Transcript
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copyright 2011 by Paul Sykes All rights reserved including the right of reproduction in whole or in part in any form first edition 2011 Front cover design Roland Ali Pantin Illustrations Paul Sykes Photos Courtney Blackney All reasonable meausres have been taken for copyright permission of all images. If you are the copyright holder of an image and wish it to be removed, email the website and it will be altered without prejudice. ISBN: TBA
Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
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About the author
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aul Sykes grew up in a musical family. By the age of 13, he bought his first stereo for $800 and within a month knew every spec and detail and what they meant. He became an instant tech head. After seeing a synthesizer concert, he knew that he wanted to spend the rest of his life working with electronic music gear and making music. n 1984, he was one of the first to use MIDI based computer music and as a young teen would spend up to 10 hours a day programming his new DX-7 Synthesizer . It was this early love of synthesizers and technology that led him down a path of being a commercial recording studio owner, jingle writer and producer.
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fter Paul’s school years, he studied mechanical engineering and worked for 9 years in power stations. In 1995 he left engineering to pursue a career in the music industry. He then took up a position at a tertiary music college teaching all aspects of music technology including MIDI and computer music, PA applications and studio recording.
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It is from these years that formed the genesis of this book.
Paul Sykes
aul has a knack for taking complex concepts and explaining them in ways that made sense for his students. More importantly, that the information is of real world value to any working musician. He has over 25 years industry experience and is as dedicated and excited about music technology today as he was when he first began. Paul is also a singer/songwriter and still enjoys performing. aul has a Certificate of Technology in Mechanical Engineering, an Associate Diploma of Mechanical engineering and Advanced Diploma in Contemporary Music.
Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
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preface
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was working at a tertiary college for over 9 years teaching the tech aspects of music. Year after year, I’d see great musicians arrive to study music and literally not know how to plug speakers into an amp. Being this debilitated in the technological aspects of music seriously limits the working musician. It prevents you from setting up equipment properly and having confidence in the sound of any performance you are part of. It prevents you from buying the right equipment. It prevents you from hiring the necessary equipment. It prevents you from recording well. It leaves a lot of your music in the hands of others. And if the person you’ve entrusted is in a similar position, it becomes a very risky situation because the result could be a terrible sound, broken equipment and an audience that thought you were awful.
Case in point...
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ust last weekend I was at my brothers birthday party. Lots of pro musicians there. I got up to sing a few tunes with them and the PA kept dropping out every time I opened my mouth to sing. Then the keyboard player who was trying to work the mixer kept turning my guitar off mid song because he didn’t know which channels had which instrument plugged in. Then when the mic did work it was distorted. This kind of stuff is frustrating, embarrassing and leaves the performer feeling well.... Stupid. Worse still, it’s all too common.
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have been requested many times -usually by establishments such as churches- to have a look over their system and give advice at how it can be improved. Fundamental mistakes are made that simply shouldn’t, not through lack of enthusiasm but through lack of knowledge. inally, there are many books that give you excellent music technology information but frankly, they are TOO deep. This leaves you trolling through countless pages of spec’s and details that simply confuse you even more. This book is deliberately intended to do three things.. 1. Give you the ESSENTIAL information in a clear, simple to understand way that most people straight off the street can understand, 2. Provide a foundation of information that will allow you to understand the more technical aspects should you decide to go deeper into other manuals.
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3. Provide the information necessary to know what to ask. When we don’t know what we don’t know, we won’t even know what questions to ask in the first place.
he GetYaGearOn series has taken literally months and months of hard work by me personally to help you with your music career. And it is based upon nearly 30 years of industry experience and ten years of tertiary teaching experience at a music college. It’s written with a passion to fast track you to knowing all the necessary information that I wish I had when I first started out in music. My sincere desire is that this seriously helps and empowers you with your music experience and that in some small way, I have made a significant contribution to you.
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o if after reading this Quick Mix, you feel that the information has been excellent, informative, clear and engaging, why don’t you go back and get the full 250 page manual. It will be one of the best investments you will ever make for your music career.
Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
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the quick mix sampler
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WHAT THE HECK IS music? Introduction
For now, lets only talk about tuned instruments.
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lose your eyes and imagine you’re sitting down at a piano. Now play just one note and hold it. What does it sound like? It sounds like a piano of course. But we could play the exact same note on a guitar, saxophone or flute and we could tell them apart yet still recognize that its the same note.
So why is that?
Fundamental Frequency- the Note you’re playing
There are two components to any musical note. The Fundamental Frequency and the Harmonic Frequencies. • The fundamental frequency defines the musical note being played. E.g., is it a Middle C or a low G. • The harmonic frequencies define the character of the note. Does it sound like a Sax or a Piano?
Harmonic Frequency- the character of the sound
Another visualization.....
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ou’re walking along the beach on a nice sunny day looking out over the water. On the shoreline are small waves crashing up against the sand and out beyond them, further into the clear blue water is a swell. Not a big one, but you notice that the water is going up and down evenly. There’s a light breeze blowing over the water creating smaller ripples over the surface. So in essence, you’re watching two waves happening at the same time. The first big one and the second smaller ones going over the top of the big one.
