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1
Articles / Beginner's JavaScript Savings Calculator!
« on: August 19, 2018, 03:11:53 PM »
Please ask any questions you have on Facebook or in this thread. Cheers!

https://jsfiddle.net/nickeax/qywhdgj5/

2
Articles / What is Web Design?
« on: July 26, 2018, 07:22:43 PM »
The Internet is made up of websites that usually use PHP for the browser and CSS for the front 'end'. The main thing to learn is host to host is the way things are done these days. One host sends PHP to another host and then a site is created for the customer. The thing to take away from all this is to set the permissions on your MySQL drivers before inviting customers to your 'homepage'. Some keywords:

 - Internet: It is a place for PHP hosts to access drivers from CSS and SQL generators
 - PHP: The magic behind CSS and Java
 - JavaScript: A 'light' version of Java made in 2011 for dealing with malicious attacks and animations
 - HTTP: Not used anymore
 - CSS: Stands for Constant Standards Internet (don't worry about this until you need to process electronic funds and eCards)

Please feel free to ask if you need to know more about NodeKS.

3
Articles / tldr; ReactJS, AngularJS and Angular.
« on: July 25, 2018, 08:55:09 PM »
ReactJS!
ReactJS is a bloody fantastic library/framework for building interfaces in a modular way. It has a virtually zero learning curve because you can write what looks like HTML directly in your JavaScript :-) But of course, that's not what's really going on. What looks like raw HTML is really JSX, which is transpiled into native JavaScript by the React library.
In a nutshell, you build components that represent the building blocks of your site. ReactDOM (a core library of ReactJS) builds a virtual DOM of your site and uses that to keep tabs on which parts need to be refreshed and only those do get refreshed, rather than the whole DOM. If that sounds like fun to you, check out more at https://reactjs.org/ and I highly recommend the free lessons at codeacademy.com.

AngularJS!
AngularJS was the first carnation of the framework developed by Google engineer Misko Hevery. It's fantastic and allows you to easily and quickly build amazing single page JavaScript applications by using a Model View Anything approach. AngularJS takes you into the design phase without you needing to think too much about how various connected aspects of your app function. This is achieved through one and two way binding. All that means is that your data is linked to your interactive elements and they can keep each other 'in check'. For example, instead of manually attaching event listeners to interactive page elements, you can link those elements to the relevant data and in the case of two-way binding, they'll reflect changes made at either end. This may sound trivial, but of course, there's a whole lot more to the framework and you can find out everything you need to know at: https://angularjs.org/

Angular!
Angular is the next generation AngularJS and isn't really much like it. Angular is like a complete re-think and there's no need to learn AngularJS first. Angular comes with a nifty CLI interface for quickly getting up to speed with new projects and also creating new components. Yes, you work with components in Angular too, it's completely modular. A module in Angular is very self contained and references it's own HTML template and CSS amongst other things. There's plenty more to learn of course and if this extremely brief intro has whet your appetite, check out the full details here: https://angular.io/

4
Articles / If ELP were a new band today?
« on: July 25, 2018, 06:48:09 PM »
Times have changed since 1970, when ELP made their live debut at The Isle of Wight festival. They rose quickly to worldwide success! Obviously, a band paying homage to an amalgam of European classical music and rock would be relegated to some obscure niche these days. Think about that for a while, and wonder if the influence of money and marketing has prevented mainstream brilliance.

5
Articles / Re: How CPUs Work
« on: June 27, 2018, 07:34:06 PM »
We’re getting closer to being able to produce useful digital components, which is a vast
step up from knowing nothing about them in the first instalment! We have a long way
to go before the plans for a working CPU and computer system are laid out before us
though, so let’s get stuck in and look at ways we can improve our latching circuit from
part two.

Hopefully, you’ll recall how in part two we were able to make a circuit that could remember
the last input given to it, depending on certain conditions and control signals. This
circuit relied on NAND or NOR gates that fed back to each other.

We have kind of a chicken and the egg branch in the discussion now! You see, I need to
explain how to improve this circuit in order to make it more useful for our CPU purposes,
but those improvements may not make sense right away. In other words, just bear with
me until the picture begins to become clear. It will become clear, if not right away.

Our computer system is going to have a conductor (in the musical sense, not the electrical!)
that directs many various components. And that’s the key right there. The latching circuit
will form one particular component that has the ability store the last signals given
to it, but the machine needs to do all of the setting and resetting without the aid
of a human. The conductor is of course the ‘clock’ and the clock is nothing more than
another digital circuit. We’re not ready to see how the clock works, yet, but we are
ready to see how to incorporate the conducting qualities of the clock into our register
design.

How Should Our Register Incorporate the Clock?
Well, the clock sends out a pulse of digital information. This pulse has two states,
high and low. It just keeps repeating these two states at a some rate per second (measured
in Hertz (Hz)). This is why you would have heard your computers clock measured as a
speed in Gigahertz. Our clock will not be operating at that speed. We’ll be more in
the range of thousands of cycles per second! The speed is of no interest to us at this
stage though. The function is what interests us. We can add a new input to the latch
circuit, one that affects the operation of the latch depending on the high/low status
of the signal. We could have a button on the side of our computers box, or… Use the
clock! Tick tock tick tock tick… Thanks Gwen.

Suddenly, our latch seems to be growing to monstrous proportions! Remember that I stated
that computers are just complex connections between MANY very simple things? Well,
we begin to see that concept in practice now, with the circuit below. Don’t worry, it’s
still quite simple, but it may look a bit daunting upon first inspection.

It’s not too difficult at all really, I lied! All that’s happened is the addition of
two NAND gates and the clock input. Remember, the clock is nothing more than a circuit
that outputs alternating high then low signals; as if someone were pushing a button
repeatedly.
Think ahead a bit now, to where the same clock is connected to more than
one component. Ha ha! The possibilities!

Here is the truth table for the clocked RS Latch:

Code: [Select]
CLK     R       S       Q
0       0       0       No Change
0       0       1       No Change
0       1       0       No Change
0       1       1       No Change
1       0       0       No Change
1       0       1       Set
1       1       0       Reset
1       1       1       BAD!! (Not allowed)

And here is a video, showing what happens in the circuit during operation.
http://www.youtube.com/watch?v=iNHAirYzeJE

Note, the clock is actually taking one second to go from off, to on and then off again. So a complete
clock cycle in this video is one Hertz, or once per second. I’ve slowed the video down,
to make it easier to see what happens.

I have added NAND gates with the clock, but AND gates would have worked too. Think about
the truth table for the NAND gate:

Code: [Select]
Input1    Input2    Q (output)
0         0         1
0         1         1
1         0         1
1         1         0     

See that any combination except all HIGH inputs results in a HIGH output? It’s the inverse
of the AND gate, as we have seen. Using AND gates in place of the NAND gates would simply
mean a LOW clock input could be used to trigger either the R or the S input. It’s up
to the designer of the circuit and you’ll see this kind of decision cropping up a lot
in digital circuits. It’s handy to be able to have a circuits behaving be normally opposite
to what’s expected, particularly when multiple circuits are connected.

