Numark's X2

A TOTAL DJ SOLUTION FOR VINYL, CDS, AND MP3 CDS.

Considering all the attention–grabbing features on this muscular hybrid turntable, make sure that you don't overlook its most important asset: a slot–load CD player, with MP3 capability. Fear not, because once you start using the X2, you won't forget all of the great attributes packed into this unit.

Numark's X2 benefits from the industry's highest torque, direct–drive motor, which provides the long–term stability and durability that professionals demand. This exceptional turntable employs adjustable pitch control for both 33 and 45 RPM vinyl records, an ultra–precise aluminum tonearm with cue, height, and anti–skate adjustments. Best of all, the X2 utilizes Numark's patented interchangeable straight or S–shaped tonearm. The X2 is specifically designed for long–term dependable operation with its 12–inch anti–drag aluminum platter and solid core construction (which minimizes vibrations and unwanted noise).

The X2's CD player offers full MP3 playback capability as well as the ability to scratch, loop, manipulate and pitch adjust both audio and MP3 data CDs. Since both the CD and turntable can be played simultaneously, this double–threat hybrid turntable represents a bridge between the comfort and control of vinyl and the expansive capabilities of

 

For a video look check this out - http://www.youtube.com/watch?v=zrPvOkXQqsI

The art of recording and processing vocals

by Rich the TweakMeister

Part I: The Recording Process

The most common mistake is recording vocals too loud or too soft. The main goal to recording a solid vocal is to get all of the performance. It's not easy to set levels with a good, dynamic vocalist. As soon as you think you have the level pegged, they do something like move a few inches and you find out they are louder than you thought and meters are in the red. So you lower the level and find out that the meters are barely moving at all. If the vocalist is nervous and moving around, you might spend hours and never find an optimum level. The human voice is extremely dynamic, from soft whispers to piercing screams. If the level is too low, you will be bringing in noise and hum if you amplify it later. However, if you record too loud, there will be times when the file goes "over" which will likely result in damage that cannot be corrected later. The solution to this madness is to use a compressor in the chain after the preamp. The compressor, essentially, automatically lowers the volume when the input exceeds a certain threshold. It's like an invisible hand on a volume control. This allows a vocalist to get louder without going into the red. One of my favorite settings is to have the input to the compressor boosted so that all the "soft" words come through with a strong level. As soon as the vocalist gets louder, the clamping down begins and if they scream, it clamps down hard. The ideal is to have more consistent loudness no matter what they are doing.

 

Large Condenser Microphones

Microphone sensitivity

The more dynamic (louder) the vocalist, the less sensitive the mic needs to be. Some condenser mics will distort like madness if the vocalist is too close when they scream and it is an awful sound, especially if you are wearing cans (headphones). There is nothing you can do to fix that audio either. Because the distortion happened before the signal hits the compressor, all the compression in the world cannot help. If there is a -10 or -20 pad on the mic, use it with untrained wild vocalists. Otherwise, use a dynamic mic which is less susceptible to break up under high sound pressure levels (SPL). Or you can have them take a step back before they commit their bellow from their personal living hell. But oops, that's in the next section.

Note: Don't think that a vocal mic has to be a large diaphragm condenser. There are many fantastic sounding dynamic vocal mics.

Proper Mic technique.

This depends on the volume of the vocalist. A soft sensitive voice requires that the vocalist nearly devour the mic. I was kidding. Don't really eat the mic. I meant 4-6 inches away. Otherwise, the rule of thumb is about 1 foot away. The vocalist should back away a few inches when they get loud and come in a few inches closer for quiet intimate parts. The vocalist should not sing directly into the mic, or bassy wind noise will get in the way. Just a few degrees to the side is better. A pop filter should always be used. This is not only a good device for getting rid of plosives and spitty sounds, but can be used to keep the vocalist from getting too close and out of the range where a proximity effect might engage excessively. So What is a proximity effect, the proximity effect is the tendency of some microphones to exaggerate the bass frequencies of a vocal when the vocalist "eats", err, gets within 1 inch the mic. Comedians, radio announcers and performers often use this to great effect, but in a pop song, you typically don't want this sudden bass enhancement.

Pre-amp Trim level

This is the amount of gain (volume) applied to the mic signal, and it is calibrated in db (decibels) from 0 to typically 60db All mics differ a bit on how much juice they need. If you have a condenser mic, phantom power needs to be engaged to power the preamp. Dynamic mics don't need phantom power. Most mics will fall between 15-40db of boost. Have your vocalist practice singing and try to get the loud peaks to peg close to 0db. This will give the compressor a healthy level to work with. If you are not using a compressor you will have to lower the trim to ensure the signal never reaches 0db. That is a much lower signal than you might think.