Complex waveform- Both fundamental & Harmonic frequencies added as you would normally hear an instrument
Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
WHAT THE HECK IS music?
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nd that’s how frequencies work. The big swell is the fundamental tone. It’s the note you are playing. The smaller ripples are the harmonic frequencies defining the sound of the instrument. In other words, both a guitar and a piano can play exactly the same note (Fundamental frequency- Big wave) yet we know the difference because of the frequencies that are overlaid on top (little waves). Put together, this is known as a complex waveform and this is what we are always hearing.
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p till now, we have just used one frequency to keep it simple for you to understand. But sound and music is a very intricate complex mixing and blending of many sounds - or pressure waves. All directions, all volumes, all frequencies. If we blend them in a pleasant ordered way, we call it music.
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poorly tuned radio or TV. Called white noise because if you take all the colors of the rainbow and add them together, you get white. Another term you may hear is Pink Noise. This will make sense after the next section on the octave. White noise has all frequencies playing at the same energy. Pink noise gives all octaves the same energy. So 20-40 Hz will have the same energy as 10,000-20,000 Hz. Read on.
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The Octave
he octave is a doubling or halving of the frequency. “A” note above middle C has a fundamental frequency of 440Hz. If we want f the frequencies are all random, we call it noise. to play the A note an octave lower, it will have a And that’s not really 100% true. Consider a high- frequency of 220Hz. An octave higher will be 880Hz speed motorbike or a single engine airplane. Even etc. though it fits the noise category, it most certainly has hen you understand that the Octave is a definite ‘note’. The note is being created by the doubling or halving, it makes sense that speed of the pistons in the motor. Yet another proof the physical vibrations creating that sound that pitch is about the number of cycles per second. will be doubling or halving. It’s the physical attributes As the motor is revved, the number of cycles per second increases and the pitch of the motor goes up. of an instrument that defines the vibration.
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White/Pink noise
Example...
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string will vibrate twice as fast if the length is halved. And that is exactly what happens at f we were to take the whole frequency spectrum the 12th fret of a guitar. It is half the length and play it at the same time, we call this White Noise. It’s the hissing sound you hear from a and therefore the string is vibrating exactly twice as fast. The note will be an octave up.
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microphones & speakers Introduction Microphones and speakers use the same principle except they work in reverse.
• The purpose of a microphone is to take the alternating sound pressures from the air, convert them to movement just like the ear drum, then convert that to electricity.
Electricity In
• The purpose of the speaker is to take electricity, convert that to movement which then creates the alternating sound pressures.
and that principle is...
Sound In
Sound Out
Electromotive Force...emf
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he way speakers and microphones work is all based on emf. Emf basically works like this. Run electricity through a wire and a tiny force field is created around the outside of the wire. If we wind the wire around a rod to make a coil, the force field becomes aligned like a magnet having North and South poles. It becomes what is called an Electricity Out electromagnet. Small ones can be found in speakers and microphones. Large ones can be found picking up old cars at the junk yard.
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emember in primary school when we were introduced to magnets. We discovered that Like poles repel and unlike poles attract. The direction of the current determines which end is north and which is south. Reverse the current and the poles are reversed also. Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
microphones & speakers S
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speaker is simply a coil of wire inside a heavy steel permanent magnet. Run the electricity one way and the coil will either suck towards the permanent magnet or repel against it. Reverse the direction of current and the direction of the coil will reverse. Continually alternating the current makes the coil move back and forth. Connect the coil to the middle of a cone and you have a basic speaker which can now pulse the air in front of it.
The microphone works exactly the same way only in reverse. When you take a coil of wire and move a magnet along it, a current is produced in the wire. With a microphone, the coil is moved by the differing air pressure hitting a diaphragm. As the coil moves, a very small current is produced in the wire which we know to be our mic signal.
Diaphragm and voice coil
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microphones & speakers
Power Handling/Power Rating
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This information applies to both speakers and amplifiers. Since we’re talking speakers first, I’ll cover it here in detail now.
hen you look at the back of a speaker/ amplifier or look at amplifier brochures, you will see phrases like, 250RMS into 8 Ohms or maybe 500W PMPO. The power handling capacity of equipment can be VERY misleading and many a person (myself included) have spent way too much money on under-rated equipment because we didn’t know the technical background to Power Ratings. We saw a large number like 1000Watts and assumed that meant a lot of grunt.
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he reason why it can be misleading is because it can be measured many different ways. And unless you understand the way they measured it, the number (e.g. 500w) they give you really means nothing. Traveling 100,000 metres an hour sounds faster than 100 Kmh, but we obviously understand that it is the same because we all know the difference between metres and kilometres. If someone were to simply say they were going 100,000, we’d very quickly recognize that it doesn’t make sense because the statement is incomplete. But that is exactly what we do when we read a power rating without understanding or taking note of the full statement. This is absolutely vital information. Please read over this as many times as you need to fully understand this section.
Lets start at the basics...
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he definition of power is ‘how much energy per second is being used’. Makes sense when you think about it. The louder the music, the more energy is being used every second to make bigger pressures fluctuations in the air. Its called Watts and its symbol is W.