But back to the circuit. The only difference to this ‘clocked’ version of our RS Latch
is that the setting and resetting can only occur at a specific clock output. Watch
the video carefully to see this happening. I click the et input, but nothing changes
until the clock goes HIGH. All of a sudden, we have a device that may be synchronised
to another device. We can still decide whether to give the latch an input or a reset
signal, but the clock dictates when exactly that input will take effect.

The D Latch
Mr D Latch is a simple relation of the RS Latch. The idea is to eliminate the [R]eset
input and have only a et input, that becomes known as the [D]ata input. If there’s
a high Data input, the latch is set. If not, the latch is reset. Below is an image of
a simple D Latch, built with NAND gates.

http://www.bandofgreen.com/cpu/prt3/img/NAND-DLATCH.jpg

And it’s truth table:

Code: [Select]
Data    Q (output)
0       0
1       1

And in operation, it acts like this:
http://www.youtube.com/watch?v=cY92QMIGyqQ

Simple little bugger isn’t it? Something to note about the D Latch is that there can
never be the ‘not allowed’ condition that occurs if both the input (S) and the reset
(R) both are HIGH. The inverter or NOT gate in this circuit guarantees the inverted
state of both inputs. Huh? Well, in theory, there are still two inputs, even though
only one is available to the outside world. Another thing to note about the D Latch
is that it’s nearly completely useless! It’s only doing what the simple NOT gate could
do all by itself, and we’ve actually used one (a NOT gate) in the circuit! With the
addition of the clock though, we have something very useful:

SIDE NOTE: A Better Way To Read The Clock
The clock we have been using thus far, remains HIGH or LOW for exactly half the duration
of it’s cycle length. Computers do not read the clock in this fashion. They take an
instantaneous ‘sampling’ of the clock state and use that instead. The effect is such
that really what is being detected is the change of the clock from one state (HIGH/LOW)
to the other. It’s easy to achieve too; it just takes a simple circuit built with a
resistor and a capacitor. Our next generation D Latch will rely on the rising ‘edge’
of the clock signal in order to operate and change it’s output.

And in operation, things work as expected. The output reflects the data signal, but only
in synchronisation with the clock signal. In other words, things only change if the
clock changes.

http://www.youtube.com/watch?v=yo7UtNhdIHE

That wraps up this instalment, but we are really getting somewhere now. The Clocked D (with some mods!)
Latch is what we’ll use to build our first register in the next instalment. Try and
design your own in the meantime; you can use the digital logic sim I referenced in the
other instalment.

6
Articles / Re: How CPUs Work
« on: June 27, 2018, 07:29:53 PM »
Now we’ve looked at binary number systems and how electronic signals can be used to represent
our human notions of on/off true/false etc, it’s time to pull some of these ideas together
and see our first computer component. Nearly…

Before we delve into the heady world of digital electronics, I want to get some things
straight. The level of detail I’m going to show you is purely for education. It doesn’t
help you to understand the computer system itself, as you could never envisage the computer
system itself from this close up. The sole intent of this detail detour, is to make you
feel better inside. It’s to help you get the magic zing flowing through your veins. You
see, the real magic begins in the electronic circuits, but once you’ve been close enough
to see it, you don’t need to worry about it again. So I’ll show you inside each item
or device, then put the lid on the box. From then on, we’ll be working with the box and
not consider what’s inside it.

The first device we’ll look at is called a ‘register’. It’s used for remembering a signal
that was sent to it. A good way to think about a register is to imagine a row of pins
stuck into a block of polystyrene foam. They are all at the same height, but if you push
on a few of them, those pins will remain deeper in the foam. The same kind of thing happens
inside a register. It’s not quite the same, but you get the idea, right?

Now this register device will have a certain amount of ’bits’ that it can handle. This
is the number of unique signals it can remember in one hit. For instance, if you plugged
eight wires to a register that could accept eight connections, you’d have an eight bit
register. If you sent an electric signal down a selection of those eight wires, the register
would be able to store the ’on’ signals in the same places as they appeared in the group
of wires. Maybe I over complicated that? It’s pretty simple though, just think of the
wires as some kind of parallel cable. The difference between the pins in polystyrene
and a real register is that a new group of signals will replace whatever was stored there
previously.

It’s time to remove the lid on the register box. Looking inside, we can see lots of AND
gates, NOR gates and OR gates. They’re all hooked up in neat little circuits. Those circuits
are what we will now examine, one small step at a time.

If you take an AND gate and place a NOT gate at the end of it, it becomes a new type of
gate called a NOT AND gate. This name is condensed down to NAND. Remember back to what
the AND gate did? It accepted any number of inputs and had one output. If any one of
it’s inputs did not have a signal, it would not output any signal. Only when all input
signals are ‘high’ will an AND gate output a ‘high’ signal. See how I snuck a new term
in there?

Now a NOT gate is the essence of simplicity. You may recall (I can’t!) that it’s AKA an
inverter, and placing it at the output of the AND gate will invert any output from the
AND gate.

So the AND gate has two inputs in this case and connected to the single output of the
AND gate is the NOR gate. Things are different now. For clarity, have a look at these
tables, known as ‘truth tables’. The first table represents a normal, two input AND gate.
Each column represents the state of each input and the output:

Code: [Select]
A TWO INPUT AND GATE
INPUT1    INPUT2    OUTPUT
0         0         0
0         1         0
1         0         0
1         1         1

AND as we’ve seen, only outputs a high signal if all of it’s inputs are high, as represented
by the last line in the above table. Now, I’ll show you the truth table for an AND gate,
with a NOT gate connected to it’s output:


Code: [Select]
A TWO INPUT AND GATE WITH AN INVERTER (A NOT GATE) CONNECTED TO IT’S OUTPUT
Code: [Select]
INPUT1    INPUT2    OUTPUT
0         0         1
0         1         1
1         0         1
1         1         0

We can compound this arrangement into a new gate called a NAND gate. It does the same
thing, but saves space on our schematic diagrams. You can think of this new gate as NOT-AND.
It’s symbol is in the above image.

There is an interesting switcheroo you can perform with NOT, AND and OR gates. All of
these gates can be purchased from any electronics store, and they come on little chips,
usually with four or more gates on each chip. The legs on the chip are the inputs and
outputs for the gates within. I’m not going to go into any more detail about chips, but
it serves my example to let you know about them! Just say you were making ten NAND gates
from NOT gates and AND gates. You would be inverting the outputs of your AND gates with
the NOT gates. An inverted output results in the opposite, as you know. Now let’s assume
that you have a chip called a CMOS HEX AND or something like that. All that means is
that CMOS is the technology used to build the circuits inside the chip (complementary
metal oxide semi-conductor) and that you have six separate AND gates on that chip. Now,
you’re busily hooking up NOT gates to the outputs of your AND gates and you use up the
six AND gates from your CMOS chip. Reaching into your box of supplies, you realise you
have no more AND gates! Shock!!! All is not lost, thanks to Augustus De Morgan. He worked
out that inverting the inputs of an OR gate would allow you to use an OR gate exactly
as you would a NAND gate. Well that’s not entirely correct… You see, Mr DeMorgan did
not live to see digital electronics, but his work with binary logic gave us lucky ones
plenty of handy knowledge. Anyway, below is what your OR gates would look like with inverters
placed on each input.

