-50's [massive bass response] or any of the cheap Sennheiser range such as 202's [crisp, energetic high-end and very clear bass response]. These headphones will set you back anywhere from 50-100 dollars. One thing to look for is low sound spillage, so purchase closed back headphones which cup the ears comfortably.

[a4] Microphones: There are two main types of microphones used in musical recordings, dynamic microphones and condensor microphones. (There are other specifics based microphones such as Ribbon or Surface pressure microphones; however for the purposes of this guide these can be ignored). Remember that all my recommendations are judgements qualified only by my limited real world experience; you should use your own ears and do your own research, rather than taking my word on things.

[a4i] Dynamic microphones: are non-powered microphones like a Shure SM-58 or SM-57. You can pick up a good dynamic microphone (I would suggest an SM-58 for live vocals [$90-$200 dollars depending on where you go] or an SM-57 for beatboxing/instruments/studio vocals for [$90-$180 dollars depending on where you go]) for much less money than you can a good condensor microphone. There are also high-end Dynamic Microphones such as the Shure SM-58 Beta A; however, if your budget expands into the reaches of high-end Dynamic Microphones, I would suggest the purchase of a Condensor instead. Dynamic microphones are generally better suited for live sound reinforcement projects, rather than the studio environment, particularly as microphones such as the SM-58 are designed to reject particular frequencies.

[a4ii] Condensor Microphones: In terms of condensors and depending on your budget I'm really feeling the Rode NT1-A [$300-$350 dollars] and the Rode NT2-A [$600-$1000dollars] these microphones are very direct in the sound they give, the NT1-A especially will give you a very accurate indication of what you are putting into the Microphone. The NT2-A is much more forgiving, however still provides an accurate replication of the performers delivery. In the higher range of budget, there are also microphones such as the Neumann U-87 or U-89 [$2000-$6000] however for home recording these are most often superfluous expenses, and the money can be better invested in other sections of the signal path, remember a chain is only as strong as its weakest link. Some other brands to look out for are Audio-Technica, AKG & Shure.

The main differences between Dynamic microphones and Condensor microphones is in the pickup particulars Condensors are powered microphones and thus are generally more accurate and detailed. The flipside of this is that bad acoustics mean you will need to sacrifice a portion of your low and high-end frequencies in an attempt to retain a clean signal. Condensors are more accurate and sound 'better' (subjectively). They also require a phantom power supply (except for in the case of a Microphone like the Rode K2, which comes with a dedicated power supply).

Condensor’s are the choice you should make if your budget is a little more accommodating, and if your projects are more serious. However I own/have owned an SM-58. SM-57, SM-58A, Rode NT1-A and Rode NT2-A and each has a purpose and can be used effectively. As a side note there are also live-performance condensor microphones such as the Rode S1.

A Quick Note; There are two types of Condensor Microphone, Tube Circuit and Semiconductor. A Tube Microphone, generally gives a 'warmer' more analogue sound, a semiconductor microphone generally gives a more accurate and less coloured sound. A tube Microphone such as the Rode NTK can be a favourable sound in some instances, particularly with certain acoustic instruments and sung-vocals. For main hip-hop vocals, unless you have a relatively harmonic delivery, I would suggest the cleanest microphone within your price range. It's a matter of personal preference. To give some examples, Art of War recorded their first release on a Rode NT1-A, the Hilltop Hoods also used a Rode NT1-A for Left Foot Right Foot, however I believe Suffa now uses a Rode NTK for many of his vocals and they also work with a Neumann U87, both of which are tube microphones.

Different Microphones will suit different voices, some performers with massive budgets [such as Bono from U2] still record on Dynamic Microphones like SM-58's or SM-57's because they favour certain sounds. The most important thing is to go into a few music stores and really try out different set ups.

[a5] Preamp

The preamp is one of the most important purchases for a studio; it may be the biggest determinant in the end product of a vocalists sound. Your microphone is just a circuit that carries the vibrations of your voice/instrument to your recording system/soundcard. This signal is very weak and thus requires amplification before it reaches the recording or playback source, hence the term preamplification.

There are two main ways to preamplify a microphone signal, dedicated preamplifiers, and invisible mixer microphone preamps. As a general rule unless your mixer is say, a high end mackie or yamaha, the preamps will sound uglier than a dedicated preamp. However, mixers are useful for an array of purposes, especially for preamplifying multiple inputs [Turntables, Vocals, Instrument Microphones, D.I. boxes etc.]. For live performance a mixer is the way to go, for recording a single Microphone at home, a dedicated preamp is probably the best choice, it depends on your means, needs and budget.

With preamps you really get what you pay for, there are a huge number of brands to look out for, comb through catalogues and talk to music store staff to see what fits your budget and specifications. Avalon, Mackie, Neve, Alto, Behringer, Artcessories, Joe Meek and others all do a range of products in different price ranges, if you are going to spend a lot of money anywhere, your preamp is a good place to do so.