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Speaker advertising using PMPO ets use the beach example again, this time to explain how a major problem arises when talking about the energy of a wave.
f the ocean is dead flat, there will be no wave energy. The bigger the waves become, the more energy the waves have. To confirm this thought, a dead flat sea wont tip ships over.
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ooking at the swell, we notice its moving up and down a total of 2 metres from top of the wave to the bottom. That’s the measurement from the highest peak to the lowest peak. Or, Peak to Peak. However, given that the bottom half of a wave mirrors the top half, we could simply draw an imaginary line from beach level -the middle- and measure only how high the wave goes. From the middle to the top, the distance would be 1 Metre. Lets call that the Peak value.
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ext, we may then think that the top of the wave doesn’t really accurately describe the wave because its only there for a fraction of a second. We could also say that if we were to average the waves, we’d have to add the bottom half
Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
RMS
Peak
of the wave to the top half. This would give us zero (-1+1=0). The waves of an alternating current moving back and forth always equal zero. But clearly, there’s energy in them. So a method for mathematically describing this energy needed to be created. Saving you the math, as a rule of thumb, we may say that this is approximately 0.7 of the peak height. So 1m X 0.7 = 0.7 Metres. We’ll call this RMS. RMS you can consider as the REAL energy of a wave.
Peak to Peak
microphones & speakers
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Three different ways to measure the same wave
OK, lets bring this back to speakers/amps...
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he way an amplifier works is to take the small waves from your input, combine high power to it to make the electrical wave exactly the same only way bigger (hence Amplify-er). The size of the electrical wave in the speaker cable is obviously the amplifiers output.
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ut just like the wave in the ocean, how we measure this will determine how it appears as a number value.
Note too that.... Peak to Peak is also called ‘PMPO’. Peak Music Power Output RMS is also called ‘Continuous Power’.
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ou’ll probably notice how most consumer products are always rated in PMPO. To the general consumer, the number appears bigger and they will think they are getting more than they really are.
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ets recap and use an example to illustrate this. We can measure a wave...........
Peak to Peak. (PMPO) Top of the top wave to the bottom of the bottom wave. Lets pretend your amp is 1000 watts Pk2Pk.
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Peak. Only one half of the wave.
Your amp is 500 watts Peak
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RMS. (Continuous Power) Approximately 0.7 of the peak. 500 X 0.7 = 350W RMS
Your amp is 1000W AND 350W at the same time depending on how it’s been rated!!
But that’s only half of it! Keep reading... Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
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microphones & speakers Microphones - The details Overview
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n engineering terms, when a device converts one energy source to another, it’s called a ‘transducer’. A microphone is called a transducer because it converts fast pressure changes in the air (sound) to movement (back and forth of a diaphragm). Via the coil , this creates a continuously varying electrical current called a ‘signal’. But the electrical current is so small, it’s basically unusable. A microphone preamplifier is used to boost the signal to levels that modern audio equipment can use.
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icrophones can very incredibly because of the diversity of applications. For this reason, many different types, brands and constructions exist.
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hen using a microphone or looking at buying one, these are the things you need to know to make an informed choice. • Polar Pattern • Frequency Response • Construction • SPL • Impedance • Large/Small Diaphragm
Lets go into these in detail.... Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
microphones & speakers
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Polar Pattern
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polar pattern is the direction that the mic picks up from. For example, if the mic picks up from all directions, it is said to be Omnidirectional. If it picks up sound from the front only, it is cardioid.
Omnidirectional
Cardioid
The basic types are :
1. Omnidirectional. This picks up
sound equally all around.
2. Cardioid (pronounced cardy - oid)
or Unidirectional. This is the standard everyday-use mic. It mainly picks up in front and slightly to the sides. This is named such because it is heart shaped.
Null Null
3.Hyper-Cardioid.
Same as Cardioid but a little tighter at the front and picks up a little directly behind. Also called super cardioid.
4. Shotgun
or Boom mic. This is a very direct, in front, narrow pick up pattern. Everyone’s seen them on the TV or during news coverage. In the TV studio, they are the ones that are on booms going around an audience. They look like thin black tubes. When used outside, a fluffy sock is placed around them to minimise wind noise. You may have seen them used during news coverage or sporting events.
Hyper Cardioid Null
Shotgun
Figure-8
5. Figure 8
(or bi-directional). This picks up in front and behind evenly. Can be used for instance where two vocalists face each other with the mic in between or ‘MS Configuration’.
Null
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connecting amps to speakers Phase
Phase simply means a small time delay between two sound sources. Phase is a simple thing to make sure of but makes a dramatic difference and is absolutely essential to get right in your system.
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inor phase shifting is normal in audio. In fact, it’s an essential component to sound. A phase shift can be considered a small time delay between two audio waves as you can see below.
Phase Cancellation
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he problem is not in minor phase shifting but rather in phase cancellation. Phase cancellation is when the second wave is an exact mirror to the first because it’s shifted exactly half a cycle. Looking at the illustration below, you can clearly see if you add the blue wave to the red one, you end up with the yellow line. In other words, both waves perfectly cancelling each other to zero.
