| Author |
Thread |
|
DJ Amaze
Starting Member
United States
1 post
Joined: Aug, 2002
|
 Posted - 2002/08/01 : 13:44:56
hey, I DJ and i would like to produce my own happr hardcore/epic trance/hard house tracks. I have some computer programs-Reason, Rebirth, Fruityloops. Anyone know if i should get some other or better progs or any pages with some info on producing with comp. programs??
thanx =)
-Amaze
dance till the party stops, love till your heart stops, live life to the fullest
Alert moderator 
|
MusicTech
Starting Member
United States
12 posts
Joined: Aug, 2002
|
 Posted - 2002/08/14 : 22:31:31
1. Sonic Fundamentals
a. Loud vs. Soft
Sound is moving air. Our ears are designed to be sensitive to these vibrations and interpret them. In music, the term, "Dynamics" refers to whether a sound is "soft" or "loud". The ability of a recording medium to reproduce the difference between soft and loud is called its "Dynamic Range". Vinyl records and cassette tapes have a limited dynamic range of 20 db or so while modern CD's and Digital Audio Tape(DAT) are capable of full dynamic range; that's 100 db! The limiting factor of how much of that range you get to actually hear is determined by the speakers, and amplifiers and the room you're listening in. Read on...
b. Highs vs. Lows
We've all heard terms like "bright", "dull", "deep" and "thin" used to describe music. Two major factors complicate this affair. The first is that we all hear the same thing differently; one person's "bright" is another person's "dull". The second is the accuracy or lack thereof, of our sound source, i.e. the speakers and amplifiers. Technically, the audible frequency range for human hearing is 20 Hertz(Hz) on the low end and 20 Kilohertz(Khz) on the high end. Most people's hearing range falls between 40Hz and 16 Khz and in fact, the specified frequency range of FM radio is 50Hz to 15Khz.
A typical car radio, boom box or home stereo has two EQ knobs on it. The "Low" and "High" knobs are usually centered at 100 Hz and 10 Khz respectively with a broad "fixed Q". "Q" refers to the range of frequencies affected by the boost or cut and is expressed in octaves. Their effect is not subtle but for consumer applications this is simple, convenient and usually sufficient. The loudness button is simply a low frequency boost that compensates for the apparent lack of low frequencies at low listening levels.
c. Speakers and Amplifiers and Rooms
This is the last step before your ears get to do their thing. Any problems here affect the sound reproduced, and thusly, your ability to interpret what you hear. The amp, speakers, and the room they are in, all make up the listening enviornment. When your mix sounds great in the studio and terrible everywhere else, you know something is wrong.
"Flat" is a term used to describe a system that reproduces all frequencies, equally, more or less. Some people spend thousands trying to achieve a "flat" room. As for me, it's great on paper, but it's not always great for music! As long as I know what the speakers and room are doing, I can deal with it. I like to mix on near-field systems at moderate SPL levels. This tends to decrease the affects the room may have. My current favorites are the Genelec 1031 A's, a compact 2-way self-powered monitor. They don't lie to me. Alesis and the Event 20/20 are recent entries into the more inexpensive 2-way self-powered monitor sweepstakes.
Listening level is a very subjective matter, but the ear does respond to frequencies differently at different volumes. Constant loud levels tend to dull the high end response of the ear, while at low levels the low frequencies are not as apparent. As with other things in life, drugs and alcohol also affect the ears, and usually not in a good way. BACK TO INDEX
2.Getting Your Sound On Tape
Until recently, "Analog" was the only kind of recording available to most musicians. The wide availability of DAT recorders, Alesis ADAT 8-tracks, Tascam DA-88 8-tracks and hard disc recorders like the Emu Darwin, Akai and Vestax machines have forever changed that situation. Still, the process is the same even though there are different technical considerations and format specific issues to address.
a. The Analog Recording Process
Analog recording devices use a plastic tape coated with magnetic particles moving across a magnetic recording head at a constant speed to record and playback. There is always an "erase head", first in the tape path, to erase and re-align the tape particles before they hit the "record head". In the "two-head" machine there is one head for both recording and playback. The "three-head" design features one head dedicated to recording, the "sync head", and another for playback, "the repro head". Professional machines have three heads.
There is a limit to the intensity of the signal that the tape particles can actually absorb and reproduce. The two parameters that interact to maximize the tape's ability to correctly record and playback are "tape speed" and "bias". At a faster speed, there is more tape area for a given signal, i.e. more tape particles to record. Most professional analog multitrack recorders run at 30 ips (inches per second). "Bias" is a process that was discovered by accident. It was found that when a high frequency signal, 100 Khz or so, much higher than human hearing, was recorded along with the normal signal, the magnetic particles did a better job of recreating the higher frequencies.
It is a complicated process and there are lots of things to go wrong! The tape machine must be mechanically and electronically aligned to very fine specifications. First, to ensure that it physically handles the tape gently during shuttling, rewind and fast forward. Although tape formulations have improved greatly over the years, mechanical problems can damage the tape by stretching or wrinkling it. There is no error correction for this! Treat your tapes with care and respect. Other problems include loss of particles off the tape, called shedding, speed fluctuations which produce "wow and flutter" and improper tape to head contact.
Furthermore, the electronics have to record the input signal and play it back faithfully. This is where tones on your master tapes becomes so important. They are required to properly align the electronics in the tape machine so when you work at different studios, your tape sounds like you remembered. When all these parameters are aligned correctly, you stand a good chance of hearing back a reasonable facsimile of what you recorded previously.
b. The Digital Recording Process
The digital recording process is far simpler mechanically, but much more involved electronically. The input signal is sampled 1000's of times per second and each acoustic slice is given it's own digital number, consisting of 0's and 1's. Theoretically, the "analog-to-digital converter" (ADC) recieves the analog input and converts it into a stream of numbers and conversely, the "digital-to-analog converter" (DAC) reverses the process.
The "sampling rate", or how many times per second the sound is sliced is the main factor in how well the sound will survive its digitization. CD's are sampled at 44.1 K or 44,100 times per second, and that has become an industry standard. Some formats offer 48 K sampling as well. DAC's and ADC's aren't created equally however and there are differences in how these machines sound, despite the theoretical consistency of 0's and 1's!
Digital tape machines use mechanical transports and plastic tape as a storage medium for the digital information. The Alesis ADAT and Tascam DA-88 are examples of new inexpensive digital multi-tracks. Another approach gaining acceptance are hard disc recorders. Some have computers with software as front-end controllers, like the Digi-Design and Soundscape machines, while others are dedicated boxes you plug hard discs into for storage, like the EMu Darwin, Vestax and Akai recorders.