Quick Note: if you are running a condensor microphone you will need phantom power, this is a standard on most mixers and preamps but make sure to check. If you have the option do not turn on phantom power if you are using a dynamic Microphone. This can potentially cause a high pitch ringing feedback. As another note, some preamplifiers have 'channel -strip’ functions, such as compression, phase inversion or EQ. These are useful for a clean and safe recording, however remember that every component in a piece of equipment is being paid for somehow, juggle what you are getting with how much you are paying, there is no such thing as a free lunch.

[a6] Soundcard:

Soundcards are very important to your recording quality, they are the base quality at which your computer can input and output sound. As a beautiful added bonus they are also the headroom and clarity at which your computer can transfer other peoples dope music to your speakers, and with a decent subwoofer and a good set of satellites you can shatter windows with your treble and shake the house down with your bass.

My main piece of advice for soundcards; don't buy off shops, buy off the internet, particularly off of eBay (albeit from reputable sellers). you will get much better deals on your products. Also, unless you have weighed up and considered an array of options, I would not recommend buying an m-box2 or other pro-tools based models just to gain pro-tools LE. Pro-tools is a decent program, but it's not worth getting lesser quality equipment for a higher dollar, in an effort to get a limited low-quality version of a great program.

The main brands (I know of to look out for) are Creative, Mackie, ESI and M-audio, though there are a number of other companies that do the same/better stuff for varying prices. The main things to look for are; 24bit 96-192 kHz AD/DA I/O, warranty and good quality connectors. Generally avoid any soundcards with Stereo-mini jack inputs or outputs as this is usually a sign of consumer, rather than studio grade hardware. If you are planning on doing a lot of music production - rather than just recording - Midi I/O can be a bonus, so too is Digital I/O if you plan on working with digital hardware.

Some decent and affordable Models are the M-Audio Delta 44, Delta 66, Audiophile 2496 or Audiophile 192, ESI's Juli@ or MAYA 44. There are also more expensive cards such as Creative's 1212 Emu, M-audio's Delta 1010LT or Mackie’s Onyx 400F or 1200F. What you buy should depend on your needs and the provisions of the card. In the case of a card like the Onyx 400F, good quality preamps are also provided. Whether you need a firewire or a PCI card, any plans to upgrade and your current outboard equipment should determine what you buy.

Quick Note: Make all efforts to avoid the purchase of consumer grade soundcards, such as those in the Creative Audigy range. These soundcards are made for multi-media viewing and gaming, not for the professional production of sound. If you want to use Creative, go with their professional range, not with their consumer grade cards, unless you are only recording as a hobbyist.

[a7] Monitors

As stated Earlier, Monitors in music refer to playback devices designed to accurately replicate the sound source, Monitors have a few objectives in mind, creating an accurate playback with linearity in mind, and creating an exceptionally detailed and extensive soundscape with a wide frequency range.

If the Microphone and Preamp are incredibly important in the recording process, because they are the start of the recording chain, then the monitors are just as important because they are the end of that chain, remember that they are the only part of your recording system that you actually hear. Monitors are so valuable, because they allow you to create an unbiased mix which will translate well to a wide range of systems, if you are mixing on a biased sound system, invariably you will overcompensate, or under judge elements of a mix, and thus your mix will translate poorly to other peoples systems. Unlike HI-FI equipment, the sign of a good monitor is often that it doesn't sound good, pleasantness shouldn't be an objective in the creation of a monitor, but rather absolute accuracy and detail, mixing without monitors is like painting without eyes.

Your monitors will probably be the most expensive purchase you make in constructing a project studio, any monitors under $500 AU, simply put won't possess the quality components required to complete their job accurately. There are a wide spectrum of monitors available in the market place, for the home studio you are most likely to desire nearfield monitors, which will generally resemble decent sized bookshelf speakers, larger fullfield monitors are really only acceptable for a large sized control and mixing room, and as this guide targets home studio users, I'll only talk about some of the options available in the nearfield range.

The cheapest decent sized monitors on the market are the Behringer Truth series, while these monitors are flawed, and won't grant a totally accurate representation of the musical picture, they will still exceed the results you would gain from the use of more, or less expensive HI-FI speakers, expect to pay under $600 dollars a pair. Mackie make some excellent monitors, particularly the HR-624 and HR-824 nearfields, these will set you back between $1500 and $3500. KRK make a set of extremely linear and affordable, if a little ugly sounding monitors in the Rokit range, the KRK-RP6 and RP8 models are both ideal for home studio uses, these will set you back a little under $1000 dollars. There are also a range of Monitors from Adam, M-audio, Event, Audio-Technica, Beyerdynamic, AKG and Alesis, all of which fit into various price ranges and are suitable for a range of environments.