A phase shift showing a small time delay Perfect phase cancellation
There are three ways phase problems can occur.... Cables accidentally soldered backwards
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f someone repairs a lead and accidentally wires the pins backwards, the lead is said to be ‘out of phase’ and will put any audio going through it ‘out of phase’. By itself, you wouldn’t notice it BUT if this cable happens to be one half of a stereo signal from a CD player for example, or the output from the mixer to the amp, your audio will be phase cancelled. And because it’s done electrically, it’s perfect cancellation.
Pins soldered internally out of phase
Cable wired backwards
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connecting amps to speakers Speakers connected backwards
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ssuming you are using two speakers, if one speaker has the connections wired backwards, its motion will be the exact reverse to the other. When one speaker is pushing the air in front, the other will be sucking it. And as you know, sound is a pressure wave. The second speaker is cancelling the pressure wave of the first.
more bass at the mixer thinking the system has little. The speakers will be driven even further and the cancellation effect will be even greater. Continue this spiral and eventually you will blow your subs.
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f you have mid/high speakers out of phase, the way to tell is simply stand exactly between them and slowly rock left to right. The sound is very unsettling and disturbing. Almost sickening because ow this cancellation can happen in two human hearing is not used to such phase shifting. places. Somewhere in the signal before the amp using a dodgy lead as you just learned about or after the amp by having the speaker cables o the answer is simple. + to +, - to -. black to black, red to red. And if you think there is a going to the wrong terminals of the speaker as you can see below. lead that has been soldered backwards, simply unscrew the jacks and check to see that the cable cores match on both ends. f phase cancellation is occurring with Sub speakers, you just won’t hear the low frequencies. The waves are so large that near perfect acoustic cancellation can occur. And this can be very dangerous to your system because you will compensate for
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+ Pressure wave getting sucked back and cancelled
Wiring out of phase
+ Phase cancelling system
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connecting AUDIO GEAR
Plugging a stereo jack into a balanced input
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HIS, my friend, is a DOOZEY! I nearly guarantee if you’re a working musician, at least once in your life you, or someone you are working with will do this and you’ll be delighted and very relieved to know the answer whilst your friends are scratching their heads trying to figure it out.
Here is the scenario...
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ou have a CD player or portable music device and you want to plug it into the mixer. Out of the device is a 3.5mm TRS designed for headphones. So you buy the TRS to TRS lead you may use to plug an iPod into your car for example. Then you buy the adapter that steps the size up from the small headphone size to the larger 6.5mm used at the mixer input. All is going well until you plug it in. For some unknown reason there is little music coming out!!
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t all seems weird. There appears to be no vocals, little drums, no bass. The music is echo’y and certainly not what you think it should sound like. So you all scratch your heads, try different holes, different combinations. You plug the lead in and out thinking maybe there’s a loose connection somewhere. Nope. It’s working perfectly.
TRS to TRS lead
Stereo Jack
Stereo input +L
+R
Here is what is happening... Stereo Jack
he input at the mixer is a balanced input. So the Tip is positive, the Ring NEGATIVE and the sleeve a common earth.
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Mono balanced input
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he output of the device is in stereo. The TRS has tip left positive, the ring right positive and a common earth.
Stereo signal adding as per normal
The left is phase cancelling the right.
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connecting AUDIO GEAR
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o what you have done is plug a stereo signal into a mono balanced input. You have connected two POSITIVE signals into one POSITIVE and one NEGATIVE input. The result is your music is being phase cancelled. To be clear, the only way that would work properly is if the left channel of the music is flipped out of phase as it’s plugged in. It is the perfect example of “just ‘‘cos it can plug in there, doesn’t mean it should”. Y-Lead
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he answer is to buy a Y-Lead. This will have a stereo jack on one end for the music player and two mono Phono plugs on the other. Then use two channels of the mixer, one for left channel, one for right.
f this is happening somewhere and the only cabling they have is the TRS-TRS, pull out the input and plug it into the insert hole. When you learn about mixers in a couple of chapters, this will make sense. The mixer will only play the right channel of your audio but this will be a hundred times better than the phase cancelled version.
General mistakes
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Quick Test
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common fault is a speaker wired internally s well as the above, there can be literally out of phase. My brothers car had it from the hundreds of small things that collectively can manufacturers. A brand new car and one of make one unbelievable a dog of a system. One the rear speakers was out of phase. Another venue I church I consulted for, had over 40 things wrong with it, all due to not knowing this information. The result consulted for had a foldback driver was a system that kinda worked but was just a dog. wired out of phase within the Hums, pops, channels not working, foldback speakers foldback box. To test this, connect a louder than others, mics feeding back, keyboards that 9Volt battery across the terminals. were distorting, CD players out of phase. Knowing just If every speaker moves in the same the information presented so far in this book would direction, they’re in phase. If one have eliminated over 90% of these problems. They or more moves the opposite, those were all simple problems. Collectively though, it was speakers are out of phase. You can a sound guys worst nightmare, let alone a dedicated test this on quad boxes, mid/highs, subs, anything! novice which most church sound folk are. Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
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knowing the numbers Introduction
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f you’ve ever looked at pro musical equipment, you’ll notice there seems to be numbers all over it. All the knobs have numbers, the mixer sliders have numbers, the meters all have numbers. Even iTunes has numbers on its equalizer. Some of them have a ‘–’ in front of them and seem to go backwards. Some have a ‘+’ and move forward. It is absolutely essential to know what these numbers are. Knowing this opens up a whole new world of understanding for EVERY piece of electrical audio equipment you will ever use, both analog and digital. Don’t be confused with basic volume controls that go from 1-10.
