HomeStudioGuide.com - Guide to Home Studio Recording

Stylus RMX/S.A.G.E.

Spectrasonics has released a FREE support update for our best-selling Stylus RMX - Realtime Groove Module on the Intel Mac platform. The new version supports all Mac OSX Universal Binary VST, Audio Unit and RTAS hosts, including Logic Pro 7.2, Digital Performer 5.1, Cubase 4, Garageband 3, Ableton Live 6 and Pro Tools 7.1.1 or higher.

The official Intel Mac 1.6.0 version is now released and located in the Updates area:

http://www.spectrasonics.net/updates/index.php

And the latest news is Atmosphere UB will be available in the FALL of this year. Long wait... but at least there's light at the end of the tunnel.

I built a new computer using the latest Core 2 Duo chip from Intel, some fast DDR2 memory and SATA drives. If you're interested in building your own computer for use as your DAW workstation, read on!

Note: This is not a tutorial on overclocking. This is just building a reliable computer setup that will be used for recording and audio production. This will be a Digital Audio Workstation. Because of this requirement, more importance has been given to reliability and lower noise than to overclocked speed enhancements, or water cooled or using big-noisy fans.

I'm using an Intel DG965WH motherboard. I specifically picked Intel instead of my normal pick (an Asus brand motherboard) because I don't want to have any problems using a new chip like the Core 2 Duo. I mean... if my chip is Intel, and my motherboard (mobo) is an Intel, I hope I wouldn't have any compatibility problems.

Since I will be housing this computer on a 2u rack case, I opted for on-board video. Again, this is not a gaming machine so I'm not too critical of video performance. We just need decent 2D performance and we don't want an additional fan for the graphics card (that will add fan noise.)

For my processor chip, I selected an Intel Core 2 Duo 2.1Ghz chip.



The retail package includes a heatsink and fan. The fan is quiet and does it job very well. In my opinion, there's no need to get a 3rd-party cpu cooler. Save money, and you don't need a noisy fan. The stock Intel fan works just fine.



Underneath the heatsink, you'd find thermal paste to facilitate heat transfer between the cpu and heatsink.



The Intel motherboard supports 2 IDE devices and 6 SATA drives. I'm using (2) 200 GB SATA hard drives.



And just to be sure, I also bought some SATA power connectors and SATA data cable. I didn't know if the motherboard came with them. For a few dollars, it's worth the peace of mind. I don't want to go online again and buy these and pay $$$ for UPS shipping because I was too cheap to buy them in the first place.



I'm using a 2u case for my computer. Just because I want it rack-mountable. Here, you can see the DVD-RW drive and PSU installed.



Took out the motherboard from it's plastic wrapping, and you'll notice there is a plastic cover protecting the cpu socket. Carefully remove this piece of plastic cover.



Lift up the lever for the cpu, and remove the plastic on top. That's the black thingy.



The Intel processor chip will also have a protective cover. That's the black thing on top. Carefully remove this. (It helps if you have a ground wrist strap. But I didn't have one when I did this... I just made sure I touch the metal case (of my rack) before I handled the cpu. Basically, discharge yourself of any static electricity.




And then we see the innards of the cpu, including all the pins. I remember handling 8086 chips during the days of the IBM XT computers... it was nothing like this. We've come a long, long way!



Carefully place the CPU into the cpu socket of the motherboard. Take your time. Make sure the orientation is correct. Don't FORCE anything. Check your chip orientation.




Then close the metal cover, and close the lever securely. This will hold the chip in place firmly.



Attach the cpu heatsink by positioning the 4 push pins on the 4 holes in the motherboard. Read the directions on how to properly attach the heatsink. Make sure you push in the locking pin. This holds the aluminum heatsink securely on the motherboard.

WARNING: Do not power up the motherboard without the heatsink attached to the cpu, or improperly attached or you risk damaging your expensive CPU.




Here's the fan securely installed to the motherboard.



The motherboard sometimes come with a sticker with the connection guide. I usually stick this on the inside case cover. It's a good idea to have a handy reference when working with the computer. No need to look for the printed manual or download the PDF manual. Open the computer case, and it's right there.



Install the DDR2 RAM sticks. Consult your mobo on where to install your RAM sticks for best possible performance. For my mobo, I need to fill-up the blue slots first.  I suggest you use a dependable brand for your RAM. This is not the time and place to skimp on RAM by buying no-name brand, generic RAM.

