Here I will cover what I’ve learned in Sound Production Theory 1 & 2 throughout the last two years.
This section will cover what I’ve learned about microphones. This will include the different type of microphones I’ve used or have been taught about in theory classes, polar patterns and microphone techniques. This clearly shows that I am capable of one of my requirements mentioned in my plan, that I understand different microphone placement techniques, and polar patterns and what they all do.
Dynamic Microphones – Dynamic microphones are well suited for recording loud instruments, such as drum kit because they have a flatter frequency response than condenser microphones do. This means that because the drums create a loud sound, the microphone can pick it up without being damaged, and more importantly, pick it up well. Dynamic microphones are also a lot more rugged than condenser microphones, meaning they can be used for numerous purposes, for example, in a live situation, because they can be dropped and carried around without fear of breaking. They also do not need power and are very popular for both recording and live performances because they are easier to set up, do not need phantom power, and do not have as much chance of switching off randomly.
How it converts sound into an audio signal:
A dynamic microphone uses mechanical movement to change acoustic energy to electrical energy. As the sound pressure changes, the diaphragm (top arrow) moves backwards and forward which then moves the coil (middle arrow) in and out of the magnetic field, (magnet being the bottom arrow) making an electrical current.
Condenser Microphones – Condenser microphones are more sensitive than dynamic microphones, which allow them to pick up a much more natural sound and clearer recording. They are best used with acoustic instruments due to the precise and wide range of frequencies they produce. Unfortunately they need batteries or phantom power to use them, and they are also very easily broken. I did not have much need for them during my recording sessions, apart from a few occasions where a dynamic microphone wouldn’t be sufficient. The picture on below is an example of a condenser microphone, coincidently; I used these for both an acoustic guitar and overheads on drums.
How it converts sound into an audio signal:
A condenser microphone uses capacitance (ability to hold electrical charge) to change electrical current. The floating metal plate acts as the diaphragm. (Top arrow) Placed behind this is a fixed metal plate. (Arrow below the top) These two together form a capacitor which stores electrical charge. As the sound pressure changes, this moves the diaphragm backwards and forwards in a similar way to the dynamic microphone. However a preamplifier is needed to drive the signal created down a microphone cable. This requires a power supply, i.e. phantom power.
Polar Patterns
A polar pattern is the way in which the microphone ‘hears’ sound from a particular direction. There are various types of polar pattern, but for me the Cardioid was the most commonly used. However, I shall still explain some of the other types for comparison.
Cardioid– A Cardioid microphone picks up most sound from the front, minimal sound comes from the rear and a small amount from the sides. This makes it an excellent choice for public speaking or recording vocals because very little background noise is picked up and maintains a clear pickup of what is in front of it. The cardioid gets its name from the heart shape pattern of sensitivity that is created when drawn. This is shown on the picture on below.
Hypercardioid – This polar pattern is very similar to the cardioid but has a much narrower pickup range than the cardioid. It also picks up more sound from the back which can be an issue depending on what microphone is being used for. However, this microphone could be considered a lot more directional than the cardioid because it picks up less from the sides and more from the front, it could be a potential candidate for use as a vocal or drum microphone.
Omnidirectional – As the name suggests, this polar pattern picks up sound from all directions equally. This is extremely useful when recording a group of singers for example; putting the microphone in the middle and the microphone will pick them all up equally. The main disadvantage of this type of microphone is that you cannot aim it away from unwanted sounds, such as a PA, which could create feedback.
Bi Directional – This polar pattern, (also known as a figure of 8) picks up sound from both the front and back equally, but nothing from the sides. It could be used for recording a snare drum, and would limit the amount of hi-hat noise considerably more than a cardioid microphone for example.
Shotgun – The shotgun polar pattern is used to pick up a sound source directly in front of it, and picking up very little from the left, right and rear. This type of pattern is usually used to pick up specific sounds that are slightly farther away, that other polar patterns can’t pick up effectively, such as a wildlife recording.
Microphone Techniques
Spatial Techniques
These techniques use two microphones and aim to create an accurate stereo image. They are primarily used for recording instruments or small live performances.
Coincident Pair – With this technique, two directional microphones are placed together, one above the other, facing left to right. Due to the microphones being so close, there are no phase issues with this technique. However, because the microphones are directional, the microphones will pick up the sounds on their respective sides faster than the other.
Spaced Pair – The spaced pair has two omnidirectional microphones placed an equal distance from each other, both facing forward. Due to the microphones being omnidirectional, both mics pick up the sound at the same level. There is phase with this technique because each microphone is closer to either the left or the right meaning that the sound will reach that microphone before reaching the other.