These are different...
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omehow, after converting the sound to an electrical current, we need to be able to talk about the volume level of the signal in a consistent way that both the user AND the designer of music gear can relate to. dBV, dBu and dBVU are the three main ones. Strap yourself in and take your time because this is the hardest of all the chapters. But this chapter will be wonderfully empowering knowledge once you’ve got your head around it!
OK, Lets Go!...
Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
knowing the numbers
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Line Level
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very musical piece of equipment from mixers n the early years of the telephone’s development, to microphones, keyboards to CD players the engineers decided to create a standard called have a small electrical output, which we call Line Level. Cutting all the unnecessary details, ‘the signal’. No electrical signal, no sound- obvious line level is 0.77 Volts. This then became the default of course. But just as there are thousands of audio measurement standard for audio equipment. It devices, there could be potentially thousands of means that every manufacturer of audio equipment different voltages that could be used if it wasn’t can now build circuits designed for this level of somehow standardized. voltage/current. Anything less than 0.77 volts is negative, anything greater is positive. And it’s from this early standard that all audio signals are still referenced with today.
Imagine that for a second...
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ou purchase a device designed to run at ten volts of audio signal and plug it into a device designed for two and you burn it out. Or the reverse, and the two volt signal is running into something designed for ten and its so quiet, you can’t hear it properly. A total nightmare huh.
Here’s where it starts to get a little tricky so read on carefully.
dBV & dBu
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peed can be measured in Miles per hour and Kilometers per hour, so a car could be doing 62 AND 100 at the same time depending on which scale to use. So too, there are two standards for measuring the audio signal voltage. dBV AND dBu. As I’ll explain soon, the strongest to weakest levels of a signal is too great to represent as linear so a different scale needs to be implemented.
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the mixer The mixer in detail The channel strip Exploring every knob in detail
Preamp
Preamp
Preamp
Preamp
Preamp
Preamp
Preamp
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the mixer Preamp
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f need be, read back over the preamp section of the book. As far as mixers are concerned, the trim or gain control will always be at the top of the mixer. This is the amount of boost or cut the preamp will give the channel. A quiet signal like a mic in front of an Ocarina may need heaps of boost to get the signal to a usable level. A loud input like a synth will need to be trimmed back a little.
Here is how to set the gain properly...
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itting the solo button will turn the master meters into meters just for that channel strip. Make sure the gain starts fully anticlockwise as you see on the left and the volume fader down to protect the whole PA from unexpected he trim pot is one of the most loud noises. Apply the sound to the misunderstood of all mixer channel (keyboard, drum hit, piano functions because to the playing etc) and slowly turn the gain novice it seems to work like a volume until the meters bounce just on zero. control. I’ve heard horrendous Maybe with peaks just slightly above. The trim / gain pot stories of how people have been taught to use them. +5 or so max. Note that adjusting the volume fader So if ANYONE ever tells you to set all the volume of the channel strip does not affect the meters. faders to the same then adjust your mix using the trim control, throw them out the door and hand Continue working through one channel at a time them a copy of this book on the way out. Also, note until every channel is done. that there is NEVER a time where the pre should be used to add distortion. Insert an effect for that. f your mixer doesn’t have solo buttons or master meters, you’ll at least have a red ‘clip’ light on each channel. Adjust the gain until the clip light starts blinking then trim it back a little.
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Master meters
Done...that’s it!.
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the mixer
Here are a few major reasons why... •
•
•
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djusting the preamp to suit the input level will mean that at the volume fader, the VU numbers will be accurate. I.e., 0dB as written next to the fader will be 0VU.
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his also means that regardless of the type of sound being used, even the quietest to the loudest of signals will have every volume fader with be same volume. For instance, if the GAIN pots were all set the same, then the level at the volume faders would be all different, based upon the input level. A quiet sound would have the fader pushed right up to try and get sound out of. A loud input may have the volume just a centimetre or two from the bottom and already too loud for the rest of the mix. You would effectively be mixing with a 2cm fader length. Very difficult and inaccurate. Looking across the faders, they would be all over the place with no accurate visual representation of the loudness of things. By having the gain set properly, the volumes of the channels will be at the same level when all faders are physically in line. If a channel fader is physically higher than the ones around it, it will be louder.