I seriously recommend taking a CD you are familiar with to a store and checking it against a range of monitoring systems, most decent sized stores will provide a listening room with a switchable set of Monitors. DO NOT buy Monitors without listening to them first, you would not spend thousands of dollars on a painting without seeing it first. Do your research and find a pair of monitors that work well to your ears, remember, don't go for the pair that sounds best, but for the pair that are the most revealing and most accurate.

[a8] Software

[a8i] Sequencer

Your sequencer [or DAW; Digital Audio Workstation] is the heart of your studio, it is the platform through which you record, mix and master your end product. The power of the most humble modern sequencer outweighs the best hardware studios of less than 10 years ago; never underestimate what can be achieved in a home bedroom with only a few thousand dollars worth of equipment behind you.

Now for some programs; most of these programs can be downloaded with cracks off of Limewire, Bittorrent or any other P2P software. Be wary of viruses, do your research. Anything with the name H20 or AIR attached to it is reliable and will work effectively. I strongly encourage the purchase of any programs that you use a lot, the technical-support, warranty, manuals and legality is really worth the money, that is, if you have it to spend. If not, don't let a lack of money stop you from making good quality music. Always remember that supporting music companies leads to the development of good quality, progressive, reliable equipment and more frequent updates.

Adobe Audition [The recent version of what used to be 'Cooledit' - Personally I dislike this program, the awkward controls and non intuitive menus make it difficult to utilise the advanced and more complex features, and the default plugins aren't flash hot. It has an easy learning curve, is a small download, and is decent as a starter program, but serious musicians who want to put out a solid release should invest the time into learning a bulkier piece of software.

Acid Pro - This program is specifically designed for working with loops, whilst it's more than capable of handling ghetto-recordings and even decent mixtapes, it's directed towards the chopping and mapping of loops. If you already use it for production then familiarity with the interface might make it a viable solution for recording and mixing, but otherwise there are better alternatives on the market.

Soundforge/Wavelab - These programs are specifically designed for the creation and alteration of samples, chopping etc. and for mastering completed works. They are effective for their purpose and are good for dealing with fine details, but I have found that using a heftier program such as Cubase/Logic/Protools will provide the same features in a similar interface, and will make the mixing of larger projects much easier.

Cubase SX - [Cubase 4 as of 2007 - there are also LE/SE versions, SE is quite close to SX in terms of functionality, but LE should probably be avoided] - This sequencer is my personal favourite. The default filters are quality, it supports a seemingly infinite number of channels, limited only by your processors capabilities, the controls are intuitive and where you would expect them to be and there are leagues of helpful manuals and tutorials available on the internet. The learning curve is a little steep, and you will really need to invest a fair amount of time before you can harness its full potential, but the rewards are worthwhile. There are virtually an unlimited number of possible functions/expansions; the real limitation is in your personal creativity and understanding.

Pro Tools - This comes in a number of versions HD, LE & M-Powered, depending on the limitations of the software version, it can be very, very powerful, or very, very overrated. I personally have avoided using Protools because of the necessitation of accompanied hardware. It is generally much harder than other software to pirate and generally much more expensive than the other software alternatives. Protools is a viable option if you have a very large budget and can afford the HD version, or are planning on having your work mixed at professional studios where Protools is the house system.

Logic - Logic is intuitive and easy, and if you have enough money to migrate to a powerful Mac, purchase the program and all the plugins and relearn the system, then it's a very viable choice. A lot of well recognised independent and commercial artists use Macintosh systems with Logic, and there really is not much to be said against the program. If Macintosh is your native system already, this is the most viable choice for anyone wanting to produce a serious release on a Mac, however with the release of Intel processors and Mac Windows the Macintosh limitations of the past are quickly disappearing.

If you are already familiar with PC's I will simply say that Cubase/Protools HD/Sonar can do everything Logic can do, and the extra money is probably not worth it. It's a personal choice and a personal preference; I would definitely suggest trying Logic out, it really is a powerhouse.

Sonar - This program is much of a muchness with any of the other high-end sequencers already mentioned. It's capable of managing everything Cubase or Protools is. Sequencer decisions should come down to interface preference and external factors. I chose Cubase, Sonar, Logic or Tools may be perfect for you.

[a8ii] Plugins

Plugins are extremely important; these are essentially software replications of Hardware modules (preamplifiers/compressors/effects units/Equalisers etc.). Whilst you may be limited in what hardware you can afford, with a bit of computer know-how almost every plugin on earth is at your disposal. There are too many on the market to name, there are some free, quality plugins such as the Ozone range, or you can crack or purchase high-grade industry standard plugins like Ozone, Native or Waves. I personally use Waves Diamond Bundle plugins for nearly all my mixing and mastering. I find they offer everything I want to do, however the kind of interface you may wish to use can vary. Every sequencer comes with its own plugins and is capable of running hundreds of compatible plugins, so doing research based on your platform is the best option.

As a personal piece of advice, don't download thousands of plugins just because you can. Read up on what you are getting, get it because you need or want it, and make sure you are familiar with what your plugins are capable of.