If you go to reputable RAM sellers (Crucial, Kingston, etc...), select your motherboard model and they'll give you a list of recommended RAM models for your motherboard.





Connect the 2x24 motherboard power connectors. Don't forget to connect the 2x2 power connector too on the motherboard. Shown on this pic is the 2x24 power connector. (Sorry, I forgot to take pictures of the 2x2 square connectors.)



IDE devices like your CDROM/DVD drive need to be connected via IDE. That's the gray ribbon connector. On this shot, you can also see the pins where you can attach additional USB connections.



Hard disk drives (SATA) need to be connected to the motherboard using SATA data cables (the red cable).  You'll also see on this picture the connectors for the POWER SWITCH, LED, HD LED, RESET switches (multi-color wires). Consult your mobo guide on where to connect these jacks.



This particular motherboard model supports 6 SATA drives (see the 6 on-board SATA connectors) and you can even setup hardware RAID!



The other end of the SATA data cable is plugged into your hard drive. It's a good time to also attach the SATA power connectors to the hard drive.



Here's the complete unit.



Install your operating system and you're done... then you can install your DAW software, hook-up your firewire interface and you've got yourself a Digital Audio Workstation.

I noticed a lot of people asking this question "How do I setup MIDI on my Mac OSX computer (Macbook, Mac Pro, or PowerPC)" so I hope to write a simple and short tutorial for this.

This procedure or steps has been tested on OSX running on PowerPC and Intel chips. It has worked from a simple 2x2 interface to a full-blown 8x8 midi router/merger interface.

1. Connect your midi interface to your Mac computer. Most probably, this will be done via USB. Some models are connected via firewire, but usually, these are combo units. i.e. audio interface and midi interface.

If you need to install drivers for your midi device, go ahead and install it and follow the manufacturer's direction. Depending on the manufacturer, it could be a simple click on a Setup program, or manually copying files to your hard drive.

2. Open up Applications > Utilities




3. Click on Audio Midi Setup.



4. Click on Midi Devices



5. At this point, if the device driver was installed successfully, (and the device is powered on), you will see it represented on your Audio Midi Setup screen.  You can change the icon representation by double-clicking on the device, and selecting a photo from the list of options provided.

Note: If you want "custom" icons for your Audio Midi Setup, I'll show that to you on another lesson.



On the above screenshot, you can see my Edirol 8x8 midi interface, and my MOTU 828mk2 audio/midi interface. For this setup, we'll just the midi ports on the Edirol 8x8 unit.

Click on "Add Device"

This is just my opinion and I'm no means the final authority.... so take my opinions and verify them with your own experience.

With compressors, the sound characteristics doesn't come from whether it uses tubes or not. You see, the tube is mainly used as a gain stage. After compressing the signal, you have reduced level, so you need some amplifying device to boost the levels up again before coming out of the equipment. This is where vacuum tubes and solid states come in.

Granted, they can *color* the sound somewhat... but coloration has little to do with whether there are tubes or not... it's got more to do with audio transformers in the signal path. With modern solid state design, they've eliminated the audio transformers for COST reason. In my DIY, transformers add about $300 minimum in parts alone to a stereo channel. Add the cost of manual labor to solder the transformer wires (because you can't automate these process) and prices go even higher.

With TRUE vacuum tube designs, because of the high voltage present, they need some way of isolating the high voltage DC from the audio signal output, and so the need for audio transformers is a MUST... not an option. Transformers also provide impedance matching and signal isolation.

Note there are solid-state design that still use audio transformers, (example: SSL, API, Neve... ever heard of those names?) and they are excellent designs. Of course, excellent prices too.  

With compressors, the more important question is the method they use to detect, and achieve their compression. Some use optical methods to detect loud material, some use solid-state method. And then, that is further subdivided whether they used JFET, or transistors, or ICs to reduce the level of the material.

Optical compression have a soft-knee response. With optical compression, you have a light that glows brightly in sync with the program material, which affects an LDR (Light Dependent Resistor) that varies the signal coming out. The famous and most revered Urei LA-2A uses optical compression.. i.e. the JBL T4B optical unit alone costs about $150. An LA-2A costs about $2300 per channel. This unit produces sound that is nice and fat... almost every pro studio have one or more of these units!

