Configuring of an EQ Plugin to Function Like a Large Format Mixing Console EQ Section

I’m Peyton Haynie from Bay City, Tx. This is for week six of Introduction to Music Production at Coursera.org. I will be talking about configuring an EQ plugin to function like a large format mixing console EQ section. For this assignment I will be using the settings shown in the lesson and SONAR Professional.

First, to get to the EQ plugin, right click on the audio track, select process effect, audio effects, directx, sonitus:fx, and finally equalizer.

getting to eq

The first EQ setting is the highpass filter (just highpass on this plugin). It should be set to 75 hertz and 18 decibels per octave. In this plugin you can set the decibels before selecting the highpass filter, but I do not believe it will take it into account.

highpass 1highpass 2

Next is the low shelving filter (shown as shelving low). It should be set at 80 hertz. I also set mine with a Q factor of 1.

low shelf 1low shelf 2

Up next is the low mid range filter (shown as peak/dip). It can be set from 100 – 2000 hertz and between negative 15 – 15 decibels with a Q factor of 1.

low peak 1low peak 2

Next is the high mid range filter (also shown as peak/dip). It can be set between 400 – 8000 hertz and negative 15 – 15 decibels, also with a Q factor of 1.

high peak 1high peak 2

The final EQ setting is the high shelving filter (shown as shelving high). It should be set at 12000 hertz and between negative 15 – 15 decibels.

high shelf 1high shelf 2

Any of these settings can be enabled or disabled by clicking the yellow box next to it. Clicking the “flat” button will take everything back to zero.

To save this configuration as a preset, simply name it in the preset box and select save.

preset save

To summarize: High pass filter, 75 Hz, 18 dB. Low shelving filter, 80 Hz. Low mid range filter, 100 – 2000 Hz, neg 15 – 15 dB, Q factor of 1. High mid range filter, 400 – 8000 Hz, neg 15 – 15 dB, Q factor of 1. High shelving filter, 12000 Hz, neg 15 – 15 dB.

Types and Usage of Important Studio Cables; or What Does This Cable Go To?

I will be talking about the different kinds of studio cables and what they do.

There are two main types of cables, balanced and unbalanced. When combining multiple cables it is best to use shorter unbalanced cables and longer balanced cables as balanced cables won’t pick up as much noise along the way as an unbalanced cable will.

TS Cables

An instrument (or TS) cable is exactly that, a cable that connects your instrument to its amp. They are typically used for unbalanced mono signals. The end of a TS cable consist of a tip, a sleeve, and a single ring.

TRS Cables

A TRS cable is typically used as the output line in an audio interface. They can be used for both balanced mono signals and stereo signals. The TRS cable looks almost exactly like the TS cable, the main difference being the end. The end of a TRS cable consists of a tip, a sleeve, and two rings.

XLR Cables

XLR cables are balanced and are mainly used for microphones. The output (or male) end consists of a shield and three holes. The input (or female) end consists of a shield and three pins.

RCA Cables

We all know RCA cables, they’re the cables that run from the DVD player to the TV or from the stereo to the speakers. In recording studios, however, they are used to connect the recording interface to the studio monitor. On one end there is a single black connector, this end is plugged into the line out of the recording interface. On the other end there is a yellow video connector, a white left audio connector, and a red right audio connector.

To summarize: A TS cable is used to connect an instrument to an amp. A TRS cable is used as an output line in an audio interface. An XLR cable is used to connect microphones and RCA cables are used to connect the recording interface to the studio monitors.

The Different Dynamic Processors and What They Do; or How Do I Fix This Sound?

I will be talking about the kinds of dynamic processors and what they do.

Dynamic processors change the sound of audio signals by changing the dynamic range. Some dynamic processors expand the dynamic range and some compress it.

Compressors

A compressor (or downward compressor) controls the volume level of an audio signal so as not to damage the speakers. To apply a compressor you must first set a threshold. The threshold is the level at which the compressor starts working. Any frequency over the threshold will, essentially, be turned down. How much it’s turned down depends on the ratio. For example, with a ratio of 2:1 all frequencies will be cut in half. With a ratio of 1:1 nothing will happen and with a ratio higher than 10:1 the compressor will become a limiter.

