What’s What in the Studio?

text by Madeleine Campbell for Women in Sound #2
illustrations by Maggie Negrete

Author's note: This article was originally published as a four-part series for She Shreds Magazine in Spring 2015 and has since been expanded to include more content and formatted to one comprehensive piece. 
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A PROFESSIONAL RECORDING STUDIO CAN BE AN UNWELCOMING, INTIMIDATING AND EXCLUSIONARY ENVIRONMENT, ESPECIALLY FOR THOSE WHO ARE NEW TO RECORDED SOUND.

Though gear and its arrangement vary widely from studio to studio, the fundamentals of audio interconnectivity remain consistent. Below is a primer on how sound is recorded and played back intended for beginners who want to expand their foundational knowledge but may be unsure where to start.

I: MICROPHONES

No matter the environment, your choice of microphone will impact the outcome of your recording. You have numerous options! There are many ways to achieve a great sound and just like instruments, amplifiers and human voices, no two microphones are exactly alike. Your microphone choice should ultimately depend on what meets the needs of your project. Microphones fall within a huge price range. Some are $15.00 and others are $15,000.00. Remember that using your gear well is far more important than buying expensive equipment for the sake of having fancier tools - A.K.A. just because it costs more doesn't mean it's necessarily better for your project.

All microphones serve the same basic function. They are acoustic-to-electric transducers. The term “transducer” describes any device that changes one form of energy into another. Microphones convert acoustic sound waves into a small electric signal that is then amplified, recorded and played back. This conversion is caused by the microphone's diaphragm, a thin piece of material suspended within the microphone that vibrates when struck by a sound wave. This causes surrounding internal components to vibrate and induce the electric signal. 

Microphones are commonly categorized based on their polar pattern and transducer principle. A polar pattern describes how the microphone picks up sound from all directions and is depicted by a circular graph representing 360° around the microphone's capsule. For example, an omnidirectional microphone picks up sound equally from all directions. This comes in handy if you are recording group vocals and want to use one microphone surrounded by many people. Conversely, if you are recording a solo instrumentalist performing for a live audience, placing an omnidirectional microphone in front of them will capture the sounds of the crowd equally well, including all of the coughs, sneezes and seat rustling. Microphones with a cardioid polar pattern pick up sound from the front and reject it from the back. This pattern is named for its heart-like shape. Bidirectional, also called figure 8, microphones pick up sound from the front and the back but reject it from the sides. (Think of how the numeral "8" looks for reference.)

A microphone's transducer principle describes how it converts acoustic sound waves to an electric signal. Here are a few common designs and how they differ: 

DYNAMIC microphones work by electromagnetic induction. The diaphragm is attached to a small wire coil that is fixed around a permanently placed magnet. As sound waves enter the microphone, the diaphragm vibrates causing the attached coil to move around the magnet. This produces electric signal within the coil. Dynamic microphones are rugged and durable. They handle high levels of audio signal and are often used to record drums or loud amplifiers. They are generally less expensive than condenser or ribbon microphones. Shure's SM57 is a common dynamic microphone that has been used on countless popular recordings. It will only set you back about $100.00 and can be used in a wide variety of recording situations. For drums, the 57 is a standard snare mic. Throw one on your guitar amp's speaker and move it around to see how it sounds in different positions relative to the center. If your amp has two speakers, try pointing one towards each for a thicker sound. Several manufacturers make low cost dynamic mics, like the Audio-Technica ATR2100USB and the Nady SP4C, which runs for less than $20.00, that are great for experimenting on a budget. 

RIBBON microphones also work by electromagnetic induction. The diaphragm is a small piece of extremely thin ribbon that is corrugated and suspended into a magnetic field. As sound waves enter the microphone, they cut across the ribbon’s slits causing it to move back and forth. This creates an electric signal within the magnetic field. Because of this fragile design, ribbon microphones generally are more sensitive than dynamic microphones and have a lower output. If used on a guitar or bass amp, make sure to either point it off-axis or keep the gain at a reasonable level. Many ribbon microphones offer a darker, warmer sound that can work well on certain acoustic guitars, as drum room microphones or on reed instruments. Historically, ribbon microphones feature a bi-directional polar pattern, meaning they pick up sound from the front and back but reject it from either side. However, innovations in ribbon design mean modern ribbon mics are now able to handle more signal and feature alternative pickup patterns. The Beyerdynamic M160 is a good example of this. It features a hypercardioid pattern, meaning it picks up sound from the front and sides and rejecting sound from the back. 

