With the right equipment you can set up a project studio in your home. Recording music can be a complex process. The learning curve can be pretty high when it comes to working with software and making a great song. We will cover the basics of recording music in this series. We will break the whole process down in order to better understand what happens.
Recording a song is a four step process
- Recording the tracks: This involves capturing the actual sound waves and recording different instruments. This also involves capturing midi data and working with software synths. These parts are separated into individual audio and midi tracks.
- Editing: This involves changing the audio or midi that was recorded. It can be anything from inserting fades to cutting and moving audio clips around the song. Editing is anything to do with changing the audio or midi data that has been recorded.
- Mixing: Mixing is the art of blending all the audio parts together. The goal here is to provide clarity so all the parts can be heard. Once the audio is mixed it’s converted into a single stereo track.
- Mastering: This involves looking at the final stereo mix and providing an overall polish to the song. You are improving the final stereo track and getting it ready for production via download or CD.
Sometimes the 4 steps will overlap one another. You may be recording tracks and you may edit something before you record another track. You may be mixing and you might discover that you need to go back and record another track or record a better version of a track. You might be Mastering and you discover that you need to go back and fix something in the mix. Now we will spend the rest of this series going into detail about the 4 steps of the recording process.
A Microphone or Mic is short for an acoustic-to-electric transducer that converts sound waves into an electrical signal. There are many different types and brands of microphones. Let’s break down the common concepts regarding microphones and how they work. There are basically two types of microphones that are used in the project studio,
Dynamic and Condenser
Dynamic microphones work by electromagnetic induction. They are the most rugged type of microphone out there. They are inexpensive. You will find they are the most common microphone used among live P.A. systems and live gear. When sound enters into the microphone, the sound waves move the diaphragm. When the diaphragm vibrates, a coil moves inside a magnetic field. This movement is producing a varying current in the coil by electromagnetic induction. A single dynamic membrane does not respond linearly to every audio frequency. For this reason some microphones utilize multiple membranes for the different parts of the audio spectrum. Then the membranes combine the resulting signals. It is much harder for Dynamic microphones to get the higher end of the audio spectrum.
The condenser microphone was invented at Bell Labs in 1916 by E. C. Wente. It is also referred to as a capacitor microphone or electrostatic microphone. This is because capacitors were historically called condensers. All condenser microphones require a power source. This can be provided via the audio interface. This is known as phantom power. You will see a switch on your audio interface that says 48v. This is to power the condenser microphone. Some condensers will have a compartment to insert a battery. Condensers generally produce a high-quality audio signal and are a popular choice in a recording studio environment. Condenser microphones require a very small mass that must be moved by the sound wave. This is unlike other microphone types that require the sound wave to do more work. Condenser microphones are great for capturing the higher end of the frequency spectrum. They pick up higher frequencies that dynamic microphones miss.
A microphone’s directionality or polar pattern specifies how sensitive it is to sound waves arriving at different angles about its central axis. Polar patterns include Omnidirectional, Bi-directional, Unidirectional, Sub-cardioid, Cardioid, Hyper-cardioid, and Super-cardioid. The maximum Sound Pressure Level (SPL) the microphone can accept is measured for particular values of total harmonic distortion (THD). Microphones are also subjected to something called the proximity effect. The closer a microphone is placed to the sound source the more bass frequencies are detected and exaggerated. A singer may produce more bass then normal if they are singing right on top of the microphone. This is due to the proximity effect.
A preamp is a sound engineering device that prepares a signal to be processed by other equipment. Microphone signals are too weak to be transmitted to units such as mixing consoles and audio interfaces. Preamps increase a microphone signal to line-level by providing stable gain while preventing induced noise that would otherwise distort the signal. Preamps are built into the audio interface. Normally they are found in the first part of a channel strip on a mixing board. You can also buy tube preamps that will make your microphone sound better. High quality preamps will give you a better recording or they will color a sound in a way that will sound good to the listener. Low quality preamps such as those found on computer sound cards will introduce noise and they will not provide the best sound quality available.
The VU Meter is a device that displays a representation of the electronic audio signal level. VU is an acronym for Volume Unit and they measure loudness or volume. They are found on audio equipment. If the signal is too strong then the signal will be distorted and this kind of distortion is not good. If the signal is too weak then it will introduce some noise to the audio recording. These meters were primarily used on analog equipment. The original VU meter was a passive electromechanical device. VU meters measure average signal levels. A sustained sound reads much higher than a brief percussive one. The reading is dependent on both the amplitude and the duration of peaks in the signal.
A peak programme meter or PPM is an instrument used in professional audio for indicating the level of an audio signal. There are different kinds of PPM Meters. In professional use consistent level measurements are needed across an industry. Audio level meters often need to comply with a detailed formal standard. This will ensure that all meters that comply with the standard will give the same indication on any given audio signal.
There will be audio meters on every channel in your DAW. The audio meters indicate the volume at which the audio will be recorded. The meter values will range from ‑INF which is silent to 0dB which is maximum volume. You may be able to change many options in the way your DAW meters display data depending on the DAW you are using. To maximize the dynamic range of your audio recording you need to set the levels as high as possible without clipping or going past 0.
When the audio level exceeds 0dB some of the audio information is lost. This is when the digital signal goes beyond the base color of the meter. This is known as overload. Many audio interfaces will use clipping to deal with an overloaded signal, but clipping can distort the audio signal too. As a result you should avoid letting the meter level exceed 0dB. Don’t let the signal fall to low. A good range would be minus 3 to minus 12. Recording in this range will give you a good recording. Going over 0 on digital audio will compromise your quality.
In the days of analog recording it would be ok to allow the meter to go over 0 on occasion. This is not the case with today’s digital recording technology. Setting your range between -12 and -3 should give you the best results.