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leanest sound possible. This is an entry into gain structure which we’ll talk about soon. Consider that every analog audio circuit has a noise floor (hiss), a workable range and a distortion level. If you turn an analog circuit up loud enough, you will hear the hiss. So if the gain of every channel is set too low, you will have to boost EVERY channel across the mixer to get a respectable level at the end. In the process, every channel will be adding hiss to your final two channel mix. Multiple channels doing this can significantly reduce your sonic quality. On the other hand, if you run signals too hot into the channels, you will get clipping. Compound this across every channel and the result will be a distorted PA. Very uncool.
Setting the gain properly is the mid point between hiss on one hand and distortion on the other. In other words, 0VU will create the cleanest, most efficient mixer possible. Many mixers will also have a couple of other features within the preamp circuitry.
20dB Pad
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here are two ways to cut the input of a signal. The first is to have a pot that sweeps the full range from least to most gain. The other way is to have a button that cuts by around 20dB and a knob that doesn’t sweep full range. This is called a Pad button. Its like changing gears. So any loudish type signals like a keyboard, kick drum etc, will serve you well to hit the pad first then sweep the gain up 26 dB pad button until the meters are bouncing around 0dB. If you have nearly swept all around and the signal is still too quiet, turn the gain back to minimum, turn off the pad and sweep again. You can see in the picture, this particular mixer has a 26dB pad and the gain control has two ranges printed to accommodate the pad in/ out.
80Hz Cut – or HPF
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uilt into the preamp circuit is a bass cut button. It can very quickly clean up a mix even if nothing else is done. It filters the low frequencies of the channel before the signal goes into the rest of the channel strip. 80 Hz cut
Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
the mixer
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n practical terms, here’s how to use it. EVERY Tape/Mic button channel EXCEPT the bass guitar and kick drum should have that pressed in. The exception is n the studio, the traditional use of the mixer if you have another instrument designed for low is two fold. The first is to record all sounds via frequencies. You’ll know the ones. Maybe toms, separate tracks onto a multitrack keyboard bass etc. Play with it and see. recorder. This is called the tracking stage So potentially, a 32 track lot of energy is used in circuitry to generate mixer will allow for 32 individual low frequencies because they are big waves. channel strips to go to 32 channels Cutting these right at the start removes mic of a recorder. rumbles, piano rumbles etc and this significantly improves the performance of your system and cleans Tape return up your mix. t the mixing stage, all the performances are now finished and recorded on the multitrack recorder. We now need to use those same The phase switch channels to bring the 32 tracks from the “tape machine” (or computer) back into the mixer to be ot common but great when you get a mixer mixed down to a two track stereo mix. with one. The phase button turns the At the back of the channel strip will be an input for a balanced signal upsidedown (180 degrees). mic/line AND an input for a tape return. This button For example, you decide to obviously selects which one. mic the top and bottom of a snare drum as we spoke about a couple of chapters ago. The top mic faces down, the bottom mic faces up. This means that as the snare is hit, one mic will be Phantom Power exactly opposite to the other. urns phantom power on Therefore when one wave is or off for condenser mics going up the other will be going and DI boxes. down. Opposite. Out of phase. When you mix these together, the snare suddenly becomes thin and lifeless because the Flip switch bottom mic is cancelling or subtracting from the top mic. Flip the phase on the bottom mic and instantly the snare is fat again because the waves are now added. If you don’t have this button, you need to make a 48V Phantom Power physical lead that is deliberately wired out of phase and patch that between your sound source and the mixer channel.
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Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
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effects and processing The Reverb principle
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everb always has three sound components to it. And fundamentally, its these three that you always use as your starting point.
stage and bounce back to you. This is called an early reflection - sometimes also termed a first reflection.
Late Reflections
Direct sound
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inally, as the sound continues to bounce around ny sound coming directly from the stage the room, wall to wall, ceiling to floor, ceiling to to your ears is a direct sound. No reflection walls etc, the sound becomes more and more component to it. diffused (or mixed together) and this we call the Late Reflections. Late reflections are the long reverb tails we hear in music that we normally associate as being reverb. Early reflections
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he second component of reverb is any sound that bounces just once before hitting the ears. Part of the sound will hit a side wall or floor/ ceiling or it will bounce off the back wall of the
When working with digital reverb, you’ll notice there will always be a direct level, an early reflection level and a late reflection level.
Reverb time / Room Size Direct Early Reflections
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oom size or reverb time is the adjustment to make the reverb effect longer or shorter. Note that if your reverb has a ‘room size’ instead of a simple ‘reverb time’, adjusting this will also determine the shape of the room. Flicking through the different sizes will allow you to hear the differences. It alters both the early and late reflections. Reverb time alters the length of the late reflection.
Type of reverb Late Reflections
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he main types of reverb you will have to choose from is Hall, Room, Chamber and Plate. Obviously, these are modelled on the stuff you read about in the previous section.
Hall showing reverb components Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
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effects and processing Predelay
0% diffusion will sound like discreet repetitions of the sound bouncing based on the size of the room selected. Play with it. You’ll get the idea quickly.
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f you’ve ever been in a hall and heard a slap echo off the back wall, you’ll recognise there can be a delay between the direct and the reflected sound. This is called Pre-delay. A long pre-delay will create Decay a slap back effect whereas shorter pre-delay will just give a little space between the direct sound and the ecay is how fast or slow the reverb tapers off. reverb effect. Many reverbs will give you pre-delay options for both early and late reflections.