BTW, the LA-2A uses audio transformers, vacuum tubes, and optical compression! Yay! ... no wonder it sounds great. And yes it is vintage design, if you open it up, you'll just see wires criss crossing around. No PCB.

The other way of doing compression is using an IC chip (commonly called a VCA). It's usually an all-in-one unit chip that does everything. For example, THAT corp makes these chips that are used by our most loved RNC compressor. Presonus ACP-8 ($900 for 8 channels) also uses the same IC chips from THAT Corp. The RNC is pretty transparent using it's Super Nice mode. $180 for stereo channel. These units use a VCA (Voltage Controlled Amplifier) to reduce levels of the signal. The IC chips are cheap... about $2.55 per piece. HINT: if you want to build your own, using the datasheet from THAT Corp, you could be saving a lot of money.

There is also the JFET compression used by the famous 1176 compressor. The JFET in this case is used as a variable resistor to vary the gain of the device. I've never heard of an 1176 (I'm currently building one), but most pro studios have them as well in their arsenal. Cost: $1800.

All 3 methods will produce different characteristic compression. You can't say X method is better than Y... They're all good, just depends on what will sound better on the material and track you're compressing and the effect/sound character you want to achieve.

So you have thousands $$$ of dollars in audio equipment sitting in your studio. You've got Korg keyboards, AKAI samplers, a top of the line music and gaming PC, or maybe even an Apple PowerMac (or MacPro). Life is good.

But you follow the power cables and you're plugging them into a $10 Wal-mart power strip? Are you crazy?

No sir! Don't do that. If you value the investment you've put in your gear. You need better protection from the unpredictable nature of power coming from the Electric Company. You need to protect yourself from power surges, sags, AC line noise and power failures.

You need a UPS or Uninterruptible Power System. But not any $50 from the bargain bin of CompUSA will do. Some of those bargain basement prices UPS output a modified square wave. Those are fine if you're only powering a cheap white-box computer. But if you'll be connecting delicate audio equipment and beefy dual core, dual processor behemoths with multiple hard drives, you want a true SINE WAVE output UPS. These types of UPS mimics the sine wave that utility power companies send. Not some stair-step square waves.

UPS's are rated in VA (Volt-Ampere). It goes like this.... the higher the VA, the more devices you can attach to your UPS.... OR the longer it can supply power to your equipment of smaller load.  Also, depending on the size of the UPS, it may only be able to give you so many minutes to keep everything powered up (just long enough for a safe shut down). So the bigger the UPS (which means, the bigger and more batteries it has), the longer it can supply power.  Some UPS will have like 4 batteries inside them. Some will only have one (1).

For instance, an 800VA UPS may give you 10 minutes of runtime power, while a 1500VA will give you 36 minutes. But pick another 1500VA model, and it may give you up to 55 minutes of runtime.

Here's a handy UPS selector sizing guide. Just select the type of computer you have, how many monitors, hard drives you're using, etc... and it will give you a "SUGGESTION."  It's a suggestion because obviously, you can always go higher. If in doubt, pick the higher VA UPS.

Note, your exact computer model may not be shown... just pick a similar model.

What am I using here? I'm using a Back-UPS XS 1000VA just for my PowerMac G5, and two (2) Smart UPS 1400VA for my servers.

And oh... UPS don't last forever. I mean, the batteries. These batteries are rated for 3-5 years. So after that time period, I recommend you buy new batteries for your UPS.

I'm going to start posting what I call "Zoddities*"... this is a collection of cool, weird looking gadgets, gear, stuff. I think you'll enjoy reading about it or looking at the pictures.

First issue of Zoddities! Straps... i.e. Guitar straps



Made by the Eyeland company. You'd notice a lot of "eyes" appearing on their guitar straps. Kinda creeps you out. But cool anyways.

from their website:

The materials that go into Eyeland products are essential in creating the most visually striking and functional guitar straps ever!

With an extensive background in the leather and fashion industry not to mention sculptural principal and design, we seek out the absolute best of all these media. The leathers used in our straps have come from the top tanneries in the world.

Our requirements are that it has to be beautiful to the eye, wonderful to touch and able to perform. The hardware has to be up to the task. It must be top quality, compliment the design and also hold up to the rigors of the road.