Compressor Settings

Setting Gates: Compressors do add noise to a signal, and they do destroy dynamic range. Noise is taken care of by gating the signal. When it dips below a certain threshold, the audio signal is muted. This is effective for getting rid of low level noise you do not want in the file, such as bleed from headphones, or the vocalist moving, turning pages on lyric sheets, etc. Gates have two parameters: 1) The noise floor threshold, and the Rate. The Noise floor threshold eliminates all of the signal when it dips below the threshold, which is set from -50db to -10db. I keep mine set to -30db. Yet one has to be careful. If the gate is set too high, then the attack of the vocalists words may be cut off or come in too abruptly. The Rate parameter "fades out" the audio signal as the gate come on. This is effective to prevent the gate from chopping off the tails of the words. Usually a rate of 1-1.5 sec is enough.

Setting Threshold: The Threshold is the all important level at which the compressor kicks in. If you set the threshold to -10, it will leave all of the signal under -10 alone. When the signal exceeds -10 then it starts compressing at the ratio. -10 is an excellent place to start. Don't confuse this with the fact that your gear is outputting -10 or +4 impedance wise. Though the threshold seems like it is a volume control, it is not. It is merely telling the compressor at what level compression takes over the signal.

Setting the Ratio 2:1 is probably the most common setting for a compressor recording or playing back nearly anything. A great starting point. What this means, simply, is that it takes 2 decibels of sound energy to raise the output meter by 1db. You can read the 1st number as the db IN and the second as the db OUT. Again, 2db IN equals 1 db OUT. Easy, huh? Yeah, with 2:1 you simply divide by two. You noobs now should have a little light bulb going off in yer headz. So lets test yourself, do the math, then you will grasp this fully.

Answer this: If your vocalist was singing at -10db and suddenly got 20 db louder, without compression, where would the meters post? "Uh, that's easy -10+20=+10. The meters would post at +10 Correct! Which, as you should know is way to loud and would ruin the track. Now, if you had 2:1 compression applied, where the output is half of the input, where would the output meters post?

10+(20/2)= is... is....is....zero db! Yes! [Tweak plays his halleluiah chorus sample] It's Perfection. Why is that? The vocalists 20db burst was compressed to an actual 10 db difference in gain. (the ratio 2:1 is the same as 20:10, or half). Makes sense? Cool. (Note, you don't have to record all the way up to 0db, leave a cushion for the best sonics)

Lets go one step further, make sure you got this in your head. If you had the compressor set at a 10:1 ratio what would that mean? It would mean for every 10 decibels of gain the meters would only go up one db. So in our example, then, the 20 db burst would only let the meters go up by 2db (10:1 is the same as 20:2, or 1/10th of the original sound), Since they started at -10, the overall level would be only at -8 during the sudden 20db boost. Hardly any change in the output level at all. Would that sound "squashed"? You bet. Setting Attack and Release: These settings can be tricky as they can "delay" the effect of compression on the attack and make is hold on a bit too long on release if set improperly. I suggest till you get these tricky settings figured out (which takes quite a bit of experimentation) you simple use the fastest attack and enough of a release so the vocal is not boosted as the word trails off. Otherwise a word may pump on you unnaturally.

Setting the output: This is the final adjustment as the signal leaves the compressor. It's sometimes called the "make-up gain". They call it that because compression often lowers the overall signal and you may need to boost it back up. Basically you want to optimize this so it does not ever go over 0db in the recorder. With luck you should see a consistent healthy level on the recorder's input meters regardless of how loud the vocalist is singing.

Just a final note, you can compress again after the vocal is recorded as you prepare your tracks for the mix. So, don't get too wild with settings at the input (recording) stage. You want the recorded vocal to sound natural, where the compressor just makes it an overall more useful signal to tweak later with exciters, harmonizers, pitch intonation correctors, and effects like reverb, delay. et


c.

Technics - SL-DZ1200 CD Turntable

The new SL-DZ1200 CD Turntable transplants the DNA of the World renowned SL turntables into a CD player. The development mission, now achieved, was to allow DJs to make a perfectly smooth transition from vinyl to todays musical media without disturbing the DJing technique. With the incorporation of the SL Direct Drive motor and rotating platter a DJ can create sets using CD, CD-RW, CD-MP3 and SD Audio using the same beat matching and scratching techniques. On board cueing, sampling and effects open up a whole new world of creativity to stand each DJ apart from the crowd.

