Music Technology Honours

Proximity Effect

Proximity effect: A phenomenon that happens with directional microphones when they are placed close to the sound source.

Amplification of bass frequencies: Bass frequencies can be amplified, which can have both positive and negative implications.

Spherical vs. cylindrical microphones: Spherical microphones provide smoother attenuation across phases compared to cylindrical ones.

Microphone proximity and phase variance: When a microphone is closer to the sound source, variations in pressure and particle velocities cause an increase in low-frequency output due to phase variance.

Proximity effect and microphone pickup patterns: To reduce the proximity effect in cylindrical microphones, an omnidirectional pickup pattern should be used.

What are the basics of hearing?

1. Sound waves enter the ear through the pinna and the ear canal which causes the eardrum to move.

2. The eardrum vibrates with sound, very similar to a microphone membrane.

3. Sound vibrations move through the ossicles to the cochlea like a system of leavers.

4. Sound vibrations cause the fluid in the cochlea to move forth and back (movements equalised by the round window).

5. Fluid movement causes the hair cells to bend. Hair cells create neural signals which are picked up by the auditory nerve. Hair cells at one end of the cochlea send low-pitch sound information and hair cells at the other end send high pitch sound information.

6. The auditory nerve sends signals to the brain where they are interpreted as sounds.

7. Middle ear is responsible for sound conduction, but also for the acoustic reflex (a protective mechanism)- Lower frequencies require more energy for us to hear them- Higher frequencies require less sustained energy for us to hear them- Our ears incorrectly assume frequencies in the range 1-3.5 kHz are louder than others due to mechanical construction of our ears- LPF and HPF filtering applied in wide bands either side of the 1-3.5 kHz range of a mix and then reducing the raising or lowering of any frequencies in that range by +/- 30% adjusted on the amount needed might assist in certain circumstances to alleviate this problem without affecting the frequencies outside that range. The result should be a more natural-sounding EQ balance across the frequency range

An Impact

1. The psychoacoustic response to sudden changes in loudness.

2. A sharp contrast between silence and peak levels and the rate at which that happens can create the 'impression' (trick the mind) into hearing that the sound is really loud at that point of the sharp contrast when in reality, the change in dB is not that drastic.

3.Keeping the verse section of the song free of too many extra tracks can help achieve this, and adding them at the chorus point will create an impact.

4.Adding extra percussion tracks at that point is also a useful trick.

Distortion levels

1. Distortion alters the original sound wave.

It can introduce additional frequencies or change the shape of the wave.

2. Harmonic distortion adds new tones or overtones that were not present in the original signal.

3. Amplitude distortion alters the shape of the sound wave, emphasising certain frequencies or compressing the dynamic range.

4. Distorted sounds can cause the brain to interpret the sound differently from its original form.

5. Distortion is commonly used in audio effects to create specific sounds or add texture.

6. Excessive or unintended distortion can result in unpleasant or difficult-to-understand sounds.

7. Musical preference and context play a role in how distortion is perceived.

What is RT60 (reverberation time 60)?

The time it takes for the sound level in a space to decay by 60 decibels (dB) after the sound source has stopped.

Reverberation Time (RT) 60, refers to the time a steady-energy-state signal of pink noise takes to reduce by 60dB in a space; it is used for calibration of room acoustics and determining critical distance (Dc) of microphones with respect to direct sound sources using the Sabine equation and absorption coefficients;

Reverberation

Sum-effect of differences in time and level created by multiple sound wave reflections in a space, gives it a sense of "acoustic volume."

How is RT60 (Reverberation Time Measurement) measured?

The RT60 is measured using a special test signal called an impulse, which is a short burst of sound that is played through a loudspeaker and then recorded using a microphone.

It is the amount of time it takes for the sound level to drop by 60 decibels (dB) after the sound source has stopped.

An important consideration in the design and acoustics of performance spaces, recording studios, and other spaces where sound quality is important.

