HomeStudioGuide.com - Guide to Home Studio Recording

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

Radial JDI Duplex Stereo Passive DI Box View more products from Radial A stereo version of the Radial JDI. Ideal for recording, broadcast, and live sound where extreme dynamics such as those produced by digital sampling devices and keyboards are encountered. Completely passive, the JD4 employs two Jensen JT-DBE transformers for 100% isolation and low phase distortion. Read Reviews... List Price: $350 Click for Sale price   FREE SHIPPING!

Radial JDI MK3 Passive Direct Box View more products from Radial The Radial JDI is considered by many to be the world’s finest direct box. It is a passive DI that employs a Jensen isolation transformer for optimum audio performance offering outstanding linearity at all frequencies, combined with extraordinary level handling without introducing distortion. Read Reviews... List Price: $200 Click for Sale price   FREE SHIPPING!

Radial Pro DI Passive Direct Box View more products from Radial The Radial ProDIs are high-quality, full-range passive direct boxes equipped with custom-made audio transformers for exceptional signal handling without saturation and with extremely low phase distortion in the critical bass and mid regions. The result is exceptional clarity and definition at an attractive price point. Read Reviews... List Price: $100 Click for Sale price   Shipping cost: $4.99

Just an update of my build: Click here if your want to read Part 2.

Soldered 90% of the parts, including the Lundahl transformers, JFETS and transistors. I'm awaiting some parts that were "out of stock" from my first order.

Also, I ordered some wire assemblies with male/female jacks. I'll use them for connections between the main board and xlr jacks, pots, switches, etc... that way, if I need to troubleshoot the board, I can just unhook them instead of desoldering the wire from the board.

TIP: Before you complete assembly of your project, have a suitable chassis ready for it. From my experience, if I built a project without finishing the chassis first, that project becomes half-finished, working but not in a case. Laziness I know...

Also, if everything is in a chassis, you can wire everything and not have to worry about your solder connections or wires becoming loose. Everything is already in place, plus it makes it easier to work on it.

You can get your rack chassis, from 1u, 2u, all the way to 4u and 8u at par-metal.com.  They've got good prices and have nice quality racks.

Looking for great compressors? I think these are some of the stand-outs among the pack.

dbx 160A - This is a classic, introduced way back in the 70s. Easy setup and accurate metering. This is an industry standard with the legendary "Overeasy" feature to transparently smooth out and maintain a constant level for vocals and instrument levels.

Empirical Labs Distressor - The distressor is usually always seen whenever you see photos of professional recording studios. It's unique look with those 4 big knobs is hard not to miss. This is a digitally controlled compressor, that switches different circuits in and out. So you can have programmable analog distortion and warmth. It can replicate the compression that occurs with tape using the Distort 3 mode. Side chain EQ, and eight unique compression curves, to the Nuke setting, that is awesome on drums. It also has an "opto" ratio which uses light controlled components similar to what the LA2A uses to achieve it's compression.


RNC RNC - Really Nice Compressor - This half-rack unit won't win any awards in the "looks good" category, but will blow you away in it's "sounds good" category and "right price" category. Don't let the plastic case and cheap knobs and small size fool yah! This is a high performing compressor that can deliver very transparent results.  It's got two modes.. regular and "SuperNice." Normal mode makes it behave like any other compressor, but Supernice mode gives it an almost invisible effect... i.e. there is compression going on, but you don't even notice it because it doesn't produce harsh artifacts. Great for compressing vocals, acoustic guitar, or the whole stereo mix.
RNC

Universal Audio LA-2A Teletronix limiter - A unique electro-optical attenuator system allows instantaneous gain reduction with no increase in harmonic distortion. It features 40 dB gain limiting, balanced stereo interconnection, and low noise – less than 70 dB below +10 dBm output. Controls are gain, peak reduction and meter selector, and connections are Jones Barrier terminals and XLR connectors.  This is classic... no respectable professional studio will not have it. Some  studios has 4 or 8 of these in their rack. It's that good.


More compressors for every budget and application can be found here.

How do you hook up or connect a hardware compressor? Compressors are used in SERIES with the signal you want to compress. Other signal processors that need to be connected in series are Limiters and Equalizers.