[a8iii] Production programs

Reason: This program is targeted towards producers; it possesses a powerful interface operated the same way as a traditional hardware rack. You right click and insert your effects unit/synthesiser/drum machine etc, and even connect it through a mixer to other pieces of hardware. Reason bears a relatively steep learning curve but when mastered can perform nearly any task. I prefer more organic composition techniques, such as through Cubase, however Reason is definitely a viable alternative for beat makers, and can be used in conjunction with other larger sequencers, by itself it lacks third party VST support however.

Fruity loops studio - For beginners this is an excellent program, very easy learning curve and very intuitive. Read a few guides and you will be making decent beats very easily. As a strong suggestion avoid releasing ANYTHING that uses ANY of the default sounds that come with Fruity Loops. Every single sound and effect in the program has been exhausted beyond its reach, and nothing strains my ears more than hearing a repetitive four bar loop of the same high hat. Experiment with using groove quantising, and playing with the velocity and rhythm of your instruments.

Many people are quick to write off fruity loops, but it can handle pretty much any task you throw at it. As with any program, it's what you put in that you get out. If you favour sample based production, then looping based utilities such as Acid will probably be more to your taste, but Fruity Loops is inarguably powerful and intuitive software.

[b1] TIPS FOR AROUND THE STUDIO

[b2] Be aware of your environment

Particularly when using high-end condensor microphones such as the Rode or Neumann series, it is important to take note of your surroundings when you record. With a vocal booth and/or adequate sound treatment you will be able to record in a pretty relaxed manner, but in a normal room using a high-end microphone you will find a lot of noises you had no idea existed. I can offer a couple of temporary solutions to cover these issues over, however be warned that recording with improper acoustics can lead to a hollow sounding final mix and is not advisable for anything CD quality; for ghetto productions these tips will be fine, but at the worst try to eliminate sources of noise and score yourself some blankets, couches or cheap acoustic foam from Clark Rubber to dampen your rooms reverberation. An effective solution is also the 'Reflexion' filter from sE electronics; this will set you back about $600 AU, read up on it.

[b3] Record lower

Record on a lower volume with multiple channels; this will generate a louder output feed from a quieter input feed, in this scenario the louder sounds will usually be the noises amplified (e.g. your voice vs. the dripping of a tap through the next room). Be careful using this technique however, you cannot create additional headroom than already exists in the system, and the slightest timing variations between the tracks will create phase issues.

[b4] low cut and high pass

Utilise the 'low-cut' or 'high-pass' feature on your mixer/preamp, if your preamp or mixer comes with this feature I would suggest using it around perhaps the 50Hz level, this will remove some of your low end 'room rumble' and unless you are Barry White shouldn't destroy to much of your voices natural tone and presence.

If you do not have this feature, you can equalise out your lowest frequencies through a software equaliser, and this should get rid of much of that noise; the very highest frequencies can also pick up room reverberations, so a roll off at around 15 kHz should remove some of this excess room presence. Compensate for these reductions with a small boost in your high-end around 6 kHz and 12 kHz

[b5] Microphone Placement

Use the microphone from closer up with less pre-amplification. I advise against this technique on microphones with significant bass boost (such as the sm-58 which deliberately amplifies bass signals in close quarters), but for the Rode series at least it is generally a decent, if not a little messy technique.

Usually it's advisable to keep a length of roughly nine inches from the microphone when rhyming, but if you're compensating for this distance by increasing your preamplification it is probably best to move closer and cut down on excess preamplification, this should also cut down on the room noise picked up. Be very careful not to overload the capsule and distort the microphone itself, microphones such as the Rode NT2-A provide an attenuation pad, use it!

If you are using a compressor, remember that the signal has already been input when it is being compressed, therefore acoustics are simply compressed or expanded along with the sound, compression should be used to protect your sound from spikes, not to maintain a ridiculously even level whilst amplifying background impurities, be sure to try and fix your problems using the actual gain first and the compression/expansion functions last.

Finally, placing the microphone on axis and recording across the grill as opposed to directly into the capsule, and using different polar patterns if they are provided can yield interesting and effective results.

[b6] Noise filters and Noise gates

Running a 'noise filter' in your software can also remedy much of the problem. This is done by recording a sample of your room noise, then executing a filter using a plugin such as Waves X-noise. Essentially the program lowers the frequency graph in relation to the noise recorded, it can remove some presence and body from your vocals, but in a very noisy room it's sometimes better to compromise on vocal presence than on the cleanliness of the signal.

Noise gates, these work using a form of compression called 'expansion', whereby quiet noises, rather than loud noises, are compressed. By setting a 'noise floor' lower than the quietest words in a phrase, you can often remove room noise without compromising the signal integrity, especially as ambient noise is most noticeable when no one is speaking.