CONDENSER microphones function differently than dynamics and ribbons. These microphones have two plates inside. One is stationary and one fluctuates as it responds to changes in pressure from sound waves entering the microphone. These movements between the plates induce an electric signal. Because they are generally more sensitive to the nuances of sound than dynamic microphones, condensers are often used on string instruments and vocals. They require an external power source, usually in the form of 48V phantom power which can be activated on most audio interfaces and recording consoles. Neumann’s U87 and KM184 stereo pair, as well as AKG’s C414, are all notable condenser microphones found in many professional studios. 

Slight variations in microphone placement can noticeably alter the outcome of recorded sound, however, there is no right or wrong way to mic your instrument or amplifier. Finding out how to best capture your specific sounds may take time. Don’t hesitate to try an unconventional technique. 

II: USB/FIREWIRE AUDIO INTERFACES

For decades, even the world’s most renowned audio engineers were able to record only a couple tracks at a time. As the 20th century progressed, recorders and consoles with more inputs and channels became available professionally and commercially. The development and integration of various recording softwares into our digital audio workstations (DAW) made the number of tracks available to us today essentially unlimited. If you’re interested in building or expanding your own digital recording rig, you’re going to need an audio interface you can trust.

What is an audio interface? 

An audio interface is the hardware that connects your microphones and/or instrument inputs to your computer and recording software. This is a critical part of your DAW that performs several key tasks. The first is conversion, specifically analog-to-digital and digital-to-analog conversion, or A/D and D/A for short. When audio is recorded, most interfaces will convert analog signal in the form of current/voltage into the digital language a computer can process called binary code. This digital information is sent to your computer through whichever connection and cabling your interface utilizes. This may be any variation of USB, FireWire or Thunderbolt, for example. Interfaces then perform this same function in reverse, converting the previously received digital information back into analog signal that you can hear through your headphones or speakers for playback. 

Another task of many interfaces is amplifying the small audio signal with a preamplifier. Most USB/FireWire/Thunderbolt audio interfaces have internal preamplifiers. When you speak into a microphone, the signal generated by your voice is very small. The preamplifier brings it up to a workable level that can be processed for playback. The quality and sonic characteristics of preamplifiers range widely and can affect your recorded sound for better or worse. In more developed home recording setups and professional studio spaces, you may find an assortment of external preamplifiers used for varying recording situations. For example, you may use a different preamplifier when recording a thrashing electric guitar than you would to record a delicate vocal melody. 

When choosing what audio interface is right for you, decide how many inputs and outputs you need. An input is where you plug in what you are recording to the interface. Pick an interface with XLR and 1/4" inputs so you can connect microphones and instruments/direct signals at the same time. If an interface has one input, you are able to record one track at a time. If an interface has 16 inputs, you are able to record up to 16 tracks simultaneously. The outputs allow you to connect your interface to headphones and speakers to hear a stereo signal of your tracks played back and sometimes to other interfaces for more channels.

III: SPEAKERS AND CABLES

The two previous sections focus on capturing and converting audio signal. Once these processes occur, you are able to hear your recorded sounds play back through headphones and speaker monitors.

Speakers come a variety of shapes and sizes. Some are small enough to fit in your cell phone and others are powerful enough to fill an arena with sound. Consider the electric signal of a recorded sound to be a picture of what that sound looks like. The purpose of the speaker is to make that picture much larger without negatively affecting the quality of the image. 