Option 1:

If you're using a stand-alone preamp, you would connect the compressor after the preamp. So it will be:

PREAMP OUT ----> COMPRESSOR IN
Then COMPRESSOR OUT ----> rest of your signal chain (maybe to a mixer or audio interface).

PROS: If your preamp is balanced, and your compressor also has balanced inputs and outputs, then your whole signal chain will be balanced. 

CONS: You'll need lots of cables. Especially if we're talking about a left and right channel setup or stereo mix compressor.

Option 2:

You can also use a hardware compressor with your mixer. If your mixer have INSERT jacks at the back, you can hook up your compressor via these jacks. This will save you some cabling and make your setup more neater. Cons : You'll have unbalanced connections from the INSERT jacks to the compressor, and from the compressor back to the mixer.

Is this a problem? Maybe not. We're just talking about a short run of a few feet here in a "controlled environment." So running unbalanced may not be an issue.

Below is a picture of the back of the mixer showing the CHANNEL INSERT where you can hookup a hardware compressor, EQ, limiter.



But you'd need a special type of cable.  Basically, you'd need an INSERT CABLE, something like this.




At first glance, the black plug (on the photo above) looks like a stereo jack. Nooooo... it's not a stereo jack. It's called a TRS jack. (TRS stands for TIP-RING-SLEEVE).

An INSERT cable is a special type of cable. Basically, the black TRS plug (from the photo above) is wired like this:



This single jack is carrying both the send signal (that will go to the compressor) and the return signal (from the compressor).

I repeat... it's not a stereo jack. It's not carrying 2 channels. There is no left or right channel here. This is carrying only a single channel.

At the other end of this TRS jack, you'd notice there are (2) TS (TIP-SLEEVE) jacks.  The White TS jack is labeled TIP.  The RED TS jack is labeled RING. 

The TIP TS jack (WHITE jack) goes to the input jacks of your compressor. 

The RING TS jack (RED jack) goes to the output jacks of your compressor.

HINT:  RED jack (think of the letter "R") is the RING jack... which is the RETURN path from the compressor

Make sure you don't swap the RIP and RING jacks when connecting to your compressor. Otherwise, you won't get any signal.

Compressors are one of the commonly used plugins when mixing/mastering. It's not hard to understand how they work... if you know what each button/knob does and how it affects the sound.

When compressors are used properly, the effect shouldn't be noticeable. ONLY when you compare the uncompressed and compressed signal should you notice the difference in the dynamics. Compressors are often used during tracking or mixdown. And during the mastering process, the whole stereo mix may be compressed and/or limited. 

The most common  controls in a compressor are the THRESHOLD, RATIO, ATTACK, RELEASE and MAKEUP-GAIN. Everything else is fluff... like input/output VU meters, or soft-knee/hard-knee, limiting on/off. 

Think of a compressor as an automatic variable volume control. The "volume" control's behavior is set by the attack, release, ratio knobs. This automatic volume control reduces the dynamics of any audio material that goes above the threshold. And this is where the paradox is. If compessors are used to reduce the volume/dynamics of music, how can it make the music sound loud? Well, the answer is in the MAKEUP-GAIN, where the average level of the music can be raised higher without clipping (chances for clipping are reduced because we just reduced the dynamics!).

So let's discuss the important parameters/control in a compressor...

1. THRESHOLD - This is measured in dB. It's a negative value because we're setting levels below the 0dB mark. Threshold (in dB) is that level at which the compressor should start reducing the output level. So if you set the threshold to -37dB (like in the picture above), signals below -37dB in levels remain unaffected.  Signals above -37dB (or in other words, louder than -37dB) will be reduced by the compressor. By how much should it be reduced? That's a function of the RATIO setting.

2. RATIO - So if a signal exceeds the THRESHOLD value, the compressor should start reducing it. Reduction is measured in ratio instead of a fixed numeric value. So let's say you set a RATIO of 2, any signal increase above -37dB in our example above will be reduced by half.  So 6dB above -37dB will be reduced to 3dB. A high ratio will reduce the the levels by a higher amount.