[b7] Use what you have properly

Always read the manual, I know it hurts to have to concede to reading the manual for your equipment and software, but honestly it is important. Try to get a feel for the functions first off through trial and error, however the manual will provide you with a technical explanation of the function beyond what you can actually hear. If instructions can't help you then turn to Google, Google knows everything.

[b7i] It's not broken, idiot!

Almost always if there is no sound going into or coming out of something it is YOUR fault, not the machines. One of your cords is loose or one of your drivers is busted, one of your settings is incorrect or your Phantom power is off. It's probably not physically broken, trust me, I have been through this many, many times. It's easy to write off any major errors as a matter of hardware fault, however often upon detailed inspection the issue can be remedied through a few clicks or insertions.

My suggestion is to always check the signal chain from its origin to its completion; this includes inspecting all cords, knobs and buttons. Secondly, check the settings of your software, and lastly turn to your drivers. It's usually basic things that lead to ridiculous problems; one fault in a chain can bring the entire signal to a standstill. Checking to see if your mixer or computer is receiving volume but not outputting can also save hassle, locating the source is always the best remedy.

[b8] Microphone Technique

Don't swallow the microphone, but don't be afraid of it. Test your rhymes at an easy level, be sure to monitor the sound and have your engineer/yourself adjust the volume levels and compression as you go to make sure everything is right. When you are testing equipment, rhyme the way you would rhyme if you are delivering, Check 1, 2, only goes so far.

Practice your verse a few times first to make sure you are comfortable with it, once it's mixed changing sections is a hassle so if there is a section where you speak with a lisp or a line where the syllables aren't flowing correctly, it's better to fix it prior to any mixing; I guarantee if you don't out of laziness, then six months down the track you will really kick yourself for it.

Become familiar with your voice and your various deliveries. If you are unhappy with your sound on the microphone then change it. Practice changing your emphasis and tone, rather than your voice, you can maintain your integrity and accent without having to compromise your actual sound. Also, post-recording editing can fix up anything you aren't happy with, but this is not as good a solution as becoming comfortable and familiar with how you actually develop your lines. if you feel your delivery isn't punchy enough on a line, try spitting it again as a solo-take, rather than relying on equalisation or compression to fix your errors.

Listen to the advice of the people around you, you don't necessarily have to agree but always take on board advice. At the end of the day it's your verse, your sound and your decision, but often a fresh set of ears can help make the difference in how you and others enjoy your music.

Mark where you are standing and remember how you are delivering; punchins are important in the process of recording, and taking breaks to rest your voice and get some water is important, but be sure to keep going back and listening to the sound of your voice as you go through.

There is no need to deliver an entire verse within a single take, but don't feel the need to punch in every line if you don't have to. Punch in where you run out of breath or where you need more energy in a line, or on lines that are difficult to rhyme coherently. Don’t punch in the for the sake of punching in as often this will sound choppy and unnatural. It is best to record your whole verse on the one day as well, conditions change and while today your throat might be soft, tomorrow it may be harsh and your tone won't match up. It is also important to make sure you are capable and confident of performing your verse without a massive amount of punchins, remove syllables and change structure where necessary, because performance is just as important as your recorded music.

PART 2 - MIXING

[c] AN INTRODUCTION TO MIXING TECHNIQUES

[c1] equalisation

Sound is composed of a wavelength defined by frequencies, and measured in hertz. A high-frequency sound is in the upper reaches of this wave length, and is perceived by the human ear as a treble note. A low-frequency sound is in the lower reaches of this wave length and is perceived by the human ear as a bass-note. The scale of these frequencies is generally defined in music between 5 Hz (the lowest state of rendering on most headphones) and 20 kHz [20,000 Hz] (the highest state of rendering). Whilst the actual frequency of sound extends to 0 Hz and well over 600 kHz, sounds below about35hz and above about 16khz [16,000hz] are incomparable and unrecognisable by the human ear, though it has been argued that humans can 'feel' notes down to about 20hz, and that the harmonic resonation of notes up to about 600,000hz can add depth to music.

Equalisation is one of the most important features of the mixing process. Equalisation is essentially the manipulation of the volume of decided frequencies at any given time or to any channel of a mix. Equalisation is important, because it can be used to correct imbalances in a mix, pull instruments or vocals out of a mix, liven and define certain sounds, polish a song in the mastering stage, cancel out unnecessary frequencies and remove strain off of speaker systems. An entire book could be written on the process of equalisation, so I will just give a brief overview of some general issues/techniques to look out for during mixing, and how an equaliser generally works.

To set up an equaliser, you use either a hardware or software interface to control the boosts and reductions to frequency. In its most basic form, an equaliser is the 'bass' and 'treble' knobs on most speaker systems, by turning up the treble you are increasing the high-range frequencies of the track, by turning it down you are doing the opposite.