With these random access digital recorders, the size of the hard discs limits the amount of recording time. Locating is a snap, as is editing. When this approach is combined with a computer as the interface, you have a powerful word processor for music. Anyone who has used a Mac or Windows on an IBM knows how to drag and click with a mouse and that's basically how you manipulate the sound files.
c. Theory of Multi-track Recording
Multi-track recorders are simply tape machines that allow you to record tracks and then overdub additional tracks in any order. For instance, you might record a drummer on four tracks, then go back and record a guitar part, etc. To do this, the tape machine must be able to record one track while its playing back the others. In an analog machine, it must do this from the same recording head. This is the job of the "sync head".
Digital machines don't rely on sync heads and repro heads, they're just reorganizing 0's and 1's. Depending on the device, sometimes the tape based digital machines are not as flexible as the random access machines.
It is possible to "lock up" more than one multi-track tape machine to get more tracks. This is usually done with two identical machines and SMPTE. SMPTE is an acronym for a timecode that was originally developed for the motion picture industry. It sounds like a high pitched squeel but to devices that can "read" it, it looks like a running clock. For lock up, we would "stripe" two multi-track tapes; for one song we might need five minutes, so we set the SMPTE "writer" to write from 0:00:00:00 to 5:00:00:00 minutes.
SMPTE is displayed as "hours:minutes:seconds:frames:sub frames", although not all devices read subframes (there are 80 subframes). There are four types of SMPTE. They are 30 frame per second (fps) drop frame, 30 fps non-drop frame, 25 fps, and 24 fps. In the United States, audio professionals generally use 30 fps non-drop frame and in England and Europe, they use 25 fps. The 30 fps drop frame, sometimes called "29.97", is used for video and film applications in the United States.
As has become the industry practice, we record this SMPTE on the highest edge track on each tape. For example, track 8 on an 8-track, track 24 on a 24-track, etc.. So, now we have identical "striped" tapes on their respective multi-tracks. The next step is to use a synchronization device designed to read the SMPTE off each machine and control the motors of both to keep them locked together. One machine becomes the "Master" and the other, or "Slave", chases the master machine. Two of the most popular professional systems that do this are the Lynx and Adam Smith synchronizers. This is the basic concept and it is possible, with the right interfaces and connections, to lock up different types of multi-tracks, VCR's, timecode equiped DAT machines, digital editors, etc.
The concept is simple, the execution can be complicated. The most important thing to remember is what kind of code you've got. Keep good notes! BACK TO INDEX
3. The Sound Source
There are only two ways to get your sonic information onto the tape, through a microphone or directly from an electronic output. In general, the quality of what comes back is affected by the quality of the equipment the signal passes through.
a. Voices, Horns and Acoustic Piano
While the human voice is the most dynamic, all of these instruments present a similar problem to the engineer. How can we preserve the performance, that is the soft and loud of it, and get it accurately on tape? With these instruments, we usually have to use a microphone.
The two main types of microphones are "dynamic", which have no active electronics involved in amplifying the input signal, and "condenser", which require either batteries or "phantom power" to power their electronics. Both types have a thin membrane, called the diaphragm, that vibrates and that physical vibration is translated into an electronic signal. In general, condenser mikes are brighter and have a broader frequency response, but they are more fragile. That's why you usually see an SM57, a general purpose dynamic mike, in the lead singer's hands at a concert. They can withstand a lot of abuse.
Classic condenser microphones like the Neumann U-47 and AKG C-12 use vacuum tube electronics and are treasured for their unique sound. They are rather large and have diaphragms 2 inches in diameter. Ribbon microphones are another vintage design that incorporates a thin rectangular strip as it's diaphragm, hence the name. PZM designs are a relatively new invention. They work on a completely different principle and don't look anything like traditional microphones.
The signal created by the microphone is very small and it is the microphone pre-amp that increases this level to what is known as "line-level" for interfacing with the mixing board. This is yet another link in the chain with it's opportunity to affect the sound, and they do.
Everyone has their favorite microphones and pre-amps for different situations and most do color the sound. The important thing is whether you like that color and if it's appropriate for the particular situation at hand. Here again, we run into the concept of "flat frequency response" and again it is relatively meaningless. Most microphones are not "flat" and some are better suited for certain jobs than others. As always, you need a reference and in this regard, frequency response charts and the like can be useful. Rules are made to be broken.
Signal processing is another powerful weapon for your sonic arsenal. The judicious use of compression can be a big help in lots of situations. Compressors were originally developed to compensate for the limited dynamic range of analog tape. Basically, they make the soft parts a little louder and the loud parts a little softer. This performs the dual function of keeping the soft passages higher above the "noise floor" and preventing the loud parts from getting too loud and peaking into distortion. Most compressors allow you to change the "range" (1:1, 2:1, 4:1, etc.) and the "attack time" and "release time" of the effect.
Expanders and Limiters are related to compressors. Expanders make the soft parts softer and the loud parts louder. Like compressors, you can set the range, and attack and release times of the effect. With limiters you can set a threshold that cannot be exceeded. Noise Gates simply do what their name implies by shutting off the signal path when there is no input.
b. Guitars, Basses and Things With Strings
String instruments can be recorded acoustically with microphones or directly if they have pickups. There is a different sound to each and in different situations, one may be more appropriate than the other. Often, both are recorded simultaneously and blended together in the mix. Electric guitars and basses are recorded through microphones on the amplifiers and direct to be blended or used seperately later in the mix. All the just mentioned signal processing definitely applies here, too.
The Rockman headphone guitar amp, invented by Tom Scholz in the mid 1980's, started the revolution in small, electronic amp simulators. Rackmount guitar pre-amps have gotten very sophisticated in the last few years, offering tube pre-amp stages, multi-effects, MIDI and memory capabilities. There's still nothing quite like a Gibson Les Paul into a 100 watt Marshall stack, but that can get dificult for the neighbors!
c. Samplers, Synthesizers and Drum Machines
All of these devices have direct outputs and can be connected straight to your mixer. A lot of them also include built-in effects and it's up to you whether you want to print them "wet" or "dry". It will probably depend on how many tracks you've got to play with, but you can print sounds and their effects seperately. This way if you change your mind about that "big reverb", you're not locked in.