SL-DZ1200
Technical Specification

* SL Type Direct Drive Patter with Forward and Reverse Option
* Full Scratch Capability
* CD, CD-RW, CD-MP3, SD Audio Playback
* Cue Point Functionality
* On-board Effects
* SD Card Storage for Settings and Sample

price -$899



The New Vestax QFO

Price - $1399

For scratch DJ's and turntablists, the VESTAX QFO is the ultimate go-anywhere, try anything, all in one DJ tool. Not only is it an extremely powerful turntable but it is also a full battle-worthy, 2 channel dj mixer with all of the professional features you would expect from a Vestax Mixer. The VESTAX Q.F.O. is a radical new tool for performance DJ's who are looking for a lot more creative options and who want to own the most advanced DJ workstation available in the world. With the QFO, DJ's will be able to take what DJ Qbert has started and revolutionize the Scratch DJ and Turntablist scenes.

The VESTAX QFO is a circular turntable with an integrated 2-channel mixer that was designed in collaboration with performance DJ QBert and Thudrumble's Yoga Frog. This new turntable/mixer provides DJs with a convenient, all-in-one scratch DJ tool for performance or practice.

The QFO's high-torque turntable offers an ASTS (Anti-Skipping Tonearm System) tone arm and pitch control with multiple variations to select from. Also included are two start and stop buttons, which allow the user to create percussive start/stop moves with either hand.

Each channel of the mixer section has a 2-band EQ, and a phono and input switch, allowing users to control the turntable on one channel and a CD player or other external source on the other. Additional mixer features include crossfader curve control, crossfader reverse, and input fader curve and reverse.

San Francisco-based turntablist legend QBert has been a proponent of DJ culture since the mid-1980s. QBert's talents have been recognized with numerous accolades, including four consecutive Disco Mixing Club (DMC) USA and World Champion titles, which led to his selection as a judge in this esteemed competition. In 1998, DMC also awarded QBert a place in its Hall of Fame.

* Designed in collaboration with Thudrumble's DJ Qbert and Yoga Frog
* Plug & Play portability
* Battle-ready mixer features including: cross fader curve control, CF reverse, IF curve & reverse, 2- band EQ, user-replaceable PCV performance faders
* High torque professional turntable built on the success of the PDX Series
* Unique pitch control that follows the action of the platter from one side to the other giving performers 180 degrees of pitch control with 3 individual settings

EXTERNAL INPUT MIXER Pre-loaded with the Global Professional Standard of the VESTAX PMC-05ProIII circuitry. External inputs for each 2 channels, and ability to assign sources from the QFO itself to each channel.
DYNAMIC BALANCE STRAIGHT ARM Adopting a unique method, using the power balance of a spring and not using gravity, creates a load to stabilize the needle pressure. This increases the trace ability and assists the needle with anti-skip solution especially during aggressive play.
DIRECT DRIVE DC MOTOR The motor, the heart of the instrument is the original high torque direct drive of high stability and reliance proven on the PDX-2000.
Included are 2 sets of START/STOP, 33/45RPM,
REVERSE, QUARTZ LOCK control, symmetrically laid out with a cross fader in between , giving the player a universal set-up.

180° SPIN SLIDE PITCH CONTROL
Total pitch adjustment up to ±50%, with a spin slide control placed around the platter. Linking hand movements and operative functions at the same time are made possible.
CROSSFADER Included is the CF-PCV, the fader that has established the PMC-05 and other series. The fader settings have been upgraded and designed to withstand vibration during operation of the QFO.
INPUT: LINE×2(RCA)
OUTPUT: LINE×1(RCA)
HEADPHONE OUT ×2(1/4inch PHONE×1, MINI JACK×1)
MOTOR: DIRECT DRIVEDC MOTOR
STARTING TORQUE: 2.7kg/cm
STARTING TIME : 0.5 sec(33 1/3 rpm)70°
WOW FLUTTER: 0.007%
W.R.M.S TONE ARM:
- Anti Skipping Tone arm System- Dynamic Balance Method(max height adjustment 9mm)
POWER: AC 120V, 50/60 Hz


 

The Pros and Cons of Digital Recording

Pros

Digital audio recording is cheap. Even the best, fastest new computer equipped with the niftiest digital interface and software will still be a few bucks cheaper than a 2" 24-track tape machine. Which one provides better sound is a personal opinion. Digital audio also provides more versatility. You can cut and paste sound files until you turn blue and pass out. This is difficult and sometimes impossible on an analog tape machine. However, an experienced engineer with a tape machine can do some unbelievable things. Furthermore, if your studio is at home, it is readily available. Recording is much better than scribbling samples and vocal melodies when inspiration strikes you at 3 o'clock in the morning.