[c1i] hardware equalisation

Most mixers and some preamps will come with some basic equalisation functions, unless the preamp/mixer is of extremely good quality, then you probably want to avoid using these in the process of recording, however for some purposes this equalisation will perform the task necessary to your recording. the number of points of manipulation on an equaliser are referred to as 'bands'; on a mixer with a low, medium, and high EQ control, you are presented with a three band equaliser, most mixers will provide you with either a 2 band, 3 band or 5 band interface, for the most part 2 and 3 band interfaces are not suitable for recording purposes, but for managing out-put signals they can be extremely useful.

It's self-explanatory, but in a basic sense to reduce or boost the bass of the out-put or input signal, you will utilise the low-eq knob, the same logic applies to the mid-range and high-range frequencies. To source exactly how the sound is being changed, consult the manual that came with your hardware, or use the internet to find the specifications. Hardware equalisers operates by implicating a pre-designed equalisation curve over these knobs, so you can see exactly what frequencies you are cutting and boosting, to what ratio, by consulting your manual, and by observing how this changes the character and tone of the sound, you can begin to work out what settings might aid various recordings.

[c1ii] software equalisation

Software equalisation is virtually limitless in its implications, and easily outstrips the capabilities of even the most advanced 32 band Hardware EQ strips (the type you see in most big-venue hardware racks). There are various types of Software EQ, but they all work via essentially the same principle as hardware equalisers. For beginners I would suggest getting a hold of a nice paragraphic EQ, so that you can visually see the implications of your actions. There are a number of settings you will need to take note of, that come standard on most software equaliser bands.

Frequency: This selects your base frequency, or frequency at which the strongest impact of your equalisation curve will stand.

Gain: The boost or reduction of frequency volume as a quantity you are going to implicate on the chosen area.

Bandwidth: Sometimes referenced by the symbol Q, this defines the scope of your boost or cuts effect, essentially if your boost is at 3 kHz, it sets how much of the rise will affect 2.9 kHz and 3.1 kHz etc. A lower bandwidth means a more gently curved impact; a higher bandwidth means a sharper curve (affecting less of the surrounding frequencies).

Treating vocals: There are no steadfast rules to treating vocals, as every voice, situation and verse is different, especially in terms of how it meshes with the beat being used, but there are some general directions that can be helpful in ascertaining the right sound for your mix.

[c1iii] Frequency descriptions

0-30hz - this area is an inaudible zone, rolling off frequencies below 30hz can help to relieve some of the strain off your speakers (which try to produce these low-revolution frequencies anyway), whilst not effecting the actual sound of your track.

50 Hz-120 Hz - this is where the main body of your bass sound sits, a soft roll off below 70hz can help to fix any muddiness or rumble caused by your Mic. stand and plosives, without detracting noticably from the tone of all but the deepest voices. Inversely, a slight boost at 120hz can help to bring out the deeper sounds of some voices without too negatively impacting upon the quality of the track.

120-250 Hz & 250-500 Hz - There are two possibilities in terms of how this area will affect your mix, if your recording is very clean and your beat is fairly empty in these frequencies then these areas can be a great source of presence for the human voice, particularly through 150-350 Hz. If, inversely, your recording is busy and the mix centres a lot around these frequencies, then dips in the human voice at either of these two areas (mainly 350-500), can help to create clarity and further define any surrounding instruments.

500 Hz-3 kHz - This is where much of the of the vocal tone sits, small boosts or cuts here can be very important in defining the character of a voice. 500HZ is the characteristic hot zone of 'muddiness', and many voices possess an over exaggerated midrange between 500 and 3 kHz. The way this area should be dealt with depends on the voice, preamp and microphone.

4 kHz-8 kHz - This is the home of both vocal sheen and dreaded sibilance, a boost or cut here can serve very much to establish or reduce the high-end of a voice, and can be a critical decision in determining the tone of a mix. Too little sound here can create a dull, lifeless mix, too much can strain the ears and sound harsh. A trick I use on vocals with a lot of sibilance, is to run a heavy dEsser on the problem zone, then compensate with a midsized boost in the 4-8khz range to bring the brightness of the vocals back up to the mark, without the ringing shriek associated with sibilance. Play this area by ear, but be careful, don't get addicted to the sound of too much high-end, it can translate very harshly.

8khz-12khz - whilst little of the actual vocal tone sits here, small boosts or cuts in this area can add a polish to vocal sounds, and the 12khz region is an extremely important frequency area in the mastering process, be careful not to have too much 12khz within your vocal track, otherwise when the mastering process comes around, your vocals may render the extra sheen gathered from a boost here unworkable. If you wish to remove a little of the higher frequencies around 12khz (this can help in bringing out the presence of high-hats behind the vocals), then compensate dually with a small boost around 8khz to retain presence.

16 kHz + - This area is almost wholly unnecessary to the human ear in terms of vocals, especially after a track has been properly mastered. I'd suggest a sharp roll off of all frequencies above 17 kHz, and a small roll off at 16 kHz, to remove strain from your speakers and help define the high-end instruments in your mix.