Playing one sound at a time to tape has never been a problem but these days, there's a lot of sequencing going on and most of these boxes only have 2 or 4 outputs. In 4 or 8-track recording, this is an asset and you just have to mix the sounds from inside the box. In a professional 24 or 48-track situation, you'll want individual sounds on individual tracks. Two ways to do this are mute the sequencer tracks and record each sound in successive passes or turn down the internal volumes of all the sounds and turn one on at a time as you make passes.
Drum machines are a powerful tool and have become a mainstay of modern music making. They all have unique sounds and many have become associated with specific types of music i.e. the Roland 808 and EMu SP-12 with rap and the Roland 909 with dance. Some also have sampling capabilities and built-in audio triggering for replacing sounds off tape. Earlier models were designed with pads for programming, and some people still prefer that. Now most are available as sound modules like any other synth and the programming is simply done from the keyboard and sequencer.
Samplers are basically digital recorders. The earliest models could only record and playback one sound at a time. In the early 80's, many a snare was replaced by hand with a steady index finger and the AMS sampler. Two of the earliest MIDI samplers were the Akai S-612 and Ensoniq Mirage. They were 8-bit, which refers to their sampling resolution. Technically, the higher the resolution, the more accurate the sample. Again, specs are one thing and sound is another. These boxes all sound different and they do what they do, differently. If you like the sound of your old 12-bit box, then go for it.
Polyphonic samplers began to appear in the mid 80's, among them the Casio FZ-1 and Akai S-900. Resolution evolved to 12-bit and then 16-bit and MIDI made these boxes even more powerful. In the studio, samplers are great for quick repairs, flying vocals around or moving tracks. A sequencer and a sampler locked to tape in the mix can be real handy for those last minute repairs and arrangement changes.
d. Real Drums and Real Drummers
There are two things that you need to get good live drum sounds, a properly tuned, great sounding kit and a properly tuned, great drummer. If you can only get one, go with the human. You can always replace the sounds.
The basic approach to drum miking involves a seperate mike for the kick, the snare, the hat, the toms and one or two overheads to get the cymbals and the room sound if there is one. Another snare option is to put one mike underneath and one on top. Some of the great drum sounds from classic rock records were recorded with two mikes on the whole kit! The miking techniques should reflect what kind of drum sound you're going for.
Compression can be a big help when recording drums because of the transient nature of the instruments. Depending on the parts being played, use it as needed. Sometimes gates can be helpful as well, especially when the rhythm section is being recorded in the same room. Noise gates with "sidechain" capability allow you to select what frequencies will open the gate. Another excellent device for this application is the "Kepex" expander/gate whose operation is frequency dependent.
Many engineers like to slam high levels on to analog tape to get the natural "tape compression" sound. Some even go to the lengths of recording drums on 2" 16-track analog and then transferring that to another format to complete the project. Hey, if you've got the budget and the time, go for it. BACK TO INDEX
4. The Mixing Console
The mixer is just that. We use it to organize our signals going to the tape machines, to organize what we need to hear back from the tape machines, to monitor playback from our mixdown DAT, 2-track or other stereo sources, and to add effects to whatever is needed. In short, it is the heart of the multi-track studio.
a. Inputs and Outputs
Recording consoles are designed to be connected to multi-track tape machines. They provide seperate mixer inputs for our sound sources(mics and line inputs) and the tape returns (signal playback from the multi-track) and multiple outputs from the mixer to the tape machines (both as "direct outs" from individual channels and through the "bussing matrix"). In addition, the input channels with a choice of line or microphone input also offer equalization, effects sends, pan, buss send options and a fader for volume on each channel strip.
In-line consoles include the input section and tape return level and pan on the same physical channel strip. Split console designs have seperate channel strips for inputs and tape returns, usually with less EQ and effect sends on the tape returns (the "monitor section").
Semi-pro and home recording gear operates at a -10 level while professional equipment operates at a +4 level. Without getting too technical, this means you have to pay attention to the particular input and output levels of your boxes and how you interconnect them. When it's not right, it often hums or sounds a little screwed up. It still might work, but it will give you much less than optimum performance.
To be sure that the sound at the source gets "through the gear and back to your ear", you need to check the "gain stages" in the "signal path". Distortion can crop up in several places. Step one is your ear! Make sure the sound at the source is what you want.
Step two is the microphone or direct output of the guitar, synth or whatever. In the case of a microphone, the levels must be set carefully to ensure faithful reproduction of the input. Some mikes have a "pad" switch on them as do most mixers, to prevent overloading the input level.
Step three is the input channel to the board. The level here must be set so that the signal doesn't overdrive the channel electronics. Once that is right, the signal can be sent to tape through the "bussing" matrix (or "group outs" as they are sometimes called) or the direct out on the channel. Obviously with multiple signals to one track, they must go through the group outs and the level to tape is controlled by the group output level.
Generally, in the analog world, very bright sounds get printed a little lower, to help prevent crosstalk bleed to adjacent tracks. Sharp transient sounds and low frequency stuff like bass can be printed hot to take advantage of tape compression. Slamming the tape isn't against the law, but make sure that's the sound you want. This is only an analog phenomenon, overdriving digital recorders results in highly unusable audio!
Tape return levels are optimized for direct connection to the machine so if this has been done correctly, there is only one place left for distortion to be created... your monitor system. As long as you are aware of how your speakers and listening enviornment are affecting the sound and you listen within the volume limits of your amp and speakers, you should have a baseline for clean signal reproduction. Now, you can introduce distorton at any of these points in the chain to any effect you prefer.
b. Equalization
"Equalization" is the term used to describe the process of changing the balance between high and low frequencies. Equalizers allow us to selectively boost and/or cut specific frequencies or bands of frequencies. "Q" refers to the width or range around the centered frequency that we are EQing, that is also affected when we boost or cut. A narrow "Q" would be .2 of an octave, a wide "Q" would be 3 octaves.
There are many types of equalizers and they get used in many different ways by different people. In general, "Parametric Equalizers" allow for very specific effect with adjustable Q and frequency control for each frequency band. "Graphic Equalizers" feature as many as 31 individual sliders centered on fixed frequencies. Tube equalizers utilize vacuum tubes in their circuits as oppossed to transistors ("solid state") and are often preferred for their warm sound.
All mixers provide some kind of EQ, switchable on or off, in the signal path. These days semi-pro consoles usually feature a couple of overlapping bands of semi-parametric EQ on the low-mids(200-2K) and hi-mids(1.5K-7K), and one EQ each for the low(100 hz) and high(10K) bands with shelving switches and low-frequency roll-off. Professional consoles offer fully- parametric designs and more overall flexibility, as you might expect.