Cons

There are a few disadvantages. For one, you need a fast computer. If you plan to run multi-track software with all the coolest plug-ins, your CPU will definitely choke. However, the software you run has a dramatic effect on the performance of the whole system. Without mentioning brand names, I used to run software from a reputable vendor. I couldn't even run eight tracks and reverb simultaneously. After switching to a much more efficient program, I have had much better results. I haven't been able to max out the processor with dry tracks alone and it took quite a few effects to even begin to stifle my poor computer.

 

What You Need to be On Your Way to Digital Recording


Computer

You'll need a beefy computer. What is considered beefy? Well, you should first decide what software you will use. Then build to the software vendor's recommendations. Consider what format you'll record in. Will it be 24-bit 44.1 kHz? Which plug-ins will you use? How many tracks would you like to squeeze out of your computer? For general guidelines, you can't have enough CPU. This is a major bottleneck. As for RAM, keep it between 256 MB and 512 MB. SCSI hard drives are generally faster, but U-100 EIDE drives are catching up and cost a lot less. Unnecessary expansion boards can drag down performance. If you fill every PCI slot mostly with cards that don't help you get sound into and out of your computer, you'll degrade your performance. Also, put a little extra money into a nice monitor. You don't need a 42" flat-screen, but you will be staring at this thing for long periods of time. Be kind to your eyeballs.

Sound card

A cheap sound card might not operate at full duplex efficiently. At a minimum, try a SoundBlaster Live. It has done the job for me in the past. But, if you don't mind spending a few extra beans, get a sound card designed for digital recording. There are a few good ones. Be sure to do your research at Dejanews. I've had excellent luck with the M-Audio stuff.

Software

Download as many free demos of software as you can. Software can make or break your digital recording experience. I've used software from Steinberg, Sonic Foundry and Cakewalk. There are lots of others so ask around. Consider ease of use, which plug-ins come with it and how efficiently it runs. For example, I've run software that could run more than one reverb. Different software on that same computer could run several reverbs along with compressors and EQs.

 

Digital Audio Cards by Larry McHugh (Courtesy of 8th Street.com)

In this is installment we`ll take a look at some options, and some recommendations about this very important link in your digital audio studio. One important factor is whether or not you plan on using an outboard physical mixer with your system. Some digital audio cards offer break out boxes which have loZ mic inputs, and guitar inputs, thus eliminating the immediate need for an external mixer. However, if you are using midi instruments, you will need at some point to mix and monitor the entire scenario of midi and audio, so some kind of external mixer would be the way to go. An optimum solution is a digital mixer such as the Yamaha O1V, which uses the standard ADAT lightpipe to run 8 buss audio into and out of your sound card. The point is, whatever way you are going with the mixer should be integrated with your choice in sound cards since they have to be compatible. You may need to use a sound card with a break out box until you can get the mixer.
One such way as this would be the Yamaha DSP Factory, which
incorporates a break out box system with a virtual O2R mixer. This is an expandable system with some great features and effects. As you can see, the main card fits into one of your PCI slots, and the break out boxes (which can be purchased separately) fit into your bay slots, making for easy access.
It provides the mixing power of Yamaha’s 02R digital mixer, complete with 24 channels of digital mixing, onboard digital effects and dynamic processors, plus 16 tracks of hard disk recording (eight tracks simultaneous) with up to 32-bit resolution. Unlike most audio cards, it relies on its own processing power and not the computer’s CPU, which makes all the card functions available simultaneously. A built-in audio-streaming engine provides 16 tracks of record/playback of 32-bit audio to and from the computer’s disk drive. It operates on the Windows 95 platform and comes with controlling software or allows you to use your own software if you prefer.