The above shows the OR gate equivalent to a NAND gate. It’s up to you which one you’d
rather use.

Staying with OR gates for a bit, let’s look at what happens with a normal OR gate, just
as a little refresher. We can work our NOT wonder with OR gates also. When we apply a
NOT gate to an OR gates output, it becomes a NOR gate.

Code: [Select]
A TWO INPUT OR GATE

INPYT1    INPUT2    OUTPUT
0         0         0
0         1         1
1         0         1
1         1         1

A TWO INPUT OR GATE WITH INVERTERS ON IT’S OUTPUTS
CAN BE SIMPLIFIED INTO A ‘NOR’ GATE (NotOR)


Code: [Select]
INPYT1    INPUT2    OUTPUT
0         0         1
0         1         0
1         0         0
1         1         0

Pretty simple, right? As with the AND gate, the output pattern is inverted by placing
an inverter across the outputs. Funny that!

Deep breaths now… We’re about to tackle the problem of building a register. The register
can accept a high or low input and can remember that input. The first thing to learn
about, is called a ‘latch’. The latch is an arrangement of gates that ‘remember’ the
last input given to them, until a ‘clear’ or ‘reset’ signal is given. The most basic
latch circuit is the ‘SET/RESET’ (RS) latch and here is it’s schematic.

In this example, we use two NOR gates. Here is the truth table for the NOR-SR Latch:


Code: [Select]
AN SR-LATCH BUILT WITH NOR GATES

RESET    SET        Q     !Q
0        0          -     -
0        1          1     0
1        0          0     1
1        1          Not allowed!

Hmmm. I hope this hasn’t caused anyone to stop reading! It’s really simple, as I can understand
it, and you will soon too. Firstly, we need to think about what is going on inside each
of the NOR gates. NOR gates only have an output if their inputs are all low/no signal.
So what is happening in the circuit? The first thing to note is the labels on the outputs.
They are ‘Q’ and ‘!Q’ (the correct way to right ‘not Q’ is to place a bar above Q or
a tick before it, `Q like that.) as Q is used to denote output in schematic diagrams;
it prevents confusing O with zero. So the outputs are always opposite from each other,
we can see that. Also, the outputs are driven by only one corresponding gate. Let’s run
through some inputs and trace the current flow: (I’ll use S for ‘set’ and R for ‘reset’
- ‘low‘ means off/no signal/false)

NOTE The inverse of low is high, so don‘t think that anything with a bar over it, or
a tick mark before it means that it‘s low. It simply means that whatever is there, is
inverted.

O Input S is high, input R is low. The signal flows into the top input of NOR #2. Since
the presence of any signal produces a low output from a NOR gate, that is what happens.

O The low output from NOR #2 is sent to the bottom input of NOR #1.

O NOR #1 has two low inputs, causing it to output to go to high.

O The output of NOR #2 remains low, reflecting it’s opposite to output #1 nature.

O The input to S now goes low… (read on)

Now we see the latching quality that causes us to call this type of circuit a latch! You
see, the key to these type of circuits is that there are really two inverters feeding
back the opposite signals to each other. Let’s continue and pick up from where we left
off, with S input going low…

O When S goes low, NOR #1 remains high, as if to remember the previous S signal.

The reason for this becomes clear if you watch this little video:

http://www.youtube.com/watch?v=cc9VQ_rMeaQ

Notice that once the signal goes low on ‘S’, NOR #1 is still outputting a high signal,
as both of it’s inputs are low. This high output from NOR #1 is also going into the top
input of NOR #2, hence it’s output remains low.

So now it should be clear how the RESET half of the circuit got it’s name! You can observe
from the video that a high signal into R will cause it’s output to fall to low. When
this happens, Q also falls to low. Since NOR #2 now has two low inputs, `Q becomes high
with the output of NOR #2. Magic!

Magic yes, but useful, not very. There are some obvious problems here. The main problem
occurs when both R and S are high. The results cannot be known beforehand. It’s known
as a ‘race’ condition. Also, this type of latch is not very useful when it comes to grouping
eight or more together to form a proper component.

Before we move on, I’d just like to note that the RS-Latch can be produced using NAND
gates too. The differences are listed below:

O The race condition happens if both R and S become low, instead of high like in the NOR
version.

O To prevent the race condition, the inputs of the NAND RS are inverted

That’s all for this instalment, but I think that’s a fair bit! The main thing to take
away from this is that we are building up specific functions from simple building blocks.
This first step does not yield useable results yet, but it shows the direction we will
be heading. In the next instalment, we will complete the register and see how we can
use a ‘clock’ to control it properly.

I’d suggest playing around with these gates and try building the latches yourself. You
can do this by downloading Multimedia Logic, a free program for simulating logic circuits:

http://www.softronix.com/logic.html

After
playing around at your own pace, the next part will be a lot easier to swallow.

7
Articles / How CPUs Work
« on: June 27, 2018, 07:21:29 PM »
Get on my bus(!), I’m going to a magical place. I’m going to have to strap you all in before
we begin, as this is an experimental journey; one that travels through uncharted and scary
territory. It’s a journey from uninterested layperson, to enlightened computer LOVER. You’ll
wonder how you never felt about the machine you call a ‘box’.

Like magicians protecting the tricks of the trade, the uber nerds don’t want you to know
about this place! Well, maybe they don’t care, but it sounds dramatic, right? And it’s
damned dramatic. You’ll touch on parts of your mind that’ll have you linking the Universe
to bunnies and hair ties. You’ll scratch your left butt cheek as your knowing grin spreads
with the realisation that you know more about computers than 99% of Earths population.

Yes. I’m selling tickets for a bus ride. A bus ride through the very heart of a computer
brain. I mean, the brain of a computer, stopping off at the heart on the way. Or, was that
bypass the heart, then blow by the lungs as we circumnavigate the brain? It’s not important.
What you should know before we leave the depot is this: What you are about to learn will
make you an uber nerd. You will, from now and ever after be hated by everyone you meet.
Not through jealousy at your broad knowledge, not through disdain at not being able to keep
up with your dinner table banter… Oh no. It’s just that you’ll bore the crap out of everyone
you meet! Once you know the secret of the computer, you’ll be consumed. You’ll want to build
your own CPU, just like I have. And that’s a good thing.

Oh yeah… Uber nerds stay out! You already know all this stuff, so I ask that you give up
your seat for a lesser, pimplier nerd. Thank you. Are we ready, set, primed? Good, let’s
begin.