Near Coincident Pair – Similar to the coincident pair, the near coincident pair uses two directional microphones but they are spaced slightly apart. They have a combination of the problems faced by the spaced and coincident pairs, phase issues and level differences are created due to the microphones facing different ways and the sound reaching them at different times, respectfully.
Multi-microphone Techniques
As the name suggests, there are multiple microphones used with these techniques. They allow recordings in more than one location, which allows them to create more detailed recordings than spatial techniques. Surround recordings are also possible due to the number of microphones used.
Blumlein – The Blumlein pair uses two coincident bi-directional microphones positioned at a 90º angle from each other. Due to the bi-directional microphones 180º polar pattern, they become out of phase with each other. This can be rectified by splitting the signal with a ‘Y’ cable and reversing the connection in one of the cables. Despite it’s complex setup and strict requirements, this technique creates an excellent stereo image, both in regards to realism and quality.
Binaural – The binaural technique uses a dummy head which holds two condenser microphones, one in each of the ‘ears’. It is possibly the most natural technique due to it being the same as actual human hearing, however it is not as detailed because the microphones do not necessarily point at the sound source.
Mid + Side – The M+S pair uses one coincident cardioid microphone and one bi-directional microphone. The cardioid is used to pick up the front and side audio where the bi-directional mic is placed sideways so that it picks up both sides that the cardioid can’t reach. It is a useful technique because it allows clear mono recordings when mixed down, but it also allows the stereo recording to be changed after it has been recorded by alternating between the two microphones.
This is what the Mid + Side looks like when recorded straight into Sonar.This was a practice to test out the technique in class.
Double Pair – The double pair uses two of the spatial pairs, most commonly, a spaced pair at the rear, and a coincident pair at the front. This allows a more surround recording combined with the natural and efficient spatial techniques.
Decca Tree – Three omni-directional mics are used, two are placed two metres apart, with the third being placed in the centre, about one metre in front. The front microphone is used as the primary mic and captures the mids, whereas the two rear microphones capture the sides and reverberation, creating a natural and spacious sound.
Below are some images of what a Decca Tree looks like when recorded straight into Sonar.This was a practice to test out the technique in class.
Hamasaki Square – Four bi- directional microphones are placed in a square, all two to three metres apart. The left mics are set to pick up the left, and right to right. It is primarily used alongside another technique, because it is mainly used to record ambience for a surround recording.
Below is what the Hamasaki Square looks like when recorded straight into Sonar. This was a practice to test out the technique in class.
IRT Cross – This uses four cardioid microphones, facing away from each other. They are placed about 20cm away from each other in a cross shape, hence the name. It is usually used for ambient recordings to capture a 360º image.
Double MS – Two cardioids, one facing forward and the other facing the rear, and one bi-directional covering the sides make up the Double Mid Side. This is used to capture both the sound source and the surroundings. This pair can also be created by having two M+S pairs with some space between them, one facing forwards, the other backwards.
Ambisonics – Possibly the most complex and unusual, the ambisonic technique is created by using a tetrahedral collection of mics. It contains three bi-directional microphones placed 90º from each other and one omni-directional microphone. The recording is split into four channels, so that different mixes can be created, left-right or up-down for example. The main advantage of this technique is that it allows full surround sound recordings, including up and down, which means that it can be used for 5.1 or 7.1 setups.
This is the essay I did for the Sound Production Theory 1 Audio Processing Outcome. It demonstrates my understanding of audio processing in regards to how it works and what it is used for. It explains that I understand the recording process, but this also applies to live sound as well, which creates a more rounded demonstration of my abilities and understanding.
Gain Structure
Gain is an important step when setting up a recording session. Gain is the loudness, or the volume, that is applied to the sound being picked up by the microphone when it comes through the monitors, or speakers, at the mixing desk.
What would it be used for?
In a recording studio gain would be used to turn up the volume on the instrument that is being recorded so that it will be loud enough and clear enough to edit later on. For example, using gain on a 16-track analogue recording system (studio mixing desk) could be turning up a microphone used for vocals that have been put into channel two. (Picture below)Turning the loudness up will increase the signal that comes through, which means that the more signal that comes through, the louder it becomes. This signal coming from the microphone to the mixing desk is known as the ‘mic level’. However, due to the signal being lower than other electrical sources, such as a synth or a CD player, it needs to be brought up to the same level/volume so that they are equal. This is known as pre-amplification. When gain is applied and the loudness/volume is suitable, the signal now going to the mixing desk is louder than it was at the beginning, so this signal is now called a ‘line level’. The diagram below shows this process in a simplified way.