Since we can't all afford Neve VR consoles at home, another option for small studios is outboard equalizers. Get a couple of good ones and insert them into the signal path and print through them to tape. This will definitely take your sounds up a notch without totally blowing your college fund.
c. Effect Sends and Returns
There are several ways to get signal to your effects and to hear those effects back. The easist is the dedicated sends and returns on the mixing board. Sends can usually be switched between pre and post EQ. Returns generally have little or no EQ, so if you want to EQ effects, that's one more reason to have more channels than tape machine tracks. If you have more effects than sends, repatch the busses as sends or use the direct outs to get into your other effects.
d. Insert Points and Patchbays
It's really nice to plug everything into the board and not have to mess with it. This is where patchbays are a necessity and incredible convenience. Every in and out on the board, all tape machine inputs and returns and all inputs and returns from your effects are duplicated in the patchbay. Every channel also has an insert point as well for individually accessing the signal path. When it's all plugged in, you can change, rearrange and repatch it all from here. BACK TO INDEX
5. Mixer Automation
Initially available only on profesional consoles, automation systems have evolved over the last 25 years, becoming less expensive and more powerful in the process. Today there are moving fader and VCA based systems available. Many only require insert points on the console and can be up and running fairly quickly. Cheap VCA's and MIDI have brought automation to the masses.
a. SMPTE
We discussed SMPTE earlier in reference to locking up tape machines. It has also become the industry standard for running automation systems. An obvious advantage is that one track of SMPTE is all you have to give up on your multi-track to give you both synchronization and automation capabilities.
b. VCA vs. Moving Faders
I'm a big moving fader fan, particularly the "Flying Fader" system used on Neve VR consoles. Moving fader systems utilize sophisticated mechanical, motorized faders running under the control of a computer and software that keeps track of the physical position of the faders relative to the SMPTE timecode on the tape. After recording a move, you simply rewind the tape and the faders replay your moves. With this type of automation, the fader is actually in the audio path controlling the output of the signal.
Initially only available in very expensive hi-end consoles, motorized faders are showing up in low priced consoles like the Yamaha 01v(about $1800), the O2R rev2(about $6000) and O3 consoles, Mackie has the d8b (about $9000), the Panasonic DA7(about $5000), the TASCAM TM-d4000(about $4000) and the Roland VM-72 system(about $4000). In 2000, Sony came out with its little "Oxford", the DMX-r100 for about $25,000. None of these machines existed in 1995!
A VCA or "voltage controlled amplifier" is the heart of the SSL automation process. In this type of system the fader controls the output of the VCA which is actually passing the signal. As in moving fader systems, a computer and software keep track of the fader movements relative to SMPTE. Instead of the faders physically replaying your moves, their movement is represented on the computer monitor as a vertical bar moving up and down. Many people don't like VCA's in the audio path, because they can color the sound. This was more of a problem 25 years ago when VCA technology was young. The fact that SSL is such a major player in big budget mixing says a lot about how little a problem that is. You can decide for yourself.
c. MIDI Automation
The wide acceptance of MIDI and low cost, high quality VCA's has fueled the development of inexpensive MIDI based automation systems. The mixing board gets a MIDI plug!
The TASCAM M-3700 is a good example of this approach. It's on-board system has a disc drive and small LCD screen and allows you to automate channel volume, channel mute, monitor mute, EQ on/off and effects send on/off. With additional software and an external computer you can have the moving bars on the screen and some useful utility options.
Add-on outboard systems like the Mackie system utilize, a VCA package that ties in via the insert points on your mixer and a smaller fader pack of 16 faders with mute buttons to record your moves on. This is all translated into MIDI data which can be displayed on a Mac computer. Other systems come with software or are already configured to work with several popular sequencer programs like Vision, Cubase and Logic. These add-on VCA systems are a powerful and relatively inexpensive way to add sophisticated automation capabilities to your mixer. BACK TO INDEX
6. Effects Devices
This is the cool stuff; reverbs, phase shifters, delays, chorus', harmonizers... and combinations never heard before. These devices have come a long way in 20 years. Analog electronics and spring reverbs have given way to very powerful digital multi-effects units with MIDI capabilities, memory for your favorite patches and wide dynamic range. Five hundred dollars today, will buy some awesome sound power that didn't even exist 20 years ago, at any price!
a. Reverb
Reverb units attempt to recreate the sound of a particular space. The way a "space" sounds is a product of it's size, whether or not it's interior surfaces are hard and reflective or soft and absorbant, and how these interior surfaces are arranged. All these factors interact to produce the reverberant sound. Two spaces can have the same interior volume, but be shaped very differently and that makes all the difference.
The primary types of spaces are rooms, halls and plates but also can include chambers, churches, clubs and any number of wild spaces. Some units give you a few parameters to tweak, others give you pages of possibilities. All start with at least the "size" of the space. Other tweakable options include the volume and intensity of "early reflections", the amount of pre-delay to the reverb, which delays the input send into the reverb and "diffusion" and "depth" settings which have to do with how intensely the reverb spreads out in the stereo field.
Special types of "reverse reverbs", where the sound envelope is turned around and ramps up in volume rather than trailing off, are actually inspired by the analog trick of "backwards reverb". The important distinction is that "reverse reverbs" occur after the sound just like regular reverbs. "Backwards reverb" occurs before the sound and seems to ramp up to the sound.
Backwards effects is an old analog recording technique. The record head is a fixed alignment of tracks, i.e. 1 - 8. When you turn the tape around on the tape machine, track 8 becomes 1, track 7 becomes 2, etc. If you record your stereo backing vocals on tracks 1 and 2, for instance, then, at the end of the song, turn the tape around and play it back, you will hear those vocals playing backwards on tracks 7 and 8. Send them to a reverb while you're playing it backwards and record the reverb on 3 and 4. Think about it; the end of the sung phrase hits the reverb first and it trails off after the start of the phrase.
When you turn the tape around and play it forward, the vocals will still be on 1 and 2, playing correctly, but now there will be "backwards reverb" ramping up to the start of the sung phrases on track 3 and 4. Check out Pink Floyd's "Dark Side of the Moon" for some tasty examples of backwards effects.
b. Echoes and Delays
An echo is an acoustic phenomenon where a sound is repeated. The classic example in mother nature is an echo canyon. Delay units recreate this capability and give us control of several different parameters. The two most basic are "delay time", the period between the input and the delayed signal's output, and "feedback" or "regeneration" as it's sometimes called, which is how many times the signal repeats as it fades away.