The glass eyes used in some of our designs are actually hand-painted and so realistic you expect them to blink! Eyeland straps are not your typical assembly line "what ya see is what ya get" situation. Each strap is created with a conviction to detail.

Adjusting the length of an Eyeland strap is a joy! All Eyeland straps are designed so they can be easily adjusted. There are a variety of methods we utilize that compliment the designs. We don't think you should have to have an Eagle Scout knot manual to lengthen or shorten your strap.

You may notice that some Eyeland guitar straps incorporate curves. The Eyeland Ergo-Curve evolved from customer input. This design conforms to the shape of your body to distribute weight and allow freedom of movement.

*Why am I calling it zoddities? Because I need a word that starts with the letter "Z" so it will show up at the end of my categories... and combining it with the word "Oddities" for odd stuff. Happy now?

So you just got a brand new pair of Behringer, KRK, or Event active monitors. Or if you're one of the lucky few with deep pockets, maybe you got a Yamaha, Mackie or Genelec active monitors. Nice!

So how do you place your new monitor on your desk for proper monitoring of your music?

The rule of thumb is imagine an equilateral triangle where the tips of the triangles touch the left and right monitors, and your head.  The distance between the left and right monitors, and the distance between the monitors and your head (i.e. where you're sitting, i.e. the SWEET SPOT) should be approximately equal.  Now, point the monitors inward so the tweeters (or high-frequency drivers) approximately hit your ears, i.e. at ear level. At this point, you may do some fidgetting if you want it more toed-in or less. But this is a good starting point when placing your monitors.

This assumes that your head and monitors are on the same plane/height. If your monitors are placed higher than your head, tilt it down so the tweeter's line of fire is approximately hitting your ears. We're not too worried about the woofers or low frequency drivers since they won't be as directional as the high frequency drivers.

What can you use to tilt the monitors down?

 

You can use these Isolation pads. They're also good since they decouple (or isolate) the base of your monitors from your desk. That way, you don't get symphatetic vibrations (especially of the low frequencies) which results in unclear stereo imaging. Also, sometimes when a monitor is sitting directly on your desk, you can feel the whole desk vibrating due to mechanical vibrations of the monitor being transferred to the desk. These pads reduce those vibrations too. For less than $40, these pads are a great addition for your studio.

A Limiter and Compressor belong to the same family of dynamics processors. You can say the Limiter and Compressor are close cousins.

If you don't know what a compressor is, or need a review of it's basics, click here.

A Limiter is just like a compressor that it prevents the signal from exceeding a threshold value. While you can use low to mid compression ratio with ordinary compressors, limiters work with very high compression ratio... like 20:1 or infinity:1. Think of it like a glass ceiling. You ain't going above the glass ceiling no matter how hard you jump up. That's what a Limiter does. It prevents the signal from going beyond that point. Just like a compressor, it also has an ATTACK time on how fast it should clamp down or limit that signal.

Very expensive limiters offer 100% guarantees that the signal will not exceed the set threshold you've set... These high-end type limiters are usualy used in the broadcast industry, PA systems and most importantly, in-ear monitoring systems. (You don't want that million dollar singer of yours suddenly go deaf because of feedback because some intern manning the FOH fell asleep.)

A good example of a limiter is the 1176LN by UREI/Universal Audio. You can also buy plugin versions of the 1176 via the UAD-1 package. Click here to search our gear database for limiters/compressors.



Related Articles
How to connect a compressor
The best compressors you can buy
Build your own 1176 Compressor/Limiter
Build your own SSL Buss Compressor

Crossgrading to SONAR 6

If you've considered switching to SONAR 6 from another DAW platform, you're in good company. With unique features, amazing sound quality and top shelf plug-ins, SONAR 6 is hard to ignore, and we hear from customers daily who are interested in finding out how SONAR 6 can make a difference in their studios.

Cakewalk is now making it easier than ever to switch by offering special "crossgrade" pricing on SONAR 6. So if you're ready to step up to SONAR 6, visit these retailers today to recieve special competitive upgrade pricing. Eligible customers can buy SONAR 6 Producer Edition for only $399*, or SONAR 6 Studio Edition for just $199*.