New from MOTU is the 828 Firewire Digital Hard Disk Recording System, perfect for any FireWire-equipped Mac or Windows PC. This Audio interface features eight channels of ADAT optical digital input and output (switchable to S/PDIF optical) and RCA S/PDIF. ADAT SYNC input allows precise synchronization. The 828 has eight 24-bit analog outputs on balanced/unbalanced 1/4" TRS jacks, six 24-bit analog inputs on balanced/unbalanced 1/4" TRS jacks, and two 24-bit analog inputs on balanced/unbalanced 1/4"/XLR combo Neutrik connectors. Two mic pre-amps with switchable 48-volt phantom power on the front panel, and CueMix® Plus zero-latency monitoring of live input. This unit takes only one rack space and works with any FireWire-equipped Mac or Windows PC. Includes 12' FireWire cable, software drivers, and Audiodesk recording/mixing software.

The Lexicon Core2 PCI Card is an affordable alternative to jumping into one of these larger systems, and can be expanded down the road by adding other Lexicon Core components. It offers four channels of analog in, eight channels of analog out, eight channels of ADAT digital I/O, and S/PDIF I/O. 24-bit A/D-D/A converters, featuring selectable DBX limiting on every channel for simulating tape compression and improving headroom. Couple this with their optional MP-100 effects board, and you have a great basic system.

One of the most popular larger systems is the MOTU core 2408mkII which includes a PCI card (the PCI-324), a rack space I/O unit (the 2408mkII I/O), a 12-foot cable to connect them, software drivers for both Mac OS and Windows, and a complete audio workstation software package for Mac OS called AudioDesk.

The 1U rack-mountable 2408mkII actually provides 7 banks of 8 channel I/O: 1 bank of 24-bit analog on balanced TRS connectors, 3 banks of ADAT optical, 3 banks of Tascam TDIF, plus stereo S/PDIF. You can choose any three banks (24 channels) to be active at one time. This means you can hook up three ADATs, three DA-88s, and eight analog devices all at the same time and access any three banks - in any combination of formats - at any time. And you can freely switch formats at any time. This is a great system that is expandable, and provides a lot of bang for the buck. It can be easily configured for almost any kind of studio situation, and is wll worth the investment.


 

Behringer B-2 PRO Studio Condenser Microphone


Basically, there are two types of technologies used in microphones: dynamic and condenser. Comparing the two types, condenser microphones have a wide-range frequency response as well as linear response, while dynamic microphones do not have such characteristics. In contrast with dynamic microphones, the condenser microphones require an external power supply, usually phantom power or battery. Due to their excellent frequency response, the condenser microphones have a wider view of applications, from studio vocals and vocal sound reinforcement to acoustic instruments.

You also must have read about omnidirectional, unidirectional, and cardioid. These are pickup patterns which refer to sounds coming from the side of the microphone. Omnidirectional microphone picks up sounds equally well from all directions. Unidirectional microphone picks up sound waves from one direction only, and cardioid microphone usually has a pick up angle, much alike to unidirectional type microphones. The typical pick up angle of a cardioid microphone is 131°, whereas the omnidirectional microphone has a 360° angle. Omnidirectional pattern best suits with picking up both signal source and ambient signals. To pick up vocals or specific instruments it is recommended to set up the B-2 PRO to cardioid pattern. The figure eight pattern, which the B-2 PRO has, is suitable for ambient signals like choirs. With this pattern, you will be able to pick up ambient sounds better than with the omnidirectional pattern.

The connectors are in different flavors too. The unprofessional side includes microphones with 1/4” and 1/8” jacks. The professional and higher quality microphones have XLR plugs, just like the B-2 PRO. The cables connected to such microphones use three thick conductors less likely to pick up interferences or to break. Condenser microphones usually have such plugs.
Earlier we mentioned about phantom power source. Most, if not all, condenser microphones require an external phantom power supply. Professional gear imposes high fidelity and a simple battery or a thin cable through which the microphone draws its power from the soundcard or any other source is a no go.

The Behringer B-2 PRO features 1” large dual-diaphragm. This makes the microphone suitable for higher fidelity applications. The diaphragm is large and it features a dual design which makes the microphone compatible with omnidirectional and cardioid specifications. By having a dual-diaphragm, the B-2 PRO is capable of covering a full 360° angle. The B-2 PRO features a rugged construction with nickel plated brass body. The microphone has a solid feel and it is quite heavy, too. Even if it should not be dropped, Behringer anticipated such unwanted experiences and created a solid body for the microphone against any bumps or direct hits.

The 3-pin XLR connector is gold plated. By connecting balanced cables to the microphone through the XLR connector you will have lots of advantages. Most important is the less probability of picking up interference. The thicker the cables are the more cable length you can use and higher quality will be achieved. That is why balanced cables must be used with the microphone.