3. ATTACK - So we now understand that input levels above the THRESHOLD value will be reduced. This means, the compressor is always watching the input and detecting it's levels. The question now is, if a signal exceeds the THRESHOLD level, how fast or slow should we take action (of reducing it)?  This is a function of the ATTACK setting, measured in milliseconds. A fast attack (i.e. low millisecond) will mean the compressor will take action faster than a slow attack speed. Therefore, a slow attack (higher millisecond) setting means the signal that peak may be allowed to pass through. What does this mean? It means the dynamics of the signal may be allowed to pass through normally without being compressed... think of a snare hit on a drum. We don't want to kill the initial dynamics of the snare hit. 

4. RELEASE - Now, if a signal exceeds the THRESHOLD level, at some point in time it will go down below  the threshold level again. If a signal goes below the THRESHOLD level, at what point should we stop compressing? Should we do it as soon as it goes below the THRESHOLD, or a few milliseconds afterwards? This is a function of the RELEASE setting, measured in milliseconds.  A longer setting, i.e. a slow release means the compressor is still holding a "grip" on the signal, i.e. still reducing it. A short or fast setting means the compressor will let go of the signal. Long settings (i.e. a slow setting) produces a gradual less noticeable effect in the sound. Short release times are good for percussion instruments, and long settings are good for vocals. 

5. GAIN - This is also known as MAKE-UP GAIN. If you set a low THRESHOLD value, and a high RATIO amount, the overall signal will be reduced.  So we would want to amplify that signal and this is the function of the GAIN setting. It boosts the compressed signal to a sufficient level that we want.

Great, now that we know the different settings. But what can we use compressors for?

Compressors can be used to "even" out the volume by reducing the peaks in the signal (thanks to the RATIO control).  For example, your singer has bad mic technique and their volume varies a lot. We can use compressors to even out the levels so the vocal levels don't vary a lot.

Compressors can be used to avoid overloading/peaking during recording. (Of course, when used during recording, the change is permanent.)

By setting the appropriate ATTACK and RELEASE settings, we can add punch to our drums, adding impact to our beats.

And compressors are also used in mastering (sometimes too much) which adds punch and volume to the track.  But if a compressor is overused, you could also end up with a crunched sound... losing all your dynamics which makes for a "boring" listen. So don't overdo it. Sometimes, slight compression settings is all that is needed.

Sample Procedure for Compressing

1. First, set the bypass switch on the compressor. Listen to your levels... note how loud they are. Take a look at your VU meters and note the peak signal level.

2. Then take off the bypass (i.e. let the compressor affect the signal).

3. Set the ratio knobs... 2:1 or 4:1, depending on how much you want to reduce the levels

4. Set the attack and release time.  Let's use a fast attack time of 5ms and slow release time of 100 or 150ms.

5. Now, set the threshold level until the compressor is showing a gain reduction of 4, 6 or 8dB.

6. Now, use the make-up gain and set it so that we're showing the same peak signal as before compression (see step #1).

At this point, the sound should be punchier. It's like magic... they both have the same peak signal levels but the compressed signal has more weight and feels more solid.

If the sound is too compressed, and all the dynamics of the signal seemed lost, raise the threshold level (which means you have to lower the makeup gain), or increase the attack time (i.e. make it slower to react). 

The Bypass button is your best friend. Sometimes, subltle compression is all you need so make sure to hit the bypass on/off button often to hear the difference between no-compression and compression.

If you're using compression for the final mix on a stereo track, you may want to combine the compressed and non compressed signal.  This will make the music "thick" while still retaining the dynamics of the original music.


Typical Compression Settings

These are just typical settings, and don't make this the "rule." The exact setting will depend on the material you're compressing and the effect you want to achieve.

Delicate Vocals - We want to set the RATIO to a high value and high TRESHOLD. By using these settings, the softer vocal sections will remain uncompressed and only loud, ear shattering vocals will be compressed.

Threshold - set the threshold (i.e. lower it) so that the loudest vocal section is reduced by 6dB.
Ratio - Set to 6:1 reduction

Make sure threshold setting is not too extreme. We only want the loud vocal sections to be compressed by 6dB and quiet vocals remain uncompressed.