Limiters

A limiter functions much like a compressor. The only difference being that with a limiter everything above the threshold will be stopped instead of just turned down. They are sometimes referred to as brick wall limiters.

Expanders

An expander is basically the opposite of a compressor. Expanders increase the dynamic range of an audio signal by reducing the gain of the signal below the threshold. Again, how much it is reduced depends on the ratio. For example, with a ratio of 2:1 the gain in the audio signal would be reduced twice as much as it already was.

Gates

Gates (or noise gates) reduce the amount of unwanted noise, such as the hum of an amp, in an audio signal. To do this you must set a threshold above the sounds you want to eliminate but below the sounds you want to hear. How well the gate works depends on the range, attack and release. The range determines which frequencies will be blocked and which frequencies will be let through. The attack determines how quickly the gate will open and the release determines how quickly it will close. If the attack is set too low the gate will open too slowly, cutting off the beginning of the wanted sounds. If it is set too high it will open too quickly, letting in unwanted sounds. If the release is too short it can cause chattering. If it is too long it will close too slowly, letting in unwanted sounds.

To summarize: A compressor reduces the gain of signals above the threshold, essentially turning them down.

A limiter stops all signals above the threshold.

An expander reduces the gain of signals below the threshold.

A gate reduces the amount of unwanted noises by stopping all signals below the threshold.

Describe the Concept Behind Dynamic Processors and Describe Threshold, Ratio, Attack and Release

I’m Peyton Haynie from Bay City, Tx. This is for week five of Introduction to Music Production at Coursera.org. I will be talking about the concept of dynamic processors and describing threshold, ratio, attack and release.

The concept behind a dynamic processor is simple: if it’s too quiet, make it louder. If it’s too loud, make it quieter. If there’s too much noise, bring it down. If there’s not enough noise, bring it up.

Threshold

The threshold is the level at which the processor will start working. In a compressor, the threshold is the level where compression begins, with anything over that level being compressed. In a gate (or noise gate) the threshold is the level at which the unwanted noise is stopped, with anything above that level going through and anything below that level being reduced.

Ratio

The ratio is the amount the audio signals are affected by the dynamic processors and will always be shown as a number:1. For example, a compressor with a ratio of 2:1 will reduce a signal that passes the threshold by half.

Attack and Release

When a processor starts working the attack determines how quickly the effect starts. The attack helps to control the transients of the audio signal, the spike in amplitude that comes as soon as the sound occurs.

The release determines how quickly the effect ends. In a gate, a shorter release time would help clean up the noise in an audio signal, but may cause chattering in percussive instruments, while a longer release time could eliminate the chattering. In a compressor, a shorter release time can produce a choppy sound, especially with low-frequency instruments, while too long of a release time can result in an extremely compressed sound.

To summarize: a dynamic processor changes the dynamic range of an audio signal by changing the amplitude over time.

The threshold is the level at which the processor starts working.

The ratio is the amount the audio signal is affected by dynamic processor.

The attack determines how quickly the effect starts and the release determines how quickly the effect ends.

Monitor Mix Concepts: Why Producers Need Them and the Signal Flow Involved; or Everybody Needs To Hear The Drummer

I’m Peyton Haynie from Bay City, Tx. This is for week four of Introduction to Music Production at Coursera.org. I will be talking about monitor mixes and the signal flow involved.

What Is a Monitor Mix and Why Do We Need Them?

Monitors allow performers to hear themselves onstage. A monitor mix is the sound that comes out of each monitor. Each performer needs a different mix. For example, singers need to be able to hear themselves to make sure they are on key and the drums to make sure they are on beat. Likewise, a guitarist would need to hear themselves and the drummer, but wouldn’t necessarily need to hear the singer. In short, monitor mixes are used to ensure the performers know what they are doing.