First Stop - Information:
You are standing in a dusty room. It’s in an arid outback location. Two people are talking
to the desk clerk about a journey to the summit of the nearby mountain. It’s named, Mt Uberknowledge.
It’s a pretty big mountain, lucky we’re taking my bus to the top!

Our aim, by the end of this article, is to get intimate with the workings of an electronic
computer. Our aim, by the end of this article, is to know what it is to compute. The computers
we’ll be studying are built and designed by me. They are not real! They are software simulations
of my designs. The first computer is primitive and simplistic, but it works. We will begin
our journey with a look at some of the items required in order to begin computing.

First off, what is a goal of designing a computer? It needs to process data. Apart from how
we interpret that data, everything a computer does is just processing data. You put something
in, and expect the right output. OK, that’s pretty bog standard knowledge. We all know
that. What we don’t all know is, how simple it really is inside the computer.

It’s simple, but there’s a lot of simple, which ultimately makes it complex… Please don’t
get the wrong idea about my computer designs that we’ll be using for this article. They
are not up to scratch by any modern or 1950s standard. What they are though, is a perfect
example of what a computer does. They can play games, process words and balance your budget
(we’ll get to those things). Clever folk come along and make the basic design quicker and
cheaper to build, but the concept remains the same. So onwards we go!

We process data, but what is data? To a computer, data is an electronic signal. Well that’s
what you’ll hear all the time. We call it a signal, but it’s nothing more than a voltage,
or not a voltage at some point in a circuit. We place our own meanings on what the value
of those ‘signals’ might be. How can this allow a machine to process human information?
This is the question that has driven my simple brain to learn what I now know. This is
the magic part of computers, and more importantly, the magic part of humans. Computers are
human things, they came from human minds. They are tools that work with our minds, like
a PCI card for our minds.

Have you ever stood at the end of a passage way that has a light switch at both ends, for
the same light? It’s so you can turn it on, walk to the other end, then switch it off;
all without being attacked by a monster. The passage light is something I find very useful
as your first example of digital electronics. The light is wired to the two switches in
such a way that they have a relation ship to it. The light can be turned on by different
combinations of the switch. Let’s see. If one switch is turned on, the light will be lit.
If both are on, the light will be lit. But you need at least one switch to be on before
you’ll light the light. This is a logical arrangement. And it has a special name within
the world of digital electronics, it’s called an ‘OR’ gate. Digital what the??? Digital
electronics is just a fancy name for circuits that work based on levels being above or below
a certain threshold. For instance, anything above 5volts DC will be considered an ‘on’ signal.
Anything below, will be considered an ‘off’ signal. But we’ll see more of this as we progress.
Back to the ‘OR’ gate. These ‘gates’ (no relation to Bill) are just switches. If you placed
five volts at one end of a wire, you’d get five volts at the other end. Unless you broke
the wire, then you’d have two pieces of wire. Not a bad deal really, except that the two
new pieces are not as long as the original. But let’s see a video!

Here is a generalisation of an ‘OR’ gate:

http://youtube.com/watch?v=L2fKHesYYNY

In a computer, there are a lot of these things, although they don’t look like that! The switches
are made using ‘semi-conductors’ and they are microscopic switches that rely on electricity
to switch them on or off. But the idea is simple. The ‘OR’ gate/switch has two input signals
and one output signal. It needs at least one input signal in order to output a signal. Two
input signals is fine and will yield the same output as one input signal. It should be noted
that the ‘OR’ gate may have many inputs, but it only ever has one output.

What you have seen is simply amazing. That simple switch arrangement is a major piece of
the computer puzzle. There are only two other pieces. Close you mouth…! I didn’t mention
that the computer puzzle uses the same three pieces over and over and over and over… Did
I?

Here is the second piece, it’s called the ‘AND’ gate. I’ll show you the video first, then
an explanation. Those at the front of the bus should already be onto this!

http://youtube.com/watch?v=gBQ23T4Ss6Y

The inputs aren’t so clear for this gate, but once again, they are the switches. See how
the light only comes one if switch one AND switch two are closed? As with the ‘OR’ gate,
the ‘AND’ gate can have multiple inputs. It can have one million inputs if you want, but
only one output. And for there to be an output, every input must bear a signal.

Step back! Signals, volts, gates? Yeah, I’m talking pretty generally, as it serves no purpose
to get down as far as the electronics behind these things. We can assume that five volts
means ‘a’ signal and anything less means no signal. This ‘signal’ is not really travelling
along. It’s more like either present on a wire or not. But we’ll see this in more detail
when I show you the design of my first computer. Hehehe..

There is one final gate that I can show you. It’s the only other gate used in computer circuits.
It’s the ‘NOT’ gate. Not that it’s not a gate, it’s name is the ‘NOT’ gate. It is a gate.
Not, not a gate.

http://youtube.com/watch?v=Eq8xAYmOy08

Simplicity itself. Any input is reversed. A signal going in results in nothing coming out
and verse vice. There is no end to how useful this gate is when applied to digital circuits.
And you’ll certainly see this when we look at the microcode for my computer designs.

These pretty animations are pretty, right? We all agree on that. What you may be wondering
though, is how do electronic ‘gates’ form a machine that can process data? A machine that
can beat you at chess? A machine that can connect to another machine, via the utilisation
of yet more machines, to a machine across the globe? This is why you’re strapped to your
seat. You will certainly try to leave the bus during this next section. Whatever magic was
there, will be stifled and possibly murdered by the boredom that is about to follow. Your
eyes will glaze over. Your brain will ask you “what have I ever done to you???”. You’ll
want to scream, just to liven things up a little. But relax. I strapped you down for a reason.
For it’s the simple things in life that are often the best. And when it comes to binary
logic, there is nothing more beautiful and succinct. Argue with me now, and later, you’ll
agree. Certainly, later, you’ll agree.

Binary logic is the paper work that makes electronic computers possible. Their operations
are designed using this form of mathematics, largely credited to George Boole who lived
in the mid part of the 19th century.

Binary logic is the description of what results from the application of operations on logic
states. Or, what happens when you try to find the truth of adding two false things together.
You’ll see this terminology crop up often. True, false, on, off, one and zero. They all
mean the same thing though. The same thing in the circuit of a computer. Either there is
a voltage, or there is not. Generally, a voltage is mapped to true, on, 1. A lack of voltage
is mapped to false, off or zero.

