What does it all mean?
Most people when it comes to Audio Equipment are faced with a barrage of terms and expressions that seem to have no real meaning. There is in fact a simple explanation for any term that is used and you will find that things start to become clearer when the abbreviations and words used by manufacturers in their manuals are explained. This dictionary is a simple way to start understanding what it's all about. Start here.

Also you may find some other things a bit daunting, such as room acoustics or calculating wavelengths or anything to do with sound. These links will help provide some enlightenment to the frustrated home engineer.
Where can I learn to use Cubase?

The Steinberg Training Center. Steinberg Training Centers the world over run courses in music technology for musicians, songwriters, arrangers, engineers, composers and teachers. Rainbow Recording & Education Services (ABN:73 462 910 725) are authorised to run three certified Steinberg courses in both versions of Cubase for home and school studio s.

Tel: (02) 9388 7222
Web: www.derekwilliams.net
Explain MIDI? 

MIDI files have been common on the Web since the beginning, but MIDI is often misunderstood. Long before MP3 was developed, people used MIDI to sonify their Web pages. MIDI is popular because the files are so small - a 20-kilobyte file can play for as long as five minutes. A five-minute MP3 file would be about 5 megabytes, which is 250 times as big!

So, if MIDI files are so small, why don't people use them instead of MP3? And can MP3 files be converted to MIDI? These are some of the more common questions encountered on this site.

Unlike MP3 and other digital audio files, MIDI files do not contain actual recorded music. Instead, the music sequence is recorded as a series of numbers, which tell how the music is to be played back. For example, to reproduce the sound of a piano playing a C note, the MIDI sequence contains digital information that says, "this is a piano sound." Another number says, "a note has been played," and other numbers convey information such as, "the note is middle C," "the key was struck very softly," "the note has now stopped," etc. Musicians love MIDI because it's easy to edit the files.

A MIDI file must be played back on a synthesizer. You may not know it, but the sound card in your computer also contains a MIDI synthesizer. Therefore, MIDI can sound great on a high-end system with a good sound card. If you think MIDI sounds tinny, like a cheap video game, then you have an old sound card that uses "FM synthesis." Newer and better cards use "wavetable synthesis."

Since MIDI must be played on a synthesizer, you can't record vocals or other sounds that are not available on the synthesizer. That's why you can't convert an MP3 file to MIDI.

There are many different types of MIDI connections. The MIDI Out sends digital messages from one MIDI device and into the MIDI In of another MIDI device


The MIDI Thru connector receives a copy of any digital message coming into the MIDI In connection and sends a duplicate of this information out of the MIDI Thru port into the MIDI In of a third MIDI device. This allows the user to have more than two MIDI devices connected as a studio. The MIDI Out port from the second or third device in the diagram below would not work because it is sending MIDI information from that particular synthesizer. The MIDI Thru port is receiving the MIDI In information and passing it on to the next device.

When MIDI devices are linked together by a series of MIDI In and MIDI T hru connections, it is referred to as a Daisy-Chain Network.
Copies of MIDI In information may be sent to numerous devices by using MIDI Thru Boxes, known as Star Networks.


A MIDI interface may only have two connections, a MIDI In and a MIDI Out. This gives the user the ability to send information to a computer, while the computer sends MIDI information out from a software program. Some music software programs will have a MIDI Echo device that allows a copy of the information to be merged with the information that is leaving the computer from the MIDI Out ports. This enables the sequencing software to simultaneously record the information that it is receiving at the MIDI In port, while sending a copy of that information out ofthe MIDI Out port along with previously recorded tracks of MIDI events.

Explain MP3?
MP3 is just a method of compressing audio files. Its name is an abbreviation for the much more unwieldy MPEG (Moving Picture Experts Group) 1 Layer 3. If you just love long, technical explanations, visit the MPEG Home Page. All you really need to know, though, is that MP3 files are about one-tenth the size of uncompressed audio files. On a CD, each minute of music takes up about 10 megabytes of space. Very few people would want to download a 40-megabyte file just to hear four minutes of music. With MP3 compression, the same four-minute song would take up about four megabytes, and would sound almost as good as the original, 40-meg file. Four megabytes is still significant, but is a practical size for sending over the Net, especially for those with fast connections.

for more info visit > http://electronics.howstuffworks.com/mp3-player1.htm
what is a midi interface ?
MIDI, the Musical Instrument Digital Interface is a protocol developed in 1983 by major synthesizer manufacturers to allow one synth to play another remotely. They quickly found out a computer could record and playback this data and it revolutionized the way music is produced.

IMPORTANT:  MIDI works by sending NOTE ON and NOTE OFF events down a midi cable as well as timing information and controller (knobs, wheels, and sliders) information.  Read that again, OK?  MIDI works by sending NOTE ON and NOTE OFF events down a midi cable as well as timing information and controller (knobs, wheels, and sliders) information.

Understand that MIDI is NOT an audio signal!!

The sound of the keyboard or module does not go down the MIDI cable, only these computer encoded "events" do. This is NOT digital audio data, its MIDI data.  Its just ones and zeros that say when you pressed down which key (a note ON event), how hard you pressed it (velocity number), when you let the key up (a note OFF event), pressed the next key, moved a knob (controller data), changed a program (program change command).  That is the basic point you have to fully understand.  The cool thing is that all this MIDI data, once recorded, is fully, totally, completely editable, malleable, changeable, transformable, reassignable, erasable, replaceable and it all happens on the edit screens of your sequencer.

MIDI Sequencers (i.e., Sonar, Logic, Cubase)  work by recording these NOTE ON/OFF events along a timing grid that can be played back at any tempo.  When you press Play on a sequencer, it simply sends the data back to the synth, which plays it just like a player piano. Get it?  Because the Synth receives the exact same events that it sent when you played the track, it sounds exactly the same. If it doesn't sound like you want you can edit it on the screen.  Makes sense so far?  Cool. 