The B-2 PRO studio microphone has three switches on the body. The first is for switching between cardioid, omnidirectional and figure eight pickup patterns. You can adjust it according to your application. The second switch if for the 10 dB input attenuation. You can set it on or off. It is very useful for live applications such as concerts whereas the loud audio sources may seem a little bit too much. And the third switch is the low-cut filter which reduces the low frequency response to compensate for proximity effect. There are even microphones twice expensive than the B-2 PRO that do not have them. Price -approx $150


 

Microphones

Microphones are transducers (a device that transforms one type of energy into another. The basic design of a microphone consists of a diaphragm housed in a protective container (capsule), which also houses the electrical body. The concept here is that sound waves from an instrument or voice is projected into the diaphragm and the diaphragm thru its vibration mechanically and electronically recreates sound into electrical energy. There are three popular classes of microphones, Condensers, Dynamic and Ribbon.

Condensers are defined as a type of microphone in which the diaphragm is one plate of a capacitor (condenser) containing an electrical charge. An electrical output signal is generated by detecting the variations in the charge present in the capacitor resulting from movement of the diaphragm by sound waves. Some are called capacitor microphones. Condensers mics are very sensitive some have to be powered from the console, which is why you usually see the term " phantom power" on some mixers. Some condensers have a built in "Electret" which is a permanent charge connected to the capacitor. These mics are most times very accurate meaning they tend to recreate correctly the sound source with out many alterations.
A Dynamic Microphone is defined as consisting of a diaphragm mechanically attached to a coil operating in a magnetic field. Sound pressure variations cause movement of the coil within the magnetic field, producing a small voltage across the coil terminals. As you can see by the definition the dynamic mic is a lot more durable. And also less sensitive to sounds out side its parameter. This might be to your advantage, if you're recording several vocalists in a small room without a divider for instance, you can close mic each person to prevent leakage.
The Ribbon microphone is an early form of dynamic mic. Ribbons are known for having a flat frequency response with a smooth, softened high end. Ribbon mics can also be very effective for medium-distance miking. Fragility of the ribbon must be protected from explosive signals and other strong blasts of air. Extreme sound pressure, such as that from a bass drum, a loud amplifier, a vocalist's popped "P"-or even from slamming the lid on the mic case-can stretch or destroy a ribbon. Phantom power can also ruin a ribbon and therefore should not be used. In addition, ribbon mics do not have a high output level, so it is important to pair them with a quiet microphone preamplifier with lots of gain.

Choosing a microphone depends on what you're recording (soloist, groups, instruments etc.) and how much you can afford. There are great mics in all types. In choosing a microphone, it is to your advantage to choose one with a flat response. (Meaning what goes in is as close as possible to what comes out) The goal here is that if the mic captures the recording as accurate as possible then you can always add effects later,such as with equalization, delay, reverb, compression etc. I recommend that before buying a mic, listen to it first. Try to set up in the store a similar setup to your own studio and listen to the characteristics( personality) of several mics before choosing one . Also remember a microphone is a very sensitive tool so treat it with care. Also remember that the cable connecting it to your mixer has to be shielded, (heavy rubber, protecting it from any outside signals), or it will start to act like an antenna. So get a good cable to connect to your microphone.

Microphone Patterns: omni, bi- directional, cardioid, hyper cardioid, and shotgun.



MICROPHONE PLACEMENT ON VOCALS

By Robert Dennis

ADMINISTRATOR, RECORDING INSTITUTE OF DETROIT


The sound wave from the vocalist projects over a wide angle. In addition to projecting out, the vocal both projects up and down (to the ceiling and floor) and to the sides of the singer. Thus you can put the microphone up, down and to the sides of the vocalist and still get a good sound.

 

When the singer sings "hard consonants" (like words beginning with B, C, T or D) there is a blast of air out of the singer that projects out and down. Putting the microphone in this blast of air can cause the microphone to "pop."

 

Vocal S sounds make a high-frequency (7 kHz) blast of sound that can distort analog tape and doesn't sound very good on a digital recording. Because of the alignment of a singer's teeth, most vocalists "S" louder on one side of the mouth than on the other. Have the singer say "Sally Sucks Soup" as you walk around the singer and find the spot where there is the least projection of the S sounds. This is a great ice-breaker with the singer and gives you valuable information as to where to place the microphone.

Place the body of the microphone up above the blast of air from the consonants and off to the side where the signer has the lowest volume "S" sounds. Point the front of the mic at the lips of the singer. Use a distance of 6 to 8 inches. The singer should sing "straight ahead" and not "into" the microphone.