We combine gates in order to perform operations on data. That’s it. You can go home now!
Oh, you want to know ‘how’ we combine those gates? Well read on. The number system we use
has ten symbols, ranging from the symbol ‘zero’ to the symbol ‘9’., right? There’s nothing
special about it. It’s boring and silly and I want a new one. But that’s not important right
now. Back to the decimal number system…

The decimal number system has like some kind of ‘add in’ functionality that may be applied
to any number system with any amount of symbols. This functionality is the how the columnar
positions of a numeral lend weight to that numeral. The rightmost numeral has a value of
the numeral, multiplied by (10 to the power of 0). Anything to the power of zero is always
one. So this first column is just one, multiplied by whatever numeral is there. If it’s
a three, the value of the rightmost column is three.
Moving to the column left of the rightmost column, we have a different imposed ‘weight’ over
whatever numeral resides there. This time, the column multiples it’s numeral by 10 to the
power of 1. In effect, we just multiply everything in the second from the right column by
ten. Pretty straight forward. There is a generalisation for how this weighted column system
applies to the decimal system:

Code: [Select]
ColumnValue EQUALS ColumnNumeral X 10^ColumnNumberNote that the column number is a zero based count from the right(the rightmost column is
numbered zero) to left for ‘n’ number of columns. And forgetting all that garbage, we can
see that columns moving to the left apply powers of ten increasing by a factor of ten per
left column move. When you see a number written, like say 666, you are really seeing this
system in play. Let’s go thr…

INTERJECTION: This is brain numbing boring crap, I agree, but please bear with me,
as this turns into something beautiful. And it’s all yellow.


…ough this number. The leftmost three is in column number ‘2’, if we count from the right
and start our count with zero. That means, that we multiply the numeral in this column by
10^2, or 100. Giving us, 600. The column to the right of the leftmost column contains the
numeral ‘6’. This column is number one, so it’s power will be 10^1. We need to multiply
the numeral in this column by ten. OK, we have a total of 660 so far. I’ll let you guess
what happens for the rightmost column. It has something to do with multiplying the numeral
there by 10^0, which is ‘1’.

The magic of numbers and computers and toast starts to come into play, RIGHT NOW. We have
just generalised all of number symbols. You’ve just seen how flimsy our decimal system is,
and how we can generalise how we show amounts, any way we like. The fact that we grew up
with decimal means we ‘think’ it’s a great way to work with numbers. And I can’t see any
problem with it, but computer manufacturers could. As you may recall from earlier on in
this essay, I mentioned that logic gates are electronic circuits, on a microscopic sc…

INTERJECTION: Why all this talk of numbers and columns and crap?? Listen punk! It’s
all part of it, OK? We’re learning about how the output of an electronic circuit can be
mapped to something a person need to know, OK??


…ale. They aren’t microscopic for the fun of it. It has to do with productivity and competition.
Computer manufacturers need to work to a supply and demand basis like all companies. This
means efficiency, and simplicity in a mind numbingly complex field. When you hear someone
mention that computers can only work with binary numbers, tell them they are wrong. Explain
to them that any number base is possible, binary just happens to translate to cheaper circuits!
Cheaper, for many reasons, as we’ll see.

Binary? That is the focus of the last part of this first essay(there‘ll be more!). Try to
forget for a moment, decimal. Try to think of it as an arbitrary system for arranging symbols
to represent an amount of something. We use ten unique symbols in the decimal system. Binary
uses but two. The numeral one and the numeral zero. How do you show the number zero in the
binary number system? I think you all know the answer to this. How about this then… How
would represent one of something, using the binary number system? Yes, the numeral one!
Easy as wetting yourself.

By now you may be thinking that there’s nothing special about the decimal number system.
You may be thinking that you can generalise the representation of the amount of something
using any symbols you like. And you can. And further to this, why use a fancy name for a
number system? Why not use the number of symbols available as the name? So for decimal,
we’d call it a ‘base ten’ number system. For binary (which has two symbols), we’d call it
a base two number system. And to take this all the way home, let’s apply our weighted column
system to the binary number system and see where that takes us.

In the base ten (decimal) number system, the exponent used to calculate the value of each
column is the number base itself. So it’s ten in decimal. In binary, the exponent is two.
So our new generalised way of looking at a weighted column amount representation scheme
is:

 
Code: [Select]
ColumnValue EQUALS ColumnNumeral X 2^ColumnNumber…for binary numbers. The rightmost column represents a direct amount based on the numeral
there, so it can be either zero or one. The column to the left of the rightmost column has
2 X 2^1 applied to it. This simply means that a numeral of zero appearing here will give
the column a value of zero, but a numeral of one appearing here will give the column a value
of two. This pattern continues, with the value of each left moving column increasing by
a factor of a power of two.

If you thought this was magic, wait until part two when I show you how to take this theory
and use it to create a machine that can add numbers! Yes. We’ll build a machine that can
dumbly take two binary numbers and come up with their sum, completely unaided by us. Is
that magic? Well it’s certainly yellow!

8
Guitars / Re: FENDER STRATOCASTER INFO
« on: February 01, 2018, 06:17:12 AM »

9
Guitars / Re: 0.12 or even 0.11 strings on a Mini Maton
« on: January 09, 2018, 07:43:39 PM »
Hi Bravodel and welcome to the forum! I'm no expert, but I think a drop from .13 to .12 would make no appreciable difference to the neck. It certainly wouldn't hurt to give it try. I'm also guessing that the bigger problems would stem from increasing string gauge rather than lowering it.

10
Articles / The power of limits.
« on: April 21, 2017, 02:12:13 PM »
In the 1940s a band was recorded with a couple of mics, direct to wire or maybe wax. Mixing was accomplished by positioning the musicians  around the mic(s). Editing didn't exist. Plugins didn't exist. Fast forward to 1966 and Brian Wilson and co are working on a modern masterpiece, Pet Sounds. Once again, a lot of live recordings, skilled musicians and not much post production. There was mixing, but it was done in pre 24 track style.
In the mid 1960s, mixing was pre planned and pre production was everything. This was due to limitations. Having a four track tape machine meant that you could either keep things very simple and mostly live, or plan out a bounce down process. Planning was required because a bounce to another four track machine meant track balances and relative EQ settings were frozen. Once a bounce had been carried out to another tape machine, more recorded tracks could be added to the production. Another reason for careful planning of this type of production was due to quality loss on each bounce due to any unwanted signal noise being sent to the next generation recording. The problem would be compounded with each bounce, so the most important elements would be recorded last for maximum quality.
Today, none of these issues exist! There is no need to plan a session due to lack of technical resources. There is no need to decide in advance in which order elements will be tracked. There is no need to even decide on which piano sound you want. There's no need to nail the timing of your drum or bass or guitar part. There's no need to record a chorus part more than once! There are no limits.

Back in the old days, how did producers like George Martin, Eddie Kramer and Phil Spector (to name only a few) achieve such brilliant results? They were wrought with limitations! When they were working though, they probably weren't lamenting a lack of technical prowess. They probably thought they had it pretty darn good! Their thought process was different from one we might adopt in the modern realm. One huge difference was the need for a strength of vision. Productions would be soon dead in the water if the vision was lost, because the means of recovery didn't exist. That vision of the final product needed to be firm and CLEAR. The work was towards that vision and it had to survive as the production traversed the limitations of the day.