Generally, each successive repeat decreases in volume. When the feedback control is raised past a certain point, the repeats get louder and louder. This is called "Runaway Feedback". Some units have a "Infinite Hold" or "Freeze" feature which captures the input signal and keeps repeating it until you stop it. To the untrained ear, discrete echoes start to be distinguished at about 20 milli-seconds. Delays lower than this start to sound more like flangers and chorus effects. Most delay units also give you this capability.
The classic analog delay unit from the 1960's was the Maestro "Echoplex" and in the 1970's, the Roland "Space Echo". Both used an endless tape loop and had a fixed record head and movable playback head. You simply moved the playback head farther from the record head to achieve a longer delay. Early analog electronic delays began to show up in the 1970's. The fidelity was somewhat limited and delay times only went to 600 or 700 milli-seconds at most. Digital delays began to show up in the 1980's and delay times of several seconds became more common. Modern units offer superb sonics, patch memories and MIDI implementation.
c. Flanging, Chorus and Phasing
Like "reverse reverbs", these effects were inspired by analog recording techniques. The famous "slap-back echo" of the 1950's and early 60's was created by sending the vocal to another tape machine in record and mixing the signal coming off the repro-head back in with the original. The distance between the record head and repro-head and the speed of the second machine determined the time of the "slap-back".
"Flanging" was a variation on this technique. The process starts similarly, but by rubbing the edge of the flange of the tape reel on the 2nd machine ever so slightly, the characteristic "Flanging" sound was produced. The time delays involved with flanging, chorus and phase shifting are usually well below 15 milli-seconds.
Digital delay units simulate these effects by incorporating an oscillator into their circuitry which allows you to control the speed and depth of the signal being recombined with the original. Modern devices give you mono and stereo flanging and chorus effects. Phase Shifters introduce slight time delays that also change the phase of the signal being recombined. This gives the distinctive deep sweeping effect that they are known for.
d. Harmonizers and Exciters
Harmonizers are basically like digital delays except they allow us to tune the pitch of the delays. When slightly detuned and added back in with the original signal, they are a useful tool for electronically thickening vocals, guitars, etc. 'Smart' harmonizers are available now that can add designated pitches according to different musical scales in their internal processors.
The most over-used effect of the 2000-2001 recording season has to be the Antares AutoTune. Originally developed as a plugin for the ProTools recording format, its now available for most major software recording packages and as a stand alone hardware unit, the ATR1. It can indeed repair pitch problems in a performance, but it is now being used as an effect on pop, rap, dance and even country songs. When it is overdriven, it resembles a vocorder but with more of the human voice intact.
The famous Aphex "aural exciter" was the first device of it's kind. The effect is often described as giving "presence" to whatever it's used on. These kinds of devices all work slightly differently by adding and subtracting harmonics, or readjusting the balance of phase relationships. It's definitely "voodoo", but very useful at times. BACK TO INDEX
7. The MIDI Revolution
This truly has been a revolution of music power and access to it. It began in the early 1980's, as small computer chips were finding their way into synthesizers. Several forward looking manufacturers of real vision had the brilliant idea to establish a standard for communication between devices. This became the first MIDI protocol. The MIDI standard has been expanding and improving since then and just about every electronic musical device made today comes with MIDI plugs and some kind of implementation.
a. Sequencers
The first "sequencers" in the 1970's were analog. They were initially designed to playback a pattern that you "stepped" in one note at a time and had a control to set the speed of the playback. They had a very limited pattern memory and there was no standard way to lock any other device to them. MIDI sequencers changed all that.
Hardware MIDI sequencers began to appear in the early 1980's. They were cheap and easy to use. The Roland MC-500 and Yamaha QX series were very popular early models. As personal computers became cheaper and more widely available, "software" sequencers began to appear. Dr.T for the Commodore C-64 was an early pioneer. Soon the Apple MacIntosh and Atari 1040 ST came on the scene and names like "Performer", "Notator", "MasterTracks", "Vision" and "Cubase" became widely known software titles.
Software based sequencers are versatile, powerful devices which have several advantages over their hardware cousins. Easy updates through software, a much larger monitor screen to view information on and more intensive editing capabilities are just a few. The range of options and features is vast. A recent innovation is the incorporating of digital recording into the sequencing program itself.
Confusion about Midi Clock and Midi Time Code (MTC) is very common. MTC and Midi Clock are related but actually intended for different purposes. Midi clock came first and its principle role is to tell listening midi devices what the tempo is(primarily sequencers and drum machines). Midi Song Pointer came next and it tells other midi devices where bar 1 is, where bar 2 is, etc. As you can imagine, a high degree of accuracy (we're talking milliseconds here) is needed for consistent control and lockup between video decks, audio machines and midi equipment running together. Midi Time Code(MTC) was developed to give midi devices an absolute reference point, much finer than bars or beats. Midi machine control(MMC) allows the sequencer to chase the audio recorder OR for the audio recorder to chase the sequencer! Midi Machine Control uses MTC to keep things locked up to a very tight degree of resolution.
Most newer programs and MIDI devices support MTC and MMC. JL Cooper makes several different boxes for communicating across formats. I use their DataMaster to read SMPTE off my 1/2" 16-track and convert that to "ADAT speak" to lock my ADAT to the 16-track. The DataMaster also supports MTC and MMC, but my sequencer doesn't. That's okay, I'll keep my Notator and Atari 1040-ST for now!
b. Synthesizers
Synthesizers have changed dramatically since computer chips began showing up in their circuitry in the early 1980's. Before that, they were rather cumbersome machines given to tuning instability, usually two voices at best and had to be reprogrammed manually for each new sound. The famous mini-moog and Arp synthesizers were popular models in the 1970's.
The first truly polyphonic modern synthesizer with patch memory was the Sequential Circuits "Prophet 5" which debuted in 1980. It had analog osciltors and their usual tuning problems but it's patch memory and programming versatility revolutionized the industry and the use of synthesizers for live performance. For the first time, keyboard players could change sounds with the touch of a button.
Digital oscilators soon followed and all variations of synthesis techniques were exploited in one form or another. The "polyphony", or how many simultaneous voices could be produced by these units, also began to increase. The MIDI protocol allowed you to have a different sound on each of 16 MIDI channels limited only by the capabilities of your MIDI synthesizer. The Yamaha FB-01 was one of the earliest synth "modules" to take advantage of this MIDI feature.