In order to translate an electrical signal into an audible sound, speakers contain an internal electromagnet - a metal coil that generates a magnetic field when current flows through it. This electromagnet sits in front of a permanently placed magnet. As signal passes through the coils of the electromagnet, the direction of the magnetic field changes rapidly allowing it to quickly move towards and away from the magnet, causing small vibrations. A cone made of thin material, often paper or plastic, attached to the electromagnet amplifies these vibrations causing it to pump sound waves into the air surrounding your ears for you to hear. The frequency of these vibrations determines the pitch of the produced sound. If you listen back at a loud enough level, you will be able to see and feel the cone of the speaker moving back and forth. 

All speakers, regardless of size, make or model, fall into one of two categories: active and passive. Active speakers contain a built-in amplifier. They are often more expensive than passive speakers, which require a separate external power amplifier. Power amplifiers connect to speakers by speaker cable, which can handle more signal that a standard 1/4” instrument cable. 

Speakers are the final stop in your basic digital audio workstation setup. Let’s review. The sound source or microphone connects to the audio interface which links to the computer and recording software, often through a USB or FireWire cable. The stereo outputs of the interface plug into to the inputs of the speakers which amplify the recorded signal for you to hear. 

Several different kinds of cables are commonly used for interconnecting studio gear. Balanced three-pin XLR cables, also known as microphone cables, are most often used to connect a microphone to a recording input. These three pins correspond to the three wires within the cable: a positive leg and a negative leg, which carry the same signal in opposite phase to one another, and the shield, which protects the audio signal from external noise and interference. If the receiving end of this audio signal - the input on your interface, for example - is also balanced, it will flip the phase of one of the legs causing both to be in phase with one another again and canceling out any noise obtained along the way. 

There are two different kinds of 1/4” cables. Tip/Sleeve (“TS”) cables have two internal wires: one inductor carrying the signal and one acting as a shield. This is considered unbalanced. You can identify this by looking at the end of cable to see that there is one ring around the connector. These cables are often used to connect a guitar or bass to the input of an amp. Tip/Ring/Sleeve (“TRS”) cables have two conductors carrying signal plus a shield. This means they’re considered balanced and have two rings around the shaft of the connector. 

There are several combinations of connectors and adapters that can be used to hear your sounds through various devices. Use a mono 1/8” to stereo 1/4” cable to connect your phone or iPod directly into speakers or an amplifier. RCA to 1/4” cables allow you to connect the RCA outputs of a record player to speakers that may not have RCA inputs. Not all speakers are the same so take a look and figure out what kind of inputs and outputs your equipment has. You’ll likely be able to find any kind of connection cable at your local music store. 

IV: A RECORDING SESSION

Recording sessions are broken up into several different parts: 

A) PRE-PRODUCTION: Pre-production is an often overlooked but crucial element to a successful recording. This is the planning period. Map out as many details as you can before you hit record. When and where are you going to record? How many tracks? What is your budget? What are your deadlines? Do you have a contract involved? If you’re the engineer, listen to demos from the artist beforehand to help narrow down what gear available to you best suits their sound. 

B) TRACKING: The term “tracking” simply means recording your tracks. Take ample time in setup to make sure you feel confidently about the sound of your tracks. Experiment with different microphones and signal processors to see what gear you like best. Some bands choose to record live, meaning everyone plays together just as they would at a live show, and some choose to record in stages starting with drums and guitar or bass. It’s common for bands to record many takes of each song before deciding on one performance so never hesitate to take a break if you are tired or frustrated. 

C) MIXING: After tracks are recorded, the engineer begins editing and mixing. The goal of mixing is to sculpt and blend all recorded tracks into one cohesive entity. They may use four microphones on a single guitar amp, but ultimately those four tracks will be mixed together so the listener hears one recorded sound. Regardless of experience or existing expertise, there are several things all mixers should keep in mind: 

1. Before you begin, make sure you are content with your raw (unmixed) tracks. If you don’t like the tone coming from your amp, fix it. If you aren’t satisfied with how you played, record another take. Mixing won’t change your performance. It’s better to rerecord a sound than to spend hours working to manipulate it into something entirely different. 