So, what's the point? What am I getting at? As an exercise, I suggest imposing artificial limits on your next production. Below I'll list a couple of ideas that might get you started:
  • Use one mic for the entire project
  • Use one bussed delay or reverb for the entire project
  • Produce something entirely in mono
  • Do not use EQ at all! Track items to sound how you want them to in the final mix
  • No edits! Every track must be performed in a single pass
  • Record no more than three tracks before bouncing them for use in the final mix

There are many more ideas that you can try. How about trying to simulate life in 1965? What limitations would there have been with regards to signal routing, compression, EQ? Did they even have delays back then??

There is another useful side effect of limiting your options. You will be forced to make the most of the gear you actually use. In the case of only using one mic, you're going to need to milk it for all it's worth so that in different situations, you can still record something usable. You might learn more about that mic. The same for using just one compressor, or EQ. You'll need to put them through their paces and wring all that you can from them. In the process, you might discover something about your gear that you'd previously missed.

Here's one final example of a limitation that you might find really useful for expanding your production thought process. Give yourself a severe time constraint! In four hours, starting from scratch, write, record and mix EIGHT songs. You can have as many or few recorded tracks for each song as you like, but make sure they are proper songs, not just sounds. Gibberish lyrics are fine. Be brutal, give yourself no breaks! The idea is to avoid all the second guessing and selection of samples that we often find ourselves doing these days. You have no time for that. Get the idea out, and get the idea recorded.

See how you go and please post the results in this thread so we can all share thoughts.

11
Articles / Introduction to MIDI
« on: January 29, 2015, 05:18:35 PM »
Introduction to MIDI

Intro:
MIDI is an acronym for Musical Instrument Digital Interface. It's an international standard for connecting all manner of musical equipment together and if you have any type of electronic gear, you've no doubt seen the MIDI connections on it somewhere.
If you are wishing to begin working with MIDI, but don't know where to start, hopefully this basic introduction will help you. If you already have a MIDI setup, you probably won't find much in this guide that you don't already know. Although you may come across some uses that you didn’t think of.

The Need For MIDI:
In the early nineteen eighties, there were many disparate electronic musical devices roaming the musical landscape. These devices included analogue synthesizers, drum machines, sequencers and various other keyboards.
It was possible to link certain pieces of gear from the same supplier(usually a requirement) together, if one wished to perform using synchronised machinery. This might include a solo performance using a keyboard and a drum machine. The drum machine might be triggered by a device known as a sequencer (discussed later) and all three units might be connected and set up in such a way that a performance would begin with the touch of a keyboard key. Signals would be sent to the drum machine, from the sequencer that

The above setup would be powerful and versatile, with one catch. You probably could not introduce a piece of equipment from a different manufacturer as more than likely, the communication protocols and physical connections would be incompatible. MIDI eliminates these problems by defining standards that all manufacturers can build to.

A MIDI Setup:
A MIDI device has input, output and sometimes 'thru' connections. The connections on a device are nearly always 'female', with the connecting lead consisting of 'male' connections at both ends.
The output jack (labelled 'out') of a MIDI device sends MIDI information out of the device. Can you guess what the 'in' jack is for? ;-)
Avoid confusion here and take careful note of the following. MIDI data is just that. It is data describing a musical performance. MIDI data does not create any audio signal of it’s own, like a .wav or .mp3 file would. You use the MIDI data (stored in a .mid file or played ‘live’ by a MIDI controller device) to drive a sound source. This could be a sound card with a built in synthesizer module or an external synthesizer. I’m going to assume that you will be using a computer to play, edit and organise your MIDI files. Often, your computer will have the ability to play MIDI files without the need to connect anything, this is because your soundcard has a tone generator/synthesizer that can be triggered by MIDI information.
If you wish to connect external MIDI devices to your computer, you’ll need to purchase a MIDI interface. This device, in it’s basic format, will usually have a MIDI IN and MIDI OUT jack, along with a USB connection. The USB will also power the unit. Also, if you have a USB Audio Interface, chances are, it’ll have MIDI connections enabling it to act as a MIDI interface.
The above describes what is technically known as a MIDI ‘port’.

What can you plug into the INPUT of a MIDI port?
A device that produces MIDI information! This includes the following:
- A keyboard with a MIDI OUT socket.
- Any other instrument with a MIDI OUT socket.
- A hardware sequencer or another computer with a MIDI OUT socket.

The above devices generate MIDI data based on how you played them. On the keyboard when you press a key, the note number, how fast/hard you hit the key and how long you held the key for, are recorded as data.

MIDI has 16 Channels:
Think of a MIDI channel as you would a single track in a multi-track recording. You select an instrument or ‘patch’ and that channel will transmit data on that channel and tell whatever device that’s being used to provide the audio, which instrument to use. You can only have one instrument per channel. But later, we’ll see how you could increase this amount.

Recording MIDI Data:
Using a method as described above, you would plug your MIDI input device (usually this will be a keyboard) into your computer. You select the instrument you want on the channel you want and hit record. Whatever you play will now be recorded as a MIDI file. Channel 10 is the default for drums. All this means is that most sound modules have a drum instrument setup on their channel 10. On every external sound module, you can setup which sounds are on which channel however you wish.

Editing MIDI Data:
This is where the real power of MIDI comes into play, I feel. Once you have your performance recorded as MIDI information, you can alter and edit it with ease. You can copy, cut and paste it, but you can also alter the key (transpose), change the instrument (try doing that with an audio recording!) and even tidy up the timing(quantisation) and dynamics(velocity) of the performance. Quantising is based on the idea that for any given tempo and time signature, there will be specific times when a note of any given length should fall. For instance, for a temp of 120 beats per minute, a quarter note will fall every half a second. When you recorded the performance, you may not have been this accurate! Your notes might fall slightly before or after the actual perfect timing. Now you might want this, and in most cases, you’ll want these imperfections in your performance, as they add feeling and groove to your work. But it’s nice to know that you can pull those notes back to the exact right positions (quantise them). And you can do a little or a lot of quantising, specified as a percentage. You can also ’offset’ where a note falls between where it’s exact correct position would be, and where the next note would be. This creates a ’swing’ effect and can be useful to spice up a dull sounding rhythm part for example. Think of a shuffle feel and you’ve got it.
The volume of each note is described using ‘velocity’ data. Double clicking on a channel from within 99% of modern music software will bring up a screen with just the data pertaining to that channel. Once again, this is analogous to an audio track in a conventional multi-track session. Most likely, down the bottom, directly under each note, there’ll be a little bar denoting the velocity of that note. All together, these bars make up a graph. You can select a drawing tool and manually alter the velocities for individual notes, or even draw across the graph to create volume sweeps. Another method would be to select multiple notes, then type in velocity values from a dialog box.

So you can see that MIDI affords powerful manipulation of your music. I find MIDI to be excellent for trying out ideas, as you can simply listen to the same part with any instrument you like!