Pop Vocals/Commercials (yes, those annoying TV and radio commercials) - Okay, we want obvious compression here. So we want compression always going on for almost the entire material, and a very big reduction during loud masterial.

Threshold - set the treshold so the the softest vocal section is reduced a little bit. (-1dB or -2dB)
Ratio - Set to 2:1 reduction
Attack - set to fast setting
Release - set to a little bit slow
Make-up gain - increase gain to increase volume again.

At these settings, we're raising the average level of the whole material... i.e. making everything sound loud!

Drums - or any percussive instrument. We want a punchy, thick sound in the mix.

Threshold - set threshold so all drum materials are compressed by -3dB.
Ratio - Set to 4:1
If you have a hard/soft knee switch, switch to "soft knee"
Attack - set to a fast setting
Release - set to a mid setting

By lowering the threshold setting, you can make the drums more compressed. But be careful not to overcompress it too much or you'll lose the dynamics and punch.

Bass guitar/bassy synths - we want a fairly constant level for the bass material so we have a "thick" sound in the mix.  We don't want bass volume going loud and soft, we want it consistent.

Threshold - set so that only loud bass material (i.e. the peaks) are compressed.
Ratio - Set to 4:1
Attack - set to slow/mid
Release - set to slow/mid

Electric lead/rythm guitar - we want compression on these material. We want the guitars to be punchy and level out the volume.

Threshold - set to a low setting, to achieve constant compression
Ratio - Set to 6:!
Attack/Release - play with it

Looking to save some money? Create your own cables! By spending a few dollars on jacks, shielded cables, solder and soldering iron, you can save a ton of money in cables.

It's easy. Here are some step by step photos. 

At the bottom of this article, is a wiring guide for XLR to XLR, XLR to TRS, XLR to TS, and whatever combination you can think of.

On this guide, I'm wiring up an XLR cable.  First, we disassemble the cable by unscrewing the lock screw. Pull out the rubber plug (black thingy) and then you'll be able to push out the 3-pin connection assembly.

DYNAMIC RANGE

The simplest explanation of Dynamic range is the difference between the loudest and quietest parts of the music, as measured in decibels, is called the Dynamic Range.

If you need a review of what is a decibel, click here.

However, there is a dilemma here. How do you define what is the quietest part of the music? It seems an easy definition, but if you consider that electronic audio equipment have inherent self noise, then the quietest part of the music is fainter than the self noise produced by the equipment. i.e. the quietest part of the music is below the "noise floor" of the equipment.

So in this case, the dynamic range is the difference between the loudest part and the noise floor, in decibels.

Are you still confused? okay.... let's say you're attending a symphony concert. Sometimes, the orchestra will be playing full blast! Let's consider this the "loudest part of the music."  Then in one part of the music, only the flutes can be heard... let's call this the "quietest part of the music."  The difference in volume between the solo flute and the whole orchestra playing is the "dynamic range" of the orchestra.  But by some bad luck, you're seated to an obnoxious person that's talking on his cell phone.  You can't hear the flute solo clearly because your seatmate is making so much "noise."  That obnoxious guy, is your "noise floor."  So that means, the dynamic range of your music has been reduced. Because now, all you can hear is anywhere between the loudest part of the orchestra playing and the noisy guy talking on his cell phone.

The above illustration illustrates how "noise floor" can rob you of available dynamic range. 

In real world scenario, we'd want equipment that gives us the lowest noise floor possible. Because this makes the dynamic range available to us bigger.

Just a continuation of the saga of building my 1176 clone compressor/limiter. Click here for Part 1

Parts arrived a few days ago, and other than the audio input and output transformers and chassis, I think I have enough parts to build (2) units. Here are some pics...

First, we have the output LL5402 Lundahl transformer and the LL1540 input transformer. Made in Sweden.  These things are small.

With its exciting new music creation tools, powerful customization and speed-enhancing features, and streamlined post production workflows, Pro Tools® HD 7.3 software is a creativity-charged upgrade that’s built for speed, empowering you to work smarter and faster than ever. Whether you’re composing music or mixing audio for post production, this essential upgrade is packed with a wide range of new features and enhancements to help you make the most of your creativity.