 

Some singers will still try to look into and sing into the microphone, undoing all of your fine placement. In this case give the singer a microphone to sing into. Have them get two inches away from the microphone and sing directly into the mic. This is a "placebo" microphone that wouldn't even be brought up at the console, but it will keep the singer at the right angle and distance from the actual mic you are using.

And Good Vocal Recording!!!!!!


The new Akai MPC 1000

In 1988, Akai professional introduced the now legendary MPC60 to the world. The MPC60 didn’t just change the way we worked - it actually created new styles of music never before imagined, styles of music that have evolved over the years but are as relevant today as they were 15 years ago when the MPC concept was born.

However, in building bigger and better machines, we realised that, size isn’t everything when customers started asking for a compact, portable 'laptop' size MPC!

Enter the MPC1000.

Small and perfectly formed (and not much bigger than an A4 sheet of paper), this diminuitive MPC is not some compromised little runt of the litter as you might expect. On the contrary - the MPC1000 inherits many of the major features of its older, bigger siblings but in a compact form factor that makes it ideal for carrying around… to your friends, to a gig, to a session… wherever. And it’s a creditable alternative to computer based systems.

As part of its lineage, the MPC1000 features the sixteen characteristic velocity and pressure sensitive pads that have been an established (and essential) component of the MPC series since its inception – arguably, they are the best pads on any drum machine past or present! Also retained is the MPC's legendary 'feel' and 'groove' so that you can be sure that your beats and sequences just swing.

Add to these a well established, friendly and intuitive user interface, two separate multi-effects processors (plus a master output effects processor!), resonant multi-mode filters, 4-way sample layering and velocity switching per pad, two MIDI ins and 32 MIDI channels via the two MIDI outputs, multiple audio outputs.

But there are new things too:

The use of compact flash as a storage medium, for example, makes for more than enough room to save your sounds and grooves in an extremely compact, portable and readily available format and present testing has verified the use of up to 2 Gigabyte cards. With regard to internal memory, you get 16Mb of on-board RAM as standard but this can be expanded to 128 MB.

Price - $880

 



Getting Your Ears Ready

When I thought about what the first article should be? I figured lets start with the fundamentals, and that would be your ears. Listening is a talent; all producers must have "ears". Your ears are an integral (important) part of the tools you'll need to be a successful producer, engineer? Your ears are probably more important than the equipment you will be using. Your ears have to decipher what the hell is going on in the studio. You'll need to know, what was the artist, producers, mix engineer or recording engineer is thinking and how did all these efforts come together (or basically, how the hell does all this shit sound?). Recording is a team effort (even if it's a team of one So I'll warn you what you are about to do might change the way you listen to music forever.

Okay you've been warned, let's get started. Grab 2 CDs, (I recommend that you have two songs to listen to compare them. I think it should be from 2 different artists on different labels with different producers). Put the first CD on and start listening. Listen to the snares the producer uses, listen the bass sounds, listen to the lead and background. Listen to the left and right headphone separately, can you hear a difference, are some of the sounds recorded to come out left or right? What is it that you like? What did the producer do, or didn't do that you like? This is the difference; you have to tune up them ears. So start by listening, try to figure out what you like and don't like and how the sounds you select determine what type of song you're going to create.


Building a Studio


Want to record music in your bedroom/bathroom/basement? Then you need to build a small studio that'll let you do this. What you want is going to be a function of

  • (i) how much you can afford, what you want to do with your music, and the quality/flexibility of recording.
  • What to use to make music

    The following are some examples of what a small home studio could/should have.
  • Microphone: You're going to pay about $150 for a decent recording mic, and about $250 for a decent Condenser mic.
  • Effects processor(s): Processors will run about $100-$200 each. (Of course they can be much more expensive also, these are basic prices) The usual suspects are Delay/Reverb Multi Effects processor, Compressor/Limiter, Mic Preamp, Sonic Maximizer/Exciter.
  • Rhythm: Since you're doing it yourself, you need to decide what to do for the rhythm section. You could get a bass guitar, a drum machine, a really cheap keyboard, or you could get a decent keyboard workstation (approx $900-$1500) where you can program all of these. You can get a used drum machine for about $150 -$300 bucks.

