Dynamics units are to either plug-ins or outboard gear (sometimes even inline in a desk) that are used to control the levels of sounds, or more specifically the range of levels a track can have. There are four main categories of dynamics units, but they are all closely related, these include: Compressors, Limiters, Expanders and Gates. All of these affect what is referred to as the dynamic range. The Dynamic range can be defined as the range of volume any sound has from it’s quietest part to it’s loudest part. Being able to control the dynamic range of a signal offers the engineer a powerful tool when balancing a mix.

A good starting point to explain dynamics units is compression. As the name implies compression compresses or shrinks the size a signal is allowed to operate in. The main advantage of being able to to this is that a signal can be made more “consistent.” By more consistent we mean that because the signal now has a restricted dynamic range the overall signal level will be more predictable. For example if a vocal seems get louder and softer during a song (and that is not the artistic intent) compression could be used to make the softer parts louder and the louder parts softer all at the same time. Once applied the overall level of the signal (the track fader or gain control on compressor) can often be raised to suit the mix.

If you haven’t realized it yet you will soon, audio engineering is all about compromise because “there is no such thing as a free lunch.” The compromise with compression is that by compressing the signal you are also raising the noise floor. Notice in the image below that by making the quietest parts louder we have also made the noise that was quiet, now louder. That is if the noise floor was originally between 0 and 2 dB, it is now between 10 and 12 dB. This means that if you compress very heavily you will start to hear more of the noise in the recording.

Now that we have discovered the balancing act that we must play with compression, let’s take a look at what we can actually control to keep our footing. Compressors will almost always have the following controls. Threshold, Ratio, Attack time, Release Time, Input level, and Output level (Gain). We will follow the signal flow through the compressor to explain what each control accomplishes.

Input level controls the level of the signal as it enters the compressions stages of the compressor. If the control is left at 0dB the signal hitting the compressor should be the same as what is coming off of the recorded media or previous effect. If you want to raise or lower the signal level prior to compression this is where you would do it.

Threshold is one of the controls you will use to control how much compression is to be applied. This control allows you to decide when (that what level) the compression will kick in. Setting a very low threshold will result in heavy compression, while a higher threshold will result in lighter compression. The threshold basically sets the point that the compressor says oops the sound is getting too loud, time to apply some compression to keep things under control.

Ratio, is the other control that determines how much compression is to be applied. If the ratio is 2 to 1 (2:1) that means that signal above the threshold is halved. That is if the threshold is set at 60 dB and the signal that is input peaks at 80 dB the amount of signal above the threshold is 20 dB. With a ratio of 2:1 the peak will be reduced by 10 dB (20 divided by 2) and the resulting signal will peak at 70dB rather than 80dB. A ratio of 5:1 would yield a 16 dB decrease or a peak at 64dB rather than 80dB (20 divided by 5 = 4dB, 20-16=4dB). If the ratio is greater than 50:1 the compressor is considered a limiter because at this point the compressor is basically “clipping” off the peaks and making them level with the threshold.

Attack time refers to the amount of time it takes the compressor to react to signals that breach the threshold. I could also be thought of as how sensitive the compressor is. The image below illustrates that a longer attack time allows more of the full transient (a sudden change in dynamic or level) through before clamping down on the signal. This is typically good for drums and percussive elements as the attack or transient of these sounds are essential to their character. A shorter attack time allows more control over the overall dynamic range this can be good for things like rhythm guitars and fingered bass.

Release time refers to the time that it takes the compressor to release control of the dynamic range back to the original signal. Say for instance the guitarist plays some very heavy loud stabs followed very quickly by some soft palm muted strums. The waveform may look something like the image below. Notice the effects of the different release times. A slow release time still compresses the quieter parts and does not allow them to be as loud as they would naturally be. With the medium attack time you can see that the quieter parts gradually increase in level to their natural level. Finally a fast attack time allows the quiet parts to return to it’s natural level very quickly. The slow and medium attack times in this case are examples of what is commonly referred to as a pumping or breathing compressor (sometimes this is a desired effect, but commonly it is a sign of over compression).

Output level or (make up) gain allows you to control the output signal from the compressor back into your channel. Often after compression the overall level of the signal is decreased. Most times this is the intent, if you used compression to tame an extra loud part of signal you can now raise the overall signal in the mix. Sometimes there is an auto gain setting that attempts to match the output signal to the loudest part of the input signal. This never seems provide a level I’m happy with so I prefer to set the output level manually. Remember the issues of the noise floor when setting the output gain, if there is too much of the noise floor in the signal after the make-up gain is applied, you may have to go back and re-adjust the ratio and threshold to make a compromise.