MIDI instruments (i.e., keyboards and hardware sound modules) typically have 16 channels they can send and receive on.  Much like your TV, you can have a different program on each channel.  The only difference is that MIDI can play all 16 channels at the same time.  These channels correspond to tracks in your sequencer.  On each track, you can assign a different instrument.  You record each track while the previous tracks play back.  Because MIDI has 16 channels, you can have 16 instruments playing together, like a 16 piece band, if you want, that is totally of your own design.

MIDI PORTS

There are 16 channels for every hardware MIDI PORT on your computer.  Your computer's soundcard may have a MIDI port, and if not you can buy a MIDI interface to add one.

There are INTERNAL (virtual) MIDI ports and EXTERNAL (hardware) MIDI ports.  Let's talk about the differences briefly.

External Ports: There are 3 basic ways computers incorporate hardware MIDI Ports. You need at least one hardware port to connect a keyboard or controller.

Internal Ports: An internal port may go to a MIDI synthesizer that is a a device "on" your soundcard or to a small virtual synth installed in your operating system. Your sequencer will create a virtual MIDI port for each software synthesizer on your system. Up to 16 channels can be available for each virtual port, though a typical softsynth may no use them all.

The External MIDI port will show up in your sequencer application after you install the drivers.  When you use a software synth, an internal MIDI port is created and will show up in your sequencer application.

Again, Why do you need a MIDI interface? You need it because that's how you connect a keyboard (or controller)  Any keyboard with a MIDI out will work, even if it sounds like crap.  Remember, you only need it to trigger the NOTE ON NOTE OFF event to the sequencer, which will send them to your available synths, which will send the sound to the speakers.  Get it?  Though the playing happened at the keyboard, the sound can be triggered at any MIDI sound source, whether they exist in hardware or software.  You can even use an old CASIO with the speakers ripped out as long as it has a MIDI out.

The external MIDI out port can connect back to your keyboard's MIDI IN and play up to 16 channels of voices if the keyboard is "multi-timbral" (meaning it can sound many voices at once).  Some synths are not.  Old keyboards, like a DX7, only respond to 1 channel unless you hot rod it.  You can usually set the keyboard to respond to only one channel if you want or to only channels 1,2,3,7,8, for example, or to all 16.   Turning off channels allows you to daisy chain more keyboards or modules by cabling up the 2nd machine from the MIDI THRU on the 1st machine (which is an exact copy of the data coming in the MIDI IN) to the MIDI IN of the second machine. 

It is possible to have one MIDI port to control 16 different keyboard s if you want it too!  Usually, if your rig gets this large you will buy a MIDI interface with 4 ports so you can play all 16 channels of your Triton on one port. On a second port you might have with 10 channels from your Proteus 2000, a couple for your Electribe, one for each of your 3 e ffects box, and the last for your drum machine. Port 3 may use all 16 channels for a control surface and Port 4 might use all 16 more for a digital mixer...ooops sorry, I forgot we are just getting starte d.

MIDI In/Out/Thru Brain-work    

Lets say you have 2 machines connected on the same port.  If, for example, you are sending a track down channel 4  from the computer and have the 1st machine is turned OFF on channel 4, it will ignore all that data.  Because all of the data coming in to the 1st 'board is also going out the MIDI THRU, the second machine will play the track, if you have channel 4 turned ON.   So you ask, what happens if channel 4 is set to ON for both devices?  Bingo dude!  You got it, they both play. 

Now if each synth was on a different PORT, would both synths make a sound.  Nope!  Each port is totally discrete.  The data on Port A does not go down Port B. Remember, each port has 16 channels.  The Port is the parent; the channel is the child.  So if you get a midi interface that has 8 ports, how many channels do you have?  Yes, you math genius, 128.  Does that mean you can connect 128 keyboards to ONE 8 port midi interface.  YES!  But don't call me to wire it ok? Most people put each synth on it's own dedicated port so they don't have to turn channels on and off all the time.
What is General MIDI ?
Most basic synths follow a specification called General MIDI or GM.  This protocol specifies what programs are available for the 16 channels.  Drums are typically on channel 10.  The rest you define how you like.  The basic GM set consists of 128 programs which always have the same program number.  Piano is always program 1.  34 is always Picked Bass, 67 is always Baritone Sax.

Because there is this common reference, any file you make with GM can be played on any computer or synth that follows the GM spec.  That's what a MIDIFILE is.  Its a sequence made up of tracks filled with timing information and NOTE ON/OFF information. A "MIDI" or Midifile has no audio data again.  It only has note and controller and time data.  But because manufacturers follow this standard spec, it sounds the same, or similar, on different soundcards.  Its possible to do a great composition with GM, but the limits are in the limited palette of 128 sounds. But its an excellent way to get started.  After a while you might want better, more varied, or more specific sounds--that's when you know its time to move on to standalone modules that give you often thousands of sounds, or focus on specific genre's like dance sounds, orchestral, hip hop, world fusion, R&B, etc..
what is an audio interface ?
A soundcard or other audio interface is a requirement for any computer-based home studio setup. Today's computers all come with built in audio hardware, but for true CD quality or better results you will want to upgrade to a higher quality soundcard or audio interface — either a PCI plug-in card (called a "soundcard") or a PCMCIA, USB or FireWire interface.There are portable digital mixers that can double as a control surface for your computer's digital audio software and provide multiple digital audio and MIDI I/O options.

Over the past few years, musicians have been opting for "Audio interfaces" over standard soundcards.  The AI typically has a breakout box that sits outside the computer and a cable that is connected to either a PCI card in the computer or a USB or Firewire cable.  On this box are a number of connectors for cables that go to your mixer or monitoring system.  The conventional soundcard just sits in the computer in a PCI slot.