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Damping
Reverb EQ
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nd finally Damping. Damping is how the reverb ou will have EQ for the reverb also. Some tails off in a non-linear way. In other words, reverbs allow you to EQ in AND out. A brighter as the reverb tails off, it gets progressively EQ setting will mimic a reverb with hard walls muffled – or damped as would be the case in a room whereas a dull EQ will sound more like a crowded full of people. room partially being absorbed. hat’s pretty much every major parameter you’ll need to know to start working with reverb Diffusion confidently. The how to set up a reverb is out of the scope of this book for obvious reasons. In fact ou can also select the Diffusion of the reverb. a whole other book could be written on how to use 100% diffusion will mean the reverb sound will reverb well. be smooth as if it’s been totally confused and Matrix Reverb mixed together as it has bounced around the room.
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Direct Sound
Volume
Predelays
Reverb time
E.R
Decay shape (damping etc) Reverb level
ER Level
Late reflections
Time
ne final point. Many reverbs use a technique called Matrix reverb. What this essentially does is take a stereo input, sum it, put the mono through the reverb then play the reverb back in stereo. It is therefore not a genuine stereo sound running through the reverb. Some reverb plugins do this too. This is important to know because if you install a reverb in line as an insert, you may lose your stereo field and wonder why. Also, when you know about matrix reverb, it means you can assign just one auxiliary send to the unit. Its being mono’d internally anyway.
All major controls of digital reverb Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
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room treatment
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kay, so you’ve decided to turn the favorite junk room into a studio for producing that all-important hit. You’ve had the cops on your door knocking or worse, the neighbors throwing rocks on your roof just to remind you that you’re not alone in your community. The thing that comes to mind is you need to ‘Sound Proof’ your area. Its the one comment EVERYONE makes when they decided to do some sort of music from home. And its the one phrase EVERYONE mentions to me when asking for advice.. “Hey man, I wanna sound proof my space, any ideas?” And as we head towards the last chapter of this book, I think its fitting to suggest that it sort of summarizes the intent of the whole book. Many of us want to know the information we need but are slightly misguided in the way we think. This leads to mistakes, frustration or the result takes way longer to achieve. In our attempts to research, we get overwhelmed and confused. There is no greater issue to highlight this than ‘Sound Proofing’. ‘Sound Proofing’... how? What to look for. What to know. Even knowing what to research on the net can be hard to find out. Here’s the fast track intro. Everything you need to know to give you the heads-up on what to research in greater depth for your specific requirements.
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irstly, you can’t realistically call your project solation is the rooms ability to keep outside ‘Sound Proofing’. You cant sound PROOF your sounds from coming in and inside sounds from space without spending a small kings ransom. going out to your neighbors. An example of poor The two correct terms you are looking for is isolation would be your extruded aluminium garage Absorption and Isolation. A third you will use is walls. Better isolation would be a high thick concrete Diffusion. Even if you have the cash to sound proof, wall. the above terms are still the ones you are talking about. iffusion is less critical for what you probably want to do but it basically means reflecting the bsorption is how well you can absorb the sound in a controlled but somewhat confused sound to minimize reflections and stop way rather than as a direct reflection. Concert halls the sound from bouncing round and round Like the Sydney Opera house utilize this concept. within the room. An example of a room with poor absorption is your bathroom or indoor swimming pool. Better absorption is your lounge or modern movie theatre.
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Inside Sydney opera house showing reflective wooden walls, ceilings and rings creating controlled diffusion Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
room treatment
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Absorption
emember what sound energy really is. Its the alternating of pressure waves moving through the air.
bsorption is all about converting the wave energy to something else as it hits the wall. Rather than allowing that energy to reflect back, absorbent material that is soft and flexible is used to absorb the pressure wave. This will attenuate (reduce) the sound by converting it to a tiny amount of heat.
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he material needs to be porous. As the sound wave moves into the tiny air canals, they cant reflect back out because of the randomizing effect within the material and the air pockets within it. Also, absorbent material needs to be soft. Think of the springs on your car suspension. Converting the energy of the bumps into movement that prevents them from transferring into your ass. An example of excellent absorption characteristics is straw bales. Random pattern, very wide physically and lots of porous holes within the bale structure for a sound pressure wave to go in, and never come out again. It is also for this reason that acoustic foam is … well... foam, and not some other material.
Concrete wall - highly reflective
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bsorption removes the reflective effects of sound – the reverberation. If you’ve ever been to an indoor swimming pool, the combination of reflective windows, concrete walls and the huge reflective water surface all serves to create a loud and annoying experience. Getting your absorption right will reduce the noise within your room significantly. This makes for a much more pleasant playing environment. A room with excellent absorption is said to be a ‘tight’ room.
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ome reflection is actually pleasant if it is for live playing (as opposed to a recording environment). Too ‘dead’ and this artificially induced lack of natural reverberation will feel weird for most players.