Conclusion: All of this is only a rough guide, equalisation is an extremely integral and very important stage in the mixing process, and tracks should be dealt with on a case by case basis. Remember that unless you have the very best monitors, are familiar with their working and are mixing in a relatively dull room, your speakers are probably lying to you, so equalisation on one set of speakers might translate very poorly elsewhere. The key to proper equalisation is to experiment and use common sense, if you are recording a bass player, then you might roll off your frequencies above 600 Hz in order to get rid of string hum, in turn, high hats might need no sound below 500 Hz.

[c2]Compression

Compression is second to none in terms of important processing in the studio environment, it falls in the family of 'dynamics' processing, and relates to the gain, or loudness of a signal. Essentially, compression does what its name implies, it compresses the signal, making the troughs higher and the peaks lower. This is very important for a number of reasons, firstly as a processing tool to make instruments, or the human voice appear relative and even, allowing for an uninterrupted and clean listening experience. It is also important as a final stage, or protective tool called 'limiting', a barrier which stops signals exceeding the capacity of analogue equipment, or exceeding 0dbfs in the digital domain.

Compression consists of a number of settings, all of which are very important in determining both the tonal character, and average signal noise of the sound. These are divided into functions which specify when the compressor should affect and stop affecting the signal, and how exactly it should do so.

[c2i] Threshold: This function determines at what level the compressor should be activated, generally keeping this setting higher than the average noise of the sound is advisable. For limiting it should be stationed at 0, or -0.2db, for all other forms of compression the placement should vary depending on the strength of the sound source.

[c2ii] Attack: this function determines how quickly the compressor begins compressing once a signal has broken the threshold, generally set in MS or milliseconds, a shorter attack means the compressor will clamp down immediately, a longer attack means it will allow initial transients through, this can be important in retaining the character of instruments such as kick drums, which possess both an attack, and decay stage, compressing the attack stage of the sound will flatten the tonal character of the drum.

[c2iii] Release: self-explanatory, this controls how quickly, or slowly the compressor should stop compressing the signal once it has broken this barrier, this to can be very important in retaining, or altering the character of a sound.

[c2iv] Ratio: The ratio by which each decibel exceeding the threshold is compressed, a ratio of 2:1 will yield 1db of additional gain for every 2db of loudness present in the original signal. Limiters should have a very high ratio at 10:1 or above, compression for different instruments should vary substantially, and depend on the sound source, and placement of the threshold.

[c2v] Knee: essentially a group term for the character of a sound, a fast attack with a low threshold and a high ratio is referred to as a hard knee, a slow attack and high threshold with a low ratio is a soft knee. Sometimes this is an option which will group all the determinant factors in a compressor together for ease of use, sometimes it refers to the character of processing, with the hardest knees allowing a rough and pumpy form of lossy distortion to occur, this can sound great in some kinds of music and works well with industrial style drums.

[c2vi] Character and misc. settings: often a compressor will have settings such as smooth, warm or sharp. This determines the tonal characteristics of the compressed sound and works differently with different source material, experiment. Some exclusive plugins such as Waves compressors feature special tonal adjustment algorithms, such as ARC or Auto Release Compression, read up on these individually, they can often do a better job than manual attack and release times.

[d] Panning

There are few things more important in a mix than creating a spatial soundscape, convincing the listener that what they are hearing is a musical piece being played in the space around them, rather than a recording being played through speakers. Auditory space is generally created through a number of deliberate techniques, however it should be noted that things as basic as volume and frequency can give a feeling of depth and direction to music, remember that every action has an equal and opposite reaction, and moving one instrument in a mix shifts the balance of all instruments.

Panning is the most basic spatial technique available to a sound engineer; it is essentially changing the balance of a sounds representation between left and right, or, in surround sound, between left, right, backwards and forwards. Panning is important because it can be used not only to create an artificial sense of direction for the listener, but also to render two instruments with similar frequency ranges more defined. By sending one instrument left and one instrument right, you allow each instrument the focus of an individual ear.

When you pan a sound to centre, 50 percent of the signal is in the left Speaker, and 50 percent is in the right speaker, this creates the auditory illusion of a centralised sound. Thus when you pan 1 left, (represented as L1 on most sequencers) you are really panning 51 percent of the signal to the left speaker, and 49 percent to the right.

If your mix sounds undefined and muddy, try changing the spatial balance of your instruments between left and right, if a solo or occasionally added layer in a mix is burying the vocals, try moving the vocals L10 and the solo R10, watch the increase of clarity from even such a minor directional shift.

[e] Reverb

Reverb is a naturally occurring phenomena harnessed by the human ear to determine the distance and location of a sound or the size and depth of our surroundings. It is easy to discern the difference between the reverberation of a cupboard, bathroom and a full size cathedral, but the subtleties of reverb can also be very important in shaping the listening experience of a song. Providing musical space to an instrument helps to define and excite a mix, as well as to create a sonic listening field. If panning denotes the direction of an instrument, reverb provides the ear with its environment and distance.