All the connections are on the back, or sometimes on an additional card called a "daughter board" which either attached to the main card or sits in its own PCI slot. The audio interface type of system is often preferred because the cables stay away from the fierce electrical fields that surround the modern computer.  Audio interfaces also sometimes allow for balanced audio connectors instead of the unbalanced 1/8" stereo jacks one sees on most cards.  If you want the cleanest possible reco rdings, then go with an audio interface that has balanced connectors.
what is a driver ?
So what is a software driver, and how does it affect audio performance?  The driver is the critical code that manages the traffic of data going from the cpu to and from the DAC. It organizes the data so the CPU can fetch it when the audio application says it is needed.  There are certain rules the driver has to follow that are set by the operating system.  The application and the CPU have to follow the rules, and if the driver interprets the rules properly, all will work.  If there is a flaw on the application side or the driver side, things are going to mess up. 

If the poor CPU can't get the data the 1st time, it might try again.  And Again.  And Again.  What just happened to your audio, dude, it stopped?  "CPU overload, eh?"  Wonder why comp anies are often finger pointing at each other when users protest?  Hehe  "Company Z's driver is is garbage", or "The problem is in Company X's application, we asked them to fix it but they have not responded".  Sound Familiar? (Those still awake grunt in agreement) 

So what does this mean in terms of buying a soundcard?  It means to look for evidence that the card

1. Has a driver for your OS O(perating) S(ystem)
2. Has a driver that claims it works with your application, like Logic, Cubase SX or Sonar. 
3. Has a site where you can download drivers 
4. Has an updated driver within the last 6 months.
5. Looks like driver development is something they take seriously.
6. Looks like they have a cooperative relationship with software makers, with product bundles. 

Drivers are very hard to perfect. What might work great on some machines, work's terribly on others. This is what makes the decision so maddening. If you are on a PC and Microsoft introduces a new OS, or changes direct X, then your soundcard may stop working well. It's like building a hou se in an area that has an earthquake every 3 years or so.  Some motherboards actually get in the way of audio flowing to the CPU and if the driver is not aware of these issues, the audio will get hurt.
I don't want to mess with all this driver stuff, what can I do?
You could go with a stand alone digital multi-track and sync it to MIDI time code so you could use all the great MIDI features of today's computers.  With excellent MIDI synths and modules,  a good midi sequencer, a hardware sampler and effects boxes you won't be missing much. 

However, the intelligent thing to do is some research before you buy. Check on Google Groups for problems users are having with your motherboard and the device you want to buy.  Avoid stuff that was just released last week. Wait a few months till you start reading about the card in discussion groups. Evaluate the horror stories you read. You don't want to be t yping out the next one.
What is the thing with Latency everyone makes such a big deal about ?

Latency can refer to the amount of time it takes from when you press a key on your midi keyboard to when you hear the sound, when playing a softsynth, or when routing back out to a midi driven sound module or keyboard. Also it is the time it takes for your audio interface to convert analog sound to digital, get it into your computer process it and then send it back out again as analog, so you can hear what you are singing. All computers have to process sound and the time it takes to process the sound is called latency. Playing back sound or recording sound is still considered a process for the computer. The less time it takes to process, the closer to real time your monitoring of input to the computer becomes.

A few years ago there were big problems with soundcards that did not support ASIO.  The windows MME drivers had terrible latency.  Now with WDM and Asio drivers, on Windows, latency is at an all time low. The Mac's Core Audio has good latency out of the box.  The improvements over the last few years in soundcard drivers actually make playing the soft synths useable.

In a nutshell, low latency means that my singing does not sound delayed, high latency means I hear myself after I have already finished!!

If you have a slow computer, a slow sound or midi interface you are going to end up with high latency - if you try to reduce your latency in your interfaces' control panel by too much, the processor load may end up too high and your computer can crash!
How can I tell what my latency is ?
Find out the size of your audio buffer in samples.  It should be listed in your soundcard's control panel. (Consumer cards may not do this) You can usually select among buffer sizes of 512, 384, 256, 128 and even 64.  This number of samples divided by the sample rate (i.e., 44,100, 96,000, etc) is the latency in microseconds (ms) or thousandths of a second.
PCI vs Firewire ?
What about PCi cards? Aren't they "old tech?"

Of course an old PCI card is old, and should be avoided in a modern machine. It was probably built for an older op erating system, older PCI data busses and older CPUs. However, modern PCI cards are the fastest audio devices on the planet, faster than Firewire, USB2, and of course USB 1.1. You can slam more audio data through a PCI card than any of the above. With an excellent driver PCI can outperform the rest. This does not mean that a consumer PCI is "better" than a Firewire interface. Remember, drivers and DACs are the critical variables. It does mean that a well heeled PCI interface and breakout box will be able, generally speaking, to outperform a similarly well-heeled Firewire interface. However, PCI has some pitfalls. You have to open the computer and place it in a "good" slot. Not all PCI slots in a PC are created equal. Some share resources with other slots. Sometimes a PCI video card in the machine interferes with proper operation. Sometimes you have to re-arrange the cards till you find the slot where the card works better. Getting a headache with all these "sometimes, this, sometimes that". Yeah, sometimes its a pain! Especially if you are on Win 98 or (gasp) Win ME. And sometimes it just works perfect the first time. Generally, the more modern the computer and OS, the more modern the PCI card, the less troubles you will have. Once correctly installed and driver parameters tweaked in your application, the PCI soundcard or audio interface will be rock solid reliable, fast, and be able to spit out audio tracks in great number.  

Is Firewire Better?

There is no doubt that firewire is easier to install. Given of course that your computer has a firewire port. You can install a firewire interface on a pre-firewire computer if you add a PCI firewire card. However, some older computers may not have fast enough data busses on their motherboards for it. If your computer already has a firewire port generally speaking you should be OK. A good firewire interface will offer plenty of tracks to do serious work, almost as many as PCI, with less hassle. With good DACs and Drivers, the Firewire interface is a great way to go.