Cakewalk will honor crossgrade pricing for all Cubase and Nuendo customers from all versions of the Steinberg programs, including Cubase LE, SL, SX and older Cubase titles on all platforms. In addition, owners of Ableton Live; Apple Logic; Magix Samplitude; MOTU Digital Performer; Pro Tools LE, M-Powered, and |HD; SONY ACID Pro and Vegas will be also able to take advantage of this special deal.

This is a repost of my SSL Build Thread. Thanks to Gyraf.dk for the schematic plans.

The compressor is a clone based from the SSL4044E desk (1985 era). The SSL mixbus is actually an integrated part of the desk's main quad fader and autofade system. Schematics and mods can be found below the article.

I also did some mods on my unit. Mainly, adding transformers at the output. Originally, the unit uses opamps for balanced outs. I have some MCI 416 output trafos that I used for my SSL unit.

The PCB...



Screenshot of my PCB stuffed with components. A few missing parts but almost there.

On another topic, I talked about how you connect a compressor. This time we'll be talking about how to connect an effects processor.

What is an effects processor? These are all-in-one units that can do reverbs, delay, chorus, flanging. A good example of this is the Lexicon MPX-550 or the TC Electronic M-One.

Here are some pics of my Lexicon MPX-550.

Photo of a Lexicon Effects processor. Also shown is a MOTU 828mk2 audio interface.



The LCD panel of the Lexicon MPX-550.

The numbers 1,2,3,4 and the text above it (which changes depending on what efects you'e selected... in this example it's "Small Hall") corresponds to the parameter knobs where you can easily change these parameters.




How do you connect these units to your home studio setup? Effects processors (or stand-alone Reverb, Delay, Chorus units) are connected in PARALLEL to the signal path.

Here's a diagram to help you picture it.



Basically, part of the original sound or signal (which is called the "DRY" signal in studio parlance) is sent to the FX/Reverb unit via the SEND jacks.  How much of the original signal is sent to the FX? This is controlled via the "Send" or "Fx" knobs on your channel mixer.

So this "DRY" signal goes to the Effects Processor where the hardware does it's magic of giving it reverb, chorus or delay. The output of the effects processor is called the "WET" signal.  This signal is sent back to the mixer via the RETURN jacks. Your mixer in turn mixes the signal from the RETURN jack to the original "DRY" signal, so you get a combination of "DRY" and "WET" signal. 

By varying the amount of "DRY" or "WET" signal with respect to each other, you can control how obvious or subtle is the effect you're trying to add.


On this diagram of a Mackie 1402VLZ mixer, you can see (2) Aux SEND jacks... labeled (1) and (2).

You'd also see 2 PAIRS of RETURN jacks... labeled (1) Left & Right and (2) Left & Right.

I know what you're thinking... how come there is only one jack for the SEND and two jacks for the RETURN?  Here's the thing... most effects processors only need (1) channel to work it's magic. The resulting signal is often stereo. For example... you sang into your SM57 and now need to add a hint of reverb on your vocal tracks. You only need (1) channel to send the vocal track to the FX unit. Add some reverb... and the FX processor gives you a Left and Right "WET" signal.  Now, when you listen to your vocal tracks (with the reverb added) it's in stereo and sounds very nice, full, and you can picture yourself singing in a concert hall or something.

Some additional notes....

The above method is just one way of using FX Processors. There are other ways to use it, but this is the most common way of using it.

The ALT3/4 jack on your mixer will always be DRY. So if you have the ALT3/4 jack outputs of your hardware mixer connected to your soundcard inputs, you won't get the FX.

However, the MAIN OUTS of your mixer will have the DRY + WET signal. Of course, it also has the "mix" of all your channels.

Some high-end FX processors (like the MPX550) can also accept digital inputs (via SPDIF) and output the WET signal also as a digital signal.

Yes, you can use a hardware FX processors in your Cubase, Sonar, Logic, DP or whatever sequencer you have.  In this case, the SENDs jacks will be coming from the analog output of your soundcard, going to the FX unit, then back to spare analog inputs on your soundcard. This is assuming you have multiple analog in/outs soundcard (or audio interface). The disadvantage of this setup is it's going to eat some of your audio interfaces input and output channels. The advantage is you can have a nice quality FX without using up any cpu resources. (Plus some say hardware is better.)

You can avoid using precious analog inputs and outputs on your soundcard, and still use your hardware FX unit by using digital SPDIF in and outs. Of course, this is assuming your soundcard also has SPDIF inputs and outputs. This is how I'm using my Lexicon MPX550 connected to my MOTU 828mk2.

