Acquiring Audio
Data
Topics
What is audio?
A/D Converter - Sampling
Why digitization?
Sampling Interval
Storage Requirement
Acquisition and Storage
Compression
Audio Data
Types : Speech, Music, Noise
Common feature of sound: Variant with time
Basics of Sound
Audio input is acquired using a microphone.
Microphone converts sound waves into an electrical voltage.
The electrical voltage varies with time and it is called as an audio waveform.
A continuous waveform is called as an analog signal.
To store this analog signal in memory, it should be converted into digital signal.
The conversion from analog to digital signal is done by an analog to digital converter.
(A/D converter)
The conversion is done by the process of sampling.
Replacing a continuous amplitude value by an approximated digital equivalent is called
amplitude quantization.
Why Should Audio be Digitized?
Digitized audio provides immunity from noise (not affected by small
disturbances).
Digital processing is performed using numeric calculations which are not
constrained by physical properties of components.
Digital processing is more powerful.
Digital processing is cheaper
Digital storage media is more durable.
Representing Sound
Sound needs to be converted into binary for computers to be able to process it.
To do this, sound is captured - usually by a microphone - and then converted into a digital
signal.
An analogue to digital converter will sample a sound wave at regular time intervals.
For example, a sound
wave like this can be
sampled at each time
sample point:
• The samples can then be converted to binary. They will be recorded to the
nearest whole number.
Time
1 2 3 4 5 6 7 8 9 10
sample
Decimal 8 3 7 6 9 7 2 6 6 6
Binary 1000 0011 0111 0110 1001 0111 0010 0100 0110 0110
• The sampling interval should be chosen in such a way that it captures all
information with the minimum number of samples.
• If the sampling interval is very small, the size of the sampling table will
become very large.
• If it is very large, the sound wave may be distorted and information will be
lost.
Calculation of Sampling Interval
A pure tone is an analog signal which is a sine wave.
fL - Lowest
Nyquist’s Sampling Theorem
If fhigh is the highest frequency present in the signal, then sampling interval should be slightly
less than 1/(2*fhigh)
Ex. fL=30 Hz; fH=3000 Hz
Sampling Interval ~ 1/(2*fH) = 1/(2*3000) = 1/6000 ~ 1/6250 = 0.00016 sec
= 0.00016 * 1000 = 0.16 milliseconds
Number of samples per second = 1/sampling interval = 1/(0.16 * 10-3) = 6250
samples/sec
Sampling interval for music may be much higher than that for speech
Storage needed for Digitized Audio Signals
Number of bits required to represent amplitudes (for speech) = 8 bits
Number of levels that can be represented = 28 = 256 levels
Number of bits required to represent music = 16 bits
Number of levels that can be represented = 216 > 64,000
If a telephone message of up to 1 minute is to be stored, how many bytes of
memory are required?
Solution
Number of samples per second = 6250
Number of samples per minute – 6250 x 60 = 375000
Number of bits/sample = 8 bits = 1 byte
Memory required to store 1 minute message = 375000 x 1 bytes
= 375000 bytes
= 375000/1024 = 367 KB
How many samples per second must be taken for audio signals suitable for a high
fidelity audio system?
Solution
For a high fidelity audio system,
fL = 20 Hz
fH = 22 kHz = 22000 Hz
Sampling interval ~ 1/(2*22000) = 1/44000.
Samples per second = 44100
High fidelity stereo music is to be represented in digital form. How many bytes are
needed to store 1 minute of music?
Solution:
Number of samples/second = 44100
Bytes/sample per channel = 2
In stereo systems, there are two independent audio signals.
Therefore, number of samples needed to store 1 minute of music = 2 x 44100 x 2 x 60
= 10584000 bytes = 10584000/(1024 x 1024) = 10.1 MB
Acquisition and Storage of Audio Signals
Sound waves are converted into electrical signals by a microphone.
If the audio signal is stored in a playback media, the audio output is fed to an A/D
converter.
The digital output of A/D converter is stored in memory.
The A/D converter is built into an electronic circuit called sound card.
Sound card can be added as an add-on card to the computer.
The digital file has the extension (.wav)
It is called a wave file.
The storage required depends on the type of input audio signal and the duration for
which the values are stored.
In a multimedia PC, good quality loud speakers are also connected.
The digitized audio signal can be played back using the sound card.
The card has an Digital to Analog converter (D/A converter) which reads the wave
file and converts it into an analog signal, amplifies it and feeds it into the loud
speakers.
Compression of Audio Signals
Compressed format : MP3
Moving Pictures Experts Group – Layer 3 audio compression standard.
Algorithm basic: When two sounds are played together the louder sound is heard
rather than the softer sound.
MP3 audio players consists of a processor which decompresses the compressed
file to original file, converts it to analog and plays it back using speakers.
Load Music Amplifier
MP3 files Store Processor D/A and
from PC MP3 Files Speaker
High Speed
Memory Decompression Conversion to
Plays music
(Random Program Analog Signal
Access
Advantages of MP3 Player
All the components used by MP3 player are integrated circuits. There are no
moving parts. So it will withstand jerks unlike a portable CD player.
Low power requirement.
Compact and light weight.
High memory size
Low cost
Part A & B Questions
Which is the input device used to acquire audio data?
----------------- converts sound waves into electrical voltage.
The conversion from analog to digital signal is done by ------------------.
The conversion from digital data to analog signal is done by ------------.
What is sampling?
What is amplitude quantization?
List the characteristics of digital audio.
State Nyquist’s sampling theorem.
What is the extension of digital audio file?
How do you compress sound?
State the advantages of MP3 player
Essay Questions
How do you acquire and store audio signals in a computer?
Explain the process of sampling audio signals.