I am thinking that manufacturers will be coming out with great new firewire interfaces that will enter into the realm of digital mixers, with onboard DSP effects cards, and possibly DVDR storage, but that is the future.  For today's intermediate and advanced users, Fire wire is ready today.  Unlike USB 1.1, its a great solution for a home recording studio.  
How many inputs and outputs should an audio interface have?

The rule of thumb is that you need as many inputs as tracks that you will record simultaneously. Going to have 3 people over to jam and they all need their own track? Then you need 4 inputs. Simple, huh?  Yep!  Obviously, to do anything at all you need at least a stereo input and a stereo output.  However, when recording music in a recording studio setting, even alone by yourself, two inputs can be at best inconvenient.  You can work around this effectively by having a 4 bus (or 2+2 bus) mixer that can switch inputs of the mixer into the recording path and not affect playback of other tracks in the mixer. Regarding outputs, if you only have a single stereo out all your audio will go through it.  If you have a mixer, and 8 outputs on your audio interface you can mix them at the board. However, it is certainly possible to do professional level work with just stereo i/o. I have a class on that here later on, but you are not ready for that yet.
Is ADAT input/output necessary?
If you have an Alesis ADAT multitrack you'd be mad not to get a card with an ADAT interface.  But ADAT interfaces are found on more than multitracks.  Most digital mixers have ADAT i/o.  This interface connection allows the transmission and reception of 8 channels of audio in real time. 
whats the difference between mic line and instrument - inputs/levels ?

Mic level is usually in the range of approximately -55db (dynamic) to -35db (condenser) although usually defined as -40db or lower

Line level is -10db (unbalanced) or +4 (balanced) usually defined as -10db to +30db into a 10kohm load.
These levels are found on the outputs of CD players, Tape Decks, Keyboards, VCRs, Mixers etc. Also Mixers, Tape Decks, VCRs and other recording or audio input processing devices will have Line Inputs.

Instrument level is usually in the range of -20db to -10db and is associated with Guitars (and Bass Guitars) with passive pickups - this means no Pre-amp built into the Guitar. If you need a battery in your Guitar you have an Active Pickup
what is balanced and unbalanced ?
Balanced is better.  Balanced means it takes a 3 conductor plug like a headphone plug on eac h channel (as opposed to the little stereo jacks that carry both channels on a soundcard). They are often called TRS plugs for tip+ring+sleeve, The tip is Hot, the ring is cold, the sleeve is ground. The ring part serves to stabilize the voltage in the tip and counteracts some forms of line noise. Unbalanced is just tip and sleeve and is called a  TS  connector.  These may come on 1/4" phone jacks or on RCA (home stereo) plugs.

Can you tell the difference? If the cable runs are short, it's hard to.  If you turn the volume up all the way on your amp, you typically will find that unbalanced connections are noisier.  The longer the cable run, the more noise in the signal. Unbalanced lines may pick up electrical interference or even act as an antenna for radio stations. But most of the time, in short runs, there are no problems.

The real difference is that in a balanced system the two signal leads (hot and cold) each carry voltages of the opposite polarity, which is what the input reads as signal. If the voltages are the same you get no sound. Therefore if interference (mains or spike or Radio Frequency) leaks through t he shield of your cable BOTH connectors are affected equally and you hear no interference.

The XLR cable (3 prong)  This is the common balanced microphone cable.  May also be used as an AES/EBU cable or a balanced line level cable.

The TRS "Tip-ring-sleeve" cable (3-conductor "stereo" cable)  This is a balanced cable just like the XLR above, it just has different connectors. 

The TS "Tip-sleeve" cable (2 conductor "phone jack") This is an unbalanced cable

The RCA Cable ("home stereo" cable) Because the RCA only has 2 conductors, it is also unbalanced. 

The MIDI Cable (5 pin DIN cable)

The Insert Cable  (stereo to mono "Y" cable).  Insert jacks on mixers are not balanced.  Basically, on the "stereo" end, it carries both the input and the output of the mixer channel with a common ground.  These cables allow you to insert a device in the channel's path, like a compressor or EQ or to carry the signal to a patchbay where devices can be conveniently patched in. 
Are "Guitar cables" the same as "instrument" cables ?
Yes.  Both are TS cables.
Is a headphone balanced ?
No, it just looks like a TRS but is not. The Tip is left Ring is Right and the Sleeve is Shield/common.
whats the difference between condenser and dynamic mics ?
Most presenters don't give their microphone much thought; they just expect it to work, and rarely question how it does its job, or whether it's the right microphone for the presentation at hand. But having a good microphone you can trust, and whose performance peculiarities you understand, can be an important confidence builder. If you do voiceovers for video tracks or narration for PowerPoint presentations, a high-quality microphone can also make a world of difference in the sound quality you end up with, adding a subtle but significant level of professionalism to your overall presentation package.

What is the right mic for the job? That depends on what you intend to record, and for what purpose. Microphones come in all different shapes, sizes and styles, for a range of different applications, so it helps to know a few basics: what type of technology is inside the microphone, its pickup pattern, type and how to work with it. After that, it's a matter of matching what you know with what you need, to get the mic you want. Luckily, microphones are not nearly as complicated as some pieces of presentation equipment. Here's a short primer to help you get closer to the microphone that will fill your needs and feed your aspirations.

INSIDE THE MICROPHONE

Inside every microphone is a small, thin sheet of metal called the diaphragm, which vibrates as sound waves strike it. The microphone converts this vibrating pattern into an equivalent electrical signal that can then be amplified and output to a public addre ss system (PA) and/or recorded in some manner. There are three main types of microphones: dynamic, ribbon and condenser.

