Image And Sound Technologies (Sound)

Front 1

Discuss the two main ways in which we can visualize audio signals.

Back 1

We can view audio signals in the time or frequency domains. The time domain representation of audio signals shows how the amplitude of an audio waveform changes over time.
The frequency domain representation of an audio signal shows the spectral components contained within the audio.
The frequency domain view is the Fourier transformed representation of the time domain signal that states that any time domain waveform can be
expressed as the sum of an infinite series of sine and cosine waves.
The time domain representation is useful for editing sounds.
The frequency domain representation is useful when analyzing sounds for filtering.

Front 2

Discuss how stereo sound creates a phantom image.

Back 2

Stereo sound creates a phantom image by producing level differences at the loudspeakers. Since both ears hear both loudspeakers, these level differences
are translated to time differences at the ears, due to the crosstalk components.

Front 3

What are the three main auditory cues for sound localisation?

Back 3

Interural time difference, interaural level difference and spectral shaping due to the filtering action of the pinnae.

Front 4

If a studio has the dimensions of 6m X 6m x 5m what low frequency problems can we expect due to the axial modes?

Back 4

For the axial mode along the length of the room:
F = c/(2*L) = 343/(2*6) = 28.5Hz
For the axial mode along the breadth of the room:
F = c/(2*L) = 343/(2*6) = 28.5Hz
For the axial mode along the height of the room:
F = c/(2*L) = 343/(2*5) = 34.3Hz
We will expect to have room resonances and nulls at these frequencies atdifferent points in the room.

Front 5

What are room modes?

Back 5

Room modes are the modes of vibration of standing waves in a room.
They can be axial, tangential or oblique.

Front 6

Outline 3 different THX recommendations for improving stereo imaging, flutter echo and acoustic isolation respectively.

Back 6

Flutter echo is a distinctive ringing sound caused by echoes bouncing back and forth between hard, parallel surfaces following a percussive sound such as a hand clap. Ideally, the side walls should be non-parallel, but if this is not possible the echo can be reduced by placing some acoustic absorptive material on the side walls, usually at ear height.
The sound image at the screen can be improved by mounting the speakers in a baffle wall. This eliminates sound reflections between the loudspeakers and the front wall, which cause the sound image to be unfocused.
Sound from adjacent rooms can be reduced by isolating the auditorium with two layers of building material separated by a single air cavity to reduce acoustic transmission through the walls.

Front 7

What is the effect of a direct sound from a loudspeaker and a single reflection coming off a mixing console (>6ms after the direct sound) adding at the ear?

Back 7

A reflection less than 6ms after the direct sound will tend to color the sound unaturally. Notches will appear in the spectrum and this interference is known as comb filtering.

Front 8

What is meant by reverb time, RT?

Back 8

Reverb time, RT is the time required for reflections of a direct sound to decay by 60 dB below the level of the direct sound. It is also known as RT60

Front 9

What is an acoustic impulse response?

Back 9

An acoustic impulse response is the response of a room to a short abrupt transient. If we clap our hands for example and record the input back in, we will see a waveform that has a direct sound, early reflections and diffuse decay.

Front 10

What is the difference between balanced and unbalanced audio connectors?

Back 10

DRAW DIAGRAM

Front 11

What is Phantom power?

Back 11

Phantom power is a way to power condenser microphones without using an external power supply or batteries. 48V of electricity is supplied via the microphone cable.

Front 12

What is the difference between mic level and line level?

Back 12

Mic level and line level represent two different electrical operating levels in professional audio equipment. Mic outputs are normally quite low; around a millivolt (1 mV) . Around 60 dB lower than line level . The nominal level of line level is around 1.23 volts. Most audio professional audio equipment works around this level.

Front 13

How do condenser mics differ from dynamic mics?

Back 13

Condenser means capacitor, an electronic component which stores energy in the form of an electrostatic field. Condenser microphones require power from a battery or external source. In the condenser mic, one of these plates is made of very light material and acts as the diaphragm. The diaphragm vibrates when struck by sound waves, changing the distance between the two plates and therefore changing the capacitance. Specifically, when the plates are closer together, capacitance increases and a charge current occurs. When the plates are further apart, capacitance decreases and a discharge current occurs. A dynamic microphone uses electromagnetics to convert the sound to an electrical signal. The diaphragm is attached to the coil. When the diaphragm vibrates in response to incoming sound waves, the coil moves backwards and forwards past the magnet. This creates a current in the coil which is channeled from the microphone along wires.

Front 14

What is the difference between mic level and line level?

Back 14

Mic level and line level represent two different electrical operating levels in professional audio equipment. Mic outputs are normally quite low; around a millivolt (1 mV) . Around 60 dB lower than line level . The nominal level of line level is around 1.23 volts. Most audio professional audio equipment works around this level.