A shorter RT60 is desirable for spaces where speech or music clarity is important, such as classrooms or concert halls

Longer RT60 may be desirable for spaces where a more natural and ambient sound is desired, such as in churches or large performance halls.

What is the Fletcher-Munson curve?

Ear perceives sound at different frequencies and loudness levels.

created in 1933 by Harvey Fletcher and Wilden A. Munson

Plots the minimum SPL in dB required for a listener to perceive a sound at a given frequency and loudness.

Higher SPL's are needed at lower frequencies for a listener to perceive the same loudness as at higher frequencies, explaining why bass frequencies often need to be amplified more in music.

Psychoacoustics

The term refers to a discipline of study that considers the psycho-physical aspects of sound by bringing together psychology and acoustics; it is related to perception, cognition and acoustic properties of sound; it forms a critical aspect of active monitoring and critical listening in mixing and mastering practices

What is Pre-Delay ?

The amount of time that passes between the moment a sound is created and the moment the reverberation or delay effect is applied to it.

Creates a sense of depth and space in the sound, making it sound more natural and realistic.

By adjusting the pre-delay time, audio engineers can create different effects such as a sense of a larger or smaller room, and can also help to create separation between the original sound and the effect.

Haas Effect

Delay is used to make a track sound wider.

A Psychoacoustic phenomenon discovered by Dr. Helmut Haas in 1949. Also known as the "precedence effect,"

When one sound is followed by another with a delay time of approximately 40 ms or less, the two are perceived as a single sound.

This is a great way to make mono signals sound wider than they are without using stereo widening or duplicating mono tracks and panning each duplicate either hard left and right.

How to do Haas effect

Hard-pan a mono keyboard comp part or rhythm guitar track to the left and bus the track to an aux channel via a send in your DAW's mixer.

Insert a delay-modulation plug-in on the aux channel.Hard-pan the aux to the right, set the plug-in's output to 100% wet, and set the delay time to around 5-30ms.

Now program a very slow and shallow modulation of the delayed signal: Set the plug-in's speed or rate control to around 0.8Hz and its depth or width control to roughly 17%.

Modulating (cyclically varying) the delay time in this manner makes the track's delayed component sound alternately closer to and farther away from its dry sound in the opposite speaker, tricking the brain into constantly shifting the stereo image.

The resulting Haas effect will make your mono track sound stereo and add subtly shimmering movement between left and right speakers.

Some key principles of placement theory include:

1. Positioning instruments and speakers in a way that minimizes phase cancellation and comb filtering, which can occur when sound waves interfere with each other in a way that cancels out certain frequencies.

2. Understanding the different directional characteristics of microphones and speakers, and positioning them accordingly in order to capture or project sound in the most effective way.

3. Taking into account the psychoacoustic effects of room acoustics, such as early reflections and reverb, and using techniques such as diffusion or absorption to control these effects.

4. Paying attention to the way different instruments interact with each other in the space, and adjusting placement as needed to create a balanced and cohesive mix.

5. Adapting placement strategies to the specific goals and needs of the music production, such as emphasizing a particular instrument or creating a specific spatial effect.

Placement on stage

Vocals: The lead vocalist should be positioned in the center of the soundstage, with backing vocalists positioned slightly behind and to the sides.

Drums: The drummer should be positioned towards the back of the stage or recording space, with the bass drum and snare drum mic'd from the front and the cymbals mic'd from overhead.

Bass: The bass player should be positioned towards the back of the stage, close to the drummer, in order to establish a solid rhythm section.

Guitars: Electric guitar amps should be placed off to the sides of the stage or recording space, angled towards the center, in order to avoid bleed into other microphones. Acoustic guitars should be positioned towards the front of the stage, closer to the audience, in order to be heard clearly.

Keyboards: Keyboard players should be positioned towards the back of the stage, close to the drummer and bass player, in order to establish a solid rhythm section.