Scribd
Upload
154 views

ch4 - Acquiring Audio Data PDF

The document discusses acquiring and digitizing audio signals. It explains that audio is captured using a microphone which converts sound waves to electrical signals. These analog signals are then converted to digital using an analog-to-digital converter via a process called sampling, which approximates the amplitude at regular time intervals. The samples are then stored as binary numbers. For proper reconstruction of sound, Nyquist's theorem states the sampling rate must be at least twice the highest frequency present in the original signal. Common file formats for digital audio include WAVE files and compressed MP3 files.

Uploaded by

Beulah Christudas
Copyright
© © All Rights Reserved
Available Formats
Download as PDF, TXT or read online on Scribd
154 views

ch4 - Acquiring Audio Data PDF

The document discusses acquiring and digitizing audio signals. It explains that audio is captured using a microphone which converts sound waves to electrical signals. These analog signals are then converted to digital using an analog-to-digital converter via a process called sampling, which approximates the amplitude at regular time intervals. The samples are then stored as binary numbers. For proper reconstruction of sound, Nyquist's theorem states the sampling rate must be at least twice the highest frequency present in the original signal. Common file formats for digital audio include WAVE files and compressed MP3 files.

Uploaded by

Beulah Christudas
Copyright
© © All Rights Reserved
Available Formats
Download as PDF, TXT or read online on Scribd
You are on page 1/ 18

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.

You might also like