Front 15

What is harmonic distortion?

Back 15

Harmonic distortion occurs when the audio system does not correctly reproduce the original waveform. This can commonly occur due to low bit depth or clipping.
•We can measure harmonic distortion by putting in a pure tone into the audio system and seeing how well it reproduces the tone at the output?
•We then look at the spectrum of the output and see if there has been any nasty upper harmonics created. If so, the system has distorted the input wave

Front 16

Discuss clipping in relation to both digital and analog audio signals?

Back 16

•When an audio signal is being recorded, we have to make sure that it is in the operating range of the equipment we are using.
•If it is too loud, then clipping will occur
•This is because audio equipment can only handle a certain input voltage range
In analog media like tape, soft-clipping will occur and the result can often be psychoacoustically pleasant to listen to. In the digital domain, hard clipping occurs, resulting in unwanted harmonic distortion.

Front 17

Discuss aliasing in relation to digital audio signals?

Back 17

Signal frequencies higher than the Nyquist frequency will encounter a "folding" about the Nyquist frequency, back into lower frequencies. For example, if the sample rate is 20 kHz, the Nyquist frequency is 10 kHz, and an 11 kHz signal will fold, or alias, to 9 kHz.
This phenomenon can be avoided by brick-wall filtering the audio prior to analogue to digital conversion, to ensure that no signals above the Nyquist frequency are digitized.

Front 18

Discuss the terms sampling rate and bit depth in relation to digital audio signals.

Back 18

The sampling rate, sample rate, or sampling frequency defines the number of samples per unit of time (usually seconds) taken from a continuous signal to make a discrete signal. For time-domain signals, the unit for sampling rate is hertz (inverse seconds, 1/s, s−1). The inverse of the sampling frequency is the sampling period or sampling interval, which is the time between samples.
In digital audio, bit depth describes the number of bits of information recorded for each sample. Bit depth directly corresponds to the resolution of each sample in a set of digital audio data. Common examples of bit depth include CD quality audio, which is recorded at 16 bits, and DVD-Audio, which can support up to 24- bit audio.

Front 19

Outline at-least three main frequency ranges when using filters

Back 19

•Sub-Bass: 16-60Hz. Sense of power. Felt more than heard. Too much makes the sound muddy.
•Bass: 60-250Hz. Contains a lot of fundamental tones. Too much makes the sound boomy.
•Low-Mids 250-2kHz. Contains a lot of low order harmonics. Boosting 250-500 sounds resonant.
•High-Mids: 2kHz-4kHz. Contains speech recognition sounds such as ‘m’, ‘b’ or ‘v’. Too much can cause listener fatigue.
•Presence: 4kHz to 6kHz. Makes the sound clearer and with more definition.
•Brilliance: 6kHz to 16kHz. Controls brilliance and clarity. Too much causes vocal sibilance

Front 20

When mixing audio, we need to consider the listening level in the cinema. What happens to the tonal balance of the mix if we mix at a lower volume than the reference cinema level?

Back 20

According to the Fletcher Munson isophon curves, our sensitivity to low frequencies changes with loudness. The quieter the sound, the less sensitive we are to its bass frequencies.
If the engineer mixes at a low level, they will compensate for this apparent reduction in bass by boosting the bass frequencies. The perception at playback at the reference level was that the bass was too loud.
This can be overcome at the mixing stage by incorporating a loudness EQ
into the monitor path, that simulates the perceived boosts in frequency response at higher listening levels

Front 21

Human beings hear with a dynamic range from 0 to120dB. Discuss how this scale may be derived based on
a. Sound Intensity
b. Sound Pressure

Back 21

Sound sources radiate acoustic energy. The rate of radiation of Energy = Power output (Joules/sec or Watts)
Sound intensity is the amount of sound power radiated by a source per square metre. The threshold of hearing is at an intensity of 1 picowatt (1 x 10^-12). The threshold of pain is 1watt per square meter. Using the threshold of hearing as a reference we can see how many times louder this is:
1/(1 x 10^-12) = 1000000000000 times greater.
Not practical to use such a big number AND the human ear is nonlinear anyway.
Log10(1/(1 x 10^-12)) = 12
This is the BEL scale
Decibels are just this scale multiplied by a factor of 10 so that we
have more steps:
10*Log10(1/(1 x 10^-12)) = 12
That’s for intensity, what about sound pressure?
Atmospheric pressure is the force per unit area exerted against a surface by the weight of air above that surface in the Earth's atmosphere. Sound pressure or acoustic pressure is the local pressure deviation from the ambient (average, or equilibrium) atmospheric pressure caused by a sound wave.