2. There is no single formula for creating a great mix. It takes time, trial and error. Only you can judge if your mix is finished. 

3. Every track has it’s own place. It’s important for your mix to have space so instruments aren’t crowding each other. Here are a few tools that can help achieve this: 

EQ: Sound is made up of many different frequencies measured in hertz (hZ). The range of human hearing spans from 20 Hz (almost inaudibly low) to 20,000 Hz (almost inaudibly high). Equalization, called EQ for short, is the process of balancing these frequencies within your recorded signal. Many modern EQ plugins are presented with the range of frequencies on the X-axis and gain on the Y-axis. Subtractive EQ means lowering or removing frequencies from a recorded signal. If your guitar signal has too much high end hiss, scoop out some of the high end (often between 10kHz and 14kHz) to get rid of it. Additive EQ means increasing frequencies within a recorded signal. If your kick drum sounds too thin, boost the low end (use 60-100 Hz as a starting point) to give it a deeper tone. 

PANNING: Panning is the spread of a sound within the stereo field. A mono audio signal (short for monophonic or monaural) means the entire signal is sent through one channel. For example, when you talk on the telephone, you hear the other person as a single sound source coming from one direction. If you pan a mono track entirely to the left, you’ll hear it in your left headphone or speaker only. If you pan a mono track entirely to the right, you’ll hear it in your right headphone or speaker only. A stereo audio signal is routed through two channels panned left and right. Panning can add clarity to mixes by allowing each instrument to be heard more concisely and helping to create a wider stereo image. Some mixers pan the individual pieces of a drum kit from the drummers’ perspective, meaning the snare and rack tom are panned slightly to the left and the floor tom is panned roughly halfway to the right. Panning multiple guitar or keyboard tracks is another easy way to widen a stereo image. Experiment by manipulating the panning of your tracks in your DAW during playback. On the illustrated software mixing window below, the vocals and guitars are stereo tracks fully panned left and right (100 each way) and the clave and conga mono tracks panned straight up the middle (0). 

COMPRESSION: Compression is the process of decreasing the dynamic range between the loudest and softest parts of your tracks. It is often used in tracking, as well as mixing. Consider this: when recording a vocal track that starts out very quietly, you'll need to turn up the preamplifier high enough so the signal is audible. If the singer suddenly get louder, the signal may clip and distort. Compression helps ensure that the dynamic range stays in control. 

Compressors come in several different varieties but all have a few common denominators. The threshold decides how loud a signal needs to be before compression is activated. The ratio of a compressor lets you choose how much compression is applied to your track. If the ratio is 1:1, no attenuation will happen when the signal crosses the threshold. A ratio of 2:1 means a signal exceeding the threshold by 2 dB will be attenuated down to 1 dB above the threshold. This is considered light compression whereas a ratio of 20:1 would be considered heavy compression. The attack of a compressor decides how quickly the compressor will work once it has been activated and the release will determine when the compressor stops working after the signal has returned below the threshold. Keep in mind compressing the dynamic range is not synonymous with making a track quieter. Many compressors allow you to add makeup gain for what may have lost while lessening the dynamic range. 

After you mix a song, listen back in as many different places as possible - through headphones or earbuds, speakers, your laptop or in your car for starters. These are just a few basic mixing tools but they can take your tracks a long way. There is no right or wrong way to mix tracks which means the process can be simultaneously daunting and liberating. Practice often.

D) MASTERING: Mastering is the last stop before duplication. Mastering engineers work with one stereo audio file instead of many individual tracks. These days, many albums are recorded in multiple different places with multiple different engineers. Mastering helps ensure that your songs will sound uniform in regards to volume, frequency response and dynamic range and will usually make your songs louder. If your mixes have undesirable or extraneous pops, clicks or noises, most mastering engineers are equipped to remove them. Mastering engineers may handle mixes differently for vinyl or cassette versus digital download so let them know how you intend to release your songs. 

Recorded sound as a field is constantly evolving. Don't let yourself feel overwhelmed by keeping up with the newest gear and never hesitate to try something new. It may result in something we haven't heard before.
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