[e1] Natural Reverb: Every space on earth has some form of reverberation, the natural reverberations of a great sounding room can magically transform dull and lifeless performance into a moving and airy experience, but reverb can also wash out and destroy the clarity of a great piece of music.

Natural reverb effects the process of recording in two major ways, firstly during the recording stage, when the ‘wet’ reflections of the room permanently infuse the vocals with the reflections of the surrounding area, and secondly during the mixing stage, when the reflections of the room change the spectral balance of the monitored sound. Both of these stages have permanent implications on the way a track is mixed, for better or worse.

[e1i] Treating Natural Reverb: Acoustically treating a room can be as simple as filling the space with soft objects such as sofas, beds, blankets, cushions and curtains, or as in-depth as the professional installation of floating floors, double glaze windows, acoustic foams, isolation booths, bass traps and acoustic drapes. Whatever the purposes of your studio, and whatever your budget, some form of acoustic treatment is a necessary and important step in creating a studio environment.

[e2] Artificial Reverb: Artificial reverb is a mixing tool utilised to provide space and depth to a recording, the plugin or hardware mimics real world reverberation by processing the original signal, creating an increasing number of blurred, stereo-divided and equalised delays of the original signal along a fading envelope. Artificial reverb is one of the most processor intensive effects available, so clever channel grouping, freezing and use of FX sends should be utilised wherever possible, particularly on slower systems.

[e2i] Pre-delay: The Pre-delay function determines the amount of time before the earliest and thus loudest signal reflections begin. Measured in MS (milliseconds) it acts as an artificial room size. A shorter pre-delay will create a more ‘boxy’, enclosed sounding reverb, a long pre-delay gives a sense of large, cathedral like space. A very short pre-delay can cause phase issues, so be careful and really listen for the effect and saturation of the Pre-delay and reverb balance.

[e2ii] Reverb: This refers to the later signal reflections in the feed, the reverb setting can be very important in determining the overall balance and tone of the reverberation effect, be sure to listen closely to the balance between the reverb level and the pre-delay level, listening on detailed headphones and monitors with a crisp and articulate high end is very important in determining whether your reverb sounds cavernous or builds the body of a sound.

[e2iii] Equalisation & volume envelopes: More powerful reverb processors will give you multiple envelopes; these filters determine the tonal characteristics of the artificial environment. A more metallic sounding room will give higher frequency reflections, whereas a wood based room dominate the lower midrange. Sharper volume envelopes indicate more ‘full’ and absorptive rooms and materials. Try using these envelopes to good effect, if a mix lacks high end energy; try using reverb with a lot of high end, rather than simply equalising the signal. Experiment and see what works for you, Jeff Buckley’s Hallelujah is a great example of creative reverberations, the guitar and vocals are in two totally different environments, and the vocal reverb features a lot of high-frequency pre-delay, and low-frequency reverb.

[e2iv] Character and misc. settings: A lot of more contemporary reverb processors will allow you a number of ‘engine’ or algorithm choices, this can make a great deal of difference in the tonal character of the reverb, experiment with these by turning the reverb send up very loud and listening to the changes when you flick through your options.

[END]

V 1.0

Added Equipment explanations

Added a section on software

Added a section on plugins

Updated Equipment explanations

Adding a section of studio tips

Updated section of studio tips

Cleaned up the guide

V2.0

Added a section on EQ

Added a section on compression

Cleaned up the guide

Added a table of contents and search ability

Added some shameless self promotion

Added a section on Monitors

Added a section on Panning

Added a section on Reverb

Updated and cleaned up the guide

Cleaned and altered search ability

Peace

[f] Please get at me if you need any help with anything, my email is chance.waters@hotmail.com and I use msn.

If you enjoyed this guide please check out my crew and my own music at

www.officialnaturalcauses.com

www.officialphatchance.com

As a side note, I record and mix for people in my home studio in Sydney for $20 an hour as a base rate, I'll soon be expanding the operation under the Nurcha records umbrella. I'm happy to bend my rates based on what you can afford as well as the size of your project and your exact needs. While my results will never rival say, studios 301, I can definitely record to the quality necessary for a mixtape, professional CD & radio release or simple internet distribution.

This guide has taken a considerable amount of work, so please if you use parts or all of its contents anywhere on the internet I'd really appreciate it if you gave me credit and supplied a link to my music and my contact email, thank you.

[Skank]

awsomne guide thanks for the info

[deanmaessen]

nice post phatchance! thanks

[ChevChelios]

Wow very good post.

This should be a sticky.

[Cupe]

Awesome post

Stickied

[Tomy]

rad-est-post-est.

[NathanJames]

nice post there, did it take you forever to write it?

[CBG]

bloody awesome. will read tonight

[Fail]

This guy is awsome! Great post!

[Mattye]

Dude..Will admit I didn't really read it, but when i think bout gettin into some production stuff will have a read.

+ Rep