One of the most mystifying topic of the studio is impedance. Impedance is basically what it sounds like, anything that impedes the flow electrical current. Impedance manifests itself in three ways: Resistance, Capacitive Reactance, and Inductive Reactance, the most commonly referred to of these is Resistance. The letter assigned to impedance is “Z” which is why you’ve probably heard the terms Hi-Z and Lo-Z fly around from time to time. Hopefully the following will clear out some of the fog.

For the time being let’s call the microphone the source, and the mic pre-amp the load. A typical dynamic microphone like a Shure SM58 (R), will have a source impedance of approximately 200 Ohms. As a general rule you would like to have a load impedance at least five times the value of the source impedance, otherwise signal strength is lost. The mic pre of a Digidesign(R) Mbox 2(R) is 3.5k Ohms which is well beyond the desired 1k Ohms. Similarly if you would like to connect a synthesizer such as a Yamaha Motif to a line input on the Mbox 2 (R) is good to check the impedances. The Mbox 2 (R) line impedance is 10k Ohms and the Motif’s output impedance is XXXXXX. Some equipment (especially some vintage and specialty synths) need to be passed through a DI (Direct Injection) box to achieve an appropriate mic level.

This brings us the the issue of DI boxes and guitars. The nature of guitar pickups makes it difficult to adhere to the load five times greater than the source rule. Guitar pickups are inherently much higher impedance (10k to 50k Ohms). This means even though the typical instrument cable you connect to your guitar will also fit in the line input, you will likely get little or no signal. This is because you likely have a higher source impedance the load impedance, thus grossly breaking our general rule (load impedance > 5 X source impedance). The DI box was designed as a solution to this problem.

The DI box is an impedance converter allowing you to pass the same signal through, but creating a new source impedance. The Art Zdirect passive DI box offers an input impedance of 50k Ohms and an output impedance of 600 Ohms. Due to the fact that the signal is typically hotter coming from a guitar (higher in voltage, especially from humbuckers) the 50k Ohms input is acceptable as the mic pre will not be expecting the hotter signal off of the guitar pickups. Therefore it is possible to record a guitar straight into the Mbox 2 (R) through the DI box (600 X 5 = 3k Ohms < 3.5k Ohms, OK!)

Some audio interfaces offer instrument inputs as well. These inputs are designed to accept a guitar or other hi-Z source and often have a load impedance in the ball park of 2 Mega-ohms or greater.

Having a good grip of how impedance works in the studio is the first step to attaining a good signal to noise ratio.

There are two main types of connections used in a studio to carry audio signals. These connections are balanced and unbalanced. When possible it is always better to use a balanced connection.

Unbalanced connections:
Unbalanced connections consist of two contact points, one is considered hot, the other ground (0V). Unbalanced connections transmit the signal (the desired information) across the hot wire, and the information is interpreted with respect to ground (0V). The issue with unbalanced connection is that they are more susceptible to electro-magnetic noise than balanced connections. The best way to explain why, is to explain how a balance connection works.

Balanced connections:
Balanced connections typically (there are a few exceptions) consist of three contact points. One is considered hot, another cold, and the final one a common ground. Unlike the unbalanced connection the hot and cold connections both carry a signal and it is the difference between these two signals that carries the desired information. In this way both the hot and cold connections are exposed to generally the same electro-mechanical noise. Therefore when this difference between the hot and cold signals is extracted the noise on both lines will phase cancel yielding only the desired signal information.

*Note: Using a balanced cable alone does not create a balanced connection, both pieces of gear and the cable must support a balanced connection to create a balanced connection. (Example: Using a balanced cable to plug an electric guitar into and amp will work, but not create a balanced connection because the guitar and amp will most likely not support a balanced connection.)

Before we can understand digital audio, we must first have a very general understanding of how sound occurs naturally. Sound is actually a mechanical wave that is propagating through the air (typically, but it can travel through other mediums as well). A basic visual illustration of sound is the sine wave.

A sine wave a represents a pure tone which is a single frequency. This sounds something like the emergency broadcast signal or the sound the TV makes when the colored bars appear, a long sustained beep. All sound is made up of these types of waves at different frequencies superimposed on each other (this forms the waveform you see on your DAW when you are editing).

Analog recording such as to a magnetic tape or a lathe (for vinyl records) used a microphone as a transducer (makes the same wave in the air, but with electricity), and stored the wave information either magnetically or with actual grooves as on a record.