Gain would also be used on an 8-channel sound reinforcement system (live mixing desk) in the same way that it would be used for a studio mixing desk. However instead of turning up the volume for a recording, it would be turned up to make sure that each instrument can be heard clearly in and around the performance area.
Equalisation
Equalisation is the process used to change the frequencies, or the tone of the signals within a piece of audio, and make it more suitable for whatever the situation/audio needs. These frequencies range from 20 hertz to 20,000 hertz, this is the range human ears can respond to. Equalisation is achieved through filters, which make the process of equalising possible.
Filter Types
There are two filter types that are the most commonly used and most functional. The first of these is the ‘Graphic Equaliser’, which is shown below. The graphic equaliser has a lot of options for the user to change the tone of the sound. The left side of the unit has the lower frequencies/tones, or the bass/bassy/low tones. The right has the opposite; it has the treble, bright and high tones. The graphic equaliser is ‘fixed’ which means that the only changes that are allowed are the ones provided with the equaliser. While this is a downside of this type of equaliser, it does allow more freedom in some ways due to the amount of options available. It is also easier to see what equalisation has been made because the sliders are easy to read and show what has been adjusted.
The other type of filter is the ‘Parametric Equaliser’, which is shown below. This type of filter has a lot less options to choose from, compared to the graphic equaliser; however it is much more specific, meaning that you can choose exactly what you want rather than having to settle for something else.
What would it be used for?
Equalisation would be used in a recording studio with a 16-track analogue recording system to change how an instrument or a singer sounds for a recording. For example, using equalisation, the user could make a singer’s voice sound brighter or tinnier, or make a bass drum or bass guitar sound lower or deeper so that it thumps or booms. If you wanted to make the singing sound brighter, you’d go to the high mid-range, or high range, depending on the singer, and push the frequencies/tones around this area up, if you wanted to make them sound duller, you’d push the frequencies down.
However, the 8-channel reinforcement system would use equalisation to enhance what a live performance sounds like. Since different tones behave differently, the type of room the band is playing in can change how they sound. A church for example, will have a lot of space for the sound to move in, which will make the band’s music sound bright and tinny. Using the mixer to reduce these tones and bring up the lower frequencies/tones, (bass, rumbles and thumps) can balance out the sound and bring it back to what the band originally wanted.
Compression/Limiters
A compressor is used to reduce the amplitude, (amplitude means how loud the level of the audio is) so that it does not get too loud or brings the volume down to a level more suitable for the song/mix/performance etc. A limiter holds back the signal/loudness from going beyond a set threshold. (A threshold is a point where the signal cannot go above a certain volume without being altered)
How do they work?
With a compressor/limiter setting a threshold, let’s say it is +3 decibels, means that the signal cannot go any higher than +3 decibels without being compressed. The amount of loudness that is taken away if the signal goes over +3 decibels is measured by the ratio, for example the ratio could be 4:1. This means that for every 4 decibels that the signal is over the threshold, 1 decibel will get through, and so a 3 decibel reduction has taken place, because only 1 of the 4 decibels has made it through. (4 take away 1 = 3)
The diagram below shows this process, when the compressor is active, it stops the signal from getting louder than desired (the dotted line), where without it, this does not happen.
Compression is controlled and altered by ‘attack and release’. Attack is the part of the process where the gain (volume) is reduced to where the ratio is set, and the release is when the compressor increases the gain/volume back up to the desired level after the level has fallen below the threshold/volume limit. Due to this process reducing the gain/loudness of the signal being reduced, ‘makeup gain’ can be added afterwards. Makeup gain is an extra set of gain/loudness that can be added to achieve the desired/perfect level or volume.
Another method of compression is ‘side chaining’. Side chaining is similar to the method of compression mentioned above, however this does not use the original signal being compressed, but uses an external signal from another source to control the amount of compression without actually changing the signal that is being compressed.
Compressors are connected to the mixing desk, both a recording and sound reinforcement mixing desk, via inserts. The compressor could be either put on to a single channel, such as the lead guitar to stop it getting too loud, or put on all the channels together on a channel that is not being used. As shown in the diagram above, the cable has a stereo end which plugs into the desk, and at the other side with two mono ends that goes into the compressor.
What would it be used for?
A compressor/limiter would be used in a recording studio with a 16-track analogue recording system to keep instruments/singers below a certain volume so that they don’t get too loud. However the compression added here would have to be kept with the recording and could not be removed at a later date after the session is finished.
With an 8-channel sound reinforcement system, compression would be used in a live performance to stop a lead guitar, for example, from getting too loud and covering up the other instruments. It can allow the engineer to control the loudness of the performance and make sure that the music doesn’t get too loud without having to turn everything down when it gets a bit louder.