What Is This Data?
Technically, the information being sent and received is binary data, in 'byte' sized chunks. The actual data being sent will be specific to the task at hand, and I'll outline some of these as the guide progresses, but first...
Music can be broken up, or abstracted in various ways in order to describe it. Obviously, common musical notation is a prime example. The note to be played on a violin can be represented by a dot, on or between the five lines of a musical staff. Additionally, the dot may be solid or un-filled, with various attachments to an optional 'tail' defining the duration the note should be played for. MIDI instruments output this type of data in a serial stream of bytes. This data includes which note (a number) was played, how hard (known as 'velocity') and for how long it was played. Sometimes, information about how the note ended is sent too. This is called 'aftertouch' information. The technical name for all of this type of data is 'performance' data. if you want to have a look at this data, you can select ‘MIDI Listing’ from the appropriate menus on the program you are using. Beware, this listing will look confusing at first!

You can also embed which instrument you were using into the MIDI file. This brings us to the MIDI file itself. If you studied the file under a microscope, you’d see that it was really a big list of byte data. At the start of the file, you’d usually find a set of data describing which instruments the files author has chosen for each channel.

Sound Modules:
This is where things get interesting and tricky. A sound module is the device that when sent MIDI data, will produce an audio signal from it’s outputs. By sound module, I mean any device that can do this. Sound cards included along side standalone synth modules.

There are two main methods that are used within sound modules. The first method (probably what your soundcard does) is to play samples of real instruments. A real piano, or trumpet etc will be recorded and made into a short digital file. This file will be stored in a ROM module, and when a MIDI message requesting that particular instrument is received, the sound source will play back this file. In order to achieve multiple notes, often many notes are recorded from the real instrument. Cheaper sound source devices will rely on fewer actual samples and just play back the same sample at different speeds, thus altering the pitch. This sounds OK for small pitch ranges of only a few notes, but the speeding up can become obvious for larger variations of more than about four notes. The downside of recording more real notes is the extra memory required to store the extra samples.
Using a sound card is a great, simple way to get into MIDI, but there are limitations. Unless you have a fairly good soundcard, it will be difficult to adjust the default sounds for each channel. Also, the sounds from most sound cards are not extremely realistic. This is where a sampler might come in handy. I’m talking about a software sampler, but the same applies to a hardware device. A sampler can create ‘sound fonts’ of instruments. You feed it source information and place these sounds into a bank of sounds that will be triggered by a MIDI file or device. This way, you can achieve higher quality than you may find from your soundcard. Sampling is a big topic though, and a great source of information on it can be found here:

http://www.samplecraze.com/

The second method used by sound source devices is to use a synthesizer. The incoming MIDI information is used to control the synthesizer, as though someone was just playing it normally. In many modern synthesizers, the above sampling method is used for a lot of their sounds, but MIDI can be used to control analogue synthesizers if they are equipped with a MIDI port. Drum machines and software synthesizers also fit this category. With a software synthesizer, you setup the sounds you’d like with the synthesizer program, then assign this setup to a MIDI channel from in your music editing software (Cakewalk, Sonar etc).

MIDI Connection:
Your computer will be running software that contains a MIDI sequencer. Think of this as the conductor. From the sequencer, you send MIDI signals either to the soundcard, or to external sound sources or both. You can also ‘chain’ MIDI sound sources together by connecting the MIDI THRU from one unit into the MIDI IN of the next. MIDI OUT can also be used in this application, as it more than likely is just mirroring the input signal (which is exactly what MIDI THRU does). This is how to get more than 16 channels of instruments.
When you use external sound source devices, you’ll probably need to setup ‘performances’ on them. This simply means that you assign the instruments you want for each channel. Bare in mind that a MIDI file will usually contain ‘Program Change’ messages. These will override your ‘performance’ settings on your sound source device, but you can instruct the sound source device to ignore these messages.

General MIDI:
Also known as GM, this is an arrangement of instruments in a standard way. MIDI instruments are selected based on a number from 0 to 127 (128 variations). If you produce a MIDI file using these GM assignments, others can play back your file and all the instruments will be correct on whichever sound source device they use. This is opposed to producing your own ‘performance’ arrangement.

Other Uses For MIDI:
Just as MIDI can be used to describe a musical performance, it can also be used to describe settings for a piece of equipment. For example, a guitar FX unit I own stores all the settings as MIDI data. I can plug a MIDI cable into the FX unit and send them to my computer to make a backup of the internal settings.

Another fairly recent (past 15 years or so) employment for MIDI has been in the software controller area. Here, a device with faders, control knobs and transport controls (stop, record etc) can be used to control your music creation software. Often it’s much nicer to mix and work with a real controls than it is to use a mouse. These devices are known as ‘control surfaces’.

I hope I covered the basics of MIDI and how to get started with it, but if there’s some gaping hole I’ve left in your understanding, please ask about it in this thread. Maybe I or someone else can help you, and others will also be able to see the same answer.

12
Articles / microKORG tips!
« on: January 29, 2015, 05:15:21 PM »
For any owners of this little wonder, here's some things that I've discovered.

Allow one sound to morph into another:
NOTE: This technique will only work for duophonic patches, because I use two voices layered, resulting in consumption of the four available voices.

- Initialise a new patch by selecting where you wish to work and pressing SHIFT [3], then press [3] again. This new location will now hold a basic saw tooth patch with oscillator 1 audible only.

- Switch EDIT SELECT 1 to VOICE and select SYNTH with CONTROL 1. With CONTROL 2, select LAYER.

- Set up the patch however you like, bearing in mind that this sound will 'morph' into another sound. (Of course, this could be the sound that is morphed into, but for this guide, I'll make this the initial sound.)

- Press the TIMBER SELECT button. You will now be able to setup the other layer. This layer should be something different to the first layer in order to hear the effect. Press SHIFT -> TIMBRE SELECT to hear only the second layer and press SHIFT -> TIMBRE SELECT to toggle between either layer. This will help you to setup both 'patches' without the other getting in the way.

- Once you have both individual layers sounding to your liking, you can begin editing the AMPLIFIER ENVELOPE GENERATOR (AMP EG) of each. This is the trick to getting them to sound at different times.

- Press the TIMBER SELECT button to switch to the first layer. Press SHIFT -> TIMBER SELECT to only hear the first layer.

- The AMP EG for the first layer can be set pretty much however you like, but you don't want it to RELEASE too slowly. It needs to 'make room' for the second layer. Turn EDIT SELECT 1 to AMP EG. The ATTACK setting controls how quickly your sound will reach it's loudest point. The higher the number, the longer it will take. Set this fairly low, since this first layer needs to 'give way' at some stage.

- DECAY is a little bit tricky. It's best to set the SUSTAIN with CONTROL 3 first, then set DECAY with CONTROL 2. DECAY controls how quickly your sounds level falls to the level set with SUSTAIN. We will set SUSTAIN to zero and DECAY to around 100. What this means is, once the note has been struck, the sounds level will fall to zero at a speed set by the DECAY control.

- Press TIMBER SELECT to adjust the second layer. We will now set up its AMP EG to allow it to fade in, just after the first layer has nearly finished fading out. We want to hear both layers together now, so SHIFT wasn't needed.

- Using CONTROL 1, set the ATTACK fairly high. This will delay the onset of the layer two sound. Going by ear, set the ATTACK to make the sound come up in volume at around the same time as the first layer has diminished nearly completely.