Digital audio is recorded differently. One can think of digital audio being captured and recorded by taking very fast snapshots of the waveform (on the order of more than 44000 snapshots per second). This process is called sampling.

This is where digital audio really shines. Once the waveform or “data” is captured in this digital form, it becomes very malleable when using a computer and mathematical formulas. This is how we gain the power to do non linear editing and apply convolution algorithms.

Sample Rate:
Sample rate is measured in Hertz which is a unit per second (samples/seconds). There is a direct correlation between how accurately sound is recorded and the sample rate. That is, the higher the sampling rate, the more accurately the waveform is recorded. The downside of this is that often each discreet sample takes up at least sixteen bits of storage space. That means 44.1 kHz sample rate and 16 bit bit depth (CD quality) about 88.2 KB of space is need per channel.

Bit Depth:
Bit depth correlates to the number of different levels of dynamic range (volume) possible. So the higher the bit depth the higher the resolution of the dynamic range. Another way to look at is is this. The human ear can tolerate 0dB (silence) to 120dB (threshold of pain) of dynamic range. We gain roughly 6dB of dynamic range for every bit, therefore 16 bits will give us 96dB or dynamic range. (24 bits will yield 144dB of dynamic range)

In summary higher sample rates and higher bit depths yield higher fidelity recordings, but they are also more taxing on processors, and take up more storage space on a hard drive. Remember that if you want to apply Digital Signal Processing to your mixes and you recorded your tracks at 88.2 kHz every plug-in you use will use roughly twice as much CPU time compared to tracks recorded at 44.1 kHz. 99% of people can no longer tell a difference beyond 48 kHz, 24 bit, and most are happy at CD quality 44.1 kHz, 16 bit (it will probably end up in this form anyway).

1. Know your gear

Historically this statement meant: know your outboard gear as well as your console if you had one. In the context of most home recording studios these days (where most mixing if not all is done “in the box”) it means, know the strengths and weaknesses of digital recording in general, and know your plug-ins. When it is all said and done a compressor is a compressor and an EQ is an EQ, and the main difference between EQ A and EQ B is some mathematical formula used for DSP. The controls presented to you are your only input to the equation so learning how to manipulate these controls to get the sound you want is paramount. Very seldom will you find something “out of the box” that will perform the task you need perfectly, and more importantly, just because you paid $X,XXX dollars for it doesn’t mean you can get the sound you want out of it, or if anyone can for that matter. At the end of the day you can only get the sounds out of your gear that you know how to, so the more sounds you how to get the stronger and more versatile engineer you will be. Invest your time in learning what you have well before investing your hard earned cash in too many plug-ins that will be useless until you really learn them too.

2. Use Your Ears

Now here’s an novel idea for a sound engineer/musician: Use your bleedin’ ears. With all these beautifully designed GUI’s that we are presented with these days it’s easy to get distracted and start mixing with our eyes. I’ve even caught myself thinking “yeah that looks about right.” It’s good always remember that sound doesn’t look like anything unless you are on hallucinogens.

3. Trust your Ears

Just because certain guidelines or “professionals” say to do it one way doesn’t mean that you should always do it that way. Remember that every mix is unique and every mix will have it’s own sound and feel. Read the lyrics or listen to the demo to try and understand the message that the singer is trying to portray and use the mix to enhance this message. Come up with a plan for the mix in your head and trust your ears to implement it. You can often even feel the tension in your shoulders release when the the sounds you are creating match the sounds of the plan in your head.

4. Get a Second Opinion

Your ears will undoubtedly get tired and often you will become hyper focused on tweaking a very particular part of the mix. This is always a good time to take a break, let your ears rest, and find someone who’s opinion you value to listen to what you’ve been working on. Often times people will agree that the sound you are aiming for is worth pursuing, but they won’t necessarily know how to achieve it. Typically you are trying to create a mix that will resonate with the masses. As an engineer you have the know how to get the mix to sound how you would like it to sound, but you must remember that you are providing service to a client (even if you are the client as well, it helps to consider the workflow this way).

5. Use multiple monitors

Using multiple monitor sources is very important to creating a good balance in a mix. Always consider where your mix is most likely to be played and make sure that it sounds good on that source. Use your studio reference monitor primarily, but demo you mixes on a home stereo, computer speaker and in headphones as well. Not every one has the caliber of speakers that we use for reference monitors and the most assuredly will not have the same frequency response. If you can demo the mix on a audio system with a well balanced sub woofer this can help a great deal when balancing the bottom end.