The Yamaha DX-7 became the most popular digital keyboard of the mid-1980's. Roland also produced many popular keyboards in it's Jupiter and Juno series. Korg introduced onboard effects with it's DW-6000 and DW-8000 synths. Korg hit another homerun with the M-1 synthesizer, introduced around 1990. One of the first "workstation" designs, it combined sampling technology and synthesis to produce breakthrough sonics along with an onboard sequencer and digital effects, to once again up the ante in the synthesizer race. MIDI synthesizers keep getting more powerful all the time for less money, and that trend continues.
c. Samplers
"Samplers" are like synthesizers in a lot of ways. In a synthesizer, oscilators produce the raw sound that is then modified by filters and LFO's and sent through envelopes and amplifiers, etc. In a sampler, on the other hand, the raw sound source can be anything that they sample. Then you can apply all the filters, LFO's, envelopes and amplifiers to that.
The first keyboard samplers available in the early 1980's were the EMu Emulator series. With an integrated 5 1/4" discdrive, they were big, heavy and expensive and awesome sounding. The Akai S-612 and Ensoniq Mirage were two of the first inexpensive rack mounted MIDI samplers. The prize for the first inexpensive MIDI polyphonic keyboard sampler goes to the Casio FZ-1.
It appeared in 1985 and had 8 outputs and a standard 3 1/2" discdrive for saving. Akai made the very popular S-900 rack mount samplers which evolved into the S-950. Then stereo samplers came along and like everything else, they just keep getting more powerful and less expensive. Recent options available include built-in CD Rom, SCSI hard disc and optical digital interfaces.
d. Computers
Computers have become essentiel in modern music making. They are found in synthesizers, recording devices, effects, automation and synchronization systems. They made the MIDI revolution possible.
The most popular music software in the 1980's was written for the Apple MacIntosh and Atari 1040-ST computers. In Europe, the Atari was the dominant machine and C-Lab "Notator" and Steinberg "Cubase" were two popular programs. In the US the Atari computers were much less expensive than the Macs but in the early 90's, having business dificulties, they eventually disappeared from the market here. Atari still enjoys a large presence in Europe and introduced the Falcon series there which includes 8-track digital recording capabilities straight out of the box!
The Mac has always had a rabid following despite it's more expensive hardware cost. "Performer", "Vision" and "MasterTracks Pro" were all strong sequencer packages written for it's unique operating system. The hardware prices have moderated somewhat and it still enjoys a loyal user base and large share of the music market.
The IBM was not that popular for music at first. Voyetra Systems had an early sequencing package for it but it wasn't until "Windows" came along in the late 1980's that more software was written for it. "Cakewalk" became a popular program for this platform in the early 1990's and soon others followed. E-Magic's Logic Audio is a powerful digital recording and sequencing package that was originally written for the Mac, but actually performs better on the Windows platform.
The real breakthrough for IBM has been it's rapidly expanding market share due to the popularity of "Windows", the ever increasing power of the chips that drive the PC and the price differential between PC's and Macs. The 386 gave way to the 486 which gave way to the Pentiums and their speed and efficiency keeps growing. Apple had the edge on digital recording systems at one time but the "Windows/IBM" platform has now become the dominant platform.
Hard disc recording systems are rapidly evolving and computers are either doing the recording and acting as the "front-end" interface between the operator and recorder. The "SoundScape" hard disc recording system from the UK uses a dedicated hardware recorder and a windows front-end. ProTools, which previously only developed software for Macs, released the "Session 8", an 8-track hard disc recorder, for the windows platform in 1993 and now ProTools is fully supported on the PC platform as well.
The computers have gotten powerful enough to handle it; so now you can sequence your synthesizers and program your drums, then record your guitar amp and vocals into the computer and arrange the digitally recorded tracks against the sequenced tracks all from within the same program. "Vision" for the Mac, "Cakewalk Pro" and C-Lab "Logic" for the PC and "Cubase" for the Atari Falcon all have this capability.
Digital recording and editing are logical jobs for the computer and there are many systems taking advantage of this power. Basically, the computer acts like a word processor for music. The material is recorded into the computer and then you cut it up and rearrange it, EQ it, adjust it and put it back together however you like. There are the usual complement of now standard AES/EBU digital ins and outs. Newer designs are incorporating digital interfaces for the popular Alesis ADAT and Tascam DA-88 digital recorders.
Another great use for a digital editing system is recording your mixes in pieces. If you don't have console automation and the mix is too complicated, simply record your song in segments and edit it together. Several powerful digital editing programs are available now including "SAW Plus" and "Sound Forge". BACK TO INDEX
============================================================================================================================================================================
8. Recording With Your PC
The explosion of recording hardware and software has finally collided with low prices and now everybody wants to use their computer to make music. More than 1/2 the questions I get at 'Ask the Doctor' are now computer recording related. The amount of sound power being offered from inside a PC today is incredible.
The main problem is that it's too often a complicated process to get the hardware and software installed properly and working correctly. However, once these "configuration" issues are solved, the PC recording enviornment can be a lot of fun to use and can produce very high quality recordings.
to Mac users: I have no particular bone to pick with the Mac vs. the PC. I used Macs when I first started working at Calliope Studios in NYC in 1986. Back then, for my own projects, the Atari was a lot cheaper and did what I needed, so I used it. PC's and Mac's both crash from time to time and both have their devotees. Use what you're comfortable with. The sound that comes out is only as good as the operator who puts it in. "ProTools this and plugin-X" no more guarantees a hit song than using a "Strat through a Marshall stack" means you'll sound just like Jimi Hendrix.
Remember, grasshopper, As you search for meaning in the musical universe, the computer is a tool that is only as useful as he/she who uses it ...
THE SOFTWARE:
For starters, figure out what it is you want to do. Do you already have a demo setup, but need to edit and compile your stereo mixes from DAT? or don't have a DAT and want to record your mixes directly into the computer? or want to do multi-track recording into the computer, i.e. record a drummer, bass player and guitarist all at the same time and mix on the computer? or you want to lock the computer to your ADAT and use them both to record on? or you don't have any other equipment and want to do it all on the computer?
Okay, you see that there are a lot of questions to answer before you buy the "box"! All of these software companies have websites. Look at the software you might want to use and find out what the different packages do. IMPORTANT!!! Every one of these software manufacturers lists the capabilities of each piece of software as well as minimum PC capabilities required to run that software. I can guarantee you that in the long run, you won't be happy with the way any music software runs on the "minimum machine" listed in the software specs. Talk to a rep and make them tell you what it really needs to run on.