Signal Flow

The signal flow to a monitor is much like the signal flow to a front of house speaker. The signal starts at the input of the mixing board. From there it goes to the trim where the noise is covered and then to the EQ. It then goes to the auxiliary sends which take it to the monitors.

Creating the Monitor Mix

The first step is to ask each of the performers what they want to hear. Then start with the drums. Have the drummer go through each piece of the kit and adjust the volume in each monitor according to what the performers want to hear. Then repeat with the bass, then vocals, then guitars and keyboard. Vocals and acoustic instruments tend to use the most gain, so make sure to keep that in mind while mixing.

To summarize: A monitor mix allows each performer to hear the rest clearly and keep on beat and in key.

The signal flow starts at the input of the mixing board, goes through the trim, the EQ, and the auxiliary send to the monitor.

Always ask the performers what they want to hear, have each of them play their instruments in turn and adjust the volume of that instrument in each monitor according to what the performers want.

Channel Strip and Signal Flow of an Analog Mixing Board

I’m Peyton Haynie from Bay City, Tx. This is for week four of Introduction to Music Production at Coursera.org. I will be talking about the components of and the signal flow through the channel strip of an analog mixing board.

The Components of the Channel Strip

There are many knobs and buttons on a mixing board. At first, this can be more than a little troubling. But once you look closer you can see each line of knobs and buttons represents one channel and each new channel is exactly the same as the last.

Trim knobs, found at the top of the channel strip, control the source level of the channel.

Faders, located at the bottom of the channel strip, control the volume. In addition to the fader at the bottom of each channel there is also a master fader, which controls the volume max for the whole board.

Auxiliary sends, found in the middle of the channel strip, feed the signal to other devices, such as a stage monitor.

The pan knob allows you to balance the signal by raising the volume on one side and lowering it on the other. Pan knobs are usually found either above the trim or above the fader.

EQ knobs are located in the middle of the channel strip. The low EQ knob adjusts the low frequencies of the signal, the mid EQ the adjusts the middle frequencies and the hi EQ the high frequencies.

The mute button is found directly above the fader. As the name suggests, it mutes that channel.

The solo button, usually found with the mute button, mutes all other channels.

The pad button, located at the top of the channel, is used when the trim is at its lowest and the signal range is still in the red. The pad button drops the signal range a few more decibels.

Signal Flow

The signal starts at the input. From there it goes into the trim, where the noise is covered and cannot be heard over the PA. The signal then goes through the EQ where it is “reshaped” to make it more suitable for mixing. It then moves into the auxiliary sends, where it can be sent to the monitors. After the auxiliary sends it’s sent to the pan knob to be balanced, and to the fader to adjust the volume. Once all the channels have been properly mixed they are combined into a main mix and sent to main outputs, such as speakers or headphones.

To summarize: The signal flow starts at the input, goes through the trim to the EQ, the auxiliary sends, the pan knob and the fader before being combined into a main mix and being sent to the speakers.

The Analog to Digital Conversion Process

I’m Peyton Haynie from Bay City, Tx. This is for week three of Introduction to Music Production at coursera.org. I will be talking about analog to digital conversion.

Analog to digital conversion is an electronic process in which a continuously variable (analog) signal is changed, without altering its essential content, into a multi-level (digital) signal.

Step one – Sampling

Generally done with a sample-and-hold circuit, sampling is a process which involves taking “snapshots” of audio signals at  fast intervals. The quality of the audio is largely determined by the sampling (or bit) rate.

Step two – Quantization

Quantization is the process which converts the sampled signals, which are continuous in value, into a signal that is discrete in value.

Step three – Coding

Coding is the process in which quantized signals are converted into a digital representation. Coding is performed by giving each quantization level a label. For example, when using four bits, the lowest level would be 0000, the next 0001, and so on.

Receiving and Decoding

After the signal is coded it is sent to a digital receiver. The signal is then decoded and reverted back into analog to be heard through speakers.

To summarize: to convert an analog signal into a digital signal, the signal is first sampled. Then each sample is quantized, and the values mapped into bits. The information is then sent as a digital signal to a receiver, where it is decoded and the analog signal is restored.