Dynamic mics work by connecting the vibrating diaphragm to a coil of wire. This wire surrounds a fixed magnet. As the diaphragm vibrates, it moves the coil through the magnetic field generating low-power alternating current (AC). Basically, dynamic mics are tiny electrical generators and therefore do not require a power source to work. Coincidentally, this moving coil principle, when applied in reverse, is exactly how a speaker works. Dynamic mics are also quite rugged and are preferred for stage and field use both in live and recording situations.

Ribbon microphones suspend a thin, corrugated piece of metal between a U-shaped magnet. As sound strikes the ribbon it vibrates, producing alternating current just as the dynamic microphone explained above. Because of its simpler design, a ribbon microphone is more sensitive than its moving-coil counterpart. Unfortunately, they are also more fragile and rarely seen outside of a recording studio. Ribbon mics are also particularly vulnerable to wind damage.

Condenser mics have two metal diaphragms spaced closely together with an insulating air space between them. One plate is fixed; the other is able to move freely. Positive voltage from an electrical supply is sent to one plate and negative voltage to the other. As sound strikes the diaphragm (the moveable plate) it vibrates, moving closer and farther away from the fixed plate. This changes the voltage creating a variable alternating current analogous to the sound. Condenser microphones need a power source to function properly, usually 48 volts. This power may come from a battery or, more commonly, a separate power supply. Many audio mixers can supply the necessary 48 volts over the same cable that connects to the microphone, too. This power delivery technique is called phantom power.

Condenser microphones are far more sensitive to subtle sound changes than typical dynamic mics. They excel at capturing the finer nuances of vocal performances (sung or spoken word) and many musical instruments, such as acoustic guitars and violins. Divided into two sub-categories, large diaphragm condensers tend to remain in recording studios because they, too, are fragile. However, many smaller diaphragm mics are used onstage and in the field.

The downside of condenser microphones used to be their cost. While most models remain more expensive than dynamic mics, there has been an influx of high-quality, low-cost models in recent years, opening up a wider realm of possibilities for this type of mic.

WHY PICKUP PATTERNS MATTER

Think back to the last time you were in a noisy restaurant. You were likely able to "tune-out" the general din and focus on the conversations of your dinner companions. If you choose, you can mentally "direct" your ears and focus on any sound you want, ignoring the rest. Unlike our ears, which can discriminate sounds remarkably well, microphones hear everything in the room. However, some mics are more sensitive in certain directions than others. Microphone directionality is ca lled its pick-up or polar pattern.

Omnidirectional microphones pick-up sound equally well from all directions. Whether you talk into the front or the rear of the mic, the level and sound quality will be identical. Because of this characteristic, omni's are not wellsuited for noisy environments. They are also a bad choice when used with a PA. Sound picked up by the microphone comes out the PA speakers and gets picked up again by the mic then sent to the speakers again in an endless loop. The result is the ear-splitting whine of feedback.

Unidirectional or cardioid microphones are more sensitive in one direction. They pick up sound in a heartshaped pattern, hence the cardioid moniker. Sound outside this zone (from the rear, for example), is reduced in level, or attenuated, significantly. This ability to reject unwanted sounds is the primary reason cardioid mics are the most-used for presentations and recording.

If you further narrow the pick-up pattern, the mic is called a super-cardioid, more commonly known as the shotgun microphone. Shotguns are the audio equivalent of a laser beam, as they can focus tightly on a sound source.

Bidirectional microphones have figure-8 patterns, and are often used for stereo recordings. Another variation of this type of pattern can be found in pressure-zone microphones (PZM), which have a hemispherical pick-up pattern and are designed to be placed on a flat surface. You'll often see PZMs on the floor along the front of the stage during plays and musicals. Their unobtrusive design works well in meeting rooms for audio- and videoconferences as well.

TYPES OF MICS

There are five primary microphone types: handheld, desktop, lavaliere, headset and shotgun. The most recognized is the handheld mic. As the name implies, it's designed to be carried by its user. Typically, these are dynamic mics built to withstand rough handling. Two good cardioid dynamics are the Shure SM57 or SM58, both of which are used in a wide variety of recording and performance situations. The SM58 is particularly warm-sounding with female voices.

If the mic is to reside in a desktop stand or podium mount, then condenser and ribbon mics work best. These mics provide a good sound, but don't respond well to constant juggling. for example Audio-Technica 3060.

Lavalieres are the tiny clip-on microphones worn by news anchors and used by presenters when they need to move around onstage and don't want to hold a mic. These are commonly small, unobtrusive electric-condensers (smaller, lower-quality versions of regular condensers). Most lavalieres are omnidirectional, so feedback can be an issu e if the wearer is in close proximity to the PA. Lavs are also an ideal choice for video interviews and "talking heads." For example Sony ECM-C10 or Audio-Technica 831s .

When you need freedom of movement, headset mics, popularized by many singing and dancing musical acts, are worn over the ear and suspended in front of the mouth. Because they are closer to the mouth, they reject feedback and noise well. You'll often see these used on busy tradeshow floors as well.

Shotguns are typically used at the end of a fish-pole boom in situations where mobility is needed and mics must be hidden, such as recording location dialog for movies and TV. The ubiquitous Sennheiser 416 is the de facto standard for shotgun mics.

OTHER MICROPHONE TIPS

Wired or wireless?

Handheld, lavaliere and headset microphones come in both wired and wireless incarnations. The wired variety tethers you to a cable, which can be annoying onstage, but it's a non-issue when a podium or table is involved. Wireless mics give you more freedom, but are susceptible to radio interference and dying batteries. G oing wireless also requires a separate transmitter and receiver, and these products usually cost more than wired products.

