Unit 12 Flashcards
Why more than one loudspeaker unit are generally used?
- A single loudspeaker generally cannot satisfactorily cover the whole of the audio frequency spectrum
- Different loudspeaker units are designed to handle different parts of the audio spectrum.
- Typically there will be a low-frequency drive unit and a high- frequency drive unit.
- Sometimes there may also be a drive unit for middle frequencies.
- Sub-bass units for extremely low frequencies are sometimes used, but these tend to be housed as separate units.
What are the advantages of working with sound in its
digital form?
Advantages of using digital techniques:
* Immunity from signal corruption brought about by
extensive processing or through transmission or
storage.
* Mixing and processing of sound comes down to a
simple process of computation (‘number crunching’)
rather than involving complicated analogue electronic
circuits and devices.
* Computer storage techniques can easily be used for
storing sound in its digital form.
How do we get the analogue signals given out by a microphone into a digital signal (a number form)?
This is the process of analogue-to-digital conversion which is based on two basic stages:
1. Sampling
2. Quantisation.
Sampling
sampling an analogue signal is to measure the instantaneous amplitude of the analogue sound signal at regular intervals
The result is a set of voltage levels which represent the sound signal’s level at the instants the samples were taken
Quantisation
to divide the maximum voltage range of the analogue sound signal into a number of discrete voltage bands and assimilate each sound sample into a voltage band.
Explain how converting the signal back to its analogue form works:
Converting the signal back into its analogue form is done by a process called digital-to-analogue conversion .
* The sample numbers are taken at the same rate that they were originally generated
* For each number a voltage which is the centre value of the voltage band that the number represents is created (this centre value is called “Quantisation level”)
* The transitions between the sample voltages are then smoothed out to give an analogue signal.
What happens if the minimum sampling rate is NOT respected?
When the minimum sampling rate is not respected, another sinewave with a lower frequency can be drawn through these samples called an alias
Alias: A lower frequency wave obtained after a digital
to analogue conversion
Quantisation level
samples (from the sampling process) are allocated to
a voltage band and they are then assimilated to the centre value of the voltage band. The center value of each voltage band represents a quantisation level
Audio perceptual compression
removes the parts of the signal that have been found to be inaudible to human listeners.
A decoder
reconstructs the signal which should be perceived
by the listener to be the same as the original signal.
The MPEG-2 standard is one of these standards
Different MPEG-2 layers of compression are: layers 1, 2 and 3.
Layer 3 has become the most used and is now almost universally known as MP3.
MP3 audio compression offers acceptable audio quality with a high compression ratio in the region of 11:1.
Explain MP3 coding and how it works:
The stream of digital audio samples is sent first to a filter-bank which splits the audio into 32 frequency bands that match the frequency characteristics of the human ear.
The sound content of each band is analysed and coded using a psychoacoustic algorithm such as to require the lowest possible amount of data for the given content
By varying the sample rate the coder can allocate more samples to complex sounds and fewer to a less complex sound.
Normalisation
The sound is first scanned to determine the level of the loudest part.
The factor by which this largest sample value needs to be multiplied so that it attains a predefined ‘normalization level’ is then calculated.
Every sample value is multiplied by this factor.
What Normalization is used for?
This effect increases the level of a sound so that it uses the top end of the available amplitude range.
This is done so that low-level noise and interference that may be introduced during later processing become less audible.
Equalisation
boosts frequency ranges that have been
reduced through a transmission medium or audio
recorder in order to ‘equalise’ signal frequencies in these ranges back to their original levels.