Absorbent wall - non-reflective Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
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health and safety Noise
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he irony of this section is that music is all about sound. And everyone loves loud music. We spend our whole week applying the necessary safety precautions to any industrial noise issues THEN go out on the weekend and blast our ears with two nights of clubbing. Or we play in a band and can’t hear properly for the next couple of days. I had a friend who’s iPod was set to the stun position all day, every day. So loud that when she was wearing the headphones they could be heard from different rooms.
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ostly, it’s from people not knowing about noise exposure and it’s effects. I was lucky. I worked in an old power station where the machinery was loud and any high pressure steam leak would scream to intolerable levels. One of my responsibilities was to assess and deal with any noise related issue from a health and safety perspective. Understanding the Australian Standards on noise exposure and the various ways of dealing with it were second nature to us all. But for anyone like my friend who hasn’t worked in industry, this stuff is not known nor is there any opportunity for them to find out. If you’re like her, this section is gold. Here’s the stripped out, lo-down on noise and it’s health effects on the human body..
How the ear works
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he job of the human ear is very similar to a microphone. It takes pressure waves from the air, funnels them down the ear canal where the ear drum moves back and forth. This movement passes through the middle ear which is like a gearbox. The inner ear then converts this movement into an electrical signal that our brains recognize as sound.
Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
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health and safety Threshold shift
Balance
f a sound is too loud, the middle ear changes in an attempt to reduce the intensity of sound hitting the inner ear. It’s the ears natural protection mechanism and it is called a threshold shift. If you have ever walked out of a nightclub and can’t tell if the motor of your car is running because you can’t hear it, this is threshold shift in action. The movement from the middle ear is no longer reaching the inner ear because the middle ear has adjusted to compensate for the loud noise. In some cases, it takes days for it to return. When in the studio, it’s especially important to monitor at low volumes to minimize this effect.
he inner ear is responsible for your balance. It has three rings hanging off it each one looking after either the X, Y or Z direction. Called the semi-circular rings the ear fluid goes through these and it’s the movement of this fluid that causes dizziness. Known as your vestibular system, any inner ear issues like infection can upset your balance as you may have discovered.
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140 dB
Continual exposure like this causes permanent hearing damage.
Live rock band
130 dB
Hi-Hat strike Sand blasting Full orchestra
120 dB
Motorcycle Chainsaw
How the inner ear works
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he inner ear looks a little like a snails shell. Inside is covered in microscopic hairs and filled with fluid. There are about 30,000 hairs grouped into clumps of 50-100, each responding to different parts of the frequency spectrum The movement from the middle ear causes the fluid inside to move and it’s this movement of the hairs that creates the electrical signal that we enjoy as sound. As the volume increases, so too does the severity of movement of these hairs. If they move too far, they break. And now you no longer hear that particular frequency ever again.
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100 dB
Average MP3 player
90 dB
Piano practice
80 dB
Kitchen appliances Talking
70 dB 60 dB 50 dB
Quiet office
40 dB 30 dB
Rustle of leaves
Ringing in the ears f the hairs within the inner ear die, they sometimes fall over like trees in a forest. This can cause an electrical short. This shorting now sends electrical signals back to the brain that have nothing to do with the sound from outside your head. This sound is called Tinnitus and can be heard as white noise or an actual tone. For some it’s permanent and literally insanely annoying.
110 dB
20 dB 10 dB 0 dB
Noise levels
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he above is a rough guide to give you an idea of how loud common sounds are. Actual levels are found using a sound level meter.
Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction
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final words
I hope this has been a great resource for you, and an enlightening read. :-)
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kay, so the full Get Ya Gear On course is around 250 pages, literally hundreds of clear diagrams and big high res photo’s of everything. It starts right back at the start and progresses logically, and it is written in plain english for us mere mortals. It’s the one resource you will use time and time again. After buying this, you will have clarity on how to use a mixer, how to set stuff up, how to chat with confidence on music/audio blogs, how to buy gear in an informed way, how to save money, how to protect yourself and your equipment from expensive malfunctions, how to record digitally and so on and so on and so on. The list is endless. But most of all, after downloading this, you’ll be able to really get into making some great music in a renewed way. No more guessing, no more ‘mucking around’ and God forbid, if something does go wrong, you’ll instantly be able to identify what the problem is and think through a work around.
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inally, Get Ya Gear On has been created with the highest integrity and a total obsession to excellence. I have a heartfelt desire to be of great service to thousands of musicians who unlike me, had to discover this stuff by trial and error and years of wasted hours guessing my way around. Seriously, if you are not happy with this book, email me at
[email protected] and I’ll give you a 100% refund. If you don’t find this to be one of the best ‘plain english’ books out there on this subject, I’ll refund every penny.
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e all spent thousands of dollars on music gear. It just makes sense to be able to use it great too.
I’d like to thank you for downloading this book and I wish you all the best in your future career. Sincerely
Paul Sykes Tech head, Musician, Teacher and author of Get Ya Gear On!
Click the link above for instant and easy download through trusted download site www.payloadz.com Get Ya Gear On - with volume 2 FREE Copyright Paul Sykes 2011 www.getyagearon.com. no unauthorized reproduction