For more information about Pro Tools HD 7.3 software, please visit the Pro Tools HD 7.3 product page.

This upgrade is a full, downloadable installation of Pro Tools HD 7.3 software for qualified computers running Windows XP or Mac OS X 10.4 (PowerPC- and Intel-based Macs). For detailed compatibility information, please visit http://www.digidesign.com/compato. This upgrade does not require a previous installation of Pro Tools software.

If you purchased a Pro Tools|HD or ICON system or Pro Tools HD 7.2 software upgrade on or after November 1, 2006, you may be eligible for a free upgrade to Pro Tools HD 7.3 software. To find out if you qualify, please check the email account you used when registering your product. Email notification will be sent to qualified purchasers with information on how to receive a free upgrade to Pro Tools HD 7.3 software.

Pro Tools HD 7.3 software is not compatible with Pro Tools LE™ systems (such as Digi 002®, Digi 002 Rack™, Mbox® 2, and Mbox 2 Pro) or Pro Tools M-Powered™ systems (hardware peripherals from M-Audio).

There is a known sync issue with simultaneous audio scrubbing in Pro Tools HD 7.3 software and MachineControl. Learn more.

Secrets of the Pros - DVD Pro Tools Vol. 1

And while we're at the subject of ProTools, you may want to get your hands on this as a Christmas gift for yourself  (or your studio). With over 4 hours of information this 2-DVD set will not only show you the basics, but also take you much further into advanced techniques for recording, editing, MIDI, and more. You will see how the software and hardware work, and be shown in detail exactly how to use Pro Tools like a professional.

It also goes step by step through Beat Detective which is a powerful part of Pro Tools that allows you to tighten, fix, and even change the groove of drum and percussion parts.

Your host, Ken Walden, is a former 9 year veteran Digidesign/Pro Tools Product Specialist and Tech, and a seasoned engineer with credits on recordings by several top name artists. Ken delivers all the essentials you need to set up your system and begin recording your music without compromise. Learn the ins and outs of creating and honing Pro Tools sessions, and discover techniques that will give you a huge head start.

Topics
Recording (well refined techniques to make recording easy, and keep the creativity flowing)

Editing (methods used by the best in the industry)

Mixing (signal routing, plug-in info, and more...)

Beat Detective

System set-up (how to get the most power and stability from your system)

MIDI -- featuring Reason (set up, signal flow, and detailed editing info)

Software overview (This will take your through the important parts of the Pro Tools software, and show you how to use thes...

Secrets of the Pros - DVD Pro Tools Vol. 2

And when you're done viewing Volume 1, check out Volume II. This DVD picks up where its highly successful prequel left off. It not only dives deep into advanced mixing and editing techniques, but also includes an overview of exciting new 7.x software features, and provides clear instruction pertaining to file management, latency, buffer settings, and more.

Contents
Advanced Pro Tools DVD Volume II is geared toward those who have some previous Pro Tools experience. Explained with the same clarity as the other Secrets of the Pros DVDs, this volume addresses a variety of the more powerful and esoteric Pro Tools features and recording techniques.

With over two hours of information, this DVD will take you to the next level in terms of your Pro Tools-based audio engineering capabilities and understanding.

Hosted by Secrets of the Pros founder Ken Walden, Advanced Pro Tools DVD Volume II provides a wealth of knowledge and technical prowess from a power user's standpoint. These insights will not only help you to get your head around the whole of Pro Tools, but you'll find yourself moving through projects more effectively and efficiently, making the most of your creativity.

Topics
Advanced Mixing -- More techniques to take you deeper into the Pro Tools mixer

Advanced Editing -- Expert level features that will help release your creativity

Pro Tools 7.x Overview -- An overview of what is new and cool, and how to best use these features...

Okay, here's the second batch of Cubase video tutorials. Grab some cola/coffee/jolt, sit back and relax.

#7 Cubase Tools

#8 Cubase VST Connections

For all you Cubase fans out there... here are some video tutorials to help you make the most use of the software.

#1 Audio Midi Setup in Cubase

#2 New Project

Designed to bring the legendary sound of SSL within your audio workstation, the award-winning Duende audio processor is now available for PC. Click here!