Expansion/Gating
An expander increases the dynamic range (the difference between the lowest and highest values in an audio signal). It is used to either increase the level of the desired sound, (the instrument being recorded) or it used to decrease the background noise, such as another performer coughing or playing loudly. A noise gate is a type of expander, which stops a signal from getting through until a pre-set threshold is exceeded. (See ‘Compression’ for what a threshold is)
How do they work?
A noise gate opens when the sound is loud, such as a guitar, and closes when the sound is quiet, when the guitar stops playing and background noise can be heard. The gate can be set to close at specific volumes with a threshold, like in compression. When the volume of audio falls below this threshold the gate will close and stops any sound from getting through. It will open again when the signal goes back up above the threshold and will allow the signal to be picked up again. The ratio, like in compression, is used to determine how much gain/loudness is applied to the signal. Attack and release is similar to how it works with compression, however with gating if the attack/release times aren’t quite right, it is more obvious because the sound will cut in and out and sound as if the audio is damaged. Gating also has a hold function, which sets the amount of time the gate is left open after the signal has gone back below the volume limit. A ‘key’ can also be used when using a noise gate. This, like the side-chain process used in compression, uses an external source to control the gate. This allows the original signal to be untouched, but can be gated through the expander separately and the result can be included if it is needed.
Expanders also allow filtering which is used if a sound exceeds the threshold. A filter allows the user to tune out (or sometimes tune in) a frequency range, so that full gain/loudness is only applied when sounds in a particular frequency/tone range exceed the volume limit.
What would it be used for?
With a recording system mixing desk, expanders/noise gates would be used to make a clearer recording. Background noise such as an amp fuzzing, or someone coughing, can be picked up during the recording and ruin the quality of the music. Using a noise gate to cut out low level noises such as coughing will ensure that only the desired sound, the instrument or singer, will be picked up.
With a sound reinforcement mixing desk, this would be used to clean up the sound during a performance, for example, the crowd cheering or the instruments being picked up in different microphones, could be lowered and removed, especially if the live performance was being recorded.
Reverberation
Reverberation, usually shortened to ‘reverb’, is when a sound reflects (bounces) from the different surfaces in a room and keeps echoing until it finally decays and disappears. These reflections include ‘Early reflection’, which is the term given to the first few reflections of reverb that bounce back to the listener from nearby surfaces, such as the wall or floor.
How is it used?
A reverb unit is used to achieve these results and ‘reverb time’, which is the amount of time that the reverb will last, is turned up or down, depending on how long the user wants the reverb to last. The amount of reverb is determined by the ‘balance’. The balance is the difference between the original unaffected signal, and the effected signal that has been created due to the reverb unit. The reverb unit is connected to the desk (studio or live) in a few steps.
- Connecting ¼ inch audio cables [an example of what they look like is shown below] into the two outputs (sockets) on the back, one goes into the one labelled left or ‘L’ and the other into the one labelled right or ‘R’.
- These two cables are then put into the same places (left to left, right to right) on the desk, behind the desk on the stereo/aux return section.
- After this has been done, connect another 1/4-inch audio cable from one of the inputs labelled “Aux/Send” on the back of the mixer, to the aux(iliary) input (socket) on the reverb unit.
What would it be used for?
Reverb can be added to audio either through a recording system’s mixer, or a sound reinforcement’s mixer. Reverb would be applied in a studio to create a mood or an effect of the song being performed in a different place, such as a cathedral or a concert hall. Its use in a live situation could also be used for a similar purpose, but can also be used to reinforce vocals for example, and make them sound a lot better than they were originally by making it ring throughout the room.
Flanging
Flanging is an effect that is created by mixing two identical audio signals together. One of these signals is delayed slightly and gradually changes which creates a chorus effect (a harmony effect). Mixing these signals together and delaying one of them is known as ‘Modulation’. The delay that is used is referred to as ‘delay time’, which is how long the signal is made to hold off playing back the audio. ‘Phasing’ is very similar to flanging; however the signal does not gradually change in phasing, but does in flanging.
What would it be used for?
Flanging is usually used to create a jet-like sound, due to its strange and whooshing sound. It can also be used on vocals, since it makes a harmony effect, to add the illusion of multiple singers. If the user wanted to use this and edit how much flanging was used, they’d use ‘feedback’. Feedback alters how much of the effect gets through into the original audio, so a lot of flange or very little flange can be applied to the audio using this. On a recording system’s mixer and a sound reinforcement’s mixer, flanging would be used to edit the sound of the instrument/singer performing, which would be used for both a performance and a studio recording.