Recording Software comes in several flavors. There are 2-track stereo programs like Sonic Foundry's Sound Forge. It's a very powerful 2-track editing package but mainly designed to work on one stereo song at a time. With a Multi-track Recording program you could work on several songs at once, do cross fades and overlap endings and beginnings, etc. or actually record and overdub parts as you would with any multi-track recorder. There are lots of multi-track programs out there like Syntrillium's Cool Edit Pro, IQS Saw Plus and SEK'd Samplitude, to name a few.
There are also powerful MIDI Sequencing programs with Digital Audio capabilities available like MOTU Digital Performer, E-Magic Logic Audio and Opcode Vision DSP. With these you can build MIDI sequences and record your audio and place the digital audio right onto the sequences. The audio becomes another sound that is triggered along with the rest of the MIDI sequence.
Here is where the capabilities of your soundcard come in. If you're only doing guitar and vocal recording, then you'll probably be fine with a simple ANALOG L/R in/out soundcard. For overdubbing, it must be "full duplex" which means the soundcard can input and output at the same time. A DIGITAL connection on the soundcard means you can send your mix from inside the computer, out to a DAT machine's DIGITAL input. To take advantage of the multi-track recording programs, you'll want multiple ANALOG ins and outs and that's why they cost more. Look for the number of simultaneous record and playback channels capabilities as well as the types of ANALOG and DIGITAL connections.
To burn cd's on your computer, you need cd-r software. A lot of software companies now have seperate add-on cd-r software to go with their recording packages. In general, these are fairly sophisticated and allow you to move id markers, change subcode info, etc. The cheapest stand alone cd-r software I've seen is Adaptec's Easy CD-Pro. It came bundled with my cd-r. It simply allows you to arrange the order of your audio files and burn the CD. Each seperate audio file gets an id on the CD. You can't place id's or access subcode info.
Easy CD-Pro is also file backup software and this is really helpful when you've filled up that big hard drive with audio files. Believe me, it will fill up faster than you can imagine! When I'm done with a project on the computer, I save all the related files onto a cd-r and erase the old files on my hard drive. When I need to work on it again, I reload the backed up files from cd-r and continue my slicing and dicing.
So, you've checked out the software, hopefully even had a chance to speak with someone who is using it or even better, actually seen it running. Now you need the hardware to run it on.
THE HARDWARE:
The Computer: Get the fastest Intel processor you can afford. The AMD and Cyrix chips are cheaper but you might want to go with Intel. Specs I've seen from several companies won't guarantee software performance on non-Intel chips, so read the fine print. Also check the motherboard for system bus speed (66mhz on older models, 100mhz and 133 mhz on newer ones), amount and type of onboard RAM it can hold(more is better) and number of open slots for PCI/ISA cards(more is better). If you have older ISA cards you want to use, double check because many new motherboards only offer PCI.
The RAM: Stuff at least 128 meg of RAM in there, 256 meg or more would be even better. It used to be 30 and 72-pin 60ns memory chips. Newer motherboards can take 168-pin SDRAM memory chips that run at 7ns on the 100mhz bus machines. Check the motherboard manuals for the type, speed and physical parameters of the memory you should use. Some machines take (2) pairs, some can handle different sizes (a 64 and 128), some can't.
The Monitor: All of these software recording packages suffer from the same problem. They fill the screen with too much information constantly. 17" monitors are getting down below $300, so go for a 19" or 21" if you can. You'll thank me later. As of 2001, 15" flat screen monitors offer almost the same screen real estate as a 17" monitor, and are in the $600 range. They take up a lot less space and give off lots less heat.
PCI/ISA slots: Most music software is now taking advantage of the PCI slots. You'll want at least (4) of them. Some PCI cards are long and some are short, so the interior physical dimensions of your computer enclosure could come into play. Get a tower enclosure if possible, it'll give you room to add hard drives and more stuff later on.
Power Supply: You'll want the biggest you can get because you'll have lots of peripherals stuffed in there. Definitely opt for the quietest one you can get as well. Those fan(s) can be a problem.
Hard Drives: Get a seperate Hard Drive(s) that will only be used for digital audio. Audio files are huge and you don't want to be recording on the same hard drive that your operating system is on. IDE drives will work and the new Ultra-ATA (UDMA)IDE drives are fast(a 66MB/sec and even newer 100MB/sec version) and very cheap.
A PC motherboard has sockets for 4 IDE drives (Primary master and slave AND Secondary master and slave) although some may only allow 2 of the faster UDMA drive types. To connect additional hard drives takes a little work. You'll have to make sure your system BIOS can recognize the drive. There's usually some kind of message when you boot up about entering "setup", press F1 or something like that. The instructions with the drive and your motherboard will help you with these settings. You'll also need to set some pins on the drive itself to set it as the master or slave.
I'm using 3 hard drives on my current system. One is dedicated to the operating system and programs. The other 2 are for digital audio. One is an EIDE 1.6 gig and the other is a UDMA 6.4 gig. The EIDE drive works fine, its just slower. CHECK THE MOTHERBOARD SPECS! A lot of software problems are actually due to hardware issues.
It used to be that for more than 4 connection capabilities, your only choice was a SCSI card which allows at least (7) SCSI devices to be connected at once. SCSI comes in several flavors and is a little faster, but SCSI peripherals generally cost 2 or 3 times as much as their IDE cousins. The chart below shows the respective transfer speeds.
IDE/UltraDMA SCSI
IDE/ATA 2.1 - 8.3 MB/sec SCSI 5 MB/sec
EIDE 11.1 - 16.6 MB/sec Fast SCSI 10 MB/sec
Ultra ATA(UDMA) 33.3 MB/sec Fast Wide SCSI 20 MB/sec
New Ultra-DMA 66 MB/sec Ultra SCSI 20 MB/sec
New Ultra-DMA II 100 MB/sec Ultra 2 SCSI 40 MB/sec
USB is now an option to SCSI. Theoretically, you can daisy chain 127 devices with USB. There's no special card needed because it's a built-in format on most newer computers (post 1999) and its plenty fast to handle digital audio. It's convenient because the USB peripherals are "hot-swappable" (meaning you can plug and unplug them with the computer on) and the cabling requirements are simpler than with SCSI devices. USB hard drives, cd-r's, cd-rw's and removeable drives are all available.
