Introduction to Music Production Week 3: Effects

This week I’ll give a quick explanation to the three effects categories of DSP (Digital Signal Processing) and they are, dynamic, delay, and filter effects. These three categories are generally related to three of the principles of sound, amplitude, propagation, and timbre.

Dynamic Effects:

  • Compressor  – Narrows a signal’s dynamic range. Amplifies low sounds and a attenuates loud sounds. Making the audio have a more normalized feel.
  • Limiter – Keeps the audio level from going over a pre-set amount.
  • Expander –  Decreases the level of signals that fall below a certain threshold to a specified range. For instance, if the threshold is set to -6dB, once the audio is below that threshold, the audio can be pushed down to -10dB, but not complete silence.
  • Noise Gate – Decreases the level of signals that fall below a certain threshold to complete silence.

Dynamic effects automatically control amplitude over time.

Delay Effects:

  • Reverbs – Reflections of sound within a space that slowly decay
  • Delays – Recorded sound that is either played back multiple times or fed back into the recording to create a repeating decay.
  • Phasers – Audio that is split into two separate paths. One path is an all pass filter which preserves amplitude. When the two paths are mixed there is cancellation of notched frequencies creating a phasing effect.
  • Flangers  – The same as a phaser, although the all pass filter is replaced by a delay.
  • Choruses – Two sounds that are almost similar in timbre and pitch converge and are perceived as one.

Delay effects are related to propagation sound. They can create the illusion
of three dimensionality. They can also make the audio sound like it was recorded in either a small or large room.

Filter Effects:

  • High Pass Filter – Allows all frequencies above the cut off point to pass. HPF cuts the lows and allows the highs to pass.
  • Low Pass Filter –  Allows all frequencies below the cut off point to pass. LPF cuts the highs and allows the lows to pass.
  • Band Pass Filter – Allows a specified band of frequencies to pass. All other frequencies above and below the specified band are then cut.
  • Parametric EQ – A more precise eq that controls three parameters, amplitude, center frequency, and bandwidth.
  • Graphic EQ – An equalizer with pre-set bands that can be boosted or cut in a range of -/+ 6dB to -/+12dB.

Filter effects control the timbre of sound.

If you have any corrections, please feel free to comment.

Introduction to Music Production Week 1 Assignment: Recording Signal Flow

Hello, my name is Harry. I live in North Eastern, Pennsylvania in the USA. For Week 1 of Berklee’s Introduction to Music Production at Coursera.org, I’ll do an overview of a typical recording signal flow for my assignment.

Signal Flow In

So to start with, we need a sound source, such as a voice or speaker. This source creates pressure variations in the air, called sound, which is then captured by an input transducer, in this case, a microphone. Those pressure variations are then turned into voltage variations, which is our audio signal. The low level audio signal then travels through a balanced XLR cable to an audio interface.

A microphone has a male XLR connector (left) that connects to the female XLR connector (right) of an XLR cable.
Lexicon Alpha Audio Interface
Lexicon Alpha (front)
Lexicon Alpha (back)
Lexicon Alpha (back)

At the audio interface the signal goes through a number of components before it reaches the computer.

First, the signal is sent through a microphone preamp that brings the low level signal up to the standard operating level. This level can be amplified or attenuated by controlling a trim knob, such as the Line 2/Mic on the picture above. (labeled as front)

Next the signal is sent through an A/D converter (analog to digital). Which converts the analog signal to a digital signal of 1’s and 0’s that the computer can understand, called binary data.

The binary data travels through either a Firewire or USB cable into your computer to be received by your DAW (Digital Audio Workstation) of choice. This data can then manipulated and mixed in with other audio within your DAW.

Signal Flow Out

The data is then sent back from your DAW via Firewire or USB cable to your audio interface

Here it is ran through the D/A converter (digital to analog) where it becomes , once again, an analog signal.

Lastly, the newly converted analog signal is sent through an amplifier to the output transducer, such as monitors or headphones.

Overview

To make a quick overview of what we just covered.

  • Pressure variations (sound) are created in the air and captured by input transducer (microphone)
  • Sound is turned into voltage variations (audio signal) and travels through XLR cable to audio interface
  • Low level audio signal is amplified by a mic preamp and adjusted by a trim knob
  • Analog signal is converted to digital (binary data) and travels via Firewire or USB cable to the computer
  • The data reaches DAW (digital audio workstation) and is manipulated
  • Binary data is sent back from DAW to audio interface and goes through digital to analog conversion
  • Analog signal is amplified and sent to output transducer, headphones/monitors

 

Thanks for reading and please feel free to give any feedback relating this topic or corrections to the material covered.