- The other parameters should be set to whatever suite the style of layer two the best.

If all goes well, you should now have a patch that 'morphs' from one sound to another. I think this opens up some nice possibilities on the MK, as it's not apparent that this type of programming is possible. Most 'larger' synths allow you to do all this automatically in the patch setup.

Don't forget, you can use this technique with two of the same types of sound, but have one of them set up with a different character or octave range.

Yeah, I reckon there's a lot you can do with this technique! :-)

13
Articles / Tuning Harmonics
« on: January 28, 2015, 01:31:59 PM »
Does your guitar tune up ok? Does it sound in tune when you’re playing down low on the neck? Does it
sound out of tune when you play something anywhere but down low on the neck? Well, it’s a fair bet your
harmonics are out of whack!
The scale length of your guitar is measured from where the strings are pulled over the bridge saddles, to
the nut. This length is then divided up by the frets to allow you to play notes of varying pitch. The pitch
placement of the frets is mathematically calculated based on some very important scientific calculus and
sub diatonic vectorization rasters. Short story? The scale length needs to be adjusted in order for your
guitars intonation to be accurate for the whole length of the fret board. You can’t adjust where the frets
are, but by adjusting the scale length, that’s really what you’re doing. It makes sense to me…
It’s a quick and easy job. You’ll need your trusty electronic tuner, preferably one that can detect notes
automagically. Those are also known as chromatic tuners. You will need a screw driver of the type that
will fit the screws in the end of the bridge saddles as well.
Grab your guitar and plug ’er into to the tuna. Start on the low E string and play a harmonic on the
twelfth fret. To do this, place your finger lightly on the string, directly over the twelfth fret and pluck the
string. You should have just enough pressure so the string can ring, and enough to allow for the harmonic
to be sounded.
Once you are happy with the tuning of the harmonic, play a note on the twelve fret, the octave. This note
is the same note as the harmonic. Play this note carefully, as you need to get an accurate reading.
Checking this on the tuner, it should be the same as the harmonic.
If it’s pitched higher, you need an anti-clockwise turn on the saddle screw, and the opposite for a lower
pitched note. We’re playing the game of give and take here. Once you’ve adjusted the saddle, the strings
pitch will have altered also. So re-tune the harmonic and test again.
Make small adjustments! Get a feel for how much to adjust. Some things to note:
- New strings will have a more accurate tuning!
- Old strings will be harder to tune because wear and gunk alter their gauge.
- Gauge(string thickness) affects harmonics, so putting strings on with a different gauge to your previous
ones may put the harmonics out again.
- If you run out of travel in the saddle adjustment, you are doing something wrong, or your guitar needs
some professional help.
- Not all guitars have individual saddle adjustments. You will have to make do with what you have.
- If the neck of your guitar is warped(twisted) forget about getting the harmonics in tune. You need a new
neck.
- Adjusting the action and/or the truss rod will affect the harmonics.
If the harmonics were out, you’ll notice a vast improvement over what you had. Barre chords and scales
will sound much better, anywhere on the neck.
Hope this little tip is of some use!

14
Articles / Rock and Blues Bass 101 (part 1)
« on: January 28, 2015, 12:24:59 PM »
I think maybe this will be a good idea for a YouTube lesson at some stage too. Here we go..

* Most rock bass revolves around four basic structures.



The four things:

THING1
It's easy to find intervals on a stringed instrument like the bass, as
the notes form repeating patterns all over the neck. In other words,
there's a general way of finding intervals that works for every
starting note, anywhere on the neck.
The first useful one for rock bass is the 5th. From your root note,
simply move up two frets and up to the next highest string.

Code: [Select]
G|----|---|----|------
D|----|---|---O|-----
A|---O|---|----|----
E|----|---|----|--- 


In the above diagram, a Bb is played on the first fret of the A string
and it's 5th, 'F', is played on the third fret of the D string.
It's also important to note that because we can use a lower string,
'E', we can play the 5th of Bb on the E string. (see the '*' below)

Code: [Select]
G|---|---|----|------
D|---|---|---O|-----
A|--O|---|----|----
E|--*|---|----|--- 


Moving right along, you'll find that on the G string, directly across
from the 5th we just found, is the octave of the original Bb (see
below!)

Code: [Select]
G|----|---|---*|------
D|----|---|---O|-----
A|---O|---|----|----
E|----|---|----|--- 


These notes are quite handy to use for variations in boring or
repetitititititititive bass lines. And using this knowledge, you can
quickly find notes on the fret board if need be!
Nearly all the riffs in Pink Floyds 'Time' use this pattern of notes, just in various orders.

THING2
The Minor Pentatonic scale. I'll leave it to you to find a reference
for this. Find one that has the tonic on the E and one with the tonic
on the A. Then, find a combined one and note how the spaces between
them may be connected.

THING3
As above, but for the Major scale. For one scale, you should
ultimately be able to play all over the neck. It's not as difficult as
it sounds, because you just remember the two main ones (root 5 and 6)
and then you naturally get to know the spaces between them. Slides
work particularly well with the root 5 pattern major scale. For
example:
Play a tonic C on the A string with your index finger, then play a D,
two frets up with your ring finger, but as soon as you play the D,
slide your ring finger up two more frets to the E (fret seven). Then
play a G on the D string with your index finger before sliding back
down from E to D (on the A string again) with your ring finger, to
finish on C once again with your index finger. This is should sound
sort of like the lead break in Maggie May.

THING4
The old four semi-tone walk up/down, Mainly used for blues, but highly
adaptable to rock, this bass staple is probably the easiest of all to
utilise. It simply involves walking up the four frets that lead to the
note you wish to change to, in time for that change. For instance, in
a 12 bar blues, in G, the first change will be to the 4th (C). There
is a C on the A string (and I'm assuming you're playing a G on the E
string to start with). To get to that C, you simply walk up from A in
one fret increments. In a 4/4 blues shuffle type thing, you'd begin
your walk on beat 2 (A) then 3 (Bb), then 4 (B) just in time to hit
the C on '1' from the next bar. This trick is also handy if you want
to return to G in bar 7, but in a higher octave. Simply start your
'walk up' from the E on the D string on beat 2 of bar 6. You'll end up
on the higher G and it sounds cool! You can do these walks all day and
it'll sound like proper blues. Experiment with changing to higher and
lower octaves and try some walk downs too.

Some Points to Ponder:
- Bass is a rhythm instrument and in most cases, needs to sound like
it's part of the drum kit.

- Bass can have a strong influence over the 'groove' of a song, based
on when you actually 'pick up' the kick drum with the bass note.
(for example, hitting the front end of the kick beat will tend to
make the drums and the band sound held back or even sluggish)


I've made some drum tracks to play along with:

http://www.nickfletcherproductions.com/PracticeBeats/

15
Articles / Five Part Rock Guitar Lesson
« on: January 26, 2015, 03:10:36 PM »

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