The 3-to-1 rule

Multiple mics are often used during speaker's panels, and the mics are usually positioned close together. Be aware that when two or more microphones are close to each other, strange audio artifacts can result. If the audio sounds hollow and distant, it may be because the mics are interfering with one another. To avoid this problem, remember the 3-to-1 rule: Place mics at least three times the distance apart as they are to the nearest sound source. For example, if a mic is six inches away from a presenter, the nearest mic to it should be 18 inches or more away.

Recording via the desktop

It's no secret to sound aficionados that the microphones embedded in desktop computers and laptops often produce crummy sound. However, the average user may wonder why the quality of the microphone matters as long as it picks up the sound? As with any technology, there are good microphones and bad ones. If you choose to match a professional PowerPoint presentation with narration from the tinny microphone on your computer, your shortsightedness will be noticeable to everyone who views and hears this presentation. Also, be aware that the microphone input on the computer's soundcard is connected to the world's worst preamp — the component that boosts the mic's power signal to a usable level

Good microphone techniques

Entire books have been written about microphone technique, and they all boil down to a few simple rules. First, find the right microphone for your voice. You need to try out a few different kinds, makes and models until you discover the choice that captures your voice best.

Second, find the right microphone for the environment in which it will be used. Are you relegated to the stage in the typical ho tel conference room? A hand-held cardioid will suffice, as it can be carted around or attached to a stand. On a noisy tradeshow floor? Consider a headset design. Recording voice-overs? Try a studio condenser.

Thirdly, when you use a microphone, get close to it. You want to maximize the level of your voice and minimize all the other sounds around you (remember, microphones hear everything). The proper distance from mouth to the mic is the width of your hand — four to six inches. Close mic technique results in a warm, clean, natural vocal quality that is pleasing to the ear. In a live situation, placing a stand, podium, or desk mic slightly below and to one side, 10-15 degrees, prevents the mic from blocking your face. When recording from a script, offset the mic slightly to reduce pops and whistles and make it easier to read the script. This will help reduce popped "P's" and "B's" in your script, but ultimately you will need to be aware of these trouble spots and practice eliminating them from your delivery.

Staying close to the mic is also important when it's connected to a PA. If the microphone volume must be boosted because your voice level is too low, feedback may occur, creating its annoying squeal. Getting closer boosts your level and reduces the risk of feedback significantly.

If you are working with a cardioid microphone, keep in mind that there is a proximity effect that boosts the bass in your voice. This can make men, especially, sound boomy in certain rooms. This artifact can also make a thin voice sound more authoritative, though. When choosing a microphone, don't be afraid to take the time to be choosy. Unlike computer systems and other presentation peripherals, a microphone is a purchase that will not be outdated in five years time. So pick your microphone as you would a good friend, because it will be with you for years to come.
Why are valves / tubes good for ?

Valve amplifier is an English expression for what Americans call a tube amplifier.
The valve sound or tube sound is the sound often associated with music amplified by a

valve amplifier, although this association is not rigorous. The valve sound is often descrived as being "warm", "rich", "relaxed" etc, in contrast to a so-called transistor sound that is sometimes said to be "faster" or even "harsh" and "brittle". Both audio enthusiasts and musicians with amplified instruments have a strong interest in these questions.

One important aspect of tube sound is the soft ceiling in the dynamic range of tube amplifiers. A tube radio or tube amplifier will increase in volume to a point, and then as the volume is further increased, it gently reduces in volume. Unlike transistor amplifiers, tube amplifiers don't tend to produce the harsh clipping of saturation and cutoff that transistor amplifiers do. Thus turning up the volume control, the sound will get louder and then quieter again in a very smooth and gentle sort of way. This soft ceiling behaves very much like a dynamic range compressor/limiter. In the same way that photographic film has a filmic look especially owing t o a soft compression of highlights (i.e. even when massively overexposed, the film still responds, to some degree, to increases in light), tube amplifiers provide a nice compressive effect. For this reason, tubes are often still used to impart a pleasant response characteristic to solid state amplifiers (e.g. through the use of one 12AX7 tube within an otherwise transistorized amplifier). Alternatively, one may use a light bulb in the feedback loop of an infinite gain multiple feedbac k (IGMF) circuit. The sluggish response of the light bulb can thus be used to moderate the sound and attain a valvelike "soft ceiling" in dynamic range. Microprocessor-based or microcontroller-based solid state signal processing devices can also be used for this purpose, and there are a wide variety of products that can shape the dynamic range of an audio signal. These typically operate in four regimes: expansion at low levels; no change at mid levels; compression at high levels; limiting at extremely high levels.

Some other aspects of valve sound have more to do with the circuit topology and circuit design of the amplifier, than with the use of valves rather than transistors as the active gain devices. There is also a degree of influence from the passive components used.

Early amplifiers were by definition valve amplifiers since the transistor did not become common in consumer amplifiers until the late 1960s. The very earliest ampifiers usually had single ended topologies with the most basic type of valve.

A thermionic vacuum tube having three electrodes; fluctuations of the charge on the grid control the flow from cathode to anode which making amplification possible triode. An audio amplifier using this topology will always be in class A.

Class A single-ended triode ampliers (Known as SET's) have a characteristic distortion spectrum, a simple and monotonically decaying series of harmonics, dominated by modest levels of second harmonic distortion.

Second harmonic distortion is musically equivalent to adding the same tone one octave higher, to form a chord. in this case the added tone is at a lower level (typically 5% or less at full power) but the effect is to "fatten" the sound.

Another aspect of the valve sound is that early valve amplifiers often had only limited bandwidth, in part due to passive component technology available at the time, notably capacitors. It must be remembered that the source material available at this time (usually vinyl records) lacked the definition of today's CDs. There was no electronic music with extreme bass, and high frequencies mostly contained noise and distortion, so it was advantagous to not emphasise these.