I'll show you how to setup your sequencer in your computer to use your hardware effects processor this way. But that will be on another day's topic.

A mic can have different pickup patterns. What do I mean by that? I mean, a mic does not only pickup the sound that is in the immediate front of it, it may also pick up sounds behind, or to the sides of the microphone.

Here are the different kinds of patterns. Understand them when looking for a mic to buy and your intended application.



Bi-Directional

This is also called figure-8. It picks up sound in front and rear of the diaphragm, but does not pickup sounds from the sides. These types of mics are often used above an instrument or used for "stereo" recording in an M-S matrix technique.

Cardioid

This is the most popular mic pattern. Basically, it looks like a heart.  It picks up sound where the mic is pointed at, but some of the sounds from the rear are also picked up, though not as much. Usually about -10 to -30dB lower.

Just be careful though, because the shape of the cardioid isn't fixed. It varies it's shape depending on the frequency. So it could be sensitive for some frequencies and not for others. This can be used for good use because the mic imparts some "coloration" to the sound.


Sample Cardioid pickup pattern of an MXL 604 mic.

This type of mic is also good for "proximity effect." i.e. the closer you get to the mic, the lower frequencies are hyped up, adding "body" and fullness to the sound. However, sometimes it's too much. So you can do 2 things... one, move the mic farther away, (or you move farther away), or if the mic has a built-in low-pass filter, activate that. (if your preamp has a low-cut filter, you can also activate that).

Omni-Directional

From the word "omni", this mic responds as evenly as possible from all directions.

Stereo Mics

Nowadays, you can buy stereo mics. These microphones have 2 diagphragms in the same body, usually, they're angled toe-in. The capsules are matched for even frequency response for both Left and Right channels. Now... there are 2 kinds of stereo mics.. the cheap ones that are mainly used for videocam cameras (usually, 1/8" jack), and the professional ones complete with XLR jacks.

Rode NT4 Stereo Condenser Microphone

"ALL OF THE ABOVE" Pattern

Due to great advances in technology, there are now mics that are "all of the above" (except stereo). A single mic can be switched to omni, cardioid or bi-directional by the simple flick of a switch. If you're on a budget, these mics are a good buy since it's like buying 3 mics for the price of one.

Rode NT2-A microphone featuring switchable patterns, low cut filter, and pads.

VMware Virtualization for Mac

Woohoo! Looks like Parallels Desktop has competition. I have Parallels and use it to run a Windows dev server on my MacBook. It will be interesting to see what VMware has to offer above Parallels.



What is VMware's virtualization product for Mac?
The new VMware desktop product for the Mac, codenamed Fusion, allows Intel-based Macs to run x86 operating systems, such as Windows, Linux, NetWare and Solaris, in virtual machines at the same time as Mac OS X. It is built on VMware's rock-solid and advanced desktop virtualization platform that is used by over four million users today.

With Fusion, you can run traditional PC applications on your Mac: if you need to run PC applications, you can now do so by leveraging the power of virtual machine technology.

Fusion allows you to:

• Create and run a wide variety of 32- and 64-bit x86 operating systems on OS X without rebooting. You can simultaneously run PC applications next to your OS X applications.
• Leverage Virtual SMP capabilities to gain additional performance improvements. On any Mac with dual-core processors, you can assign multiple CPUs to your virtual machine to gain additional performance for CPU-intensive workloads.
• Access physical devices from the virtual machine: read and burn CDs and DVDs, and use USB 2.0 devices like video cameras, iPods, printers, and disks at full speed.  Even devices that do not have drivers for OS X will work in a virtual machine.
• Drag and drop files and folders between OS X and virtual machines to easily share data between the two environments.
• Leverage the cross-compatibility of VMware virtual machines. VMware virtual machines created with existing VMware products are all cross compatible, including virtual machines created by VMware Workstation, VMware Player, VMware Server and VMware Infrastructure 3.
• Run any of the 360 virtual appliances available from the Virtual Appliance Marketplace (http://vam.vmware.com).

http://www.vmware.com/products/beta/fusion/

There are 2 general types of microphones in the market today. They are 1) Dynamic microphones and 2) Condenser microphones.

There are a few other types like piezo, electret, etc... but we'll discuss those later. For now, let's talk about these two.