What to use for recording the music

Digital/hardisk recording

There are many hard disk recorders around, but again, price is an issue.
Some Prices:

  • 4 Tracks BOSS BR532 - $399 KORG PXR4 Pandora - $399
  • 8 Tracks - FOSTEX VF08 - $599 BOSS - BR8 $699 Roland VS -880/890 $990 ($650 used)
  • 18 Tracks - Roland VS1800HD $1,500
  • Monitors: I recommend you invest in a pair of excellent headphones (spend about $50). A general rule---try your mixdown on as many machines as possible). Passive Monitors (Amplifier needed) such as: the Alesis One Mk2 $250 per pair, Event 20/20- $300per pair, Powered monitors will run approx $250 each (no amp needed)
  • Mixdown sources: Sony / Fostex DAT (Digital Audio Tape) approx $650, A CD Recorder such as the Phillips CDR600 will run about $300. A good tape deck will be around $100 dollars. (Tip- if u can't afford a CDR or DAT, a Stereo VCR will fill in just fine)
  • Recording Media: A CD recorder is essential, if not a tape recorder will have to do

Using Computers to make and record music

Computers (different kinds) can be used at various stages of your recording. At the extreme, you can use a computer to do digital hard disk recording, editing, and adding effects. You can use a Personal Computer for sequencing with a Pentium III machine with 120 GB of RAM (Random Access Memory) and 40GB of hard disk space that you can use for sequencing. In my view, instead of going for an expensive synth/sampler combination, a PC is the thing to go for since you can sample any instrument and use it in your sequencer. Expect to spend anywhere between $1000-$2000 for the whole set up. If you already have a PC, a soundcard might cost you anywhere from $200-$1000.
I am not being specific here since computer technology changes so rapidly. Here are a few tips to keep in mind when purchasing a computer to record music:

  • Buy a computer with a large hard drive. (at least 40.0 GB)
  • Look for at least 512mb RAM (Random access Memory)
  • Pentium 3 or higher recommended
  • Make sure the software you like is supported by the hardware you buy (especially with regards to soundcards).
  • Buy a quality sound card if you can afford it. (approx. $150-$700)
  • Some other expenses, Amplifier ($200), Mixing board ($400), Keyboard module or keyboard ($400), EQ ($100)
  • Wires ($100), Software ($300), CD's ($20), Mic Stand ($15), Pop Filter ($20)

Official Technics SL1200 Update

DJ's - Technics has released three new flavors of 1200's! Here's the inside story:

Technics new SL1200MK5 Direct Drive Pro Turntable
The SL1200MK5 replaces the popular SL1200M3D. The SL-1200MK5 Direct Drive Turntable offers a number of groundbreaking features along with the time-tested functions made famous by the original "wheel of steel." Significant tone arm modifications have greatly reduced skipping and with customizable braking speeds, you can really fine-tune your performance. There is also a Quartz-Lock (zero pitch) button and a selectable pitch range of ±8%. The long-life LED stylus illuminator is also a fun new innovation.

Technics new SL1210MK5 Direct Drive Pro Turntable - Black Finish
This new model replaces the SL1200M3DB. It is identical in features to the new SL1200MK5 expect with a black finish instead of the standard silver finish.

Technics new SL1210M5G Direct Drive "Grand Master" Turntable - Black Finish
As the new "flagship" turntable from Technics, the SL-1210M5G "Grand Master" Direct Drive Turntable offers even more features. Check out the technologically advanced tone arm modifications, which virtually eliminate skipping. This is awesome news for scratch DJs. And now you can customize braking speeds to suit your applications. There is also a Quartz-Lock (zero pitch) button and expanded pitch range options: ±8% or ±16% via computerized pitch control. Blue LED numbers indicate the amount of pitch adjustment currently applied. A second button next to the pitch-reset button allows you to switch between the two levels of pitch adjustment. There are two columns of numbers, 2/4/6/8 and 4/8/12/16, and each press of the button switches between the two columns and illuminates the current setting. Another new feature is the long-life blue LED stylus illuminator.

More SL1200 scoop:
o The SL1200MK2 and SL1200MK2B (black finish) will continue in production. These models are now the "entry level" SL1200's.
o ProSound has a limited number of the discontinued SL1200M3D Turntables in stock at the special price of $397.99 while supplies last.
o Accept no substitutes! With the new SL1200's now out we are already seeing a number of non-Technics dealers attempting to sell "gray market" versions of the new 1200's without disclosing this information. "Gray market" 1200's are not designed for the U.S. market. These SL1200's are designed for other countries, usually with different AC requirements. Many of these turntables have also been modified to work on U.S. voltages. In these cases, Technics will NOT honor the warranty. Most gray market turntables also have instruction manuals in Japanese or other languages - not English or Spanish.