Tascam and Event both have come out with small digital audio mixers that use the USB interface. They're both about $600 and provide 8 faders, EQ, digital input and output, etc. When you get tired of recording with the mouse, you can check out one of these as an interface between you and your favorite recording software.
The newest audio interface on the block is FireWire and this one will fuel the next big audio hardware revolution. No PCI cards, just plug in and start recording. Its right on the motherboard like USB and ready to rock and roll. Devices that take advantage of FireWire are just starting to show up in 2001.
Removable Drives: A standard Zip drive cartridge will hold about 10 minutes of 44.1 stereo digital audio. A 1 meg Jazz or Syquest Sparq drive cartridge holds about 100 minutes of 44.1 stereo digital audio. If you're doing a lot of different projects, a removeable storage medium can be very helpful. The zip cartridges are about $10 apiece, Jazz carts are $90-100 each and Sparq carts are $33 apiece.
Unfortunately, Syquest is out of business as of summer '99 , so Iomega doesn't really have any competition and can charge what they want. There is a new removeable drive out there called the Orbit that looks a lot like the Syquest Sparq. It holds 2.2gig per cartridge and they're also about $35 apiece.
CD-r's and CD-rw's: 12x speed cd-r's are now in the $300 range! They can be SCSI or IDE as well. Bulk cd-r's on a spindle are down to $.20 each. Get a decent inkjet, do your own labels and make your own cd's one at a time; voila, instant record label. I picked up an 8x SCSI cd-r for $200 in Nov.99. I actually have a 2x SCSI cd-r and the 8x SCSI cd-r both installed on my system. I have some old backup software that works with the 2x cd-r and I use the 8x for burning cd's. I love burning a 50 minute cd in 8 minutes!
CD-rw is great for backing up your files. It looks like another hard drive to the computer and you can reuse the medium. The physical cd's are more expensive, but it's a no-brainer.
The Soundcard: This is the device that gets the sound into and out of the computer. There are lots of soundcards out there and you need to pick the right one for doing what you want to do. Again, go to the maufacturers' websites. Make sure the software that you have picked will work with the soundcard you're looking at. The software should list soundcards that have been tested and work correctly with it.
Don't get a card that only has digital in/outs unless you have a DAT machine or other box that reads a digital signal. There are two flavors of digital in/out. One is S/PDIF which uses an RCA style connector (like your cassette deck) and the other is AES/EBU which uses a cannon connector (like a microphone cable). Some cards have one, some have both. You can't plug them into your home stereo and hear them! Whether its simply for stereo editing or full blown multi-track recording, get a soundcard with at least a stereo analog monitor output so when all the digital bells and whistles quit on you, you can plug this into your stereo and hear what the hell is going on OR not going on!
A Word About Wordclock: If you're only using the computer to record and mix on, this doesn't apply to you. If on the other hand, you're planning on using the computer with other digital divices such as ADAT's or DA-88's, or a digital mixer like the Panasonic DA7 or Yamaha 02R, then this is very important to you.
All digital devices have computers inside them running their own internal digital clocks. To work properly together, there has to be a MASTER WORDCLOCK device with all the others being connected to it. When you start to interconnect them and move the digital audio between devices, and don't properly connect the wordclock ins and outs, these different digital clocks begin to cause problems which can include pops and clicks, random noise, timing discrepancies, audio drift or no audio at all. Furthermore, different combinations of digital equipment will require some experimentation as to which particular device should be the wordclock master.
The Drivers: These are little software programs that communicate with the operating system, be it Win95, Win98, NT or Win3.11 and facilitate the smooth interaction between your H/W and S/W. Win95 had a ver. 1 and a version 2 as did Windows 98 and now we have Windows 2000 to contend with. You need to make sure that your software and hardware BOTH are supported by whatever version of Windows is on your computer.
NT is particularly unforgiving in this area because it does not support plug and play. NT is very robust but the hardware has to be approved specifically for the NT platform. Updated drivers are always promised but if they aren't available, you're out of luck.
COMPATIBILITY ISSUES: Most software manufacturers will list systems and hardware that they have tested to work with their products. Find this information and use it. It could save you a lot of headaches. Particularly because the PC is a "roll your own" box, the money you save building a system yourself, is worthless if the software won't perform properly on it.
So let's review. To get into the computer recording game, you'll need:
COMPUTER HARDWARE
the computer $1000-2000
the monitor $ 200-800
the soundcard $ 350-1000
dedicated hard drive(s) $ 200-600
cd-r/cd-rw $ 200-600
recording software $ 100-800
MIDI sequencing software (?) $ 100-800
cd-r/cd-rw software (?) $ 100-500
SCSI card (?) $ 120-300
UltraDMA-66/100 card (?) $ 100
Removeable drive (?) $ 300
AUDIO HARDWARE
microphones/preamps (?)
external signal processors (?)
external mixer (?)
speakers (?)
external recorders (?)
external digital synchronization (?) (PRICES VARY)
__________________________________
A Successful Producer Is Someone Who Can Encourage The Genertation Of Many Ideas, And Then Discard The Vast Majority Of Them Leaving Only The Ones That Will Blend Together To Create The Perfect Sound
Alert moderator
|
Underloop
Advanced Member
United Kingdom
3,895 posts
Joined: Mar, 2002
91 hardcore releases
|
Posted - 2002/08/15 : 03:56:15
WOW!!!! Thats a post and a half!!!! Welcome to HappyHardcore.com you two.
Matthew aka DJ Underloop
--------------------------
"Turn that shit up!!!!!"
__________________________________
"We don't stop playing because we grow old;
we grow old because we stop playing."
- George Bernard Shaw
Alert moderator
Edited by - Underloop on 2002/08/15 03:57:17 |
Soren
Senior Member
United States
499 posts
Joined: Mar, 2001
|
Posted - 2002/08/15 : 04:27:49
DJ Amaze: You have lots of good programs there. Plenty of power for getting a great start at producing. Just read the manuals that came with them when you bought them and you'll be all set :)
__________________________________
_______________________________________________________________
Happy Hardcore makes me feel like a Koala bear just crapped a rainbow in my brain.
Alert moderator
|
Kaffine
Senior Member
United States
474 posts
Joined: Jun, 2002
91 hardcore releases
|
Posted - 2002/11/04 : 17:54:49
Assuming he bought them ;)
And musictech, plagarizing isn't cool. Link to the site if you must reference it...
Or at least remove the "BACK TO INDEX" from the end of each paragraph.
Alert moderator
|
|
computer program producing
Printer friendly