With a simple FireWire connection, Duende delivers up to 32 channels of processing at sample rates up to 96khz. The EQ and Dynamics Channel provides a single plug-in slot EQ & Dynamics processing solution with all the power and processing flexibility of an SSL console channel.

Key features include:

EQ & Dynamics Channel

4-band EQ, two shelving sections and two parametric
Variable low-pass and high-pass filter
Switchable EQ characteristics between E Series and G Series EQ
“Over-Easy” soft ratio compression characteristic for smooth transitions
Variable process order routing
Dynamics side-chain processing with independent side chains for compressor and expander/gate
Bus Compressor

The Duende Bus Compressor delivers the punch and drive of the classic SSL Master Bus Compressor, a key element of many legendary recordings. Widely regarded as ‘audio glue’ for a mix, its sound is an essential component in creating a great mix.

More information and the v1.5 software is available at: www.solid-state-logic.com/duende

I've seen the new Zoom H4 Handy Recorder. I think it's a neat little thing! And best of all, it's "cheap" at $299. Of course, "cheap" is a relative word, but compared to other gear out there, this seems to be one of the low priced new entry. It looks good too!

While reading the H4 specs, this caught my eye... "The H4 features 2 studio-quality electret condenser microphones configured in an X/Y pattern for true stereo recording."  The keyword here is "electret microphone." And the picture seems to confirm that it's really just an electret mic.

So I'm thinking, maybe I can DIY my own stereo mic.

I've used an electret microphone on a non-audio project. I made a sound-triggered flash sync for my Canon EOS30D camera. In this case, I just used the audio output from the mic to trigger an SCR, which then triggered the camera flash.

Can we used the same electret mic to create a good enough sounding stereo mic? I think we can.

So I begin adapting my sound-triggered flash sync and while doing some Googling around, I came across this product.
Stereo Super Ear Amplifier Kit

It's a kit made by Velleman and it uses 2 electret mics, and a headphone output jack. Hmmmm.... interesting. So I ordered a kit (hey, it's less than $10 so if it didn't work, it's not too much of a loss). You can use the link above to order your own kit.

The circuit is simple... the Left and Right channel is the same, so the operation of the circuit is identical. Basically, an electret condenser microphone is biased for operation using a single resistor and capacitor. The output of the microphone goes to a potentiometer which acts as the volume control. This is then amplified by the NE5532 opamp, which is then routed to a headphone jack. Since the whole circuit operates from a single supply voltage (4.5Volts), the opamp uses a virtual ground by the use of 2 resistors in it's input pin. This of course, means that each leg of the split power supply will be only Vcc/2, where Vcc=4.5Volts.

I built this kit in less than an hour... while watching TV. And construction isn't hard and it's very simple. So if you're ready to jump into this Electronics hobby and looking for your first audio project, why not try this stereo mic kit? It's only less than $10.

Here are some pics during construction... enjoy.

What is the difference between dBu and dBV?

If you've followed my previous articles, dBu is 0.775 Volts at 0dB.

But just to let you know, dBu is the same as dBv (with the lower case "v"). i.e. 0dBv = 0.775 Volts. The National Association of Broadcasters adopted the dBv term for 0.775Volts reference = 0dBv

Now, don't confuse dBv (lower case v) with dBV (upper case V) which is referenced to 1 volt = 0dBV

Now, the good news is today, dBu (with the letter "u") is the preferable usage. That is good... and that would avoid confusion with the use of the letter "V" or "v". 

It's that time of the year... here are some gift ideas for your home studio.

Furman PL8II Rackmount Power Conditioner with Lights
So you have thousands of dollars in gear and you're plugging it "naked" to your wall outlet or with a $5 Walmart power strip? I recommend you buy this power conditioner that will give you added protection to your gear. All your gear plugs to the back and you have a single switch to turn on all your gear. The lights are very handy too illuminating your rack gear so you can see what you're doing.



Korg MA-30 Metronome
Keeping time is important. And this little device from KORG packs a lot of sophisticated features at a small size. You can do rythms such as triplets and quadruplets with inner beats removed.