Transistor amplifiers are almost always class AB push pull, and this topology tends to cancel even order harmonic distortion products. The resulting distortion is therefore dominated by odd order harmonics, which to human sensibilities sound "harsh" etc. Transistor amplifiers made during the 1980s typically also had extremely high gain, but poor open loop linearity, and relied on large amounts of negative feedback in opposite phase with (decreasing) the input negative feedback (NFB). Some consider that NFB does not sound "natural" or "musical", due to errors in the way it reacts to transients. These errors also reveal themselves in very complex distortion spectra, that humans find "discordant".

In contrast audio valves typically have only modest gain, and are extremely linear. This makes it possible to design very simple valve circuits that rely on this inherent open loop linearity and have little, or indeed no, NFB, and thus have very simple distortion spectra.

The subject is further complicated by the way triodes and MOSFETS have certain similarities in their transfer characteristics, whereas later forms of the valve, the tetrode: A thermionic tube having four electrodes
tetrode and pentode have quite different characteristics, in some ways reminicient of the transistor.
All these factors of contribute to the valve sound for audio applications. Valve amplifiers however are also widely used for electric guitars, in this case for quite different reasons : the way they distort when overdriven is quite different, again leading to a very distinctive sound.

In more modern times, transistor amplifiers have become dominant, mainly be cause they are cheaper to produce, and operate on lower voltages (including batteries, in applications such as car radios.) However, valve amplifiers have retained a loyal following among audio fanatics, especially for SET's, in Japan, and in recent years there has also been a resurgence in the west.

Using modern passive components, and modern digital sources and wide band speakers, it is possible to build valve amplifiers that have the characteristic wide bandwidth and "fast" sound of modern transistor amplifiers, including using push pull circuits, class AB and feedback. And some enthusiasts have built amplifiers using transistors and MOSFETS that operate in class A, including single ended, and these often have the valve sound.
whats the difference between a synthesizer and a keyboard ?
A Keyboard is generally simply playing "samples" (recorded sounds stored digitally). In some keyboards each note , for example, of a piano sound is actually individually sampled and when you play middle C it is a recording of someone playing middle C on a piano.

A synthesizer (spelling var. synthesiser) is an electronic musical instrument designed to produce artificially generated sound, using techniques such as additive, subtractive, FM, physical modelling synthesis, or phase distortion to create sounds.

Synthesizers create sounds through direct manipulation of electrical currents (as in analog synthesizers), mathematical manipulation of discrete values using computers (as in software synthesizers), or by a combination of both methods. In the final stage of the synthesizer, electrical currents are used to cause vibrations in the diaphragms of loudspeakers, headphones, etc. This synthesized sound is contrasted with recording of natural sound, where the mechanical energy of a sound wave is transformed into a signal which will then be converted back to mechanical energy on playback (though sampling significantly blurs this distinction).

The term "speech synthesizer" is also used in electronic speech processing, often in connection with vocoders.
What are the differences between "acoustic", "digital", "electric" and "electronic" pianos ?
An "acoustic" piano is the traditional piano everybody is familiar with, that produces sounds by means of hammers striking strings. It is generally referred to as "the piano." The term "acoustic" is usually used when specifically making a distinction between other various forms of piano (digital, electric, etc.)

A "digital" piano is an instrument which does its best to duplicate the sound and feel of playing an acoustic piano. It uses digitally sampled sounds, amplifiers and speakers instead of strings and hammers to produce the piano-like sound. They have weighted key action to imitate the action of an acoustic piano.

An "electric" piano is an electro-acoustic instrument analogous to an electric guitar. It has a real action, some sort of metal tine or string which vibrates, and pickups to detect the audio signal for subsequent amplification.

An "electronic" piano, better referred to as an "electronic keyboard" is an instrument with a keyboard, but usually without the weighted key action or velocity sensitivity, and the sound is usually generated through synthesizers (computer-generated); however, some may have sampled sounds. This typ e of keyboard is the one you see most in bands playing popular music.

What's different between the different manufacturers and the different models?
As for *manufacturers*, they usually differ in:

(1) how they sampled the sounds, (2) key action design, (3) sound output design, (4) overall appearance.
Thus each company's product line has a distinctly different feel and sound. There are many makers out there.

The best way to decide what these differences mean to you is to try out the different brands.

Usually, the main differences between various models within the *same maker* are in:
(1) the weighted key action (how the keys feel), (2) amplifier power, (3) number and size of speakers, (4) number of voices (piano and non-piano sounds), (5) polyphony (how many notes you can play simultaneously), (6) the miscellaneous "gadgets" (like sequencing capabilities, metronome, etc.)

In general, the more money you shell out for a digital piano, the more closely it imitates a piano and better sound. The difference between a low-end model and a mid-range model is usually pretty dramatic.

Some low-end models do not have all 88-keys. Some may not even have a sound output system (you basically play it through your stereo speakers or use headphones). The difference between a mid-range model and a high-end model is mostly in its "features".
what does fully weighted mean / action ?
This refers to the feel of the board.  There are many boards that feature 88 key "piano" action.  And they do cost more.  This does not mean they are better for you.  If you are a trained pianist, you may want to go this route, but if you are a guitarist building a midi studio, your going to have a train a whole new set of finger muscles. 

With a synth style keyboard there are some advantages.  You can usually play faster and easier.  Much like the difference between an acoustic guitar and an electric axe. Doing synthy and techno stuff, the light touch of synth action is often better suited.  And of course, if you have dreams of playing Beethoven, better go for the weighted since you will be learning keyboard chops to get there. 

One cool thing if you do get a board with piano action is that, even if you can't play piano now, after you find your way around and your hands get used to it, you will be able to walk up to any piano and play. .
How many keys does a keyboard need to have?
Synth action keyboards come in 88, 76, 61, 49, 37, 25, even less. If you're strictly doing stuff that just requires a small range, like samples, or need a small footprint cause your on the road, you might be able to get away with a small controller. (Lots of pros with massive rigs keep a little mini keyboard next to their computers so they can quickly send off a ditty or two.)   But assuming you only want one keyboard, and you want to do typical melodic stuff, pads, leads, and sometimes pretend like you are playing a piano, then you need at least 49 and larger boards are definitely better.