How are they different? Which one is better?

First, let's talk about dynamic mics. Dynamic microphones are "mechanical" in nature. No, I don't mean they have gears in them but mechanical movement of the microphone diagphragm causes a small voltage to be induced. Basically, sound waves travelling and causes a moving coil to vibrate in and out of a magnetic field. When this happens, a very small and minute amount of voltage is created. If you amplify this small voltage (using a mic preamp or mixer), you'd hear the original sound.

Does this sound familiar to you? I hope so. Because the same principle works in REVERSE when we're talking about speakers. With speakers, current applied to the coil causes it to vibrate, which produces sound waves which eventually reaches your ear. So think of dynamic mics as very small speakers wired in reverse.

Because of the way they operate, dynamic mics don't need an external source of power like a battery or phantom power supplied by your mixer/preamp.

Also, the voltage generated tend to be on the small side and requires more amplification by your mixer/preamp. Of course, with more amplification and higher gain settings by your mixer or preamp, comes more "noise" because your preamp is going to amplify both the original signal and any noise in the system.

Also, they tend to be less sensitive to sound because it requires a good amount of sound energy to move that diagphragm to cause sufficient vibration to generate an output voltage (or signal).

Some examples of dynamic mics are the Shure SM57, SM58.

Shure SM57 Cardioid Dynamic Microphone View more products from Shure The SM57 is a cardiod (unidirectional) dynamic microphone with a contoured frequency response of 40 to 15,000 Hz, perfect for clean reproduction of vocals and instruments. Read Reviews... List Price: $146 Click for Sale price   FREE SHIPPING!

Shure SM58 Dynamic Handheld Microphone View more products from Shure Consistently the first choice of performers around the globe, the SM58 vocal microphone is a genuine world standard and a true audio legend. The Shure SM58 is a unidirectional (cardioid) dynamic vocal microphone designed for professional vocal use in sound reinforcement and studio recording. Read Reviews... List Price: $188 Click for Sale price   FREE SHIPPING!

Condenser microphones on the other hand work on the principle of capacitance. "Condenser" is another word for capacitors back in the olden days of vacuum tubes. So anyway, a capacitor has 2 plates and it's capacitance is dependent on several factors like area of plates, distance between plates, and dielectric used. Anyways, for this discussion, one of the plate is made to vibrate by sound waves. This effectively varies the capacitance of the condenser diagphragm.

This varying capacitance is proportional to the displacement of the plate, which is proportional to the strength of the sound waves. Current flows through the wire when the distance between the plates change (caused by the sound vibrations). This is a very small current that must be amplified before it even goes to your mixer/preamp. So yes, condenser microphones have little preamplifiers inside them.

Also, unlike dynamic microphones... condenser mics require power to operate. This may come in the form of a battery housed inside the mic unit, or via the microphone cables through the XLR jack, supplied by the mixer/preamp. This is called phantom power. Basically it's 48Volts supplied by the preamp to the mic.

Condenser microphones are very, very sensitive. It can pick up the sound of your breathing easily, or air coming out of an A/C duct, or the fan noise in your computer. So if your bedroom (ahem, recording studio) is very noisy... using a condenser microphone may leave you feeling frustrated as every sound will be captured by it.... even sounds/noise you don't like.

Here are some examples of condenser-type microphones.

Neumann TLM49 Cardioid Condenser Microphone View more products from Neumann The TLM 49 is a large-diaphragm studio microphone with a cardioid directional characteristic and a warm sound which is especially optimized for vocal performance. It is supplied as a set, with an elastic suspension. Read Reviews... List Price: $1699 Click for Sale price   FREE SHIPPING!

Rode NT1000 Studio Condenser Microphone View more products from RODE With its ultra low-noise transformer-less circuitry the Rode NT1000 brings new standards to the recording industry. Read Reviews... List Price: $599 Click for Sale price   FREE SHIPPING!

Rode NT1A Studio Condenser Microphone View more products from RODE The Anniversary Model NT1-A is a complete redesign of the now legendary NT1 classic studio microphone. From the new nickel-plated body to the state-of-the-art surface mount electronic circuitry, the NT1-A will leave you asking "how can RODE offer a microphone that sounds this good, for so little money?" It takes advantage of the huge RODE investment in advanced large scale manufacturing allowing premium performance, durability, consistency, and construction at a price that anyone can afford. Read Reviews... List Price: $349 Click for Sale price   FREE SHIPPING!