Classical composers and hard core tweak heads will agree--88 is best.  It maximizes the number of notes you can have on a channel and allows you to make more useful zoned presets. At 76 keys the classical dudes/dudettes start shaking their heads, wondering how in the heck they can do Penderecki on 76 keys <gg>. But your your typical hardcore tweak is still enthused with the great feeling 76 key boards for Kurzweil, Emu, Ensoniq, Korg...in fact, its hard to find a good feeling reasonably priced 88 key with synth action, they all seem to have piano action, so if you want quality synth action, look at the 76-er's. Kurzweil being on the higher end with a very sexy synthy feel, touch strips (great feature!), onboard arpeggiators, etc. 

61 keys are good for those if money is an issue (and it usually is for 95% of us involved in the MIDI enterprise) , or for those putting their money on the sound, like in with a Triton, Motif or Fantom, or for your second synth if you are expanding. Using the transpose function on your sequencer you can always get around the few problems you will encounter by your choice just by setting the midi thru to plus or minus 12 semitones.  With 48 keys you ARE going to run out of notes playing solos and bass, and it can be frustrating, especially during an impassioned recording and you run out of keys!  I know.  I started with a 48 key Six-Trak.  But with creative pre-thinking of what you intend to play, even these can work well.

If you are planning to do sound development you should bit the bullet and get the full 88.  Most pros use 88 keys, and if you want to sell sounds to them you better have it too.   Boards for live use also benefit by 88/76 keys, especially if you are called on to comp with a bass part as you slam keys.
Velocity and Aftertouch ?
Velocity, which makes the sound more pronounced (louder) the harder you play, is absolutely critical.  You have to have it.  Even if you are just buying a tiny little board just to trigger notes, make sure it does.  Aftertouch is a controller that is activated when you hold the keys down a press them into the keybed. It is good to have. Beginners can get by without it, however you will be missing one of the more expressive ways to control a synth.  I wouldn't work with out it. Velocity is usually routed to loudness and timbre.  Aftertouch may be routed to volume, timbre, vibrato, FX, depending on the patch.  Many composers turn off aftertouch using sequencers because they generate a lot of events and can clog the midi bandwidth.  In the early days, sequencers were limited to a few thousand events and you had to turn it off.  Nowadays, its usually not a problem to leave it on all the time. 
What is 128 voice Polyphony ?
Polyphony refers to how many internal available channels of aud io, or "voices",  can be output by the synth at a given moment.  Don't confuse this with MIDI channels or sounds.  Many synths function by layering these "voices" in a single program or preset.  A simple, one layer tone uses 1 voice of polyphony when you press one key.  Press down a chord of 4 keys and you have used 4 voices.  If the preset uses 4 layers, as many do today to get a "phat" sound, then your chord uses 16 voices.  A "Performance Preset", sometimes called a "multi" or "combination" may layer several 1-4 voice presets in different "zones" on the keyboard.  Hence, you might find that by using a single performance preset, latching down an arp, kicking some drums and playing a solo you might be using 20, 30, even more voices at one time.  Typically, today's synths allocate these voices as you need them so after the sound stops those voices are available for use again.  If you run out of voices in a synth, you will hear "voice stealing" going on.  The tails of the sounds might get cut off, sometimes abruptly.  This was a huge problem in the early days of 8 voice machines.  Rule of thumb, 32 voices is usually enough for 4 or 5 tracks. It's relatively easy to run out of voices in a song. 64 voice polyphony will get you close to 8-10. At the dense parts, you may notice it.  With a 128 voice machine you can usually get 16 tracks.  Of course, if you are really careful about selecting only single voice programs and are playing only mono lines (one voice) as in the case of bass and analog sounds, as in many techno oriented piec es, even 8 voice polyphony will go a long way.  This is why virtual analog modeled synths can can be useful with only 4-8 voices.

To sum up, 128 voices is important if the synth is going to be doing lots of tracks. If it is your only module, you should consider it.  If the synth is just an "add on" to your existing rig, then your polyphony requirements are not so demanding.
Explain Sequencers and arpeggiators?
If you like to work with the keyboard without a computer, or take it on gigs, then having a sequencer is important.  If you are running a computer sequencer you really don't need another.  It might come in handy on occasion as a scratch pad, but most of the time it will go unused. 

The arpeggiator, which strings notes you are momentarily holding down into a cadence, is a different story and is a welcome addition in any board.  Those that are particularly useful for sequencer applications are arpeggiators that will sync to the MIDI clock coming from the sequencer.  You press record, switch on the arpeggiator, slam down a chord and, viola, instant sequence.  Especially for techno, trance, space music, ambient, the arpeggiator will get used and you will be glad you have one. 

But if you don't have one, don't sweat.  Sequencers like Logic, Sonar, and Cubase have their own software arpeggiators that can be set up in a few seconds.  I prefer a hardware arpeggiator myself, and you probably will too.
Where can I buy Sheet Music?
Established in 1955, Hutchings Pianos is a family business with three generations of experience in selling, renting  and buying quality pianos.With a huge range of new and second-hand upright and grand pianos, available at very competitive rates for purchase or private and commercial rental.They also have piano stools, sheet music, metronomes and other accessories available.

5-7 Edgecliff Rd
Bondi Junction, NSW 2022
Mon-Fri 8:30am- 5:30pm
Sat 8:30am- 2:00pm
Tel: (02) 9387-1376
Fax: (02) 9386-0121
Email: hutpiano@pnc.com.au
Where can I find a piano teacher?
Contact Hutchings pianos for a list of piano teachers.