So which type of mic is better? Condenser or Dynamic?

The answer is.... NEITHER.

There is no clear winner with regards to which mic is better performing. There are crappy dynamic mics that will get beaten by condenser mics and there are condenser mics that will get beaten by dynamic mics.

We talked above how condenser mics require power to operate in the form of a battery or phantom power. This requires that the mixer/preamp you picked/used can supply phantom power. Otherwise, your condenser mics will be useless. In this case, if you have a non-phantom power capable mixer/preamp, you're better off with a dynamic microphone.

But dynamic microphones can be heavy since it requires better shielding. (Remember, inside the mic are coils and magnets that generate electricity.) Dynamic mics are susceptible to stray magnetic fields which will cause noise to be generated in the mic. What do I mean by stray magnetic field? hmmm... like motors, ballast in flourescent lights, high-tension power lines, and things like that.

The thing is, there are excellent dynamic mics and there are excellent condenser mics. One should look at the bigger picture, and the required application when choosing a mic. You can't make a generalized statement that X is better than Y.

On a previous post, I posted a schematic of a passive direct box using Jensen Transformers.

I have a ROLLS DB25 passive direct box in my studio so I decided to open it up to see it's "guts".

The ROLLS DB25 is an inexpensive direct box using all passive components. That's right, no need for a 9V battery or wall wart to use this thing. Plus, it's got a transformer inside that takes care of converting from an unbalanced Hi-Z connection (from a guitar) to a balanced Lo-Z connection for connection to your mixer or preamp via XLR jacks.

In addition, it has a -20dB and -40dB pad, and a ground lift switch for "stubborn hum" reduction.



This DI box is very inexpensive at less than $30 per unit. 

Input impedance is 50Kohms, with an output impedance of 600ohms. Max Output level is +4dBm (loaded). Frequency response is 50Hz to 15Khz +/- 3dB. Okay, I know what you're thinking... only 50Hz? Only 15Khz?

Before you get crazy, this unit's application is for electric/bass guitars.  You don't need 20Khz response or down to 20hz response for this application.

So let's crack open this thing and see what makes it tick.

The unit's case is made of steel. This gives you strength and it's small size is perfect for the cramped stage (or recording studio).

Here, you can see the -20dB/-40dB pad switch and the ground lift switch. You can also see the XLR male output jacks.




And these are the guts inside the unit. I've labeled them for easy identification. If you buy the unit, there is a schematic included so I won't bother posting it here.



This is a nice unit featuring transformer-based isolation. Additional features like pads and ground lift switches makes this a versatile unit. You should have at least one in your studio! Come on... it's cheap at less than $30. 

Click here to BUY this DI Box.

This is a schematic plan for building a passive, transformer-based DI Box (Direct Box) from Jensen-Transformers.com. View the original PDF here.

You plug your hi-Z (or high impedance) guitar to jack J1. From here, it goes to a pad on/off switch. You also have 2 pad choices... -10dB and -20dB depending on the position of switch S2.

S3 is a hi-cut filter switch. If S1a (i.e. the pad switch) is switched ON, and the hi-cut filter is ON, the high frequencies are shunted to ground via C3.

The signal going back to your guitar amp via jack J2 is unaffected by the pad and hi-cut switches.

Transformer T1 is a JT-DB-E transformer. This takes care of matching impedance between your guitar and the mic preamp of your mixer. It also converts the guitar signal from unbalanced to balanced connection.  The red-brwn wires of the transformer goes to pin 2 and 3 of the XLR which forms the HOT (+) and COLD (-) connections.

Ground is connected to Pin 1 of the XLR. A ground lift switch is provided, S5, which isolates the pin 1 ground of the mixer from the rest of the circuit.

If you don't want to build one, here are some passive DI boxes.

ART ZDirect Passive DI View more products from ART The Zdirect is a high-quality, totally passive interface that lets you connect instrument-, line-, or speaker-level signals to a mixer or other balanced input through a high-performance audio isolation transformer. The high impedance single-ended 1/4 in. input is converted by the transformer into an isolated balanced low impedance signal source. Read Reviews... List Price: $30 Click for Sale price   Shipping cost: $4.99